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fuchsia / third_party / android / platform / frameworks / av / refs/heads/upstream/android10-qpr2-s3-release / . / services / audioflinger / Tracks.cpp
blob: 51e57b51cb7535c1e962cf6c55b582cdcc54d2fc [file] [log] [blame]
/*
**
** Copyright 2012, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#define LOG_TAG "AudioFlinger"
//#define LOG_NDEBUG 0
#include "Configuration.h"
#include <linux/futex.h>
#include <math.h>
#include <sys/syscall.h>
#include <utils/Log.h>
#include <private/media/AudioTrackShared.h>
#include "AudioFlinger.h"
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
#include <media/RecordBufferConverter.h>
#include <mediautils/ServiceUtilities.h>
#include <audio_utils/minifloat.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::VolumeShaper;
// ----------------------------------------------------------------------------
// TrackBase
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::TrackBase"
static volatile int32_t nextTrackId = 55;
// TrackBase constructor must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase::TrackBase::TrackBase(
ThreadBase *thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_session_t sessionId,
pid_t creatorPid,
uid_t clientUid,
bool isOut,
alloc_type alloc,
track_type type,
audio_port_handle_t portId)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(NULL),
// mBuffer, mBufferSize
mState(IDLE),
mAttr(attr),
mSampleRate(sampleRate),
mFormat(format),
mChannelMask(channelMask),
mChannelCount(isOut ?
audio_channel_count_from_out_mask(channelMask) :
audio_channel_count_from_in_mask(channelMask)),
mFrameSize(audio_has_proportional_frames(format) ?
mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
mFrameCount(frameCount),
mSessionId(sessionId),
mIsOut(isOut),
mId(android_atomic_inc(&nextTrackId)),
mTerminated(false),
mType(type),
mThreadIoHandle(thread ? thread->id() : AUDIO_IO_HANDLE_NONE),
mPortId(portId),
mIsInvalid(false),
mCreatorPid(creatorPid)
{
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
if (!isAudioServerOrMediaServerUid(callingUid) || clientUid == AUDIO_UID_INVALID) {
ALOGW_IF(clientUid != AUDIO_UID_INVALID && clientUid != callingUid,
"%s(%d): uid %d tried to pass itself off as %d",
__func__, mId, callingUid, clientUid);
clientUid = callingUid;
}
// clientUid contains the uid of the app that is responsible for this track, so we can blame
// battery usage on it.
mUid = clientUid;
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t minBufferSize = buffer == NULL ? roundup(frameCount) : frameCount;
// check overflow when computing bufferSize due to multiplication by mFrameSize.
if (minBufferSize < frameCount // roundup rounds down for values above UINT_MAX / 2
|| mFrameSize == 0 // format needs to be correct
|| minBufferSize > SIZE_MAX / mFrameSize) {
android_errorWriteLog(0x534e4554, "34749571");
return;
}
minBufferSize *= mFrameSize;
if (buffer == nullptr) {
bufferSize = minBufferSize; // allocated here.
} else if (minBufferSize > bufferSize) {
android_errorWriteLog(0x534e4554, "38340117");
return;
}
size_t size = sizeof(audio_track_cblk_t);
if (buffer == NULL && alloc == ALLOC_CBLK) {
// check overflow when computing allocation size for streaming tracks.
if (size > SIZE_MAX - bufferSize) {
android_errorWriteLog(0x534e4554, "34749571");
return;
}
size += bufferSize;
}
if (client != 0) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory == 0 ||
(mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) {
ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
client->heap()->dump("AudioTrack");
mCblkMemory.clear();
return;
}
} else {
mCblk = (audio_track_cblk_t *) malloc(size);
if (mCblk == NULL) {
ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
return;
}
}
// construct the shared structure in-place.
if (mCblk != NULL) {
new(mCblk) audio_track_cblk_t();
switch (alloc) {
case ALLOC_READONLY: {
const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
if (roHeap == 0 ||
(mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
(mBuffer = mBufferMemory->pointer()) == NULL) {
ALOGE("%s(%d): not enough memory for read-only buffer size=%zu",
__func__, mId, bufferSize);
if (roHeap != 0) {
roHeap->dump("buffer");
}
mCblkMemory.clear();
mBufferMemory.clear();
return;
}
memset(mBuffer, 0, bufferSize);
} break;
case ALLOC_PIPE:
mBufferMemory = thread->pipeMemory();
// mBuffer is the virtual address as seen from current process (mediaserver),
// and should normally be coming from mBufferMemory->pointer().
// However in this case the TrackBase does not reference the buffer directly.
// It should references the buffer via the pipe.
// Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
mBuffer = NULL;
bufferSize = 0;
break;
case ALLOC_CBLK:
// clear all buffers
if (buffer == NULL) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, bufferSize);
} else {
mBuffer = buffer;
#if 0
mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic
#endif
}
break;
case ALLOC_LOCAL:
mBuffer = calloc(1, bufferSize);
break;
case ALLOC_NONE:
mBuffer = buffer;
break;
default:
LOG_ALWAYS_FATAL("%s(%d): invalid allocation type: %d", __func__, mId, (int)alloc);
}
mBufferSize = bufferSize;
#ifdef TEE_SINK
mTee.set(sampleRate, mChannelCount, format, NBAIO_Tee::TEE_FLAG_TRACK);
#endif
}
}
status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const
{
status_t status;
if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) {
status = cblk() != NULL ? NO_ERROR : NO_MEMORY;
} else {
status = getCblk() != 0 ? NO_ERROR : NO_MEMORY;
}
return status;
}
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
// delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
mServerProxy.clear();
if (mCblk != NULL) {
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
if (mClient == 0) {
free(mCblk);
}
}
mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
if (mClient != 0) {
// Client destructor must run with AudioFlinger client mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mClientLock);
// If the client's reference count drops to zero, the associated destructor
// must run with AudioFlinger lock held. Thus the explicit clear() rather than
// relying on the automatic clear() at end of scope.
mClient.clear();
}
// flush the binder command buffer
IPCThreadState::self()->flushCommands();
}
// AudioBufferProvider interface
// getNextBuffer() = 0;
// This implementation of releaseBuffer() is used by Track and RecordTrack
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
#ifdef TEE_SINK
mTee.write(buffer->raw, buffer->frameCount);
#endif
ServerProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
buf.mRaw = buffer->raw;
buffer->frameCount = 0;
buffer->raw = NULL;
mServerProxy->releaseBuffer(&buf);
}
status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event)
{
mSyncEvents.add(event);
return NO_ERROR;
}
AudioFlinger::ThreadBase::PatchTrackBase::PatchTrackBase(sp<ClientProxy> proxy,
const ThreadBase& thread,
const Timeout& timeout)
: mProxy(proxy)
{
if (timeout) {
setPeerTimeout(*timeout);
} else {
// Double buffer mixer
uint64_t mixBufferNs = ((uint64_t)2 * thread.frameCount() * 1000000000) /
thread.sampleRate();
setPeerTimeout(std::chrono::nanoseconds{mixBufferNs});
}
}
void AudioFlinger::ThreadBase::PatchTrackBase::setPeerTimeout(std::chrono::nanoseconds timeout) {
mPeerTimeout.tv_sec = timeout.count() / std::nano::den;
mPeerTimeout.tv_nsec = timeout.count() % std::nano::den;
}
// ----------------------------------------------------------------------------
// Playback
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::TrackHandle"
AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
: BnAudioTrack(),
mTrack(track)
{
}
AudioFlinger::TrackHandle::~TrackHandle() {
// just stop the track on deletion, associated resources
// will be freed from the main thread once all pending buffers have
// been played. Unless it's not in the active track list, in which
// case we free everything now...
mTrack->destroy();
}
sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
return mTrack->getCblk();
}
status_t AudioFlinger::TrackHandle::start() {
return mTrack->start();
}
void AudioFlinger::TrackHandle::stop() {
mTrack->stop();
}
void AudioFlinger::TrackHandle::flush() {
mTrack->flush();
}
void AudioFlinger::TrackHandle::pause() {
mTrack->pause();
}
status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
{
return mTrack->attachAuxEffect(EffectId);
}
status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) {
return mTrack->setParameters(keyValuePairs);
}
status_t AudioFlinger::TrackHandle::selectPresentation(int presentationId, int programId) {
return mTrack->selectPresentation(presentationId, programId);
}
VolumeShaper::Status AudioFlinger::TrackHandle::applyVolumeShaper(
const sp<VolumeShaper::Configuration>& configuration,
const sp<VolumeShaper::Operation>& operation) {
return mTrack->applyVolumeShaper(configuration, operation);
}
sp<VolumeShaper::State> AudioFlinger::TrackHandle::getVolumeShaperState(int id) {
return mTrack->getVolumeShaperState(id);
}
status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
{
return mTrack->getTimestamp(timestamp);
}
void AudioFlinger::TrackHandle::signal()
{
return mTrack->signal();
}
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioTrack::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
// AppOp for audio playback
// -------------------------------
// static
sp<AudioFlinger::PlaybackThread::OpPlayAudioMonitor>
AudioFlinger::PlaybackThread::OpPlayAudioMonitor::createIfNeeded(
uid_t uid, const audio_attributes_t& attr, int id, audio_stream_type_t streamType)
{
if (isServiceUid(uid)) {
Vector <String16> packages;
getPackagesForUid(uid, packages);
if (packages.isEmpty()) {
ALOGD("OpPlayAudio: not muting track:%d usage:%d for service UID %d",
id,
attr.usage,
uid);
return nullptr;
}
}
// stream type has been filtered by audio policy to indicate whether it can be muted
if (streamType == AUDIO_STREAM_ENFORCED_AUDIBLE) {
ALOGD("OpPlayAudio: not muting track:%d usage:%d ENFORCED_AUDIBLE", id, attr.usage);
return nullptr;
}
if ((attr.flags & AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY)
== AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY) {
ALOGD("OpPlayAudio: not muting track:%d flags %#x have FLAG_BYPASS_INTERRUPTION_POLICY",
id, attr.flags);
return nullptr;
}
return new OpPlayAudioMonitor(uid, attr.usage, id);
}
AudioFlinger::PlaybackThread::OpPlayAudioMonitor::OpPlayAudioMonitor(
uid_t uid, audio_usage_t usage, int id)
: mHasOpPlayAudio(true), mUid(uid), mUsage((int32_t) usage), mId(id)
{
}
AudioFlinger::PlaybackThread::OpPlayAudioMonitor::~OpPlayAudioMonitor()
{
if (mOpCallback != 0) {
mAppOpsManager.stopWatchingMode(mOpCallback);
}
mOpCallback.clear();
}
void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::onFirstRef()
{
getPackagesForUid(mUid, mPackages);
checkPlayAudioForUsage();
if (!mPackages.isEmpty()) {
mOpCallback = new PlayAudioOpCallback(this);
mAppOpsManager.startWatchingMode(AppOpsManager::OP_PLAY_AUDIO, mPackages[0], mOpCallback);
}
}
bool AudioFlinger::PlaybackThread::OpPlayAudioMonitor::hasOpPlayAudio() const {
return mHasOpPlayAudio.load();
}
// Note this method is never called (and never to be) for audio server / patch record track
// - not called from constructor due to check on UID,
// - not called from PlayAudioOpCallback because the callback is not installed in this case
void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::checkPlayAudioForUsage()
{
if (mPackages.isEmpty()) {
mHasOpPlayAudio.store(false);
} else {
bool hasIt = true;
for (const String16& packageName : mPackages) {
const int32_t mode = mAppOpsManager.checkAudioOpNoThrow(AppOpsManager::OP_PLAY_AUDIO,
mUsage, mUid, packageName);
if (mode != AppOpsManager::MODE_ALLOWED) {
hasIt = false;
break;
}
}
ALOGD("OpPlayAudio: track:%d usage:%d %smuted", mId, mUsage, hasIt ? "not " : "");
mHasOpPlayAudio.store(hasIt);
}
}
AudioFlinger::PlaybackThread::OpPlayAudioMonitor::PlayAudioOpCallback::PlayAudioOpCallback(
const wp<OpPlayAudioMonitor>& monitor) : mMonitor(monitor)
{ }
void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::PlayAudioOpCallback::opChanged(int32_t op,
const String16& packageName) {
// we only have uid, so we need to check all package names anyway
UNUSED(packageName);
if (op != AppOpsManager::OP_PLAY_AUDIO) {
return;
}
sp<OpPlayAudioMonitor> monitor = mMonitor.promote();
if (monitor != NULL) {
monitor->checkPlayAudioForUsage();
}
}
// static
void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::getPackagesForUid(
uid_t uid, Vector<String16>& packages)
{
PermissionController permissionController;
permissionController.getPackagesForUid(uid, packages);
}
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::Track"
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
pid_t creatorPid,
uid_t uid,
audio_output_flags_t flags,
track_type type,
audio_port_handle_t portId,
size_t frameCountToBeReady)
: TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
(sharedBuffer != 0) ? sharedBuffer->pointer() : buffer,
(sharedBuffer != 0) ? sharedBuffer->size() : bufferSize,
sessionId, creatorPid, uid, true /*isOut*/,
(type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK,
type, portId),
mFillingUpStatus(FS_INVALID),
// mRetryCount initialized later when needed
mSharedBuffer(sharedBuffer),
mStreamType(streamType),
mMainBuffer(thread->sinkBuffer()),
mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false),
mPresentationCompleteFrames(0),
mFrameMap(16 /* sink-frame-to-track-frame map memory */),
mVolumeHandler(new media::VolumeHandler(sampleRate)),
mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(uid, attr, id(), streamType)),
// mSinkTimestamp
mFrameCountToBeReady(frameCountToBeReady),
mFastIndex(-1),
mCachedVolume(1.0),
/* The track might not play immediately after being active, similarly as if its volume was 0.
* When the track starts playing, its volume will be computed. */
mFinalVolume(0.f),
mResumeToStopping(false),
mFlushHwPending(false),
mFlags(flags)
{
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
ALOGV_IF(sharedBuffer != 0, "%s(%d): sharedBuffer: %p, size: %zu",
__func__, mId, sharedBuffer->pointer(), sharedBuffer->size());
if (mCblk == NULL) {
return;
}
if (sharedBuffer == 0) {
mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, !isExternalTrack(), sampleRate);
} else {
mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount,
mFrameSize);
}
mServerProxy = mAudioTrackServerProxy;
if (!thread->isTrackAllowed_l(channelMask, format, sessionId, uid)) {
ALOGE("%s(%d): no more tracks available", __func__, mId);
return;
}
// only allocate a fast track index if we were able to allocate a normal track name
if (flags & AUDIO_OUTPUT_FLAG_FAST) {
// FIXME: Not calling framesReadyIsCalledByMultipleThreads() exposes a potential
// race with setSyncEvent(). However, if we call it, we cannot properly start
// static fast tracks (SoundPool) immediately after stopping.
//mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
int i = __builtin_ctz(thread->mFastTrackAvailMask);
ALOG_ASSERT(0 < i && i < (int)FastMixerState::sMaxFastTracks);
// FIXME This is too eager. We allocate a fast track index before the
// fast track becomes active. Since fast tracks are a scarce resource,
// this means we are potentially denying other more important fast tracks from
// being created. It would be better to allocate the index dynamically.
mFastIndex = i;
thread->mFastTrackAvailMask &= ~(1 << i);
}
mServerLatencySupported = thread->type() == ThreadBase::MIXER
|| thread->type() == ThreadBase::DUPLICATING;
#ifdef TEE_SINK
mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
+ "_" + std::to_string(mId) + "_T");
#endif
if (channelMask & AUDIO_CHANNEL_HAPTIC_ALL) {
mAudioVibrationController = new AudioVibrationController(this);
mExternalVibration = new os::ExternalVibration(
mUid, "" /* pkg */, mAttr, mAudioVibrationController);
}
}
AudioFlinger::PlaybackThread::Track::~Track()
{
ALOGV("%s(%d)", __func__, mId);
// The destructor would clear mSharedBuffer,
// but it will not push the decremented reference count,
// leaving the client's IMemory dangling indefinitely.
// This prevents that leak.
if (mSharedBuffer != 0) {
mSharedBuffer.clear();
}
}
status_t AudioFlinger::PlaybackThread::Track::initCheck() const
{
status_t status = TrackBase::initCheck();
if (status == NO_ERROR && mCblk == nullptr) {
status = NO_MEMORY;
}
return status;
}
void AudioFlinger::PlaybackThread::Track::destroy()
{
// NOTE: destroyTrack_l() can remove a strong reference to this Track
// by removing it from mTracks vector, so there is a risk that this Tracks's
// destructor is called. As the destructor needs to lock mLock,
// we must acquire a strong reference on this Track before locking mLock
// here so that the destructor is called only when exiting this function.
// On the other hand, as long as Track::destroy() is only called by
// TrackHandle destructor, the TrackHandle still holds a strong ref on
// this Track with its member mTrack.
sp<Track> keep(this);
{ // scope for mLock
bool wasActive = false;
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
wasActive = playbackThread->destroyTrack_l(this);
}
if (isExternalTrack() && !wasActive) {
AudioSystem::releaseOutput(mPortId);
}
}
forEachTeePatchTrack([](auto patchTrack) { patchTrack->destroy(); });
}
void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
{
result.appendFormat("Type Id Active Client Session Port Id S Flags "
" Format Chn mask SRate "
"ST Usg CT "
" G db L dB R dB VS dB "
" Server FrmCnt FrmRdy F Underruns Flushed"
"%s\n",
isServerLatencySupported() ? " Latency" : "");
}
void AudioFlinger::PlaybackThread::Track::appendDump(String8& result, bool active)
{
char trackType;
switch (mType) {
case TYPE_DEFAULT:
case TYPE_OUTPUT:
if (isStatic()) {
trackType = 'S'; // static
} else {
trackType = ' '; // normal
}
break;
case TYPE_PATCH:
trackType = 'P';
break;
default:
trackType = '?';
}
if (isFastTrack()) {
result.appendFormat("F%d %c %6d", mFastIndex, trackType, mId);
} else {
result.appendFormat(" %c %6d", trackType, mId);
}
char nowInUnderrun;
switch (mObservedUnderruns.mBitFields.mMostRecent) {
case UNDERRUN_FULL:
nowInUnderrun = ' ';
break;
case UNDERRUN_PARTIAL:
nowInUnderrun = '<';
break;
case UNDERRUN_EMPTY:
nowInUnderrun = '*';
break;
default:
nowInUnderrun = '?';
break;
}
char fillingStatus;
switch (mFillingUpStatus) {
case FS_INVALID:
fillingStatus = 'I';
break;
case FS_FILLING:
fillingStatus = 'f';
break;
case FS_FILLED:
fillingStatus = 'F';
break;
case FS_ACTIVE:
fillingStatus = 'A';
break;
default:
fillingStatus = '?';
break;
}
// clip framesReadySafe to max representation in dump
const size_t framesReadySafe =
std::min(mAudioTrackServerProxy->framesReadySafe(), (size_t)99999999);
// obtain volumes
const gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
const std::pair<float /* volume */, bool /* active */> vsVolume =
mVolumeHandler->getLastVolume();
// Our effective frame count is obtained by ServerProxy::getBufferSizeInFrames()
// as it may be reduced by the application.
const size_t bufferSizeInFrames = (size_t)mAudioTrackServerProxy->getBufferSizeInFrames();
// Check whether the buffer size has been modified by the app.
const char modifiedBufferChar = bufferSizeInFrames < mFrameCount
? 'r' /* buffer reduced */: bufferSizeInFrames > mFrameCount
? 'e' /* error */ : ' ' /* identical */;
result.appendFormat("%7s %6u %7u %7u %2s 0x%03X "
"%08X %08X %6u "
"%2u %3x %2x "
"%5.2g %5.2g %5.2g %5.2g%c "
"%08X %6zu%c %6zu %c %9u%c %7u",
active ? "yes" : "no",
(mClient == 0) ? getpid() : mClient->pid(),
mSessionId,
mPortId,
getTrackStateString(),
mCblk->mFlags,
mFormat,
mChannelMask,
sampleRate(),
mStreamType,
mAttr.usage,
mAttr.content_type,
20.0 * log10(mFinalVolume),
20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))),
20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
20.0 * log10(vsVolume.first), // VolumeShaper(s) total volume
vsVolume.second ? 'A' : ' ', // if any VolumeShapers active
mCblk->mServer,
bufferSizeInFrames,
modifiedBufferChar,
framesReadySafe,
fillingStatus,
mAudioTrackServerProxy->getUnderrunFrames(),
nowInUnderrun,
(unsigned)mAudioTrackServerProxy->framesFlushed() % 10000000
);
if (isServerLatencySupported()) {
double latencyMs;
bool fromTrack;
if (getTrackLatencyMs(&latencyMs, &fromTrack) == OK) {
// Show latency in msec, followed by 't' if from track timestamp (the most accurate)
// or 'k' if estimated from kernel because track frames haven't been presented yet.
result.appendFormat(" %7.2lf %c", latencyMs, fromTrack ? 't' : 'k');
} else {
result.appendFormat("%10s", mCblk->mServer != 0 ? "unavail" : "new");
}
}
result.append("\n");
}
uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
return mAudioTrackServerProxy->getSampleRate();
}
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
ServerProxy::Buffer buf;
size_t desiredFrames = buffer->frameCount;
buf.mFrameCount = desiredFrames;
status_t status = mServerProxy->obtainBuffer(&buf);
buffer->frameCount = buf.mFrameCount;
buffer->raw = buf.mRaw;
if (buf.mFrameCount == 0 && !isStopping() && !isStopped() && !isPaused()) {
ALOGV("%s(%d): underrun, framesReady(%zu) < framesDesired(%zd), state: %d",
__func__, mId, buf.mFrameCount, desiredFrames, mState);
mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
} else {
mAudioTrackServerProxy->tallyUnderrunFrames(0);
}
return status;
}
void AudioFlinger::PlaybackThread::Track::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
interceptBuffer(*buffer);
TrackBase::releaseBuffer(buffer);
}
// TODO: compensate for time shift between HW modules.
void AudioFlinger::PlaybackThread::Track::interceptBuffer(
const AudioBufferProvider::Buffer& sourceBuffer) {
auto start = std::chrono::steady_clock::now();
const size_t frameCount = sourceBuffer.frameCount;
if (frameCount == 0) {
return; // No audio to intercept.
// Additionally PatchProxyBufferProvider::obtainBuffer (called by PathTrack::getNextBuffer)
// does not allow 0 frame size request contrary to getNextBuffer
}
for (auto& teePatch : mTeePatches) {
RecordThread::PatchRecord* patchRecord = teePatch.patchRecord.get();
size_t framesWritten = writeFrames(patchRecord, sourceBuffer.i8, frameCount);
// On buffer wrap, the buffer frame count will be less than requested,
// when this happens a second buffer needs to be used to write the leftover audio
size_t framesLeft = frameCount - framesWritten;
if (framesWritten != 0 && framesLeft != 0) {
framesWritten +=
writeFrames(patchRecord, sourceBuffer.i8 + framesWritten * mFrameSize, framesLeft);
framesLeft = frameCount - framesWritten;
}
ALOGW_IF(framesLeft != 0, "%s(%d) PatchRecord %d can not provide big enough "
"buffer %zu/%zu, dropping %zu frames", __func__, mId, patchRecord->mId,
framesWritten, frameCount, framesLeft);
}
auto spent = ceil<std::chrono::microseconds>(std::chrono::steady_clock::now() - start);
using namespace std::chrono_literals;
// Average is ~20us per track, this should virtually never be logged (Logging takes >200us)
ALOGD_IF(spent > 500us, "%s: took %lldus to intercept %zu tracks", __func__,
spent.count(), mTeePatches.size());
}
size_t AudioFlinger::PlaybackThread::Track::writeFrames(AudioBufferProvider* dest,
const void* src,
size_t frameCount) {
AudioBufferProvider::Buffer patchBuffer;
patchBuffer.frameCount = frameCount;
auto status = dest->getNextBuffer(&patchBuffer);
if (status != NO_ERROR) {
ALOGW("%s PathRecord getNextBuffer failed with error %d: %s",
__func__, status, strerror(-status));
return 0;
}
ALOG_ASSERT(patchBuffer.frameCount <= frameCount);
memcpy(patchBuffer.raw, src, patchBuffer.frameCount * mFrameSize);
auto framesWritten = patchBuffer.frameCount;
dest->releaseBuffer(&patchBuffer);
return framesWritten;
}
// releaseBuffer() is not overridden
// ExtendedAudioBufferProvider interface
// framesReady() may return an approximation of the number of frames if called
// from a different thread than the one calling Proxy->obtainBuffer() and
// Proxy->releaseBuffer(). Also note there is no mutual exclusion in the
// AudioTrackServerProxy so be especially careful calling with FastTracks.
size_t AudioFlinger::PlaybackThread::Track::framesReady() const {
if (mSharedBuffer != 0 && (isStopped() || isStopping())) {
// Static tracks return zero frames immediately upon stopping (for FastTracks).
// The remainder of the buffer is not drained.
return 0;
}
return mAudioTrackServerProxy->framesReady();
}
int64_t AudioFlinger::PlaybackThread::Track::framesReleased() const
{
return mAudioTrackServerProxy->framesReleased();
}
void AudioFlinger::PlaybackThread::Track::onTimestamp(const ExtendedTimestamp &timestamp)
{
// This call comes from a FastTrack and should be kept lockless.
// The server side frames are already translated to client frames.
mAudioTrackServerProxy->setTimestamp(timestamp);
// We do not set drained here, as FastTrack timestamp may not go to very last frame.
// Compute latency.
// TODO: Consider whether the server latency may be passed in by FastMixer
// as a constant for all active FastTracks.
const double latencyMs = timestamp.getOutputServerLatencyMs(sampleRate());
mServerLatencyFromTrack.store(true);
mServerLatencyMs.store(latencyMs);
}
// Don't call for fast tracks; the framesReady() could result in priority inversion
bool AudioFlinger::PlaybackThread::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) {
return true;
}
if (isStopping()) {
if (framesReady() > 0) {
mFillingUpStatus = FS_FILLED;
}
return true;
}
size_t bufferSizeInFrames = mServerProxy->getBufferSizeInFrames();
size_t framesToBeReady = std::min(mFrameCountToBeReady, bufferSizeInFrames);
if (framesReady() >= framesToBeReady || (mCblk->mFlags & CBLK_FORCEREADY)) {
ALOGV("%s(%d): consider track ready with %zu/%zu, target was %zu)",
__func__, mId, framesReady(), bufferSizeInFrames, framesToBeReady);
mFillingUpStatus = FS_FILLED;
android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
return true;
}
return false;
}
status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
audio_session_t triggerSession __unused)
{
status_t status = NO_ERROR;
ALOGV("%s(%d): calling pid %d session %d",
__func__, mId, IPCThreadState::self()->getCallingPid(), mSessionId);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
if (isOffloaded()) {
Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
Mutex::Autolock _lth(thread->mLock);
sp<EffectChain> ec = thread->getEffectChain_l(mSessionId);
if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
(ec != 0 && ec->isNonOffloadableEnabled())) {
invalidate();
return PERMISSION_DENIED;
}
}
Mutex::Autolock _lth(thread->mLock);
track_state state = mState;
// here the track could be either new, or restarted
// in both cases "unstop" the track
// initial state-stopping. next state-pausing.
// What if resume is called ?
if (state == PAUSED || state == PAUSING) {
if (mResumeToStopping) {
// happened we need to resume to STOPPING_1
mState = TrackBase::STOPPING_1;
ALOGV("%s(%d): PAUSED => STOPPING_1 on thread %d",
__func__, mId, (int)mThreadIoHandle);
} else {
mState = TrackBase::RESUMING;
ALOGV("%s(%d): PAUSED => RESUMING on thread %d",
__func__, mId, (int)mThreadIoHandle);
}
} else {
mState = TrackBase::ACTIVE;
ALOGV("%s(%d): ? => ACTIVE on thread %d",
__func__, mId, (int)mThreadIoHandle);
}
// states to reset position info for non-offloaded/direct tracks
if (!isOffloaded() && !isDirect()
&& (state == IDLE || state == STOPPED || state == FLUSHED)) {
mFrameMap.reset();
}
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (isFastTrack()) {
// refresh fast track underruns on start because that field is never cleared
// by the fast mixer; furthermore, the same track can be recycled, i.e. start
// after stop.
mObservedUnderruns = playbackThread->getFastTrackUnderruns(mFastIndex);
}
status = playbackThread->addTrack_l(this);
if (status == INVALID_OPERATION || status == PERMISSION_DENIED) {
triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
// restore previous state if start was rejected by policy manager
if (status == PERMISSION_DENIED) {
mState = state;
}
}
if (status == NO_ERROR || status == ALREADY_EXISTS) {
// for streaming tracks, remove the buffer read stop limit.
mAudioTrackServerProxy->start();
}
// track was already in the active list, not a problem
if (status == ALREADY_EXISTS) {
status = NO_ERROR;
} else {
// Acknowledge any pending flush(), so that subsequent new data isn't discarded.
// It is usually unsafe to access the server proxy from a binder thread.
// But in this case we know the mixer thread (whether normal mixer or fast mixer)
// isn't looking at this track yet: we still hold the normal mixer thread lock,
// and for fast tracks the track is not yet in the fast mixer thread's active set.
// For static tracks, this is used to acknowledge change in position or loop.
ServerProxy::Buffer buffer;
buffer.mFrameCount = 1;
(void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);
}
} else {
status = BAD_VALUE;
}
if (status == NO_ERROR) {
forEachTeePatchTrack([](auto patchTrack) { patchTrack->start(); });
}
return status;
}
void AudioFlinger::PlaybackThread::Track::stop()
{
ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
track_state state = mState;
if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
// If the track is not active (PAUSED and buffers full), flush buffers
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
mState = STOPPED;
} else if (!isFastTrack() && !isOffloaded() && !isDirect()) {
mState = STOPPED;
} else {
// For fast tracks prepareTracks_l() will set state to STOPPING_2
// presentation is complete
// For an offloaded track this starts a drain and state will
// move to STOPPING_2 when drain completes and then STOPPED
mState = STOPPING_1;
if (isOffloaded()) {
mRetryCount = PlaybackThread::kMaxTrackStopRetriesOffload;
}
}
playbackThread->broadcast_l();
ALOGV("%s(%d): not stopping/stopped => stopping/stopped on thread %d",
__func__, mId, (int)mThreadIoHandle);
}
}
forEachTeePatchTrack([](auto patchTrack) { patchTrack->stop(); });
}
void AudioFlinger::PlaybackThread::Track::pause()
{
ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
switch (mState) {
case STOPPING_1:
case STOPPING_2:
if (!isOffloaded()) {
/* nothing to do if track is not offloaded */
break;
}
// Offloaded track was draining, we need to carry on draining when resumed
mResumeToStopping = true;
FALLTHROUGH_INTENDED;
case ACTIVE:
case RESUMING:
mState = PAUSING;
ALOGV("%s(%d): ACTIVE/RESUMING => PAUSING on thread %d",
__func__, mId, (int)mThreadIoHandle);
playbackThread->broadcast_l();
break;
default:
break;
}
}
// Pausing the TeePatch to avoid a glitch on underrun, at the cost of buffered audio loss.
forEachTeePatchTrack([](auto patchTrack) { patchTrack->pause(); });
}
void AudioFlinger::PlaybackThread::Track::flush()
{
ALOGV("%s(%d)", __func__, mId);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
// Flush the ring buffer now if the track is not active in the PlaybackThread.
// Otherwise the flush would not be done until the track is resumed.
// Requires FastTrack removal be BLOCK_UNTIL_ACKED
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
(void)mServerProxy->flushBufferIfNeeded();
}
if (isOffloaded()) {
// If offloaded we allow flush during any state except terminated
// and keep the track active to avoid problems if user is seeking
// rapidly and underlying hardware has a significant delay handling
// a pause
if (isTerminated()) {
return;
}
ALOGV("%s(%d): offload flush", __func__, mId);
reset();
if (mState == STOPPING_1 || mState == STOPPING_2) {
ALOGV("%s(%d): flushed in STOPPING_1 or 2 state, change state to ACTIVE",
__func__, mId);
mState = ACTIVE;
}
mFlushHwPending = true;
mResumeToStopping = false;
} else {
if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED &&
mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) {
return;
}
// No point remaining in PAUSED state after a flush => go to
// FLUSHED state
mState = FLUSHED;
// do not reset the track if it is still in the process of being stopped or paused.
// this will be done by prepareTracks_l() when the track is stopped.
// prepareTracks_l() will see mState == FLUSHED, then
// remove from active track list, reset(), and trigger presentation complete
if (isDirect()) {
mFlushHwPending = true;
}
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
}
}
// Prevent flush being lost if the track is flushed and then resumed
// before mixer thread can run. This is important when offloading
// because the hardware buffer could hold a large amount of audio
playbackThread->broadcast_l();
}
// Flush the Tee to avoid on resume playing old data and glitching on the transition to new data
forEachTeePatchTrack([](auto patchTrack) { patchTrack->flush(); });
}
// must be called with thread lock held
void AudioFlinger::PlaybackThread::Track::flushAck()
{
if (!isOffloaded() && !isDirect())
return;
// Clear the client ring buffer so that the app can prime the buffer while paused.
// Otherwise it might not get cleared until playback is resumed and obtainBuffer() is called.
mServerProxy->flushBufferIfNeeded();
mFlushHwPending = false;
}
void AudioFlinger::PlaybackThread::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before
// the audioflinger thread detects the track is stopped.
if (!mResetDone) {
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
mFillingUpStatus = FS_FILLING;
mResetDone = true;
if (mState == FLUSHED) {
mState = IDLE;
}
}
}
status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs)
{
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
ALOGE("%s(%d): thread is dead", __func__, mId);
return FAILED_TRANSACTION;
} else if ((thread->type() == ThreadBase::DIRECT) ||
(thread->type() == ThreadBase::OFFLOAD)) {
return thread->setParameters(keyValuePairs);
} else {
return PERMISSION_DENIED;
}
}
status_t AudioFlinger::PlaybackThread::Track::selectPresentation(int presentationId,
int programId) {
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
ALOGE("thread is dead");
return FAILED_TRANSACTION;
} else if ((thread->type() == ThreadBase::DIRECT) || (thread->type() == ThreadBase::OFFLOAD)) {
DirectOutputThread *directOutputThread = static_cast<DirectOutputThread*>(thread.get());
return directOutputThread->selectPresentation(presentationId, programId);
}
return INVALID_OPERATION;
}
VolumeShaper::Status AudioFlinger::PlaybackThread::Track::applyVolumeShaper(
const sp<VolumeShaper::Configuration>& configuration,
const sp<VolumeShaper::Operation>& operation)
{
sp<VolumeShaper::Configuration> newConfiguration;
if (isOffloadedOrDirect()) {
const VolumeShaper::Configuration::OptionFlag optionFlag
= configuration->getOptionFlags();
if ((optionFlag & VolumeShaper::Configuration::OPTION_FLAG_CLOCK_TIME) == 0) {
ALOGW("%s(%d): %s tracks do not support frame counted VolumeShaper,"
" using clock time instead",
__func__, mId,
isOffloaded() ? "Offload" : "Direct");
newConfiguration = new VolumeShaper::Configuration(*configuration);
newConfiguration->setOptionFlags(
VolumeShaper::Configuration::OptionFlag(optionFlag
| VolumeShaper::Configuration::OPTION_FLAG_CLOCK_TIME));
}
}
VolumeShaper::Status status = mVolumeHandler->applyVolumeShaper(
(newConfiguration.get() != nullptr ? newConfiguration : configuration), operation);
if (isOffloadedOrDirect()) {
// Signal thread to fetch new volume.
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
thread->broadcast_l();
}
}
return status;
}
sp<VolumeShaper::State> AudioFlinger::PlaybackThread::Track::getVolumeShaperState(int id)
{
// Note: We don't check if Thread exists.
// mVolumeHandler is thread safe.
return mVolumeHandler->getVolumeShaperState(id);
}
void AudioFlinger::PlaybackThread::Track::setFinalVolume(float volume)
{
if (mFinalVolume != volume) { // Compare to an epsilon if too many meaningless updates
mFinalVolume = volume;
setMetadataHasChanged();
}
}
void AudioFlinger::PlaybackThread::Track::copyMetadataTo(MetadataInserter& backInserter) const
{
*backInserter++ = {
.usage = mAttr.usage,
.content_type = mAttr.content_type,
.gain = mFinalVolume,
};
}
void AudioFlinger::PlaybackThread::Track::setTeePatches(TeePatches teePatches) {
forEachTeePatchTrack([](auto patchTrack) { patchTrack->destroy(); });
mTeePatches = std::move(teePatches);
}
status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
{
if (!isOffloaded() && !isDirect()) {
return INVALID_OPERATION; // normal tracks handled through SSQ
}
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
return INVALID_OPERATION;
}
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
return playbackThread->getTimestamp_l(timestamp);
}
status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
{
sp<ThreadBase> thread = mThread.promote();
if (thread == nullptr) {
return DEAD_OBJECT;
}
sp<PlaybackThread> dstThread = (PlaybackThread *)thread.get();
sp<PlaybackThread> srcThread; // srcThread is initialized by call to moveAuxEffectToIo()
sp<AudioFlinger> af = mClient->audioFlinger();
status_t status = af->moveAuxEffectToIo(EffectId, dstThread, &srcThread);
if (EffectId != 0 && status == NO_ERROR) {
status = dstThread->attachAuxEffect(this, EffectId);
if (status == NO_ERROR) {
AudioSystem::moveEffectsToIo(std::vector<int>(EffectId), dstThread->id());
}
}
if (status != NO_ERROR && srcThread != nullptr) {
af->moveAuxEffectToIo(EffectId, srcThread, &dstThread);
}
return status;
}
void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
{
mAuxEffectId = EffectId;
mAuxBuffer = buffer;
}
bool AudioFlinger::PlaybackThread::Track::presentationComplete(
int64_t framesWritten, size_t audioHalFrames)
{
// TODO: improve this based on FrameMap if it exists, to ensure full drain.
// This assists in proper timestamp computation as well as wakelock management.
// a track is considered presented when the total number of frames written to audio HAL
// corresponds to the number of frames written when presentationComplete() is called for the
// first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time.
// For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used
// to detect when all frames have been played. In this case framesWritten isn't
// useful because it doesn't always reflect whether there is data in the h/w
// buffers, particularly if a track has been paused and resumed during draining
ALOGV("%s(%d): presentationComplete() mPresentationCompleteFrames %lld framesWritten %lld",
__func__, mId,
(long long)mPresentationCompleteFrames, (long long)framesWritten);
if (mPresentationCompleteFrames == 0) {
mPresentationCompleteFrames = framesWritten + audioHalFrames;
ALOGV("%s(%d): presentationComplete() reset:"
" mPresentationCompleteFrames %lld audioHalFrames %zu",
__func__, mId,
(long long)mPresentationCompleteFrames, audioHalFrames);
}
bool complete;
if (isOffloaded()) {
complete = true;
} else if (isDirect() || isFastTrack()) { // these do not go through linear map
complete = framesWritten >= (int64_t) mPresentationCompleteFrames;
} else { // Normal tracks, OutputTracks, and PatchTracks
complete = framesWritten >= (int64_t) mPresentationCompleteFrames
&& mAudioTrackServerProxy->isDrained();
}
if (complete) {
triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
mAudioTrackServerProxy->setStreamEndDone();
return true;
}
return false;
}
void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type)
{
for (size_t i = 0; i < mSyncEvents.size();) {
if (mSyncEvents[i]->type() == type) {
mSyncEvents[i]->trigger();
mSyncEvents.removeAt(i);
} else {
++i;
}
}
}
// implement VolumeBufferProvider interface
gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
{
// called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
float vl = float_from_gain(gain_minifloat_unpack_left(vlr));
float vr = float_from_gain(gain_minifloat_unpack_right(vlr));
// track volumes come from shared memory, so can't be trusted and must be clamped
if (vl > GAIN_FLOAT_UNITY) {
vl = GAIN_FLOAT_UNITY;
}
if (vr > GAIN_FLOAT_UNITY) {
vr = GAIN_FLOAT_UNITY;
}
// now apply the cached master volume and stream type volume;
// this is trusted but lacks any synchronization or barrier so may be stale
float v = mCachedVolume;
vl *= v;
vr *= v;
// re-combine into packed minifloat
vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr));
// FIXME look at mute, pause, and stop flags
return vlr;
}
status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event)
{
if (isTerminated() || mState == PAUSED ||
((framesReady() == 0) && ((mSharedBuffer != 0) ||
(mState == STOPPED)))) {
ALOGW("%s(%d): in invalid state %d on session %d %s mode, framesReady %zu",
__func__, mId,
mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady());
event->cancel();
return INVALID_OPERATION;
}
(void) TrackBase::setSyncEvent(event);
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::Track::invalidate()
{
TrackBase::invalidate();
signalClientFlag(CBLK_INVALID);
}
void AudioFlinger::PlaybackThread::Track::disable()
{
// TODO(b/142394888): the filling status should also be reset to filling
signalClientFlag(CBLK_DISABLED);
}
void AudioFlinger::PlaybackThread::Track::signalClientFlag(int32_t flag)
{
// FIXME should use proxy, and needs work
audio_track_cblk_t* cblk = mCblk;
android_atomic_or(flag, &cblk->mFlags);
android_atomic_release_store(0x40000000, &cblk->mFutex);
// client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
(void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
}
void AudioFlinger::PlaybackThread::Track::signal()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
PlaybackThread *t = (PlaybackThread *)thread.get();
Mutex::Autolock _l(t->mLock);
t->broadcast_l();
}
}
//To be called with thread lock held
bool AudioFlinger::PlaybackThread::Track::isResumePending() {
if (mState == RESUMING)
return true;
/* Resume is pending if track was stopping before pause was called */
if (mState == STOPPING_1 &&
mResumeToStopping)
return true;
return false;
}
//To be called with thread lock held
void AudioFlinger::PlaybackThread::Track::resumeAck() {
if (mState == RESUMING)
mState = ACTIVE;
// Other possibility of pending resume is stopping_1 state
// Do not update the state from stopping as this prevents
// drain being called.
if (mState == STOPPING_1) {
mResumeToStopping = false;
}
}
//To be called with thread lock held
void AudioFlinger::PlaybackThread::Track::updateTrackFrameInfo(
int64_t trackFramesReleased, int64_t sinkFramesWritten,
uint32_t halSampleRate, const ExtendedTimestamp &timeStamp) {
// Make the kernel frametime available.
const FrameTime ft{
timeStamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL],
timeStamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]};
// ALOGD("FrameTime: %lld %lld", (long long)ft.frames, (long long)ft.timeNs);
mKernelFrameTime.store(ft);
if (!audio_is_linear_pcm(mFormat)) {
return;
}
//update frame map
mFrameMap.push(trackFramesReleased, sinkFramesWritten);
// adjust server times and set drained state.
//
// Our timestamps are only updated when the track is on the Thread active list.
// We need to ensure that tracks are not removed before full drain.
ExtendedTimestamp local = timeStamp;
bool drained = true; // default assume drained, if no server info found
bool checked = false;
for (int i = ExtendedTimestamp::LOCATION_MAX - 1;
i >= ExtendedTimestamp::LOCATION_SERVER; --i) {
// Lookup the track frame corresponding to the sink frame position.
if (local.mTimeNs[i] > 0) {
local.mPosition[i] = mFrameMap.findX(local.mPosition[i]);
// check drain state from the latest stage in the pipeline.
if (!checked && i <= ExtendedTimestamp::LOCATION_KERNEL) {
drained = local.mPosition[i] >= mAudioTrackServerProxy->framesReleased();
checked = true;
}
}
}
mAudioTrackServerProxy->setDrained(drained);
// Set correction for flushed frames that are not accounted for in released.
local.mFlushed = mAudioTrackServerProxy->framesFlushed();
mServerProxy->setTimestamp(local);
// Compute latency info.
const bool useTrackTimestamp = !drained;
const double latencyMs = useTrackTimestamp
? local.getOutputServerLatencyMs(sampleRate())
: timeStamp.getOutputServerLatencyMs(halSampleRate);
mServerLatencyFromTrack.store(useTrackTimestamp);
mServerLatencyMs.store(latencyMs);
}
binder::Status AudioFlinger::PlaybackThread::Track::AudioVibrationController::mute(
/*out*/ bool *ret) {
*ret = false;
sp<ThreadBase> thread = mTrack->mThread.promote();
if (thread != 0) {
// Lock for updating mHapticPlaybackEnabled.
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if ((mTrack->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
&& playbackThread->mHapticChannelCount > 0) {
mTrack->setHapticPlaybackEnabled(false);
*ret = true;
}
}
return binder::Status::ok();
}
binder::Status AudioFlinger::PlaybackThread::Track::AudioVibrationController::unmute(
/*out*/ bool *ret) {
*ret = false;
sp<ThreadBase> thread = mTrack->mThread.promote();
if (thread != 0) {
// Lock for updating mHapticPlaybackEnabled.
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if ((mTrack->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
&& playbackThread->mHapticChannelCount > 0) {
mTrack->setHapticPlaybackEnabled(true);
*ret = true;
}
}
return binder::Status::ok();
}
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::OutputTrack"
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
PlaybackThread *playbackThread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
uid_t uid)
: Track(playbackThread, NULL, AUDIO_STREAM_PATCH,
audio_attributes_t{} /* currently unused for output track */,
sampleRate, format, channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, nullptr /* sharedBuffer */,
AUDIO_SESSION_NONE, getpid(), uid, AUDIO_OUTPUT_FLAG_NONE,
TYPE_OUTPUT),
mActive(false), mSourceThread(sourceThread)
{
if (mCblk != NULL) {
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("%s(): mCblk %p, mBuffer %p, "
"frameCount %zu, mChannelMask 0x%08x",
__func__, mCblk, mBuffer,
frameCount, mChannelMask);
// since client and server are in the same process,
// the buffer has the same virtual address on both sides
mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
true /*clientInServer*/);
mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
mClientProxy->setSendLevel(0.0);
mClientProxy->setSampleRate(sampleRate);
} else {
ALOGW("%s(%d): Error creating output track on thread %d",
__func__, mId, (int)mThreadIoHandle);
}
}
AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
{
clearBufferQueue();
// superclass destructor will now delete the server proxy and shared memory both refer to
}
status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event,
audio_session_t triggerSession)
{
status_t status = Track::start(event, triggerSession);
if (status != NO_ERROR) {
return status;
}
mActive = true;
mRetryCount = 127;
return status;
}
void AudioFlinger::PlaybackThread::OutputTrack::stop()
{
Track::stop();
clearBufferQueue();
mOutBuffer.frameCount = 0;
mActive = false;
}
ssize_t AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.raw = data;
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
if (!mActive && frames != 0) {
(void) start();
}
while (waitTimeLeftMs) {
// First write pending buffers, then new data
if (mBufferQueue.size()) {
pInBuffer = mBufferQueue.itemAt(0);
} else {
pInBuffer = &inBuffer;
}
if (pInBuffer->frameCount == 0) {
break;
}
if (mOutBuffer.frameCount == 0) {
mOutBuffer.frameCount = pInBuffer->frameCount;
nsecs_t startTime = systemTime();
status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs);
if (status != NO_ERROR && status != NOT_ENOUGH_DATA) {
ALOGV("%s(%d): thread %d no more output buffers; status %d",
__func__, mId,
(int)mThreadIoHandle, status);
outputBufferFull = true;
break;
}
uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
if (waitTimeLeftMs >= waitTimeMs) {
waitTimeLeftMs -= waitTimeMs;
} else {
waitTimeLeftMs = 0;
}
if (status == NOT_ENOUGH_DATA) {
restartIfDisabled();
continue;
}
}
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize);
Proxy::Buffer buf;
buf.mFrameCount = outFrames;
buf.mRaw = NULL;
mClientProxy->releaseBuffer(&buf);
restartIfDisabled();
pInBuffer->frameCount -= outFrames;
pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize;
mOutBuffer.frameCount -= outFrames;
mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize;
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
free(pInBuffer->mBuffer);
if (pInBuffer != &inBuffer) {
delete pInBuffer;
}
ALOGV("%s(%d): thread %d released overflow buffer %zu",
__func__, mId,
(int)mThreadIoHandle, mBufferQueue.size());
} else {
break;
}
}
}
// If we could not write all frames, allocate a buffer and queue it for next time.
if (inBuffer.frameCount) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0 && !thread->standby()) {
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
pInBuffer = new Buffer;
pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize);
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->raw = pInBuffer->mBuffer;
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize);
mBufferQueue.add(pInBuffer);
ALOGV("%s(%d): thread %d adding overflow buffer %zu", __func__, mId,
(int)mThreadIoHandle, mBufferQueue.size());
// audio data is consumed (stored locally); set frameCount to 0.
inBuffer.frameCount = 0;
} else {
ALOGW("%s(%d): thread %d no more overflow buffers",
__func__, mId, (int)mThreadIoHandle);
// TODO: return error for this.
}
}
}
// Calling write() with a 0 length buffer means that no more data will be written:
// We rely on stop() to set the appropriate flags to allow the remaining frames to play out.
if (frames == 0 && mBufferQueue.size() == 0 && mActive) {
stop();
}
return frames - inBuffer.frameCount; // number of frames consumed.
}
void AudioFlinger::PlaybackThread::OutputTrack::copyMetadataTo(MetadataInserter& backInserter) const
{
std::lock_guard<std::mutex> lock(mTrackMetadatasMutex);
backInserter = std::copy(mTrackMetadatas.begin(), mTrackMetadatas.end(), backInserter);
}
void AudioFlinger::PlaybackThread::OutputTrack::setMetadatas(const SourceMetadatas& metadatas) {
{
std::lock_guard<std::mutex> lock(mTrackMetadatasMutex);
mTrackMetadatas = metadatas;
}
// No need to adjust metadata track volumes as OutputTrack volumes are always 0dBFS.
setMetadataHasChanged();
}
status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
{
ClientProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
struct timespec timeout;
timeout.tv_sec = waitTimeMs / 1000;
timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000;
status_t status = mClientProxy->obtainBuffer(&buf, &timeout);
buffer->frameCount = buf.mFrameCount;
buffer->raw = buf.mRaw;
return status;
}
void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
for (size_t i = 0; i < size; i++) {
Buffer *pBuffer = mBufferQueue.itemAt(i);
free(pBuffer->mBuffer);
delete pBuffer;
}
mBufferQueue.clear();
}
void AudioFlinger::PlaybackThread::OutputTrack::restartIfDisabled()
{
int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
if (mActive && (flags & CBLK_DISABLED)) {
start();
}
}
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::PatchTrack"
AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_output_flags_t flags,
const Timeout& timeout,
size_t frameCountToBeReady)
: Track(playbackThread, NULL, streamType,
audio_attributes_t{} /* currently unused for patch track */,
sampleRate, format, channelMask, frameCount,
buffer, bufferSize, nullptr /* sharedBuffer */,
AUDIO_SESSION_NONE, getpid(), AID_AUDIOSERVER, flags, TYPE_PATCH,
AUDIO_PORT_HANDLE_NONE, frameCountToBeReady),
PatchTrackBase(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true),
*playbackThread, timeout)
{
ALOGV("%s(%d): sampleRate %d mPeerTimeout %d.%03d sec",
__func__, mId, sampleRate,
(int)mPeerTimeout.tv_sec,
(int)(mPeerTimeout.tv_nsec / 1000000));
}
AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack()
{
ALOGV("%s(%d)", __func__, mId);
}
status_t AudioFlinger::PlaybackThread::PatchTrack::start(AudioSystem::sync_event_t event,
audio_session_t triggerSession)
{
status_t status = Track::start(event, triggerSession);
if (status != NO_ERROR) {
return status;
}
android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags);
return status;
}
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer(
AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
Proxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
ALOGV_IF(status != NO_ERROR, "%s(%d): getNextBuffer status %d", __func__, mId, status);
buffer->frameCount = buf.mFrameCount;
if (buf.mFrameCount == 0) {
return WOULD_BLOCK;
}
status = Track::getNextBuffer(buffer);
return status;
}
void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
Proxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
buf.mRaw = buffer->raw;
mPeerProxy->releaseBuffer(&buf);
TrackBase::releaseBuffer(buffer);
}
status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer,
const struct timespec *timeOut)
{
status_t status = NO_ERROR;
static const int32_t kMaxTries = 5;
int32_t tryCounter = kMaxTries;
const size_t originalFrameCount = buffer->mFrameCount;
do {
if (status == NOT_ENOUGH_DATA) {
restartIfDisabled();
buffer->mFrameCount = originalFrameCount; // cleared on error, must be restored.
}
status = mProxy->obtainBuffer(buffer, timeOut);
} while ((status == NOT_ENOUGH_DATA) && (tryCounter-- > 0));
return status;
}
void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer)
{
mProxy->releaseBuffer(buffer);
restartIfDisabled();
}
void AudioFlinger::PlaybackThread::PatchTrack::restartIfDisabled()
{
if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) {
ALOGW("%s(%d): disabled due to previous underrun, restarting", __func__, mId);
start();
}
}
// ----------------------------------------------------------------------------
// Record
// ----------------------------------------------------------------------------
// ----------------------------------------------------------------------------
// AppOp for audio recording
// -------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::OpRecordAudioMonitor"
// static
sp<AudioFlinger::RecordThread::OpRecordAudioMonitor>
AudioFlinger::RecordThread::OpRecordAudioMonitor::createIfNeeded(
uid_t uid, const String16& opPackageName)
{
if (isServiceUid(uid)) {
ALOGV("not silencing record for service uid:%d pack:%s",
uid, String8(opPackageName).string());
return nullptr;
}
if (opPackageName.size() == 0) {
Vector<String16> packages;
// no package name, happens with SL ES clients
// query package manager to find one
PermissionController permissionController;
permissionController.getPackagesForUid(uid, packages);
if (packages.isEmpty()) {
return nullptr;
} else {
ALOGV("using pack:%s for uid:%d", String8(packages[0]).string(), uid);
return new OpRecordAudioMonitor(uid, packages[0]);
}
}
return new OpRecordAudioMonitor(uid, opPackageName);
}
AudioFlinger::RecordThread::OpRecordAudioMonitor::OpRecordAudioMonitor(
uid_t uid, const String16& opPackageName)
: mHasOpRecordAudio(true), mUid(uid), mPackage(opPackageName)
{
}
AudioFlinger::RecordThread::OpRecordAudioMonitor::~OpRecordAudioMonitor()
{
if (mOpCallback != 0) {
mAppOpsManager.stopWatchingMode(mOpCallback);
}
mOpCallback.clear();
}
void AudioFlinger::RecordThread::OpRecordAudioMonitor::onFirstRef()
{
checkRecordAudio();
mOpCallback = new RecordAudioOpCallback(this);
ALOGV("start watching OP_RECORD_AUDIO for pack:%s", String8(mPackage).string());
mAppOpsManager.startWatchingMode(AppOpsManager::OP_RECORD_AUDIO, mPackage, mOpCallback);
}
bool AudioFlinger::RecordThread::OpRecordAudioMonitor::hasOpRecordAudio() const {
return mHasOpRecordAudio.load();
}
// Called by RecordAudioOpCallback when OP_RECORD_AUDIO is updated in AppOp callback
// and in onFirstRef()
// Note this method is never called (and never to be) for audio server / root track
// due to the UID in createIfNeeded(). As a result for those record track, it's:
// - not called from constructor,
// - not called from RecordAudioOpCallback because the callback is not installed in this case
void AudioFlinger::RecordThread::OpRecordAudioMonitor::checkRecordAudio()
{
const int32_t mode = mAppOpsManager.checkOp(AppOpsManager::OP_RECORD_AUDIO,
mUid, mPackage);
const bool hasIt = (mode == AppOpsManager::MODE_ALLOWED);
// verbose logging only log when appOp changed
ALOGI_IF(hasIt != mHasOpRecordAudio.load(),
"OP_RECORD_AUDIO missing, %ssilencing record uid%d pack:%s",
hasIt ? "un" : "", mUid, String8(mPackage).string());
mHasOpRecordAudio.store(hasIt);
}
AudioFlinger::RecordThread::OpRecordAudioMonitor::RecordAudioOpCallback::RecordAudioOpCallback(
const wp<OpRecordAudioMonitor>& monitor) : mMonitor(monitor)
{ }
void AudioFlinger::RecordThread::OpRecordAudioMonitor::RecordAudioOpCallback::opChanged(int32_t op,
const String16& packageName) {
UNUSED(packageName);
if (op != AppOpsManager::OP_RECORD_AUDIO) {
return;
}
sp<OpRecordAudioMonitor> monitor = mMonitor.promote();
if (monitor != NULL) {
monitor->checkRecordAudio();
}
}
#undef LOG_TAG
#define LOG_TAG "AF::RecordHandle"
AudioFlinger::RecordHandle::RecordHandle(
const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
}
AudioFlinger::RecordHandle::~RecordHandle() {
stop_nonvirtual();
mRecordTrack->destroy();
}
binder::Status AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
int /*audio_session_t*/ triggerSession) {
ALOGV("%s()", __func__);
return binder::Status::fromStatusT(
mRecordTrack->start((AudioSystem::sync_event_t)event, (audio_session_t) triggerSession));
}
binder::Status AudioFlinger::RecordHandle::stop() {
stop_nonvirtual();
return binder::Status::ok();
}
void AudioFlinger::RecordHandle::stop_nonvirtual() {
ALOGV("%s()", __func__);
mRecordTrack->stop();
}
binder::Status AudioFlinger::RecordHandle::getActiveMicrophones(
std::vector<media::MicrophoneInfo>* activeMicrophones) {
ALOGV("%s()", __func__);
return binder::Status::fromStatusT(
mRecordTrack->getActiveMicrophones(activeMicrophones));
}
binder::Status AudioFlinger::RecordHandle::setPreferredMicrophoneDirection(
int /*audio_microphone_direction_t*/ direction) {
ALOGV("%s()", __func__);
return binder::Status::fromStatusT(mRecordTrack->setPreferredMicrophoneDirection(
static_cast<audio_microphone_direction_t>(direction)));
}
binder::Status AudioFlinger::RecordHandle::setPreferredMicrophoneFieldDimension(float zoom) {
ALOGV("%s()", __func__);
return binder::Status::fromStatusT(mRecordTrack->setPreferredMicrophoneFieldDimension(zoom));
}
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::RecordTrack"
// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::RecordThread::RecordTrack::RecordTrack(
RecordThread *thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_session_t sessionId,
pid_t creatorPid,
uid_t uid,
audio_input_flags_t flags,
track_type type,
const String16& opPackageName,
audio_port_handle_t portId)
: TrackBase(thread, client, attr, sampleRate, format,
channelMask, frameCount, buffer, bufferSize, sessionId,
creatorPid, uid, false /*isOut*/,
(type == TYPE_DEFAULT) ?
((flags & AUDIO_INPUT_FLAG_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE),
type, portId),
mOverflow(false),
mFramesToDrop(0),
mResamplerBufferProvider(NULL), // initialize in case of early constructor exit
mRecordBufferConverter(NULL),
mFlags(flags),
mSilenced(false),
mOpRecordAudioMonitor(OpRecordAudioMonitor::createIfNeeded(uid, opPackageName))
{
if (mCblk == NULL) {
return;
}
if (!isDirect()) {
mRecordBufferConverter = new RecordBufferConverter(
thread->mChannelMask, thread->mFormat, thread->mSampleRate,
channelMask, format, sampleRate);
// Check if the RecordBufferConverter construction was successful.
// If not, don't continue with construction.
//
// NOTE: It would be extremely rare that the record track cannot be created
// for the current device, but a pending or future device change would make
// the record track configuration valid.
if (mRecordBufferConverter->initCheck() != NO_ERROR) {
ALOGE("%s(%d): RecordTrack unable to create record buffer converter", __func__, mId);
return;
}
}
mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
mFrameSize, !isExternalTrack());
mResamplerBufferProvider = new ResamplerBufferProvider(this);
if (flags & AUDIO_INPUT_FLAG_FAST) {
ALOG_ASSERT(thread->mFastTrackAvail);
thread->mFastTrackAvail = false;
} else {
// TODO: only Normal Record has timestamps (Fast Record does not).
mServerLatencySupported = checkServerLatencySupported(mFormat, flags);
}
#ifdef TEE_SINK
mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
+ "_" + std::to_string(mId)
+ "_R");
#endif
}
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
{
ALOGV("%s()", __func__);
delete mRecordBufferConverter;
delete mResamplerBufferProvider;
}
status_t AudioFlinger::RecordThread::RecordTrack::initCheck() const
{
status_t status = TrackBase::initCheck();
if (status == NO_ERROR && mServerProxy == 0) {
status = BAD_VALUE;
}
return status;
}
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
ServerProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
status_t status = mServerProxy->obtainBuffer(&buf);
buffer->frameCount = buf.mFrameCount;
buffer->raw = buf.mRaw;
if (buf.mFrameCount == 0) {
// FIXME also wake futex so that overrun is noticed more quickly
(void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags);
}
return status;
}
status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
audio_session_t triggerSession)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
return recordThread->start(this, event, triggerSession);
} else {
return BAD_VALUE;
}
}
void AudioFlinger::RecordThread::RecordTrack::stop()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
if (recordThread->stop(this) && isExternalTrack()) {
AudioSystem::stopInput(mPortId);
}
}
}
void AudioFlinger::RecordThread::RecordTrack::destroy()
{
// see comments at AudioFlinger::PlaybackThread::Track::destroy()
sp<RecordTrack> keep(this);
{
track_state priorState = mState;
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
RecordThread *recordThread = (RecordThread *) thread.get();
priorState = mState;
recordThread->destroyTrack_l(this); // move mState to STOPPED, terminate
}
// APM portid/client management done outside of lock.
// NOTE: if thread doesn't exist, the input descriptor probably doesn't either.
if (isExternalTrack()) {
switch (priorState) {
case ACTIVE: // invalidated while still active
case STARTING_2: // invalidated/start-aborted after startInput successfully called
case PAUSING: // invalidated while in the middle of stop() pausing (still active)
AudioSystem::stopInput(mPortId);
break;
case STARTING_1: // invalidated/start-aborted and startInput not successful
case PAUSED: // OK, not active
case IDLE: // OK, not active
break;
case STOPPED: // unexpected (destroyed)
default:
LOG_ALWAYS_FATAL("%s(%d): invalid prior state: %d", __func__, mId, priorState);
}
AudioSystem::releaseInput(mPortId);
}
}
}
void AudioFlinger::RecordThread::RecordTrack::invalidate()
{
TrackBase::invalidate();
// FIXME should use proxy, and needs work
audio_track_cblk_t* cblk = mCblk;
android_atomic_or(CBLK_INVALID, &cblk->mFlags);
android_atomic_release_store(0x40000000, &cblk->mFutex);
// client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
(void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
}
void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
{
result.appendFormat("Active Id Client Session Port Id S Flags "
" Format Chn mask SRate Source "
" Server FrmCnt FrmRdy Sil%s\n",
isServerLatencySupported() ? " Latency" : "");
}
void AudioFlinger::RecordThread::RecordTrack::appendDump(String8& result, bool active)
{
result.appendFormat("%c%5s %6d %6u %7u %7u %2s 0x%03X "
"%08X %08X %6u %6X "
"%08X %6zu %6zu %3c",
isFastTrack() ? 'F' : ' ',
active ? "yes" : "no",
mId,
(mClient == 0) ? getpid() : mClient->pid(),
mSessionId,
mPortId,
getTrackStateString(),
mCblk->mFlags,
mFormat,
mChannelMask,
mSampleRate,
mAttr.source,
mCblk->mServer,
mFrameCount,
mServerProxy->framesReadySafe(),
isSilenced() ? 's' : 'n'
);
if (isServerLatencySupported()) {
double latencyMs;
bool fromTrack;
if (getTrackLatencyMs(&latencyMs, &fromTrack) == OK) {
// Show latency in msec, followed by 't' if from track timestamp (the most accurate)
// or 'k' if estimated from kernel (usually for debugging).
result.appendFormat(" %7.2lf %c", latencyMs, fromTrack ? 't' : 'k');
} else {
result.appendFormat("%10s", mCblk->mServer != 0 ? "unavail" : "new");
}
}
result.append("\n");
}
void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
{
if (event == mSyncStartEvent) {
ssize_t framesToDrop = 0;
sp<ThreadBase> threadBase = mThread.promote();
if (threadBase != 0) {
// TODO: use actual buffer filling status instead of 2 buffers when info is available
// from audio HAL
framesToDrop = threadBase->mFrameCount * 2;
}
mFramesToDrop = framesToDrop;
}
}
void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent()
{
if (mSyncStartEvent != 0) {
mSyncStartEvent->cancel();
mSyncStartEvent.clear();
}
mFramesToDrop = 0;
}
void AudioFlinger::RecordThread::RecordTrack::updateTrackFrameInfo(
int64_t trackFramesReleased, int64_t sourceFramesRead,
uint32_t halSampleRate, const ExtendedTimestamp &timestamp)
{
// Make the kernel frametime available.
const FrameTime ft{
timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL],
timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]};
// ALOGD("FrameTime: %lld %lld", (long long)ft.frames, (long long)ft.timeNs);
mKernelFrameTime.store(ft);
if (!audio_is_linear_pcm(mFormat)) {
return;
}
ExtendedTimestamp local = timestamp;
// Convert HAL frames to server-side track frames at track sample rate.
// We use trackFramesReleased and sourceFramesRead as an anchor point.
for (int i = ExtendedTimestamp::LOCATION_SERVER; i < ExtendedTimestamp::LOCATION_MAX; ++i) {
if (local.mTimeNs[i] != 0) {
const int64_t relativeServerFrames = local.mPosition[i] - sourceFramesRead;
const int64_t relativeTrackFrames = relativeServerFrames
* mSampleRate / halSampleRate; // TODO: potential computation overflow
local.mPosition[i] = relativeTrackFrames + trackFramesReleased;
}
}
mServerProxy->setTimestamp(local);
// Compute latency info.
const bool useTrackTimestamp = true; // use track unless debugging.
const double latencyMs = - (useTrackTimestamp
? local.getOutputServerLatencyMs(sampleRate())
: timestamp.getOutputServerLatencyMs(halSampleRate));
mServerLatencyFromTrack.store(useTrackTimestamp);
mServerLatencyMs.store(latencyMs);
}
bool AudioFlinger::RecordThread::RecordTrack::isSilenced() const {
if (mSilenced) {
return true;
}
// The monitor is only created for record tracks that can be silenced.
return mOpRecordAudioMonitor ? !mOpRecordAudioMonitor->hasOpRecordAudio() : false;
}
status_t AudioFlinger::RecordThread::RecordTrack::getActiveMicrophones(
std::vector<media::MicrophoneInfo>* activeMicrophones)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
return recordThread->getActiveMicrophones(activeMicrophones);
} else {
return BAD_VALUE;
}
}
status_t AudioFlinger::RecordThread::RecordTrack::setPreferredMicrophoneDirection(
audio_microphone_direction_t direction) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
return recordThread->setPreferredMicrophoneDirection(direction);
} else {
return BAD_VALUE;
}
}
status_t AudioFlinger::RecordThread::RecordTrack::setPreferredMicrophoneFieldDimension(float zoom) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
return recordThread->setPreferredMicrophoneFieldDimension(zoom);
} else {
return BAD_VALUE;
}
}
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::PatchRecord"
AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_input_flags_t flags,
const Timeout& timeout)
: RecordTrack(recordThread, NULL,
audio_attributes_t{} /* currently unused for patch track */,
sampleRate, format, channelMask, frameCount,
buffer, bufferSize, AUDIO_SESSION_NONE, getpid(), AID_AUDIOSERVER,
flags, TYPE_PATCH, String16()),
PatchTrackBase(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true),
*recordThread, timeout)
{
ALOGV("%s(%d): sampleRate %d mPeerTimeout %d.%03d sec",
__func__, mId, sampleRate,
(int)mPeerTimeout.tv_sec,
(int)(mPeerTimeout.tv_nsec / 1000000));
}
AudioFlinger::RecordThread::PatchRecord::~PatchRecord()
{
ALOGV("%s(%d)", __func__, mId);
}
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer(
AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
Proxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
ALOGV_IF(status != NO_ERROR,
"%s(%d): mPeerProxy->obtainBuffer status %d", __func__, mId, status);
buffer->frameCount = buf.mFrameCount;
if (buf.mFrameCount == 0) {
return WOULD_BLOCK;
}
status = RecordTrack::getNextBuffer(buffer);
return status;
}
void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
Proxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
buf.mRaw = buffer->raw;
mPeerProxy->releaseBuffer(&buf);
TrackBase::releaseBuffer(buffer);
}
status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer,
const struct timespec *timeOut)
{
return mProxy->obtainBuffer(buffer, timeOut);
}
void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer)
{
mProxy->releaseBuffer(buffer);
}
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::MmapTrack"
AudioFlinger::MmapThread::MmapTrack::MmapTrack(ThreadBase *thread,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_session_t sessionId,
bool isOut,
uid_t uid,
pid_t pid,
pid_t creatorPid,
audio_port_handle_t portId)
: TrackBase(thread, NULL, attr, sampleRate, format,
channelMask, (size_t)0 /* frameCount */,
nullptr /* buffer */, (size_t)0 /* bufferSize */,
sessionId, creatorPid, uid, isOut,
ALLOC_NONE,
TYPE_DEFAULT, portId),
mPid(pid), mSilenced(false), mSilencedNotified(false)
{
}
AudioFlinger::MmapThread::MmapTrack::~MmapTrack()
{
}
status_t AudioFlinger::MmapThread::MmapTrack::initCheck() const
{
return NO_ERROR;
}
status_t AudioFlinger::MmapThread::MmapTrack::start(AudioSystem::sync_event_t event __unused,
audio_session_t triggerSession __unused)
{
return NO_ERROR;
}
void AudioFlinger::MmapThread::MmapTrack::stop()
{
}
// AudioBufferProvider interface
status_t AudioFlinger::MmapThread::MmapTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->frameCount = 0;
buffer->raw = nullptr;
return INVALID_OPERATION;
}
// ExtendedAudioBufferProvider interface
size_t AudioFlinger::MmapThread::MmapTrack::framesReady() const {
return 0;
}
int64_t AudioFlinger::MmapThread::MmapTrack::framesReleased() const
{
return 0;
}
void AudioFlinger::MmapThread::MmapTrack::onTimestamp(const ExtendedTimestamp &timestamp __unused)
{
}
void AudioFlinger::MmapThread::MmapTrack::appendDumpHeader(String8& result)
{
result.appendFormat("Client Session Port Id Format Chn mask SRate Flags %s\n",
isOut() ? "Usg CT": "Source");
}
void AudioFlinger::MmapThread::MmapTrack::appendDump(String8& result, bool active __unused)
{
result.appendFormat("%6u %7u %7u %08X %08X %6u 0x%03X ",
mPid,
mSessionId,
mPortId,
mFormat,
mChannelMask,
mSampleRate,
mAttr.flags);
if (isOut()) {
result.appendFormat("%3x %2x", mAttr.usage, mAttr.content_type);
} else {
result.appendFormat("%6x", mAttr.source);
}
result.append("\n");
}
} // namespace android