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fuchsia / third_party / android / platform / frameworks / av / b0de569edd2d27b8dae9e3048f9c131b6ebc2d45 / . / services / audioflinger / Threads.h
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/*
**
** Copyright 2012, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#ifndef INCLUDING_FROM_AUDIOFLINGER_H
#error This header file should only be included from AudioFlinger.h
#endif
class ThreadBase : public Thread {
public:
#include "TrackBase.h"
enum type_t {
MIXER, // Thread class is MixerThread
DIRECT, // Thread class is DirectOutputThread
DUPLICATING, // Thread class is DuplicatingThread
RECORD, // Thread class is RecordThread
OFFLOAD, // Thread class is OffloadThread
MMAP // control thread for MMAP stream
// If you add any values here, also update ThreadBase::threadTypeToString()
};
static const char *threadTypeToString(type_t type);
ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
audio_devices_t outDevice, audio_devices_t inDevice, type_t type,
bool systemReady);
virtual ~ThreadBase();
virtual status_t readyToRun();
void clearPowerManager();
// base for record and playback
enum {
CFG_EVENT_IO,
CFG_EVENT_PRIO,
CFG_EVENT_SET_PARAMETER,
CFG_EVENT_CREATE_AUDIO_PATCH,
CFG_EVENT_RELEASE_AUDIO_PATCH,
};
class ConfigEventData: public RefBase {
public:
virtual ~ConfigEventData() {}
virtual void dump(char *buffer, size_t size) = 0;
protected:
ConfigEventData() {}
};
// Config event sequence by client if status needed (e.g binder thread calling setParameters()):
// 1. create SetParameterConfigEvent. This sets mWaitStatus in config event
// 2. Lock mLock
// 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal
// 4. sendConfigEvent_l() reads status from event->mStatus;
// 5. sendConfigEvent_l() returns status
// 6. Unlock
//
// Parameter sequence by server: threadLoop calling processConfigEvents_l():
// 1. Lock mLock
// 2. If there is an entry in mConfigEvents proceed ...
// 3. Read first entry in mConfigEvents
// 4. Remove first entry from mConfigEvents
// 5. Process
// 6. Set event->mStatus
// 7. event->mCond.signal
// 8. Unlock
class ConfigEvent: public RefBase {
public:
virtual ~ConfigEvent() {}
void dump(char *buffer, size_t size) { mData->dump(buffer, size); }
const int mType; // event type e.g. CFG_EVENT_IO
Mutex mLock; // mutex associated with mCond
Condition mCond; // condition for status return
status_t mStatus; // status communicated to sender
bool mWaitStatus; // true if sender is waiting for status
bool mRequiresSystemReady; // true if must wait for system ready to enter event queue
sp<ConfigEventData> mData; // event specific parameter data
protected:
explicit ConfigEvent(int type, bool requiresSystemReady = false) :
mType(type), mStatus(NO_ERROR), mWaitStatus(false),
mRequiresSystemReady(requiresSystemReady), mData(NULL) {}
};
class IoConfigEventData : public ConfigEventData {
public:
IoConfigEventData(audio_io_config_event event, pid_t pid,
audio_port_handle_t portId) :
mEvent(event), mPid(pid), mPortId(portId) {}
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "IO event: event %d\n", mEvent);
}
const audio_io_config_event mEvent;
const pid_t mPid;
const audio_port_handle_t mPortId;
};
class IoConfigEvent : public ConfigEvent {
public:
IoConfigEvent(audio_io_config_event event, pid_t pid, audio_port_handle_t portId) :
ConfigEvent(CFG_EVENT_IO) {
mData = new IoConfigEventData(event, pid, portId);
}
virtual ~IoConfigEvent() {}
};
class PrioConfigEventData : public ConfigEventData {
public:
PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio, bool forApp) :
mPid(pid), mTid(tid), mPrio(prio), mForApp(forApp) {}
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d, for app? %d\n",
mPid, mTid, mPrio, mForApp);
}
const pid_t mPid;
const pid_t mTid;
const int32_t mPrio;
const bool mForApp;
};
class PrioConfigEvent : public ConfigEvent {
public:
PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) :
ConfigEvent(CFG_EVENT_PRIO, true) {
mData = new PrioConfigEventData(pid, tid, prio, forApp);
}
virtual ~PrioConfigEvent() {}
};
class SetParameterConfigEventData : public ConfigEventData {
public:
explicit SetParameterConfigEventData(String8 keyValuePairs) :
mKeyValuePairs(keyValuePairs) {}
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string());
}
const String8 mKeyValuePairs;
};
class SetParameterConfigEvent : public ConfigEvent {
public:
explicit SetParameterConfigEvent(String8 keyValuePairs) :
ConfigEvent(CFG_EVENT_SET_PARAMETER) {
mData = new SetParameterConfigEventData(keyValuePairs);
mWaitStatus = true;
}
virtual ~SetParameterConfigEvent() {}
};
class CreateAudioPatchConfigEventData : public ConfigEventData {
public:
CreateAudioPatchConfigEventData(const struct audio_patch patch,
audio_patch_handle_t handle) :
mPatch(patch), mHandle(handle) {}
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "Patch handle: %u\n", mHandle);
}
const struct audio_patch mPatch;
audio_patch_handle_t mHandle;
};
class CreateAudioPatchConfigEvent : public ConfigEvent {
public:
CreateAudioPatchConfigEvent(const struct audio_patch patch,
audio_patch_handle_t handle) :
ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) {
mData = new CreateAudioPatchConfigEventData(patch, handle);
mWaitStatus = true;
}
virtual ~CreateAudioPatchConfigEvent() {}
};
class ReleaseAudioPatchConfigEventData : public ConfigEventData {
public:
explicit ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) :
mHandle(handle) {}
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "Patch handle: %u\n", mHandle);
}
audio_patch_handle_t mHandle;
};
class ReleaseAudioPatchConfigEvent : public ConfigEvent {
public:
explicit ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) :
ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) {
mData = new ReleaseAudioPatchConfigEventData(handle);
mWaitStatus = true;
}
virtual ~ReleaseAudioPatchConfigEvent() {}
};
class PMDeathRecipient : public IBinder::DeathRecipient {
public:
explicit PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
virtual ~PMDeathRecipient() {}
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
private:
DISALLOW_COPY_AND_ASSIGN(PMDeathRecipient);
wp<ThreadBase> mThread;
};
virtual status_t initCheck() const = 0;
// static externally-visible
type_t type() const { return mType; }
bool isDuplicating() const { return (mType == DUPLICATING); }
audio_io_handle_t id() const { return mId;}
// dynamic externally-visible
uint32_t sampleRate() const { return mSampleRate; }
audio_channel_mask_t channelMask() const { return mChannelMask; }
audio_format_t format() const { return mHALFormat; }
uint32_t channelCount() const { return mChannelCount; }
// Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
// and returns the [normal mix] buffer's frame count.
virtual size_t frameCount() const = 0;
// Return's the HAL's frame count i.e. fast mixer buffer size.
size_t frameCountHAL() const { return mFrameCount; }
size_t frameSize() const { return mFrameSize; }
// Should be "virtual status_t requestExitAndWait()" and override same
// method in Thread, but Thread::requestExitAndWait() is not yet virtual.
void exit();
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status) = 0;
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys) = 0;
virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;
// sendConfigEvent_l() must be called with ThreadBase::mLock held
// Can temporarily release the lock if waiting for a reply from
// processConfigEvents_l().
status_t sendConfigEvent_l(sp<ConfigEvent>& event);
void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp);
void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp);
status_t sendSetParameterConfigEvent_l(const String8& keyValuePair);
status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch,
audio_patch_handle_t *handle);
status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle);
void processConfigEvents_l();
virtual void cacheParameters_l() = 0;
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle) = 0;
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0;
virtual void toAudioPortConfig(struct audio_port_config *config) = 0;
// see note at declaration of mStandby, mOutDevice and mInDevice
bool standby() const { return mStandby; }
audio_devices_t outDevice() const { return mOutDevice; }
audio_devices_t inDevice() const { return mInDevice; }
audio_devices_t getDevice() const { return isOutput() ? mOutDevice : mInDevice; }
virtual bool isOutput() const = 0;
virtual sp<StreamHalInterface> stream() const = 0;
sp<EffectHandle> createEffect_l(
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority,
audio_session_t sessionId,
effect_descriptor_t *desc,
int *enabled,
status_t *status /*non-NULL*/,
bool pinned);
// return values for hasAudioSession (bit field)
enum effect_state {
EFFECT_SESSION = 0x1, // the audio session corresponds to at least one
// effect
TRACK_SESSION = 0x2, // the audio session corresponds to at least one
// track
FAST_SESSION = 0x4 // the audio session corresponds to at least one
// fast track
};
// get effect chain corresponding to session Id.
sp<EffectChain> getEffectChain(audio_session_t sessionId);
// same as getEffectChain() but must be called with ThreadBase mutex locked
sp<EffectChain> getEffectChain_l(audio_session_t sessionId) const;
std::vector<int> getEffectIds_l(audio_session_t sessionId);
// add an effect chain to the chain list (mEffectChains)
virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0;
// remove an effect chain from the chain list (mEffectChains)
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0;
// lock all effect chains Mutexes. Must be called before releasing the
// ThreadBase mutex before processing the mixer and effects. This guarantees the
// integrity of the chains during the process.
// Also sets the parameter 'effectChains' to current value of mEffectChains.
void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains);
// unlock effect chains after process
void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains);
// get a copy of mEffectChains vector
Vector< sp<EffectChain> > getEffectChains_l() const { return mEffectChains; };
// set audio mode to all effect chains
void setMode(audio_mode_t mode);
// get effect module with corresponding ID on specified audio session
sp<AudioFlinger::EffectModule> getEffect(audio_session_t sessionId, int effectId);
sp<AudioFlinger::EffectModule> getEffect_l(audio_session_t sessionId, int effectId);
// add and effect module. Also creates the effect chain is none exists for
// the effects audio session. Only called in a context of moving an effect
// from one thread to another
status_t addEffect_l(const sp< EffectModule>& effect);
// remove and effect module. Also removes the effect chain is this was the last
// effect
void removeEffect_l(const sp< EffectModule>& effect, bool release = false);
// disconnect an effect handle from module and destroy module if last handle
void disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast);
// detach all tracks connected to an auxiliary effect
virtual void detachAuxEffect_l(int effectId __unused) {}
// returns a combination of:
// - EFFECT_SESSION if effects on this audio session exist in one chain
// - TRACK_SESSION if tracks on this audio session exist
// - FAST_SESSION if fast tracks on this audio session exist
virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0;
uint32_t hasAudioSession(audio_session_t sessionId) const {
Mutex::Autolock _l(mLock);
return hasAudioSession_l(sessionId);
}
template <typename T>
uint32_t hasAudioSession_l(audio_session_t sessionId, const T& tracks) const {
uint32_t result = 0;
if (getEffectChain_l(sessionId) != 0) {
result = EFFECT_SESSION;
}
for (size_t i = 0; i < tracks.size(); ++i) {
const sp<TrackBase>& track = tracks[i];
if (sessionId == track->sessionId()
&& !track->isInvalid() // not yet removed from tracks.
&& !track->isTerminated()) {
result |= TRACK_SESSION;
if (track->isFastTrack()) {
result |= FAST_SESSION; // caution, only represents first track.
}
break;
}
}
return result;
}
// the value returned by default implementation is not important as the
// strategy is only meaningful for PlaybackThread which implements this method
virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused)
{ return 0; }
// check if some effects must be suspended/restored when an effect is enabled
// or disabled
void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
bool enabled,
audio_session_t sessionId =
AUDIO_SESSION_OUTPUT_MIX);
void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
bool enabled,
audio_session_t sessionId =
AUDIO_SESSION_OUTPUT_MIX);
virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0;
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0;
// Return a reference to a per-thread heap which can be used to allocate IMemory
// objects that will be read-only to client processes, read/write to mediaserver,
// and shared by all client processes of the thread.
// The heap is per-thread rather than common across all threads, because
// clients can't be trusted not to modify the offset of the IMemory they receive.
// If a thread does not have such a heap, this method returns 0.
virtual sp<MemoryDealer> readOnlyHeap() const { return 0; }
virtual sp<IMemory> pipeMemory() const { return 0; }
void systemReady();
// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc,
audio_session_t sessionId) = 0;
void broadcast_l();
virtual bool isTimestampCorrectionEnabled() const { return false; }
bool isMsdDevice() const { return mIsMsdDevice; }
void dump(int fd, const Vector<String16>& args);
// deliver stats to mediametrics.
void sendStatistics(bool force);
mutable Mutex mLock;
protected:
// entry describing an effect being suspended in mSuspendedSessions keyed vector
class SuspendedSessionDesc : public RefBase {
public:
SuspendedSessionDesc() : mRefCount(0) {}
int mRefCount; // number of active suspend requests
effect_uuid_t mType; // effect type UUID
};
void acquireWakeLock();
virtual void acquireWakeLock_l();
void releaseWakeLock();
void releaseWakeLock_l();
void updateWakeLockUids_l(const SortedVector<uid_t> &uids);
void getPowerManager_l();
// suspend or restore effects of the specified type (or all if type is NULL)
// on a given session. The number of suspend requests is counted and restore
// occurs when all suspend requests are cancelled.
void setEffectSuspended_l(const effect_uuid_t *type,
bool suspend,
audio_session_t sessionId);
// updated mSuspendedSessions when an effect is suspended or restored
void updateSuspendedSessions_l(const effect_uuid_t *type,
bool suspend,
audio_session_t sessionId);
// check if some effects must be suspended when an effect chain is added
void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain);
// sends the metadata of the active tracks to the HAL
virtual void updateMetadata_l() = 0;
String16 getWakeLockTag();
virtual void preExit() { }
virtual void setMasterMono_l(bool mono __unused) { }
virtual bool requireMonoBlend() { return false; }
// called within the threadLoop to obtain timestamp from the HAL.
virtual status_t threadloop_getHalTimestamp_l(
ExtendedTimestamp *timestamp __unused) const {
return INVALID_OPERATION;
}
virtual void dumpInternals_l(int fd __unused, const Vector<String16>& args __unused)
{ }
virtual void dumpTracks_l(int fd __unused, const Vector<String16>& args __unused) { }
friend class AudioFlinger; // for mEffectChains
const type_t mType;
// Used by parameters, config events, addTrack_l, exit
Condition mWaitWorkCV;
const sp<AudioFlinger> mAudioFlinger;
// updated by PlaybackThread::readOutputParameters_l() or
// RecordThread::readInputParameters_l()
uint32_t mSampleRate;
size_t mFrameCount; // output HAL, direct output, record
audio_channel_mask_t mChannelMask;
uint32_t mChannelCount;
size_t mFrameSize;
// not HAL frame size, this is for output sink (to pipe to fast mixer)
audio_format_t mFormat; // Source format for Recording and
// Sink format for Playback.
// Sink format may be different than
// HAL format if Fastmixer is used.
audio_format_t mHALFormat;
size_t mBufferSize; // HAL buffer size for read() or write()
Vector< sp<ConfigEvent> > mConfigEvents;
Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready
// These fields are written and read by thread itself without lock or barrier,
// and read by other threads without lock or barrier via standby(), outDevice()
// and inDevice().
// Because of the absence of a lock or barrier, any other thread that reads
// these fields must use the information in isolation, or be prepared to deal
// with possibility that it might be inconsistent with other information.
bool mStandby; // Whether thread is currently in standby.
audio_devices_t mOutDevice; // output device
audio_devices_t mInDevice; // input device
audio_devices_t mPrevOutDevice; // previous output device
audio_devices_t mPrevInDevice; // previous input device
struct audio_patch mPatch;
/**
* @brief mDeviceId current device port unique identifier
*/
audio_port_handle_t mDeviceId = AUDIO_PORT_HANDLE_NONE;
audio_source_t mAudioSource;
const audio_io_handle_t mId;
Vector< sp<EffectChain> > mEffectChains;
static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit
char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated
sp<IPowerManager> mPowerManager;
sp<IBinder> mWakeLockToken;
const sp<PMDeathRecipient> mDeathRecipient;
// list of suspended effects per session and per type. The first (outer) vector is
// keyed by session ID, the second (inner) by type UUID timeLow field
// Updated by updateSuspendedSessions_l() only.
KeyedVector< audio_session_t, KeyedVector< int, sp<SuspendedSessionDesc> > >
mSuspendedSessions;
// TODO: add comment and adjust size as needed
static const size_t kLogSize = 4 * 1024;
sp<NBLog::Writer> mNBLogWriter;
bool mSystemReady;
ExtendedTimestamp mTimestamp;
TimestampVerifier< // For timestamp statistics.
int64_t /* frame count */, int64_t /* time ns */> mTimestampVerifier;
audio_devices_t mTimestampCorrectedDevices = AUDIO_DEVICE_NONE;
// ThreadLoop statistics per iteration.
int64_t mLastIoBeginNs = -1;
int64_t mLastIoEndNs = -1;
// This should be read under ThreadBase lock (if not on the threadLoop thread).
audio_utils::Statistics<double> mIoJitterMs{0.995 /* alpha */};
audio_utils::Statistics<double> mProcessTimeMs{0.995 /* alpha */};
audio_utils::Statistics<double> mLatencyMs{0.995 /* alpha */};
// Save the last count when we delivered statistics to mediametrics.
int64_t mLastRecordedTimestampVerifierN = 0;
int64_t mLastRecordedTimeNs = 0; // BOOTTIME to include suspend.
bool mIsMsdDevice = false;
// A condition that must be evaluated by the thread loop has changed and
// we must not wait for async write callback in the thread loop before evaluating it
bool mSignalPending;
#ifdef TEE_SINK
NBAIO_Tee mTee;
#endif
// ActiveTracks is a sorted vector of track type T representing the
// active tracks of threadLoop() to be considered by the locked prepare portion.
// ActiveTracks should be accessed with the ThreadBase lock held.
//
// During processing and I/O, the threadLoop does not hold the lock;
// hence it does not directly use ActiveTracks. Care should be taken
// to hold local strong references or defer removal of tracks
// if the threadLoop may still be accessing those tracks due to mix, etc.
//
// This class updates power information appropriately.
//
template <typename T>
class ActiveTracks {
public:
explicit ActiveTracks(SimpleLog *localLog = nullptr)
: mActiveTracksGeneration(0)
, mLastActiveTracksGeneration(0)
, mLocalLog(localLog)
{ }
~ActiveTracks() {
ALOGW_IF(!mActiveTracks.isEmpty(),
"ActiveTracks should be empty in destructor");
}
// returns the last track added (even though it may have been
// subsequently removed from ActiveTracks).
//
// Used for DirectOutputThread to ensure a flush is called when transitioning
// to a new track (even though it may be on the same session).
// Used for OffloadThread to ensure that volume and mixer state is
// taken from the latest track added.
//
// The latest track is saved with a weak pointer to prevent keeping an
// otherwise useless track alive. Thus the function will return nullptr
// if the latest track has subsequently been removed and destroyed.
sp<T> getLatest() {
return mLatestActiveTrack.promote();
}
// SortedVector methods
ssize_t add(const sp<T> &track);
ssize_t remove(const sp<T> &track);
size_t size() const {
return mActiveTracks.size();
}
bool isEmpty() const {
return mActiveTracks.isEmpty();
}
ssize_t indexOf(const sp<T>& item) {
return mActiveTracks.indexOf(item);
}
sp<T> operator[](size_t index) const {
return mActiveTracks[index];
}
typename SortedVector<sp<T>>::iterator begin() {
return mActiveTracks.begin();
}
typename SortedVector<sp<T>>::iterator end() {
return mActiveTracks.end();
}
// Due to Binder recursion optimization, clear() and updatePowerState()
// cannot be called from a Binder thread because they may call back into
// the original calling process (system server) for BatteryNotifier
// (which requires a Java environment that may not be present).
// Hence, call clear() and updatePowerState() only from the
// ThreadBase thread.
void clear();
// periodically called in the threadLoop() to update power state uids.
void updatePowerState(sp<ThreadBase> thread, bool force = false);
/** @return true if one or move active tracks was added or removed since the
* last time this function was called or the vector was created. */
bool readAndClearHasChanged();
private:
void logTrack(const char *funcName, const sp<T> &track) const;
SortedVector<uid_t> getWakeLockUids() {
SortedVector<uid_t> wakeLockUids;
for (const sp<T> &track : mActiveTracks) {
wakeLockUids.add(track->uid());
}
return wakeLockUids; // moved by underlying SharedBuffer
}
std::map<uid_t, std::pair<ssize_t /* previous */, ssize_t /* current */>>
mBatteryCounter;
SortedVector<sp<T>> mActiveTracks;
int mActiveTracksGeneration;
int mLastActiveTracksGeneration;
wp<T> mLatestActiveTrack; // latest track added to ActiveTracks
SimpleLog * const mLocalLog;
// If the vector has changed since last call to readAndClearHasChanged
bool mHasChanged = false;
};
SimpleLog mLocalLog;
private:
void dumpBase_l(int fd, const Vector<String16>& args);
void dumpEffectChains_l(int fd, const Vector<String16>& args);
};
class VolumeInterface {
public:
virtual ~VolumeInterface() {}
virtual void setMasterVolume(float value) = 0;
virtual void setMasterMute(bool muted) = 0;
virtual void setStreamVolume(audio_stream_type_t stream, float value) = 0;
virtual void setStreamMute(audio_stream_type_t stream, bool muted) = 0;
virtual float streamVolume(audio_stream_type_t stream) const = 0;
};
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback,
public VolumeInterface {
public:
#include "PlaybackTracks.h"
enum mixer_state {
MIXER_IDLE, // no active tracks
MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready
MIXER_TRACKS_READY, // at least one active track, and at least one track has data
MIXER_DRAIN_TRACK, // drain currently playing track
MIXER_DRAIN_ALL, // fully drain the hardware
// standby mode does not have an enum value
// suspend by audio policy manager is orthogonal to mixer state
};
// retry count before removing active track in case of underrun on offloaded thread:
// we need to make sure that AudioTrack client has enough time to send large buffers
//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is
// handled for offloaded tracks
static const int8_t kMaxTrackRetriesOffload = 20;
static const int8_t kMaxTrackStartupRetriesOffload = 100;
static const int8_t kMaxTrackStopRetriesOffload = 2;
static constexpr uint32_t kMaxTracksPerUid = 40;
static constexpr size_t kMaxTracks = 256;
// Maximum delay (in nanoseconds) for upcoming buffers in suspend mode, otherwise
// if delay is greater, the estimated time for timeLoopNextNs is reset.
// This allows for catch-up to be done for small delays, while resetting the estimate
// for initial conditions or large delays.
static const nsecs_t kMaxNextBufferDelayNs = 100000000;
PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady);
virtual ~PlaybackThread();
// Thread virtuals
virtual bool threadLoop();
// RefBase
virtual void onFirstRef();
virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc,
audio_session_t sessionId);
protected:
// Code snippets that were lifted up out of threadLoop()
virtual void threadLoop_mix() = 0;
virtual void threadLoop_sleepTime() = 0;
virtual ssize_t threadLoop_write();
virtual void threadLoop_drain();
virtual void threadLoop_standby();
virtual void threadLoop_exit();
virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
// prepareTracks_l reads and writes mActiveTracks, and returns
// the pending set of tracks to remove via Vector 'tracksToRemove'. The caller
// is responsible for clearing or destroying this Vector later on, when it
// is safe to do so. That will drop the final ref count and destroy the tracks.
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0;
void removeTracks_l(const Vector< sp<Track> >& tracksToRemove);
status_t handleVoipVolume_l(float *volume);
// StreamOutHalInterfaceCallback implementation
virtual void onWriteReady();
virtual void onDrainReady();
virtual void onError();
void resetWriteBlocked(uint32_t sequence);
void resetDraining(uint32_t sequence);
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
virtual void onAddNewTrack_l();
void onAsyncError(); // error reported by AsyncCallbackThread
// ThreadBase virtuals
virtual void preExit();
virtual bool keepWakeLock() const { return true; }
virtual void acquireWakeLock_l() {
ThreadBase::acquireWakeLock_l();
mActiveTracks.updatePowerState(this, true /* force */);
}
void dumpInternals_l(int fd, const Vector<String16>& args) override;
void dumpTracks_l(int fd, const Vector<String16>& args) override;
public:
virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
// return estimated latency in milliseconds, as reported by HAL
uint32_t latency() const;
// same, but lock must already be held
uint32_t latency_l() const;
// VolumeInterface
virtual void setMasterVolume(float value);
virtual void setMasterBalance(float balance);
virtual void setMasterMute(bool muted);
virtual void setStreamVolume(audio_stream_type_t stream, float value);
virtual void setStreamMute(audio_stream_type_t stream, bool muted);
virtual float streamVolume(audio_stream_type_t stream) const;
void setVolumeForOutput_l(float left, float right) const;
sp<Track> createTrack_l(
const sp<AudioFlinger::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 *pFrameCount,
size_t *pNotificationFrameCount,
uint32_t notificationsPerBuffer,
float speed,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
audio_output_flags_t *flags,
pid_t creatorPid,
pid_t tid,
uid_t uid,
status_t *status /*non-NULL*/,
audio_port_handle_t portId);
AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
virtual sp<StreamHalInterface> stream() const;
// a very large number of suspend() will eventually wraparound, but unlikely
void suspend() { (void) android_atomic_inc(&mSuspended); }
void restore()
{
// if restore() is done without suspend(), get back into
// range so that the next suspend() will operate correctly
if (android_atomic_dec(&mSuspended) <= 0) {
android_atomic_release_store(0, &mSuspended);
}
}
bool isSuspended() const
{ return android_atomic_acquire_load(&mSuspended) > 0; }
virtual String8 getParameters(const String8& keys);
virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
// Consider also removing and passing an explicit mMainBuffer initialization
// parameter to AF::PlaybackThread::Track::Track().
effect_buffer_t *sinkBuffer() const {
return reinterpret_cast<effect_buffer_t *>(mSinkBuffer); };
virtual void detachAuxEffect_l(int effectId);
status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track>& track,
int EffectId);
status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track>& track,
int EffectId);
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
uint32_t hasAudioSession_l(audio_session_t sessionId) const override {
return ThreadBase::hasAudioSession_l(sessionId, mTracks);
}
virtual uint32_t getStrategyForSession_l(audio_session_t sessionId);
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const;
// called with AudioFlinger lock held
bool invalidateTracks_l(audio_stream_type_t streamType);
virtual void invalidateTracks(audio_stream_type_t streamType);
virtual size_t frameCount() const { return mNormalFrameCount; }
status_t getTimestamp_l(AudioTimestamp& timestamp);
void addPatchTrack(const sp<PatchTrack>& track);
void deletePatchTrack(const sp<PatchTrack>& track);
virtual void toAudioPortConfig(struct audio_port_config *config);
// Return the asynchronous signal wait time.
virtual int64_t computeWaitTimeNs_l() const { return INT64_MAX; }
virtual bool isOutput() const override { return true; }
// returns true if the track is allowed to be added to the thread.
virtual bool isTrackAllowed_l(
audio_channel_mask_t channelMask __unused,
audio_format_t format __unused,
audio_session_t sessionId __unused,
uid_t uid) const {
return trackCountForUid_l(uid) < PlaybackThread::kMaxTracksPerUid
&& mTracks.size() < PlaybackThread::kMaxTracks;
}
bool isTimestampCorrectionEnabled() const override {
const audio_devices_t device =
mOutDevice & mTimestampCorrectedDevices;
return audio_is_output_devices(device) && popcount(device) > 0;
}
protected:
// updated by readOutputParameters_l()
size_t mNormalFrameCount; // normal mixer and effects
bool mThreadThrottle; // throttle the thread processing
uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads
uint32_t mThreadThrottleEndMs; // notify once per throttling
uint32_t mHalfBufferMs; // half the buffer size in milliseconds
void* mSinkBuffer; // frame size aligned sink buffer
// TODO:
// Rearrange the buffer info into a struct/class with
// clear, copy, construction, destruction methods.
//
// mSinkBuffer also has associated with it:
//
// mSinkBufferSize: Sink Buffer Size
// mFormat: Sink Buffer Format
// Mixer Buffer (mMixerBuffer*)
//
// In the case of floating point or multichannel data, which is not in the
// sink format, it is required to accumulate in a higher precision or greater channel count
// buffer before downmixing or data conversion to the sink buffer.
// Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer.
bool mMixerBufferEnabled;
// Storage, 32 byte aligned (may make this alignment a requirement later).
// Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames.
void* mMixerBuffer;
// Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize.
size_t mMixerBufferSize;
// The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only.
audio_format_t mMixerBufferFormat;
// An internal flag set to true by MixerThread::prepareTracks_l()
// when mMixerBuffer contains valid data after mixing.
bool mMixerBufferValid;
// Effects Buffer (mEffectsBuffer*)
//
// In the case of effects data, which is not in the sink format,
// it is required to accumulate in a different buffer before data conversion
// to the sink buffer.
// Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer.
bool mEffectBufferEnabled;
// Storage, 32 byte aligned (may make this alignment a requirement later).
// Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames.
void* mEffectBuffer;
// Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize.
size_t mEffectBufferSize;
// The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only.
audio_format_t mEffectBufferFormat;
// An internal flag set to true by MixerThread::prepareTracks_l()
// when mEffectsBuffer contains valid data after mixing.
//
// When this is set, all mixer data is routed into the effects buffer
// for any processing (including output processing).
bool mEffectBufferValid;
// suspend count, > 0 means suspended. While suspended, the thread continues to pull from
// tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle
// concurrent use of both of them, so Audio Policy Service suspends one of the threads to
// workaround that restriction.
// 'volatile' means accessed via atomic operations and no lock.
volatile int32_t mSuspended;
int64_t mBytesWritten;
int64_t mFramesWritten; // not reset on standby
int64_t mSuspendedFrames; // not reset on standby
// mHapticChannelMask and mHapticChannelCount will only be valid when the thread support
// haptic playback.
audio_channel_mask_t mHapticChannelMask = AUDIO_CHANNEL_NONE;
uint32_t mHapticChannelCount = 0;
private:
// mMasterMute is in both PlaybackThread and in AudioFlinger. When a
// PlaybackThread needs to find out if master-muted, it checks it's local
// copy rather than the one in AudioFlinger. This optimization saves a lock.
bool mMasterMute;
void setMasterMute_l(bool muted) { mMasterMute = muted; }
protected:
ActiveTracks<Track> mActiveTracks;
// Time to sleep between cycles when:
virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED
virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE
virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us
// No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write()
// No sleep in standby mode; waits on a condition
// Code snippets that are temporarily lifted up out of threadLoop() until the merge
void checkSilentMode_l();
// Non-trivial for DUPLICATING only
virtual void saveOutputTracks() { }
virtual void clearOutputTracks() { }
// Cache various calculated values, at threadLoop() entry and after a parameter change
virtual void cacheParameters_l();
virtual uint32_t correctLatency_l(uint32_t latency) const;
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle);
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle);
bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL)
&& mHwSupportsPause
&& (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); }
uint32_t trackCountForUid_l(uid_t uid) const;
private:
friend class AudioFlinger; // for numerous
DISALLOW_COPY_AND_ASSIGN(PlaybackThread);
status_t addTrack_l(const sp<Track>& track);
bool destroyTrack_l(const sp<Track>& track);
void removeTrack_l(const sp<Track>& track);
void readOutputParameters_l();
void updateMetadata_l() final;
virtual void sendMetadataToBackend_l(const StreamOutHalInterface::SourceMetadata& metadata);
// The Tracks class manages tracks added and removed from the Thread.
template <typename T>
class Tracks {
public:
Tracks(bool saveDeletedTrackIds) :
mSaveDeletedTrackIds(saveDeletedTrackIds) { }
// SortedVector methods
ssize_t add(const sp<T> &track) {
const ssize_t index = mTracks.add(track);
LOG_ALWAYS_FATAL_IF(index < 0, "cannot add track");
return index;
}
ssize_t remove(const sp<T> &track);
size_t size() const {
return mTracks.size();
}
bool isEmpty() const {
return mTracks.isEmpty();
}
ssize_t indexOf(const sp<T> &item) {
return mTracks.indexOf(item);
}
sp<T> operator[](size_t index) const {
return mTracks[index];
}
typename SortedVector<sp<T>>::iterator begin() {
return mTracks.begin();
}
typename SortedVector<sp<T>>::iterator end() {
return mTracks.end();
}
size_t processDeletedTrackIds(std::function<void(int)> f) {
for (const int trackId : mDeletedTrackIds) {
f(trackId);
}
return mDeletedTrackIds.size();
}
void clearDeletedTrackIds() { mDeletedTrackIds.clear(); }
private:
// Tracks pending deletion for MIXER type threads
const bool mSaveDeletedTrackIds; // true to enable tracking
std::set<int> mDeletedTrackIds;
SortedVector<sp<T>> mTracks; // wrapped SortedVector.
};
Tracks<Track> mTracks;
stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
AudioStreamOut *mOutput;
float mMasterVolume;
std::atomic<float> mMasterBalance{};
audio_utils::Balance mBalance;
int mNumWrites;
int mNumDelayedWrites;
bool mInWrite;
// FIXME rename these former local variables of threadLoop to standard "m" names
nsecs_t mStandbyTimeNs;
size_t mSinkBufferSize;
// cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l()
uint32_t mActiveSleepTimeUs;
uint32_t mIdleSleepTimeUs;
uint32_t mSleepTimeUs;
// mixer status returned by prepareTracks_l()
mixer_state mMixerStatus; // current cycle
// previous cycle when in prepareTracks_l()
mixer_state mMixerStatusIgnoringFastTracks;
// FIXME or a separate ready state per track
// FIXME move these declarations into the specific sub-class that needs them
// MIXER only
uint32_t sleepTimeShift;
// same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value
nsecs_t mStandbyDelayNs;
// MIXER only
nsecs_t maxPeriod;
// DUPLICATING only
uint32_t writeFrames;
size_t mBytesRemaining;
size_t mCurrentWriteLength;
bool mUseAsyncWrite;
// mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is
// incremented each time a write(), a flush() or a standby() occurs.
// Bit 0 is set when a write blocks and indicates a callback is expected.
// Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence
// callbacks are ignored.
uint32_t mWriteAckSequence;
// mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is
// incremented each time a drain is requested or a flush() or standby() occurs.
// Bit 0 is set when the drain() command is called at the HAL and indicates a callback is
// expected.
// Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence
// callbacks are ignored.
uint32_t mDrainSequence;
sp<AsyncCallbackThread> mCallbackThread;
private:
// The HAL output sink is treated as non-blocking, but current implementation is blocking
sp<NBAIO_Sink> mOutputSink;
// If a fast mixer is present, the blocking pipe sink, otherwise clear
sp<NBAIO_Sink> mPipeSink;
// The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink
sp<NBAIO_Sink> mNormalSink;
uint32_t mScreenState; // cached copy of gScreenState
// TODO: add comment and adjust size as needed
static const size_t kFastMixerLogSize = 8 * 1024;
sp<NBLog::Writer> mFastMixerNBLogWriter;
// Downstream patch latency, available if mDownstreamLatencyStatMs.getN() > 0.
audio_utils::Statistics<double> mDownstreamLatencyStatMs{0.999};
public:
virtual bool hasFastMixer() const = 0;
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const
{ FastTrackUnderruns dummy; return dummy; }
protected:
// accessed by both binder threads and within threadLoop(), lock on mutex needed
unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available
bool mHwSupportsPause;
bool mHwPaused;
bool mFlushPending;
// volumes last sent to audio HAL with stream->setVolume()
float mLeftVolFloat;
float mRightVolFloat;
};
class MixerThread : public PlaybackThread {
public:
MixerThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
audio_devices_t device,
bool systemReady,
type_t type = MIXER);
virtual ~MixerThread();
// Thread virtuals
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status);
virtual bool isTrackAllowed_l(
audio_channel_mask_t channelMask, audio_format_t format,
audio_session_t sessionId, uid_t uid) const override;
protected:
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual uint32_t idleSleepTimeUs() const;
virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l();
virtual void acquireWakeLock_l() {
PlaybackThread::acquireWakeLock_l();
if (hasFastMixer()) {
mFastMixer->setBoottimeOffset(
mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]);
}
}
void dumpInternals_l(int fd, const Vector<String16>& args) override;
// threadLoop snippets
virtual ssize_t threadLoop_write();
virtual void threadLoop_standby();
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
virtual uint32_t correctLatency_l(uint32_t latency) const;
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle);
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle);
AudioMixer* mAudioMixer; // normal mixer
private:
// one-time initialization, no locks required
sp<FastMixer> mFastMixer; // non-0 if there is also a fast mixer
sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread
// contents are not guaranteed to be consistent, no locks required
FastMixerDumpState mFastMixerDumpState;
#ifdef STATE_QUEUE_DUMP
StateQueueObserverDump mStateQueueObserverDump;
StateQueueMutatorDump mStateQueueMutatorDump;
#endif
AudioWatchdogDump mAudioWatchdogDump;
// accessible only within the threadLoop(), no locks required
// mFastMixer->sq() // for mutating and pushing state
int32_t mFastMixerFutex; // for cold idle
std::atomic_bool mMasterMono;
public:
virtual bool hasFastMixer() const { return mFastMixer != 0; }
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const {
ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks);
return mFastMixerDumpState.mTracks[fastIndex].mUnderruns;
}
status_t threadloop_getHalTimestamp_l(
ExtendedTimestamp *timestamp) const override {
if (mNormalSink.get() != nullptr) {
return mNormalSink->getTimestamp(*timestamp);
}
return INVALID_OPERATION;
}
protected:
virtual void setMasterMono_l(bool mono) {
mMasterMono.store(mono);
if (mFastMixer != nullptr) { /* hasFastMixer() */
mFastMixer->setMasterMono(mMasterMono);
}
}
// the FastMixer performs mono blend if it exists.
// Blending with limiter is not idempotent,
// and blending without limiter is idempotent but inefficient to do twice.
virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); }
void setMasterBalance(float balance) override {
mMasterBalance.store(balance);
if (hasFastMixer()) {
mFastMixer->setMasterBalance(balance);
}
}
};
class DirectOutputThread : public PlaybackThread {
public:
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, audio_devices_t device, bool systemReady)
: DirectOutputThread(audioFlinger, output, id, device, DIRECT, systemReady) { }
virtual ~DirectOutputThread();
status_t selectPresentation(int presentationId, int programId);
// Thread virtuals
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status);
virtual void flushHw_l();
void setMasterBalance(float balance) override;
protected:
virtual uint32_t activeSleepTimeUs() const;
virtual uint32_t idleSleepTimeUs() const;
virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l();
void dumpInternals_l(int fd, const Vector<String16>& args) override;
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
virtual void threadLoop_exit();
virtual bool shouldStandby_l();
virtual void onAddNewTrack_l();
bool mVolumeShaperActive = false;
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, audio_devices_t device, ThreadBase::type_t type,
bool systemReady);
void processVolume_l(Track *track, bool lastTrack);
// prepareTracks_l() tells threadLoop_mix() the name of the single active track
sp<Track> mActiveTrack;
wp<Track> mPreviousTrack; // used to detect track switch
// This must be initialized for initial condition of mMasterBalance = 0 (disabled).
float mMasterBalanceLeft = 1.f;
float mMasterBalanceRight = 1.f;
public:
virtual bool hasFastMixer() const { return false; }
virtual int64_t computeWaitTimeNs_l() const override;
status_t threadloop_getHalTimestamp_l(ExtendedTimestamp *timestamp) const override {
// For DIRECT and OFFLOAD threads, query the output sink directly.
if (mOutput != nullptr) {
uint64_t uposition64;
struct timespec time;
if (mOutput->getPresentationPosition(
&uposition64, &time) == OK) {
timestamp->mPosition[ExtendedTimestamp::LOCATION_KERNEL]
= (int64_t)uposition64;
timestamp->mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]
= audio_utils_ns_from_timespec(&time);
return NO_ERROR;
}
}
return INVALID_OPERATION;
}
};
class OffloadThread : public DirectOutputThread {
public:
OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, uint32_t device, bool systemReady);
virtual ~OffloadThread() {};
virtual void flushHw_l();
protected:
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual void threadLoop_exit();
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual void invalidateTracks(audio_stream_type_t streamType);
virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); }
private:
size_t mPausedWriteLength; // length in bytes of write interrupted by pause
size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume
bool mKeepWakeLock; // keep wake lock while waiting for write callback
uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback
// used and valid only during underrun. ~0 if
// no underrun has occurred during playback and
// is not reset on standby.
};
class AsyncCallbackThread : public Thread {
public:
explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread);
virtual ~AsyncCallbackThread();
// Thread virtuals
virtual bool threadLoop();
// RefBase
virtual void onFirstRef();
void exit();
void setWriteBlocked(uint32_t sequence);
void resetWriteBlocked();
void setDraining(uint32_t sequence);
void resetDraining();
void setAsyncError();
private:
const wp<PlaybackThread> mPlaybackThread;
// mWriteAckSequence corresponds to the last write sequence passed by the offload thread via
// setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used
// to indicate that the callback has been received via resetWriteBlocked()
uint32_t mWriteAckSequence;
// mDrainSequence corresponds to the last drain sequence passed by the offload thread via
// setDraining(). The sequence is shifted one bit to the left and the lsb is used
// to indicate that the callback has been received via resetDraining()
uint32_t mDrainSequence;
Condition mWaitWorkCV;
Mutex mLock;
bool mAsyncError;
};
class DuplicatingThread : public MixerThread {
public:
DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
audio_io_handle_t id, bool systemReady);
virtual ~DuplicatingThread();
// Thread virtuals
void addOutputTrack(MixerThread* thread);
void removeOutputTrack(MixerThread* thread);
uint32_t waitTimeMs() const { return mWaitTimeMs; }
void sendMetadataToBackend_l(
const StreamOutHalInterface::SourceMetadata& metadata) override;
protected:
virtual uint32_t activeSleepTimeUs() const;
void dumpInternals_l(int fd, const Vector<String16>& args) override;
private:
bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
protected:
// threadLoop snippets
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
virtual ssize_t threadLoop_write();
virtual void threadLoop_standby();
virtual void cacheParameters_l();
private:
// called from threadLoop, addOutputTrack, removeOutputTrack
virtual void updateWaitTime_l();
protected:
virtual void saveOutputTracks();
virtual void clearOutputTracks();
private:
uint32_t mWaitTimeMs;
SortedVector < sp<OutputTrack> > outputTracks;
SortedVector < sp<OutputTrack> > mOutputTracks;
public:
virtual bool hasFastMixer() const { return false; }
status_t threadloop_getHalTimestamp_l(
ExtendedTimestamp *timestamp) const override {
if (mOutputTracks.size() > 0) {
// forward the first OutputTrack's kernel information for timestamp.
const ExtendedTimestamp trackTimestamp =
mOutputTracks[0]->getClientProxyTimestamp();
if (trackTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] > 0) {
timestamp->mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
trackTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
timestamp->mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
trackTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
return OK; // discard server timestamp - that's ignored.
}
}
return INVALID_OPERATION;
}
};
// record thread
class RecordThread : public ThreadBase
{
public:
class RecordTrack;
/* The ResamplerBufferProvider is used to retrieve recorded input data from the
* RecordThread. It maintains local state on the relative position of the read
* position of the RecordTrack compared with the RecordThread.
*/
class ResamplerBufferProvider : public AudioBufferProvider
{
public:
explicit ResamplerBufferProvider(RecordTrack* recordTrack) :
mRecordTrack(recordTrack),
mRsmpInUnrel(0), mRsmpInFront(0) { }
virtual ~ResamplerBufferProvider() { }
// called to set the ResamplerBufferProvider to head of the RecordThread data buffer,
// skipping any previous data read from the hal.
virtual void reset();
/* Synchronizes RecordTrack position with the RecordThread.
* Calculates available frames and handle overruns if the RecordThread
* has advanced faster than the ResamplerBufferProvider has retrieved data.
* TODO: why not do this for every getNextBuffer?
*
* Parameters
* framesAvailable: pointer to optional output size_t to store record track
* frames available.
* hasOverrun: pointer to optional boolean, returns true if track has overrun.
*/
virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL);
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
private:
RecordTrack * const mRecordTrack;
size_t mRsmpInUnrel; // unreleased frames remaining from
// most recent getNextBuffer
// for debug only
int32_t mRsmpInFront; // next available frame
// rolling counter that is never cleared
};
#include "RecordTracks.h"
RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
audio_io_handle_t id,
audio_devices_t outDevice,
audio_devices_t inDevice,
bool systemReady
);
virtual ~RecordThread();
// no addTrack_l ?
void destroyTrack_l(const sp<RecordTrack>& track);
void removeTrack_l(const sp<RecordTrack>& track);
// Thread virtuals
virtual bool threadLoop();
virtual void preExit();
// RefBase
virtual void onFirstRef();
virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
virtual sp<MemoryDealer> readOnlyHeap() const { return mReadOnlyHeap; }
virtual sp<IMemory> pipeMemory() const { return mPipeMemory; }
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
const audio_attributes_t& attr,
uint32_t *pSampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
audio_session_t sessionId,
size_t *pNotificationFrameCount,
pid_t creatorPid,
uid_t uid,
audio_input_flags_t *flags,
pid_t tid,
status_t *status /*non-NULL*/,
audio_port_handle_t portId,
const String16& opPackageName);
status_t start(RecordTrack* recordTrack,
AudioSystem::sync_event_t event,
audio_session_t triggerSession);
// ask the thread to stop the specified track, and
// return true if the caller should then do it's part of the stopping process
bool stop(RecordTrack* recordTrack);
AudioStreamIn* clearInput();
virtual sp<StreamHalInterface> stream() const;
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status);
virtual void cacheParameters_l() {}
virtual String8 getParameters(const String8& keys);
virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle);
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle);
void addPatchTrack(const sp<PatchRecord>& record);
void deletePatchTrack(const sp<PatchRecord>& record);
void readInputParameters_l();
virtual uint32_t getInputFramesLost();
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
uint32_t hasAudioSession_l(audio_session_t sessionId) const override {
return ThreadBase::hasAudioSession_l(sessionId, mTracks);
}
// Return the set of unique session IDs across all tracks.
// The keys are the session IDs, and the associated values are meaningless.
// FIXME replace by Set [and implement Bag/Multiset for other uses].
KeyedVector<audio_session_t, bool> sessionIds() const;
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const;
static void syncStartEventCallback(const wp<SyncEvent>& event);
virtual size_t frameCount() const { return mFrameCount; }
bool hasFastCapture() const { return mFastCapture != 0; }
virtual void toAudioPortConfig(struct audio_port_config *config);
virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc,
audio_session_t sessionId);
virtual void acquireWakeLock_l() {
ThreadBase::acquireWakeLock_l();
mActiveTracks.updatePowerState(this, true /* force */);
}
virtual bool isOutput() const override { return false; }
void checkBtNrec();
// Sets the UID records silence
void setRecordSilenced(uid_t uid, bool silenced);
status_t getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction);
status_t setPreferredMicrophoneFieldDimension(float zoom);
void updateMetadata_l() override;
bool fastTrackAvailable() const { return mFastTrackAvail; }
bool isTimestampCorrectionEnabled() const override {
// checks popcount for exactly one device.
return audio_is_input_device(
mInDevice & mTimestampCorrectedDevices);
}
protected:
void dumpInternals_l(int fd, const Vector<String16>& args) override;
void dumpTracks_l(int fd, const Vector<String16>& args) override;
private:
// Enter standby if not already in standby, and set mStandby flag
void standbyIfNotAlreadyInStandby();
// Call the HAL standby method unconditionally, and don't change mStandby flag
void inputStandBy();
void checkBtNrec_l();
AudioStreamIn *mInput;
SortedVector < sp<RecordTrack> > mTracks;
// mActiveTracks has dual roles: it indicates the current active track(s), and
// is used together with mStartStopCond to indicate start()/stop() progress
ActiveTracks<RecordTrack> mActiveTracks;
Condition mStartStopCond;
// resampler converts input at HAL Hz to output at AudioRecord client Hz
void *mRsmpInBuffer; // size = mRsmpInFramesOA
size_t mRsmpInFrames; // size of resampler input in frames
size_t mRsmpInFramesP2;// size rounded up to a power-of-2
size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation
// rolling index that is never cleared
int32_t mRsmpInRear; // last filled frame + 1
// For dumpsys
const sp<MemoryDealer> mReadOnlyHeap;
// one-time initialization, no locks required
sp<FastCapture> mFastCapture; // non-0 if there is also
// a fast capture
// FIXME audio watchdog thread
// contents are not guaranteed to be consistent, no locks required
FastCaptureDumpState mFastCaptureDumpState;
#ifdef STATE_QUEUE_DUMP
// FIXME StateQueue observer and mutator dump fields
#endif
// FIXME audio watchdog dump
// accessible only within the threadLoop(), no locks required
// mFastCapture->sq() // for mutating and pushing state
int32_t mFastCaptureFutex; // for cold idle
// The HAL input source is treated as non-blocking,
// but current implementation is blocking
sp<NBAIO_Source> mInputSource;
// The source for the normal capture thread to read from: mInputSource or mPipeSource
sp<NBAIO_Source> mNormalSource;
// If a fast capture is present, the non-blocking pipe sink written to by fast capture,
// otherwise clear
sp<NBAIO_Sink> mPipeSink;
// If a fast capture is present, the non-blocking pipe source read by normal thread,
// otherwise clear
sp<NBAIO_Source> mPipeSource;
// Depth of pipe from fast capture to normal thread and fast clients, always power of 2
size_t mPipeFramesP2;
// If a fast capture is present, the Pipe as IMemory, otherwise clear
sp<IMemory> mPipeMemory;
// TODO: add comment and adjust size as needed
static const size_t kFastCaptureLogSize = 4 * 1024;
sp<NBLog::Writer> mFastCaptureNBLogWriter;
bool mFastTrackAvail; // true if fast track available
// common state to all record threads
std::atomic_bool mBtNrecSuspended;
int64_t mFramesRead = 0; // continuous running counter.
};
class MmapThread : public ThreadBase
{
public:
#include "MmapTracks.h"
MmapThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
AudioHwDevice *hwDev, sp<StreamHalInterface> stream,
audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady);
virtual ~MmapThread();
virtual void configure(const audio_attributes_t *attr,
audio_stream_type_t streamType,
audio_session_t sessionId,
const sp<MmapStreamCallback>& callback,
audio_port_handle_t deviceId,
audio_port_handle_t portId);
void disconnect();
// MmapStreamInterface
status_t createMmapBuffer(int32_t minSizeFrames,
struct audio_mmap_buffer_info *info);
status_t getMmapPosition(struct audio_mmap_position *position);
status_t start(const AudioClient& client, audio_port_handle_t *handle);
status_t stop(audio_port_handle_t handle);
status_t standby();
// RefBase
virtual void onFirstRef();
// Thread virtuals
virtual bool threadLoop();
virtual void threadLoop_exit();
virtual void threadLoop_standby();
virtual bool shouldStandby_l() { return false; }
virtual status_t exitStandby();
virtual status_t initCheck() const { return (mHalStream == 0) ? NO_INIT : NO_ERROR; }
virtual size_t frameCount() const { return mFrameCount; }
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status);
virtual String8 getParameters(const String8& keys);
virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
void readHalParameters_l();
virtual void cacheParameters_l() {}
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle);
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle);
virtual void toAudioPortConfig(struct audio_port_config *config);
virtual sp<StreamHalInterface> stream() const { return mHalStream; }
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc,
audio_session_t sessionId);
uint32_t hasAudioSession_l(audio_session_t sessionId) const override {
// Note: using mActiveTracks as no mTracks here.
return ThreadBase::hasAudioSession_l(sessionId, mActiveTracks);
}
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const;
virtual void checkSilentMode_l() {}
virtual void processVolume_l() {}
void checkInvalidTracks_l();
virtual audio_stream_type_t streamType() { return AUDIO_STREAM_DEFAULT; }
virtual void invalidateTracks(audio_stream_type_t streamType __unused) {}
// Sets the UID records silence
virtual void setRecordSilenced(uid_t uid __unused, bool silenced __unused) {}
protected:
void dumpInternals_l(int fd, const Vector<String16>& args) override;
void dumpTracks_l(int fd, const Vector<String16>& args) override;
audio_attributes_t mAttr;
audio_session_t mSessionId;
audio_port_handle_t mPortId;
wp<MmapStreamCallback> mCallback;
sp<StreamHalInterface> mHalStream;
sp<DeviceHalInterface> mHalDevice;
AudioHwDevice* const mAudioHwDev;
ActiveTracks<MmapTrack> mActiveTracks;
float mHalVolFloat;
int32_t mNoCallbackWarningCount;
static constexpr int32_t kMaxNoCallbackWarnings = 5;
};
class MmapPlaybackThread : public MmapThread, public VolumeInterface
{
public:
MmapPlaybackThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamOut *output,
audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady);
virtual ~MmapPlaybackThread() {}
virtual void configure(const audio_attributes_t *attr,
audio_stream_type_t streamType,
audio_session_t sessionId,
const sp<MmapStreamCallback>& callback,
audio_port_handle_t deviceId,
audio_port_handle_t portId);
AudioStreamOut* clearOutput();
// VolumeInterface
virtual void setMasterVolume(float value);
virtual void setMasterMute(bool muted);
virtual void setStreamVolume(audio_stream_type_t stream, float value);
virtual void setStreamMute(audio_stream_type_t stream, bool muted);
virtual float streamVolume(audio_stream_type_t stream) const;
void setMasterMute_l(bool muted) { mMasterMute = muted; }
virtual void invalidateTracks(audio_stream_type_t streamType);
virtual audio_stream_type_t streamType() { return mStreamType; }
virtual void checkSilentMode_l();
void processVolume_l() override;
virtual bool isOutput() const override { return true; }
void updateMetadata_l() override;
virtual void toAudioPortConfig(struct audio_port_config *config);
protected:
void dumpInternals_l(int fd, const Vector<String16>& args) override;
audio_stream_type_t mStreamType;
float mMasterVolume;
float mStreamVolume;
bool mMasterMute;
bool mStreamMute;
AudioStreamOut* mOutput;
};
class MmapCaptureThread : public MmapThread
{
public:
MmapCaptureThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamIn *input,
audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady);
virtual ~MmapCaptureThread() {}
AudioStreamIn* clearInput();
status_t exitStandby() override;
virtual bool isOutput() const override { return false; }
void updateMetadata_l() override;
void processVolume_l() override;
void setRecordSilenced(uid_t uid, bool silenced) override;
virtual void toAudioPortConfig(struct audio_port_config *config);
protected:
AudioStreamIn* mInput;
};