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android / platform / frameworks / av / ac76e8e9e330f27a75b0deb5734171ca3731952b / . / services / audioflinger / Threads.h
blob: ddf0669f51c2ea69fb3540c4138c2132cb2b34a0 [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.
*/
#pragma once
// ADD_BATTERY_DATA AUDIO_WATCHDOG FAST_THREAD_STATISTICS STATE_QUEUE_DUMP TEE_SINK
#include "Configuration.h"
#include "IAfThread.h"
#include "IAfTrack.h"
#include <android-base/macros.h> // DISALLOW_COPY_AND_ASSIGN
#include <android/os/IPowerManager.h>
#include <afutils/AudioWatchdog.h>
#include <afutils/NBAIO_Tee.h>
#include <audio_utils/Balance.h>
#include <audio_utils/SimpleLog.h>
#include <datapath/ThreadMetrics.h>
#include <fastpath/FastCapture.h>
#include <fastpath/FastMixer.h>
#include <mediautils/Synchronization.h>
#include <mediautils/ThreadSnapshot.h>
#include <timing/MonotonicFrameCounter.h>
#include <utils/Log.h>
namespace android {
class AsyncCallbackThread;
class ThreadBase : public virtual IAfThreadBase, public Thread {
public:
static const char *threadTypeToString(type_t type);
// ThreadBase_ThreadLoop is a virtual mutex (always nullptr) that
// guards methods and variables that ONLY run and are accessed
// on the single threaded threadLoop().
//
// As access is by a single thread, the variables are thread safe.
static audio_utils::mutex* ThreadBase_ThreadLoop;
IAfThreadCallback* afThreadCallback() const final { return mAfThreadCallback.get(); }
ThreadBase(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
type_t type, bool systemReady, bool isOut);
~ThreadBase() override;
status_t readyToRun() final;
void clearPowerManager() final EXCLUDES_ThreadBase_Mutex;
// 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,
CFG_EVENT_UPDATE_OUT_DEVICE,
CFG_EVENT_RESIZE_BUFFER,
CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS,
CFG_EVENT_HAL_LATENCY_MODES_CHANGED,
};
class ConfigEventData: public RefBase {
public:
virtual void dump(char *buffer, size_t size) = 0;
protected:
ConfigEventData() = default;
};
// 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 mutex()
// 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 mutex()
// 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->mCondition.notify_one()
// 8. Unlock
class ConfigEvent: public RefBase {
public:
void dump(char *buffer, size_t size) {
snprintf(buffer, size, "Event type: %d\n", mType);
if (mData != nullptr) {
snprintf(buffer, size, "Data:\n");
mData->dump(buffer, size);
}
}
audio_utils::mutex& mutex() const RETURN_CAPABILITY(audio_utils::ConfigEvent_Mutex) {
return mMutex;
}
const int mType; // event type e.g. CFG_EVENT_IO
// mutex associated with mCondition
mutable audio_utils::mutex mMutex{audio_utils::MutexOrder::kConfigEvent_Mutex};
audio_utils::condition_variable mCondition; // condition for status return
// NO_THREAD_SAFETY_ANALYSIS Can we add GUARDED_BY?
status_t mStatus; // status communicated to sender
bool mWaitStatus GUARDED_BY(mutex()); // true if sender is waiting for status
// true if must wait for system ready to enter event queue
bool mRequiresSystemReady GUARDED_BY(mutex());
// NO_THREAD_SAFETY_ANALYSIS Can we add GUARDED_BY?
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_t 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_t mEvent;
const pid_t mPid;
const audio_port_handle_t mPortId;
};
class IoConfigEvent : public ConfigEvent {
public:
IoConfigEvent(audio_io_config_event_t event, pid_t pid, audio_port_handle_t portId) :
ConfigEvent(CFG_EVENT_IO) {
mData = new IoConfigEventData(event, pid, portId);
}
};
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);
}
};
class SetParameterConfigEventData : public ConfigEventData {
public:
explicit SetParameterConfigEventData(const String8& keyValuePairs) :
mKeyValuePairs(keyValuePairs) {}
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "- KeyValue: %s\n", mKeyValuePairs.c_str());
}
const String8 mKeyValuePairs;
};
class SetParameterConfigEvent : public ConfigEvent {
public:
explicit SetParameterConfigEvent(const String8& keyValuePairs) :
ConfigEvent(CFG_EVENT_SET_PARAMETER) {
mData = new SetParameterConfigEventData(keyValuePairs);
mWaitStatus = true;
}
};
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; // cannot be const
};
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;
}
};
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);
}
const 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;
}
};
class UpdateOutDevicesConfigEventData : public ConfigEventData {
public:
explicit UpdateOutDevicesConfigEventData(const DeviceDescriptorBaseVector& outDevices) :
mOutDevices(outDevices) {}
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "- Devices: %s", android::toString(mOutDevices).c_str());
}
const DeviceDescriptorBaseVector mOutDevices;
};
class UpdateOutDevicesConfigEvent : public ConfigEvent {
public:
explicit UpdateOutDevicesConfigEvent(const DeviceDescriptorBaseVector& outDevices) :
ConfigEvent(CFG_EVENT_UPDATE_OUT_DEVICE) {
mData = new UpdateOutDevicesConfigEventData(outDevices);
}
};
class ResizeBufferConfigEventData : public ConfigEventData {
public:
explicit ResizeBufferConfigEventData(int32_t maxSharedAudioHistoryMs) :
mMaxSharedAudioHistoryMs(maxSharedAudioHistoryMs) {}
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "- mMaxSharedAudioHistoryMs: %d", mMaxSharedAudioHistoryMs);
}
const int32_t mMaxSharedAudioHistoryMs;
};
class ResizeBufferConfigEvent : public ConfigEvent {
public:
explicit ResizeBufferConfigEvent(int32_t maxSharedAudioHistoryMs) :
ConfigEvent(CFG_EVENT_RESIZE_BUFFER) {
mData = new ResizeBufferConfigEventData(maxSharedAudioHistoryMs);
}
};
class CheckOutputStageEffectsEvent : public ConfigEvent {
public:
CheckOutputStageEffectsEvent() :
ConfigEvent(CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS) {
}
};
class HalLatencyModesChangedEvent : public ConfigEvent {
public:
HalLatencyModesChangedEvent() :
ConfigEvent(CFG_EVENT_HAL_LATENCY_MODES_CHANGED) {
}
};
class PMDeathRecipient : public IBinder::DeathRecipient {
public:
explicit PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
// IBinder::DeathRecipient
void binderDied(const wp<IBinder>& who) final;
private:
DISALLOW_COPY_AND_ASSIGN(PMDeathRecipient);
const wp<ThreadBase> mThread;
};
type_t type() const final { return mType; }
bool isDuplicating() const final { return (mType == DUPLICATING); }
audio_io_handle_t id() const final { return mId;}
uint32_t sampleRate() const final { return mSampleRate; }
audio_channel_mask_t channelMask() const final { return mChannelMask; }
audio_channel_mask_t mixerChannelMask() const override { return mChannelMask; }
audio_format_t format() const final { return mHALFormat; }
uint32_t channelCount() const final { return mChannelCount; }
audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
uint32_t hapticChannelCount() const override { return 0; }
uint32_t latency_l() const override { return 0; } // NO_THREAD_SAFETY_ANALYSIS
void setVolumeForOutput_l(float /* left */, float /* right */) const override
REQUIRES(mutex()) {}
// Return's the HAL's frame count i.e. fast mixer buffer size.
size_t frameCountHAL() const final { return mFrameCount; }
size_t frameSize() const final { return mFrameSize; }
// Should be "virtual status_t requestExitAndWait()" and override same
// method in Thread, but Thread::requestExitAndWait() is not yet virtual.
void exit() final EXCLUDES_ThreadBase_Mutex;
status_t setParameters(const String8& keyValuePairs) final EXCLUDES_ThreadBase_Mutex;
// sendConfigEvent_l() must be called with ThreadBase::mutex() held
// Can temporarily release the lock if waiting for a reply from
// processConfigEvents_l().
status_t sendConfigEvent_l(sp<ConfigEvent>& event) REQUIRES(mutex());
void sendIoConfigEvent(audio_io_config_event_t event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final EXCLUDES_ThreadBase_Mutex;
void sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final REQUIRES(mutex());
void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) final
EXCLUDES_ThreadBase_Mutex;
void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp) final
REQUIRES(mutex());
status_t sendSetParameterConfigEvent_l(const String8& keyValuePair) final REQUIRES(mutex());
status_t sendCreateAudioPatchConfigEvent(const struct audio_patch* patch,
audio_patch_handle_t* handle) final EXCLUDES_ThreadBase_Mutex;
status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle) final
EXCLUDES_ThreadBase_Mutex;
status_t sendUpdateOutDeviceConfigEvent(
const DeviceDescriptorBaseVector& outDevices) final EXCLUDES_ThreadBase_Mutex;
void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs) final REQUIRES(mutex());
void sendCheckOutputStageEffectsEvent() final EXCLUDES_ThreadBase_Mutex;
void sendCheckOutputStageEffectsEvent_l() final REQUIRES(mutex());
void sendHalLatencyModesChangedEvent_l() final REQUIRES(mutex());
void processConfigEvents_l() final REQUIRES(mutex());
void setCheckOutputStageEffects() override {}
void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override;
void toAudioPortConfig(struct audio_port_config* config) override;
void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override REQUIRES(mutex());
// see note at declaration of mStandby, mOutDevice and mInDevice
bool inStandby() const override { return mStandby; }
const DeviceTypeSet outDeviceTypes_l() const final REQUIRES(mutex()) {
return getAudioDeviceTypes(mOutDeviceTypeAddrs);
}
audio_devices_t inDeviceType_l() const final REQUIRES(mutex()) {
return mInDeviceTypeAddr.mType;
}
DeviceTypeSet getDeviceTypes_l() const final REQUIRES(mutex()) {
return isOutput() ? outDeviceTypes_l() : DeviceTypeSet({inDeviceType_l()});
}
const AudioDeviceTypeAddrVector& outDeviceTypeAddrs() const final {
return mOutDeviceTypeAddrs;
}
const AudioDeviceTypeAddr& inDeviceTypeAddr() const final {
return mInDeviceTypeAddr;
}
bool isOutput() const final { return mIsOut; }
bool isOffloadOrMmap() const final {
switch (mType) {
case OFFLOAD:
case MMAP_PLAYBACK:
case MMAP_CAPTURE:
return true;
default:
return false;
}
}
sp<IAfEffectHandle> createEffect_l(
const sp<Client>& client,
const sp<media::IEffectClient>& effectClient,
int32_t priority,
audio_session_t sessionId,
effect_descriptor_t *desc,
int *enabled,
status_t *status /*non-NULL*/,
bool pinned,
bool probe,
bool notifyFramesProcessed) final
REQUIRES(audio_utils::AudioFlinger_Mutex);
// 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
SPATIALIZED_SESSION = 0x8, // the audio session corresponds to at least one
// spatialized track
BIT_PERFECT_SESSION = 0x10 // the audio session corresponds to at least one
// bit-perfect track
};
// get effect chain corresponding to session Id.
sp<IAfEffectChain> getEffectChain(audio_session_t sessionId) const final;
// same as getEffectChain() but must be called with ThreadBase mutex locked
sp<IAfEffectChain> getEffectChain_l(audio_session_t sessionId) const final REQUIRES(mutex());
std::vector<int> getEffectIds_l(audio_session_t sessionId) const final REQUIRES(mutex());
// 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<IAfEffectChain>>& effectChains) final REQUIRES(mutex());
// unlock effect chains after process
void unlockEffectChains(const Vector<sp<IAfEffectChain>>& effectChains) final;
// get a copy of mEffectChains vector
Vector<sp<IAfEffectChain>> getEffectChains_l() const final REQUIRES(mutex()) {
return mEffectChains;
}
// set audio mode to all effect chains
void setMode(audio_mode_t mode) final;
// get effect module with corresponding ID on specified audio session
sp<IAfEffectModule> getEffect(audio_session_t sessionId, int effectId) const final;
sp<IAfEffectModule> getEffect_l(audio_session_t sessionId, int effectId) const final
REQUIRES(mutex());
// 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_ll(const sp<IAfEffectModule>& effect) final
REQUIRES(audio_utils::AudioFlinger_Mutex, mutex());
// remove and effect module. Also removes the effect chain is this was the last
// effect
void removeEffect_l(const sp<IAfEffectModule>& effect, bool release = false) final
REQUIRES(mutex());
// disconnect an effect handle from module and destroy module if last handle
void disconnectEffectHandle(IAfEffectHandle* handle, bool unpinIfLast) final;
// detach all tracks connected to an auxiliary effect
void detachAuxEffect_l(int /* effectId */) override REQUIRES(mutex()) {}
// TODO(b/291317898) - remove hasAudioSession_l below.
uint32_t hasAudioSession_l(audio_session_t sessionId) const override REQUIRES(mutex()) = 0;
uint32_t hasAudioSession(audio_session_t sessionId) const final EXCLUDES_ThreadBase_Mutex {
audio_utils::lock_guard _l(mutex());
return hasAudioSession_l(sessionId);
}
template <typename T>
uint32_t hasAudioSession_l(audio_session_t sessionId, const T& tracks) const
REQUIRES(mutex()) {
uint32_t result = 0;
if (getEffectChain_l(sessionId) != 0) {
result = EFFECT_SESSION;
}
for (size_t i = 0; i < tracks.size(); ++i) {
const sp<IAfTrackBase>& 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.
}
if (track->isSpatialized()) {
result |= SPATIALIZED_SESSION; // caution, only first track.
}
if (track->isBitPerfect()) {
result |= BIT_PERFECT_SESSION;
}
break;
}
}
return result;
}
// the value returned by default implementation is not important as the
// strategy is only meaningful for PlaybackThread which implements this method
product_strategy_t getStrategyForSession_l(
audio_session_t /* sessionId */) const override REQUIRES(mutex()){
return static_cast<product_strategy_t>(0);
}
// check if some effects must be suspended/restored when an effect is enabled
// or disabled
void checkSuspendOnEffectEnabled(bool enabled,
audio_session_t sessionId,
bool threadLocked) final;
// 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.
sp<MemoryDealer> readOnlyHeap() const override { return nullptr; }
sp<IMemory> pipeMemory() const override { return nullptr; }
void systemReady() final EXCLUDES_ThreadBase_Mutex;
void broadcast_l() final REQUIRES(mutex());
bool isTimestampCorrectionEnabled_l() const override REQUIRES(mutex()) { return false; }
bool isMsdDevice() const final { return mIsMsdDevice; }
void dump(int fd, const Vector<String16>& args) override;
// deliver stats to mediametrics.
void sendStatistics(bool force) final
REQUIRES(ThreadBase_ThreadLoop) EXCLUDES_ThreadBase_Mutex;
audio_utils::mutex& mutex() const final RETURN_CAPABILITY(audio_utils::ThreadBase_Mutex) {
return mMutex;
}
mutable audio_utils::mutex mMutex{audio_utils::MutexOrder::kThreadBase_Mutex};
void onEffectEnable(const sp<IAfEffectModule>& effect) final EXCLUDES_ThreadBase_Mutex;
void onEffectDisable() final EXCLUDES_ThreadBase_Mutex;
// invalidateTracksForAudioSession_l must be called with holding mutex().
void invalidateTracksForAudioSession_l(audio_session_t /* sessionId */) const override
REQUIRES(mutex()) {}
// Invalidate all the tracks with the given audio session.
void invalidateTracksForAudioSession(audio_session_t sessionId) const final
EXCLUDES_ThreadBase_Mutex {
audio_utils::lock_guard _l(mutex());
invalidateTracksForAudioSession_l(sessionId);
}
template <typename T>
void invalidateTracksForAudioSession_l(audio_session_t sessionId,
const T& tracks) const REQUIRES(mutex()) {
for (size_t i = 0; i < tracks.size(); ++i) {
const sp<IAfTrackBase>& track = tracks[i];
if (sessionId == track->sessionId()) {
track->invalidate();
}
}
}
void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) override
REQUIRES(audio_utils::AudioFlinger_Mutex);
void stopMelComputation_l() override
REQUIRES(audio_utils::AudioFlinger_Mutex);
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() EXCLUDES_ThreadBase_Mutex;
virtual void acquireWakeLock_l() REQUIRES(mutex());
void releaseWakeLock() EXCLUDES_ThreadBase_Mutex;
void releaseWakeLock_l() REQUIRES(mutex());
void updateWakeLockUids_l(const SortedVector<uid_t> &uids) REQUIRES(mutex());
void getPowerManager_l() REQUIRES(mutex());
// 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) final REQUIRES(mutex());
// updated mSuspendedSessions when an effect is suspended or restored
void updateSuspendedSessions_l(const effect_uuid_t *type,
bool suspend,
audio_session_t sessionId) REQUIRES(mutex());
// check if some effects must be suspended when an effect chain is added
void checkSuspendOnAddEffectChain_l(const sp<IAfEffectChain>& chain) REQUIRES(mutex());
/**
* waitWhileThreadBusy_l() serves as a mutex gate, which does not allow
* progress beyond the method while the PlaybackThread is busy (see setThreadBusy_l()).
* During the wait, the ThreadBase_Mutex is temporarily unlocked.
*
* This implementation uses a condition variable. Alternative methods to gate
* the thread may use a second mutex (i.e. entry based on scoped_lock(mutex, gating_mutex)),
* but those have less flexibility and more lock order issues.
*
* Current usage by Track::destroy(), Track::start(), Track::stop(), Track::pause(),
* and Track::flush() block this way, and the primary caller is through TrackHandle
* with no other mutexes held.
*
* Special tracks like PatchTrack and OutputTrack may also hold the another thread's
* ThreadBase_Mutex during this time. No other mutex is held.
*/
void waitWhileThreadBusy_l(audio_utils::unique_lock& ul) final REQUIRES(mutex()) {
// the wait returns immediately if the predicate is satisfied.
mThreadBusyCv.wait(ul, [&]{ return mThreadBusy == false;});
}
void setThreadBusy_l(bool busy) REQUIRES(mutex()) {
if (busy == mThreadBusy) return;
mThreadBusy = busy;
if (busy == true) return; // no need to wake threads if we become busy.
mThreadBusyCv.notify_all();
}
// sends the metadata of the active tracks to the HAL
struct MetadataUpdate {
std::vector<playback_track_metadata_v7_t> playbackMetadataUpdate;
std::vector<record_track_metadata_v7_t> recordMetadataUpdate;
};
// NO_THREAD_SAFETY_ANALYSIS, updateMetadata_l() should include ThreadBase_ThreadLoop
// but MmapThread::start() -> exitStandby_l() -> updateMetadata_l() prevents this.
virtual MetadataUpdate updateMetadata_l() REQUIRES(mutex()) = 0;
String16 getWakeLockTag();
virtual void preExit() EXCLUDES_ThreadBase_Mutex {}
virtual void setMasterMono_l(bool mono __unused) REQUIRES(mutex()) {}
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
REQUIRES(mutex(), ThreadBase_ThreadLoop) {
return INVALID_OPERATION;
}
public:
// TODO(b/291317898) organize with publics
product_strategy_t getStrategyForStream(audio_stream_type_t stream) const;
protected:
virtual void onHalLatencyModesChanged_l() REQUIRES(mutex()) {}
virtual void dumpInternals_l(int fd __unused, const Vector<String16>& args __unused)
REQUIRES(mutex()) {}
virtual void dumpTracks_l(int fd __unused, const Vector<String16>& args __unused)
REQUIRES(mutex()) {}
const type_t mType;
// Used by parameters, config events, addTrack_l, exit
audio_utils::condition_variable mWaitWorkCV;
const sp<IAfThreadCallback> mAfThreadCallback;
ThreadMetrics mThreadMetrics;
const bool mIsOut;
// mThreadBusy is checked under the ThreadBase_Mutex to ensure that
// TrackHandle operations do not proceed while the ThreadBase is busy
// with the track. mThreadBusy is only true if the track is active.
//
bool mThreadBusy = false; // GUARDED_BY(ThreadBase_Mutex) but read in lambda.
audio_utils::condition_variable mThreadBusyCv;
// 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()
// output device types and addresses
AudioDeviceTypeAddrVector mOutDeviceTypeAddrs GUARDED_BY(mutex());
AudioDeviceTypeAddr mInDeviceTypeAddr GUARDED_BY(mutex()); // input device type and address
Vector<sp<ConfigEvent>> mConfigEvents GUARDED_BY(mutex());
// events awaiting system ready
Vector<sp<ConfigEvent>> mPendingConfigEvents GUARDED_BY(mutex());
// These fields are written and read by thread itself without lock or barrier,
// and read by other threads without lock or barrier via standby(), outDeviceTypes()
// and inDeviceType().
// 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.
// NO_THREAD_SAFETY_ANALYSIS - mPatch and mAudioSource should be guarded by mutex().
struct audio_patch mPatch;
audio_source_t mAudioSource;
const audio_io_handle_t mId;
Vector<sp<IAfEffectChain>> mEffectChains GUARDED_BY(mutex());
static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit
char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated
sp<os::IPowerManager> mPowerManager GUARDED_BY(mutex());
sp<IBinder> mWakeLockToken GUARDED_BY(mutex());
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;
// NO_THREAD_SAFETY_ANALYSIS - mTimestamp and mTimestampVerifier should be
// accessed under mutex for the RecordThread.
ExtendedTimestamp mTimestamp;
TimestampVerifier<int64_t /* frame count */, int64_t /* time ns */> mTimestampVerifier;
// DIRECT and OFFLOAD threads should reset frame count to zero on stop/flush
// TODO: add confirmation checks:
// 1) DIRECT threads and linear PCM format really resets to 0?
// 2) Is frame count really valid if not linear pcm?
// 3) Are all 64 bits of position returned, not just lowest 32 bits?
// Timestamp corrected device should be a single device.
audio_devices_t mTimestampCorrectedDevice = AUDIO_DEVICE_NONE; // CONST set in ctor
// ThreadLoop statistics per iteration.
std::atomic<int64_t> mLastIoBeginNs = -1; // set in threadLoop, read by dump()
int64_t mLastIoEndNs GUARDED_BY(ThreadBase_ThreadLoop) = -1;
// ThreadSnapshot is thread-safe (internally locked)
mediautils::ThreadSnapshot mThreadSnapshot;
audio_utils::Statistics<double> mIoJitterMs GUARDED_BY(mutex()) {0.995 /* alpha */};
audio_utils::Statistics<double> mProcessTimeMs GUARDED_BY(mutex()) {0.995 /* alpha */};
// NO_THREAD_SAFETY_ANALYSIS GUARDED_BY(mutex())
audio_utils::Statistics<double> mLatencyMs{0.995 /* alpha */};
audio_utils::Statistics<double> mMonopipePipeDepthStats{0.999 /* 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) const {
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_l(const sp<ThreadBase>& thread, bool force = false)
REQUIRES(audio_utils::ThreadBase_Mutex);
/** @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.
* true if volume of one of active tracks was changed.
*/
bool readAndClearHasChanged();
/** Force updating track metadata to audio HAL stream next time
* readAndClearHasChanged() is called.
*/
void setHasChanged() { mHasChanged = true; }
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
}
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; // locked internally
private:
void dumpBase_l(int fd, const Vector<String16>& args) REQUIRES(mutex());
void dumpEffectChains_l(int fd, const Vector<String16>& args) REQUIRES(mutex());
};
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase, public virtual IAfPlaybackThread,
public StreamOutHalInterfaceCallback,
public virtual VolumeInterface, public StreamOutHalInterfaceEventCallback {
public:
sp<IAfPlaybackThread> asIAfPlaybackThread() final {
return sp<IAfPlaybackThread>::fromExisting(this);
}
// 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 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<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
audio_io_handle_t id, type_t type, bool systemReady,
audio_config_base_t *mixerConfig = nullptr);
~PlaybackThread() override;
// Thread virtuals
bool threadLoop() final REQUIRES(ThreadBase_ThreadLoop) EXCLUDES_ThreadBase_Mutex;
// RefBase
void onFirstRef() override;
status_t checkEffectCompatibility_l(
const effect_descriptor_t* desc, audio_session_t sessionId) final REQUIRES(mutex());
void addOutputTrack_l(const sp<IAfTrack>& track) final REQUIRES(mutex()) {
mTracks.add(track);
}
protected:
// Code snippets that were lifted up out of threadLoop()
virtual void threadLoop_mix() REQUIRES(ThreadBase_ThreadLoop) = 0;
virtual void threadLoop_sleepTime() REQUIRES(ThreadBase_ThreadLoop) = 0;
virtual ssize_t threadLoop_write() REQUIRES(ThreadBase_ThreadLoop);
virtual void threadLoop_drain() REQUIRES(ThreadBase_ThreadLoop);
virtual void threadLoop_standby() REQUIRES(ThreadBase_ThreadLoop);
virtual void threadLoop_exit() REQUIRES(ThreadBase_ThreadLoop);
virtual void threadLoop_removeTracks(const Vector<sp<IAfTrack>>& tracksToRemove)
REQUIRES(ThreadBase_ThreadLoop);
// 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<IAfTrack>>* tracksToRemove)
REQUIRES(mutex(), ThreadBase_ThreadLoop) = 0;
void removeTracks_l(const Vector<sp<IAfTrack>>& tracksToRemove) REQUIRES(mutex());
status_t handleVoipVolume_l(float *volume) REQUIRES(mutex());
// StreamOutHalInterfaceCallback implementation
virtual void onWriteReady();
virtual void onDrainReady();
virtual void onError();
public: // AsyncCallbackThread
void resetWriteBlocked(uint32_t sequence);
void resetDraining(uint32_t sequence);
protected:
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l() REQUIRES(mutex());
virtual bool shouldStandby_l() REQUIRES(mutex(), ThreadBase_ThreadLoop);
virtual void onAddNewTrack_l() REQUIRES(mutex());
public: // AsyncCallbackThread
void onAsyncError(); // error reported by AsyncCallbackThread
protected:
// StreamHalInterfaceCodecFormatCallback implementation
void onCodecFormatChanged(
const std::vector<uint8_t>& metadataBs) final;
// ThreadBase virtuals
void preExit() final EXCLUDES_ThreadBase_Mutex;
virtual bool keepWakeLock() const { return true; }
virtual void acquireWakeLock_l() REQUIRES(mutex()) {
ThreadBase::acquireWakeLock_l();
mActiveTracks.updatePowerState_l(this, true /* force */);
}
virtual void checkOutputStageEffects()
REQUIRES(ThreadBase_ThreadLoop) EXCLUDES_ThreadBase_Mutex {}
virtual void setHalLatencyMode_l() {}
void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
void dumpTracks_l(int fd, const Vector<String16>& args) final REQUIRES(mutex());
public:
status_t initCheck() const final { return mOutput == nullptr ? NO_INIT : NO_ERROR; }
// return estimated latency in milliseconds, as reported by HAL
uint32_t latency() const final;
// same, but lock must already be held
uint32_t latency_l() const final /* REQUIRES(mutex()) */; // NO_THREAD_SAFETY_ANALYSIS
// VolumeInterface
void setMasterVolume(float value) final;
void setMasterBalance(float balance) override EXCLUDES_ThreadBase_Mutex;
void setMasterMute(bool muted) final;
void setStreamVolume(audio_stream_type_t stream, float value) final EXCLUDES_ThreadBase_Mutex;
void setStreamMute(audio_stream_type_t stream, bool muted) final EXCLUDES_ThreadBase_Mutex;
float streamVolume(audio_stream_type_t stream) const final EXCLUDES_ThreadBase_Mutex;
void setVolumeForOutput_l(float left, float right) const final;
sp<IAfTrack> createTrack_l(
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 *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,
const AttributionSourceState& attributionSource,
pid_t tid,
status_t *status /*non-NULL*/,
audio_port_handle_t portId,
const sp<media::IAudioTrackCallback>& callback,
bool isSpatialized,
bool isBitPerfect,
audio_output_flags_t* afTrackFlags) final
REQUIRES(audio_utils::AudioFlinger_Mutex);
bool isTrackActive(const sp<IAfTrack>& track) const final {
return mActiveTracks.indexOf(track) >= 0;
}
AudioStreamOut* getOutput_l() const final REQUIRES(mutex()) { return mOutput; }
AudioStreamOut* getOutput() const final EXCLUDES_ThreadBase_Mutex;
AudioStreamOut* clearOutput() final EXCLUDES_ThreadBase_Mutex;
// NO_THREAD_SAFETY_ANALYSIS -- probably needs a lock.
sp<StreamHalInterface> stream() const final;
// suspend(), restore(), and isSuspended() are implemented atomically.
void suspend() final { ++mSuspended; }
void restore() final {
// if restore() is done without suspend(), get back into
// range so that the next suspend() will operate correctly
while (true) {
int32_t suspended = mSuspended;
if (suspended <= 0) {
ALOGW("%s: invalid mSuspended %d <= 0", __func__, suspended);
return;
}
const int32_t desired = suspended - 1;
if (mSuspended.compare_exchange_weak(suspended, desired)) return;
}
}
bool isSuspended() const final { return mSuspended > 0; }
String8 getParameters(const String8& keys) EXCLUDES_ThreadBase_Mutex;
// Hold either the AudioFlinger::mutex or the ThreadBase::mutex
void ioConfigChanged_l(audio_io_config_event_t event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
status_t getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames) const final
EXCLUDES_ThreadBase_Mutex;
// Consider also removing and passing an explicit mMainBuffer initialization
// parameter to AF::IAfTrack::Track().
float* sinkBuffer() const final {
return reinterpret_cast<float *>(mSinkBuffer); };
void detachAuxEffect_l(int effectId) final REQUIRES(mutex());
status_t attachAuxEffect(const sp<IAfTrack>& track, int EffectId) final
EXCLUDES_ThreadBase_Mutex;
status_t attachAuxEffect_l(const sp<IAfTrack>& track, int EffectId) final REQUIRES(mutex());
status_t addEffectChain_l(const sp<IAfEffectChain>& chain) final REQUIRES(mutex());
size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) final REQUIRES(mutex());
uint32_t hasAudioSession_l(audio_session_t sessionId) const final REQUIRES(mutex()) {
return ThreadBase::hasAudioSession_l(sessionId, mTracks);
}
product_strategy_t getStrategyForSession_l(audio_session_t sessionId) const final
REQUIRES(mutex());
status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final
EXCLUDES_ThreadBase_Mutex;
// could be static.
bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const final;
// Does this require the AudioFlinger mutex as well?
bool invalidateTracks_l(audio_stream_type_t streamType) final
REQUIRES(mutex());
bool invalidateTracks_l(std::set<audio_port_handle_t>& portIds) final
REQUIRES(mutex());
void invalidateTracks(audio_stream_type_t streamType) override;
// Invalidate tracks by a set of port ids. The port id will be removed from
// the given set if the corresponding track is found and invalidated.
void invalidateTracks(std::set<audio_port_handle_t>& portIds) override
EXCLUDES_ThreadBase_Mutex;
size_t frameCount() const final { return mNormalFrameCount; }
audio_channel_mask_t mixerChannelMask() const final {
return mMixerChannelMask;
}
status_t getTimestamp_l(AudioTimestamp& timestamp) final
REQUIRES(mutex(), ThreadBase_ThreadLoop);
void addPatchTrack(const sp<IAfPatchTrack>& track) final EXCLUDES_ThreadBase_Mutex;
void deletePatchTrack(const sp<IAfPatchTrack>& track) final EXCLUDES_ThreadBase_Mutex;
// NO_THREAD_SAFETY_ANALYSIS - fix this to use atomics.
void toAudioPortConfig(struct audio_port_config* config) final;
// Return the asynchronous signal wait time.
int64_t computeWaitTimeNs_l() const override REQUIRES(mutex()) { return INT64_MAX; }
// returns true if the track is allowed to be added to the thread.
bool isTrackAllowed_l(
audio_channel_mask_t channelMask __unused,
audio_format_t format __unused,
audio_session_t sessionId __unused,
uid_t uid) const override REQUIRES(mutex()) {
return trackCountForUid_l(uid) < PlaybackThread::kMaxTracksPerUid
&& mTracks.size() < PlaybackThread::kMaxTracks;
}
bool isTimestampCorrectionEnabled_l() const final REQUIRES(mutex()) {
return audio_is_output_devices(mTimestampCorrectedDevice)
&& outDeviceTypes_l().count(mTimestampCorrectedDevice) != 0;
}
// NO_THREAD_SAFETY_ANALYSIS - fix this to be atomic.
bool isStreamInitialized() const final {
return !(mOutput == nullptr || mOutput->stream == nullptr);
}
audio_channel_mask_t hapticChannelMask() const final {
return mHapticChannelMask;
}
uint32_t hapticChannelCount() const final {
return mHapticChannelCount;
}
bool supportsHapticPlayback() const final {
return (mHapticChannelMask & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE;
}
void setDownStreamPatch(const struct audio_patch* patch) final EXCLUDES_ThreadBase_Mutex {
audio_utils::lock_guard _l(mutex());
mDownStreamPatch = *patch;
}
IAfTrack* getTrackById_l(audio_port_handle_t trackId) final REQUIRES(mutex());
bool hasMixer() const final {
return mType == MIXER || mType == DUPLICATING || mType == SPATIALIZER;
}
status_t setRequestedLatencyMode(
audio_latency_mode_t /* mode */) override { return INVALID_OPERATION; }
status_t getSupportedLatencyModes(
std::vector<audio_latency_mode_t>* /* modes */) override {
return INVALID_OPERATION;
}
status_t setBluetoothVariableLatencyEnabled(bool /* enabled */) override{
return INVALID_OPERATION;
}
void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) override
REQUIRES(audio_utils::AudioFlinger_Mutex);
void stopMelComputation_l() override
REQUIRES(audio_utils::AudioFlinger_Mutex);
void setStandby() final EXCLUDES_ThreadBase_Mutex {
audio_utils::lock_guard _l(mutex());
setStandby_l();
}
void setStandby_l() final REQUIRES(mutex()) {
mStandby = true;
mHalStarted = false;
mKernelPositionOnStandby =
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
}
bool waitForHalStart() final EXCLUDES_ThreadBase_Mutex {
audio_utils::unique_lock _l(mutex());
static const nsecs_t kWaitHalTimeoutNs = seconds(2);
nsecs_t endWaitTimetNs = systemTime() + kWaitHalTimeoutNs;
while (!mHalStarted) {
nsecs_t timeNs = systemTime();
if (timeNs >= endWaitTimetNs) {
break;
}
nsecs_t waitTimeLeftNs = endWaitTimetNs - timeNs;
mWaitHalStartCV.wait_for(_l, std::chrono::nanoseconds(waitTimeLeftNs));
}
return mHalStarted;
}
protected:
// updated by readOutputParameters_l()
size_t mNormalFrameCount; // normal mixer and effects
// throttle the thread processing
bool mThreadThrottle GUARDED_BY(ThreadBase_ThreadLoop);
// throttle time for MIXER threads - atomic as read by dump()
std::atomic<uint32_t> mThreadThrottleTimeMs;
// notify once per throttling
uint32_t mThreadThrottleEndMs GUARDED_BY(ThreadBase_ThreadLoop);
// half the buffer size in milliseconds
uint32_t mHalfBufferMs GUARDED_BY(ThreadBase_ThreadLoop);
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 GUARDED_BY(ThreadBase_ThreadLoop);
// 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 GUARDED_BY(ThreadBase_ThreadLoop);
// Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize.
size_t mMixerBufferSize GUARDED_BY(ThreadBase_ThreadLoop);
// The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only.
audio_format_t mMixerBufferFormat GUARDED_BY(ThreadBase_ThreadLoop);
// An internal flag set to true by MixerThread::prepareTracks_l()
// when mMixerBuffer contains valid data after mixing.
bool mMixerBufferValid GUARDED_BY(ThreadBase_ThreadLoop);
// 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;
// NO_THREAD_SAFETY_ANALYSIS: Spatializer access this in addEffectChain_l()
// 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;
// NO_THREAD_SAFETY_ANALYSIS: Spatializer access this in addEffectChain_l()
// Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize.
size_t mEffectBufferSize;
// NO_THREAD_SAFETY_ANALYSIS: Spatializer access this in addEffectChain_l()
// The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only.
// NO_THREAD_SAFETY_ANALYSIS: Spatializer access this in addEffectChain_l()
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 GUARDED_BY(ThreadBase_ThreadLoop);
// Set to "true" to enable when data has already copied to sink
bool mHasDataCopiedToSinkBuffer GUARDED_BY(ThreadBase_ThreadLoop) = false;
// Frame size aligned buffer used as input and output to all post processing effects
// except the Spatializer in a SPATIALIZER thread. Non spatialized tracks are mixed into
// this buffer so that post processing effects can be applied.
void* mPostSpatializerBuffer GUARDED_BY(mutex()) = nullptr;
// Size of mPostSpatializerBuffer in bytes
size_t mPostSpatializerBufferSize GUARDED_BY(mutex());
// 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.
std::atomic<int32_t> mSuspended;
int64_t mBytesWritten;
std::atomic<int64_t> mFramesWritten; // not reset on standby
int64_t mLastFramesWritten = -1; // track changes in timestamp
// server frames written.
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;
audio_channel_mask_t mMixerChannelMask = AUDIO_CHANNEL_NONE;
// 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 GUARDED_BY(mutex());
void setMasterMute_l(bool muted) REQUIRES(mutex()) { mMasterMute = muted; }
auto discontinuityForStandbyOrFlush() const { // call on threadLoop or with lock.
return ((mType == DIRECT && !audio_is_linear_pcm(mFormat))
|| mType == OFFLOAD)
? mTimestampVerifier.DISCONTINUITY_MODE_ZERO
: mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS;
}
ActiveTracks<IAfTrack> 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
// consider unification with MMapThread
virtual void checkSilentMode_l() final REQUIRES(mutex());
// Non-trivial for DUPLICATING only
virtual void saveOutputTracks() REQUIRES(ThreadBase_ThreadLoop) {}
virtual void clearOutputTracks() REQUIRES(ThreadBase_ThreadLoop) {}
// Cache various calculated values, at threadLoop() entry and after a parameter change
virtual void cacheParameters_l() REQUIRES(mutex(), ThreadBase_ThreadLoop);
void setCheckOutputStageEffects() override {
mCheckOutputStageEffects.store(true);
}
virtual uint32_t correctLatency_l(uint32_t latency) const REQUIRES(mutex());
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle) REQUIRES(mutex());
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle)
REQUIRES(mutex());
// NO_THREAD_SAFETY_ANALYSIS - fix this to use atomics
bool usesHwAvSync() const final { return mType == DIRECT && mOutput != nullptr
&& mHwSupportsPause
&& (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); }
uint32_t trackCountForUid_l(uid_t uid) const;
void invalidateTracksForAudioSession_l(
audio_session_t sessionId) const override REQUIRES(mutex()) {
ThreadBase::invalidateTracksForAudioSession_l(sessionId, mTracks);
}
DISALLOW_COPY_AND_ASSIGN(PlaybackThread);
status_t addTrack_l(const sp<IAfTrack>& track) final REQUIRES(mutex());
bool destroyTrack_l(const sp<IAfTrack>& track) final REQUIRES(mutex());
void removeTrack_l(const sp<IAfTrack>& track) REQUIRES(mutex());
void readOutputParameters_l() REQUIRES(mutex());
MetadataUpdate updateMetadata_l() final REQUIRES(mutex());
virtual void sendMetadataToBackend_l(const StreamOutHalInterface::SourceMetadata& metadata)
REQUIRES(mutex()) ;
void collectTimestamps_l() REQUIRES(mutex(), ThreadBase_ThreadLoop);
// The Tracks class manages tracks added and removed from the Thread.
template <typename T>
class Tracks {
public:
explicit 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(const 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<IAfTrack> 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 GUARDED_BY(ThreadBase_ThreadLoop); // current cycle
// previous cycle when in prepareTracks_l()
mixer_state mMixerStatusIgnoringFastTracks GUARDED_BY(ThreadBase_ThreadLoop);
// 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 GUARDED_BY(ThreadBase_ThreadLoop);
// same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value
nsecs_t mStandbyDelayNs; // GUARDED_BY(mutex());
// MIXER only
nsecs_t maxPeriod;
// DUPLICATING only
uint32_t writeFrames;
size_t mBytesRemaining GUARDED_BY(ThreadBase_ThreadLoop);
size_t mCurrentWriteLength GUARDED_BY(ThreadBase_ThreadLoop);
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;
audio_utils::mutex& audioTrackCbMutex() const { return mAudioTrackCbMutex; }
mutable audio_utils::mutex mAudioTrackCbMutex{
audio_utils::MutexOrder::kPlaybackThread_AudioTrackCbMutex};
// Record of IAudioTrackCallback
std::map<sp<IAfTrack>, sp<media::IAudioTrackCallback>> mAudioTrackCallbacks;
// 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};
// output stream start detection based on render position returned by the kernel
// condition signalled when the output stream has started
audio_utils::condition_variable mWaitHalStartCV;
// true when the output stream render position has moved, reset to false in standby
bool mHalStarted = false;
// last kernel render position saved when entering standby
int64_t mKernelPositionOnStandby = 0;
public:
FastTrackUnderruns getFastTrackUnderruns(size_t /* fastIndex */) const override
{ return {}; }
const std::atomic<int64_t>& framesWritten() const final { return mFramesWritten; }
protected:
// accessed by both binder threads and within threadLoop(), lock on mutex needed
uint32_t& fastTrackAvailMask_l() final REQUIRES(mutex()) { return mFastTrackAvailMask; }
uint32_t 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;
// audio patch used by the downstream software patch.
// Only used if ThreadBase::mIsMsdDevice is true.
struct audio_patch mDownStreamPatch;
std::atomic_bool mCheckOutputStageEffects{};
// Provides periodic checking for timestamp advancement for underrun detection.
class IsTimestampAdvancing {
public:
// The timestamp will not be checked any faster than the specified time.
explicit IsTimestampAdvancing(nsecs_t minimumTimeBetweenChecksNs)
: mMinimumTimeBetweenChecksNs(minimumTimeBetweenChecksNs)
{
clear();
}
// Check if the presentation position has advanced in the last periodic time.
bool check(AudioStreamOut * output);
// Clear the internal state when the playback state changes for the output
// stream.
void clear();
private:
// The minimum time between timestamp checks.
const nsecs_t mMinimumTimeBetweenChecksNs;
// Add differential check on the timestamps to see if there is a change in the
// timestamp frame position between the last call to check.
uint64_t mPreviousPosition;
// The time at which the last check occurred, to ensure we don't check too
// frequently, giving the Audio HAL enough time to update its timestamps.
nsecs_t mPreviousNs;
// The valued is latched so we don't check timestamps too frequently.
bool mLatchedValue;
};
IsTimestampAdvancing mIsTimestampAdvancing;
virtual void flushHw_l() {
mIsTimestampAdvancing.clear();
}
};
class MixerThread : public PlaybackThread,
public StreamOutHalInterfaceLatencyModeCallback {
public:
MixerThread(const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output,
audio_io_handle_t id,
bool systemReady,
type_t type = MIXER,
audio_config_base_t *mixerConfig = nullptr);
~MixerThread() override;
// RefBase
void onFirstRef() override;
// StreamOutHalInterfaceLatencyModeCallback
void onRecommendedLatencyModeChanged(
std::vector<audio_latency_mode_t> modes) final;
// Thread virtuals
bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) final
REQUIRES(mutex());
bool isTrackAllowed_l(
audio_channel_mask_t channelMask, audio_format_t format,
audio_session_t sessionId, uid_t uid) const final REQUIRES(mutex());
protected:
mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) override
REQUIRES(mutex(), ThreadBase_ThreadLoop);
uint32_t idleSleepTimeUs() const final;
uint32_t suspendSleepTimeUs() const final;
void cacheParameters_l() override REQUIRES(mutex(), ThreadBase_ThreadLoop);
void acquireWakeLock_l() final REQUIRES(mutex()) {
PlaybackThread::acquireWakeLock_l();
if (hasFastMixer()) {
mFastMixer->setBoottimeOffset(
mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]);
}
}
void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
// threadLoop snippets
ssize_t threadLoop_write() override REQUIRES(ThreadBase_ThreadLoop);
void threadLoop_standby() override REQUIRES(ThreadBase_ThreadLoop);
void threadLoop_mix() override REQUIRES(ThreadBase_ThreadLoop);
void threadLoop_sleepTime() override REQUIRES(ThreadBase_ThreadLoop);
uint32_t correctLatency_l(uint32_t latency) const final REQUIRES(mutex());
status_t createAudioPatch_l(
const struct audio_patch* patch, audio_patch_handle_t* handle)
final REQUIRES(mutex());
status_t releaseAudioPatch_l(const audio_patch_handle_t handle) final REQUIRES(mutex());
AudioMixer* mAudioMixer; // normal mixer
// Support low latency mode by default as unless explicitly indicated by the audio HAL
// we assume the audio path is compatible with the head tracking latency requirements
std::vector<audio_latency_mode_t> mSupportedLatencyModes = {AUDIO_LATENCY_MODE_LOW};
// default to invalid value to force first update to the audio HAL
audio_latency_mode_t mSetLatencyMode =
(audio_latency_mode_t)AUDIO_LATENCY_MODE_INVALID;
// Bluetooth Variable latency control logic is enabled or disabled for this thread
std::atomic_bool mBluetoothLatencyModesEnabled;
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 GUARDED_BY(ThreadBase_ThreadLoop); // 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
REQUIRES(mutex(), ThreadBase_ThreadLoop) {
if (mNormalSink.get() != nullptr) {
return mNormalSink->getTimestamp(*timestamp);
}
return INVALID_OPERATION;
}
status_t getSupportedLatencyModes(
std::vector<audio_latency_mode_t>* modes) override;
status_t setBluetoothVariableLatencyEnabled(bool enabled) override;
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 EXCLUDES_ThreadBase_Mutex {
mMasterBalance.store(balance);
if (hasFastMixer()) {
mFastMixer->setMasterBalance(balance);
}
}
void updateHalSupportedLatencyModes_l() REQUIRES(mutex());
void onHalLatencyModesChanged_l() override REQUIRES(mutex());
void setHalLatencyMode_l() override REQUIRES(mutex());
};
class DirectOutputThread : public PlaybackThread, public virtual IAfDirectOutputThread {
public:
sp<IAfDirectOutputThread> asIAfDirectOutputThread() final {
return sp<IAfDirectOutputThread>::fromExisting(this);
}
DirectOutputThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
audio_io_handle_t id, bool systemReady,
const audio_offload_info_t& offloadInfo)
: DirectOutputThread(afThreadCallback, output, id, DIRECT, systemReady, offloadInfo) { }
~DirectOutputThread() override;
status_t selectPresentation(int presentationId, int programId) final;
// Thread virtuals
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status) REQUIRES(mutex());
void flushHw_l() override REQUIRES(mutex(), ThreadBase_ThreadLoop);
void setMasterBalance(float balance) override EXCLUDES_ThreadBase_Mutex;
protected:
virtual uint32_t activeSleepTimeUs() const;
virtual uint32_t idleSleepTimeUs() const;
virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l() REQUIRES(mutex(), ThreadBase_ThreadLoop);
void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove)
REQUIRES(mutex(), ThreadBase_ThreadLoop);
virtual void threadLoop_mix() REQUIRES(ThreadBase_ThreadLoop);
virtual void threadLoop_sleepTime() REQUIRES(ThreadBase_ThreadLoop);
virtual void threadLoop_exit() REQUIRES(ThreadBase_ThreadLoop);
virtual bool shouldStandby_l() REQUIRES(mutex());
virtual void onAddNewTrack_l() REQUIRES(mutex());
const audio_offload_info_t mOffloadInfo;
audioflinger::MonotonicFrameCounter mMonotonicFrameCounter; // for VolumeShaper
bool mVolumeShaperActive = false;
DirectOutputThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
audio_io_handle_t id, ThreadBase::type_t type, bool systemReady,
const audio_offload_info_t& offloadInfo);
void processVolume_l(IAfTrack *track, bool lastTrack) REQUIRES(mutex());
bool isTunerStream() const { return (mOffloadInfo.content_id > 0); }
// prepareTracks_l() tells threadLoop_mix() the name of the single active track
sp<IAfTrack> mActiveTrack;
wp<IAfTrack> 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 REQUIRES(mutex());
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<IAfThreadCallback>& afThreadCallback, AudioStreamOut* output,
audio_io_handle_t id, bool systemReady,
const audio_offload_info_t& offloadInfo);
virtual ~OffloadThread() {};
void flushHw_l() final REQUIRES(mutex(), ThreadBase_ThreadLoop);
protected:
// threadLoop snippets
mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) final
REQUIRES(mutex(), ThreadBase_ThreadLoop);
void threadLoop_exit() final REQUIRES(ThreadBase_ThreadLoop);
bool waitingAsyncCallback() final;
bool waitingAsyncCallback_l() final REQUIRES(mutex());
void invalidateTracks(audio_stream_type_t streamType) final EXCLUDES_ThreadBase_Mutex;
void invalidateTracks(std::set<audio_port_handle_t>& portIds) final EXCLUDES_ThreadBase_Mutex;
bool keepWakeLock() const final { 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
};
class AsyncCallbackThread : public Thread {
public:
explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread);
// Thread virtuals
bool threadLoop() final;
// RefBase
void onFirstRef() final;
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;
audio_utils::condition_variable mWaitWorkCV;
mutable audio_utils::mutex mMutex{audio_utils::MutexOrder::kAsyncCallbackThread_Mutex};
bool mAsyncError;
audio_utils::mutex& mutex() const RETURN_CAPABILITY(audio_utils::AsyncCallbackThread_Mutex) {
return mMutex;
}
};
class DuplicatingThread : public MixerThread, public IAfDuplicatingThread {
public:
DuplicatingThread(const sp<IAfThreadCallback>& afThreadCallback,
IAfPlaybackThread* mainThread,
audio_io_handle_t id, bool systemReady);
~DuplicatingThread() override;
sp<IAfDuplicatingThread> asIAfDuplicatingThread() final {
return sp<IAfDuplicatingThread>::fromExisting(this);
}
// Thread virtuals
void addOutputTrack(IAfPlaybackThread* thread) final EXCLUDES_ThreadBase_Mutex;
void removeOutputTrack(IAfPlaybackThread* thread) final EXCLUDES_ThreadBase_Mutex;
uint32_t waitTimeMs() const final { return mWaitTimeMs; }
void sendMetadataToBackend_l(
const StreamOutHalInterface::SourceMetadata& metadata) final REQUIRES(mutex());
protected:
virtual uint32_t activeSleepTimeUs() const;
void dumpInternals_l(int fd, const Vector<String16>& args) final REQUIRES(mutex());
private:
bool outputsReady() REQUIRES(ThreadBase_ThreadLoop);
protected:
// threadLoop snippets
void threadLoop_mix() final REQUIRES(ThreadBase_ThreadLoop);
void threadLoop_sleepTime() final REQUIRES(ThreadBase_ThreadLoop);
ssize_t threadLoop_write() final REQUIRES(ThreadBase_ThreadLoop);
void threadLoop_standby() final REQUIRES(ThreadBase_ThreadLoop);
void threadLoop_exit() final REQUIRES(ThreadBase_ThreadLoop);
void cacheParameters_l() final REQUIRES(mutex(), ThreadBase_ThreadLoop);
private:
// called from threadLoop, addOutputTrack, removeOutputTrack
void updateWaitTime_l() REQUIRES(mutex());
protected:
void saveOutputTracks() final REQUIRES(mutex(), ThreadBase_ThreadLoop);
void clearOutputTracks() final REQUIRES(mutex(), ThreadBase_ThreadLoop);
private:
uint32_t mWaitTimeMs;
// NO_THREAD_SAFETY_ANALYSIS GUARDED_BY(ThreadBase_ThreadLoop)
SortedVector <sp<IAfOutputTrack>> outputTracks;
SortedVector <sp<IAfOutputTrack>> mOutputTracks GUARDED_BY(mutex());
public:
virtual bool hasFastMixer() const { return false; }
status_t threadloop_getHalTimestamp_l(
ExtendedTimestamp *timestamp) const override REQUIRES(mutex()) {
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;
}
};
class SpatializerThread : public MixerThread {
public:
SpatializerThread(const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamOut* output,
audio_io_handle_t id,
bool systemReady,
audio_config_base_t *mixerConfig);
bool hasFastMixer() const final { return false; }
status_t setRequestedLatencyMode(audio_latency_mode_t mode) final EXCLUDES_ThreadBase_Mutex;
protected:
void checkOutputStageEffects() final
REQUIRES(ThreadBase_ThreadLoop) EXCLUDES_ThreadBase_Mutex;
void setHalLatencyMode_l() final REQUIRES(mutex());
void threadLoop_exit() final REQUIRES(ThreadBase_ThreadLoop);
private:
// Do not request a specific mode by default
audio_latency_mode_t mRequestedLatencyMode = AUDIO_LATENCY_MODE_FREE;
sp<IAfEffectHandle> mFinalDownMixer;
};
// record thread
class RecordThread : public IAfRecordThread, public ThreadBase
{
friend class ResamplerBufferProvider;
public:
sp<IAfRecordThread> asIAfRecordThread() final {
return sp<IAfRecordThread>::fromExisting(this);
}
RecordThread(const sp<IAfThreadCallback>& afThreadCallback,
AudioStreamIn *input,
audio_io_handle_t id,
bool systemReady
);
~RecordThread() override;
// no addTrack_l ?
void destroyTrack_l(const sp<IAfRecordTrack>& track) final REQUIRES(mutex());
void removeTrack_l(const sp<IAfRecordTrack>& track) final REQUIRES(mutex());
// Thread virtuals
bool threadLoop() final REQUIRES(ThreadBase_ThreadLoop) EXCLUDES_ThreadBase_Mutex;
void preExit() final EXCLUDES_ThreadBase_Mutex;
// RefBase
void onFirstRef() final EXCLUDES_ThreadBase_Mutex;
status_t initCheck() const final { return mInput == nullptr ? NO_INIT : NO_ERROR; }
sp<MemoryDealer> readOnlyHeap() const final { return mReadOnlyHeap; }
sp<IMemory> pipeMemory() const final { return mPipeMemory; }
sp<IAfRecordTrack> createRecordTrack_l(
const sp<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,
const AttributionSourceState& attributionSource,
audio_input_flags_t *flags,
pid_t tid,
status_t *status /*non-NULL*/,
audio_port_handle_t portId,
int32_t maxSharedAudioHistoryMs) final
REQUIRES(audio_utils::AudioFlinger_Mutex) EXCLUDES_ThreadBase_Mutex;
status_t start(IAfRecordTrack* recordTrack,
AudioSystem::sync_event_t event,
audio_session_t triggerSession) final EXCLUDES_ThreadBase_Mutex;
// 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(IAfRecordTrack* recordTrack) final EXCLUDES_ThreadBase_Mutex;
AudioStreamIn* getInput() const final { return mInput; }
AudioStreamIn* clearInput() final;
// TODO(b/291317898) Unify with IAfThreadBase
virtual sp<StreamHalInterface> stream() const;
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status) REQUIRES(mutex());
virtual void cacheParameters_l() REQUIRES(mutex(), ThreadBase_ThreadLoop) {}
virtual String8 getParameters(const String8& keys) EXCLUDES_ThreadBase_Mutex;
// Hold either the AudioFlinger::mutex or the ThreadBase::mutex
void ioConfigChanged_l(audio_io_config_event_t event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle) REQUIRES(mutex());
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) REQUIRES(mutex());
void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override
EXCLUDES_ThreadBase_Mutex;
void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override REQUIRES(mutex());
void addPatchTrack(const sp<IAfPatchRecord>& record) final EXCLUDES_ThreadBase_Mutex;
void deletePatchTrack(const sp<IAfPatchRecord>& record) final EXCLUDES_ThreadBase_Mutex;
void readInputParameters_l() REQUIRES(mutex());
uint32_t getInputFramesLost() const final EXCLUDES_ThreadBase_Mutex;
virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain) REQUIRES(mutex());
virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) REQUIRES(mutex());
uint32_t hasAudioSession_l(audio_session_t sessionId) const override REQUIRES(mutex()) {
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;
status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) override
EXCLUDES_ThreadBase_Mutex;
bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const override;
static void syncStartEventCallback(const wp<audioflinger::SyncEvent>& event);
virtual size_t frameCount() const { return mFrameCount; }
bool hasFastCapture() const final { 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) REQUIRES(mutex());
virtual void acquireWakeLock_l() REQUIRES(mutex()) {
ThreadBase::acquireWakeLock_l();
mActiveTracks.updatePowerState_l(this, true /* force */);
}
void checkBtNrec() final EXCLUDES_ThreadBase_Mutex;
// Sets the UID records silence
void setRecordSilenced(audio_port_handle_t portId, bool silenced) final
EXCLUDES_ThreadBase_Mutex;
status_t getActiveMicrophones(
std::vector<media::MicrophoneInfoFw>* activeMicrophones) const final
EXCLUDES_ThreadBase_Mutex;
status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) final
EXCLUDES_ThreadBase_Mutex;
status_t setPreferredMicrophoneFieldDimension(float zoom) final EXCLUDES_ThreadBase_Mutex;
MetadataUpdate updateMetadata_l() override REQUIRES(mutex());
bool fastTrackAvailable() const final { return mFastTrackAvail; }
void setFastTrackAvailable(bool available) final { mFastTrackAvail = available; }
bool isTimestampCorrectionEnabled_l() const override REQUIRES(mutex()) {
// checks popcount for exactly one device.
// Is currently disabled. Before enabling,
// verify compressed record timestamps.
return audio_is_input_device(mTimestampCorrectedDevice)
&& inDeviceType_l() == mTimestampCorrectedDevice;
}
status_t shareAudioHistory(const std::string& sharedAudioPackageName,
audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
int64_t sharedAudioStartMs = -1) final EXCLUDES_ThreadBase_Mutex;
status_t shareAudioHistory_l(const std::string& sharedAudioPackageName,
audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
int64_t sharedAudioStartMs = -1) REQUIRES(mutex());
void resetAudioHistory_l() final REQUIRES(mutex());
bool isStreamInitialized() const final {
return !(mInput == nullptr || mInput->stream == nullptr);
}
protected:
void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
void dumpTracks_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
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() REQUIRES(mutex());
int32_t getOldestFront_l() REQUIRES(mutex());
void updateFronts_l(int32_t offset) REQUIRES(mutex());
AudioStreamIn *mInput;
Source *mSource;
SortedVector <sp<IAfRecordTrack>> mTracks;
// mActiveTracks has dual roles: it indicates the current active track(s), and
// is used together with mStartStopCV to indicate start()/stop() progress
ActiveTracks<IAfRecordTrack> mActiveTracks;
audio_utils::condition_variable mStartStopCV;
// 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.
DeviceDescriptorBaseVector mOutDevices;
int32_t mMaxSharedAudioHistoryMs = 0;
std::string mSharedAudioPackageName = {};
int32_t mSharedAudioStartFrames = -1;
audio_session_t mSharedAudioSessionId = AUDIO_SESSION_NONE;
};
class MmapThread : public ThreadBase, public virtual IAfMmapThread
{
public:
MmapThread(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice *hwDev, const sp<StreamHalInterface>& stream, bool systemReady,
bool isOut);
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) override EXCLUDES_ThreadBase_Mutex {
audio_utils::lock_guard l(mutex());
configure_l(attr, streamType, sessionId, callback, deviceId, portId);
}
void configure_l(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) REQUIRES(mutex());
void disconnect() final EXCLUDES_ThreadBase_Mutex;
// MmapStreamInterface for adapter.
status_t createMmapBuffer(int32_t minSizeFrames, struct audio_mmap_buffer_info* info) final
EXCLUDES_ThreadBase_Mutex;
status_t getMmapPosition(struct audio_mmap_position* position) const override
EXCLUDES_ThreadBase_Mutex;
status_t start(const AudioClient& client,
const audio_attributes_t *attr,
audio_port_handle_t* handle) final EXCLUDES_ThreadBase_Mutex;
status_t stop(audio_port_handle_t handle) final EXCLUDES_ThreadBase_Mutex;
status_t standby() final EXCLUDES_ThreadBase_Mutex;
status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const
EXCLUDES_ThreadBase_Mutex = 0;
status_t reportData(const void* buffer, size_t frameCount) override EXCLUDES_ThreadBase_Mutex;
// RefBase
void onFirstRef() final;
// Thread virtuals
bool threadLoop() final REQUIRES(ThreadBase_ThreadLoop) EXCLUDES_ThreadBase_Mutex;
// Not in ThreadBase
virtual void threadLoop_exit() final REQUIRES(ThreadBase_ThreadLoop);
virtual void threadLoop_standby() final REQUIRES(ThreadBase_ThreadLoop);
virtual bool shouldStandby_l() final REQUIRES(mutex()){ return false; }
virtual status_t exitStandby_l() REQUIRES(mutex());
status_t initCheck() const final { return mHalStream == nullptr ? NO_INIT : NO_ERROR; }
size_t frameCount() const final { return mFrameCount; }
bool checkForNewParameter_l(const String8& keyValuePair, status_t& status)
final REQUIRES(mutex());
String8 getParameters(const String8& keys) final EXCLUDES_ThreadBase_Mutex;
void ioConfigChanged_l(audio_io_config_event_t event, pid_t pid = 0,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final
/* holds either AF::mutex or TB::mutex */;
void readHalParameters_l() REQUIRES(mutex());
void cacheParameters_l() final REQUIRES(mutex(), ThreadBase_ThreadLoop) {}
status_t createAudioPatch_l(
const struct audio_patch* patch, audio_patch_handle_t* handle) final
REQUIRES(mutex());
status_t releaseAudioPatch_l(const audio_patch_handle_t handle) final
REQUIRES(mutex());
// NO_THREAD_SAFETY_ANALYSIS
void toAudioPortConfig(struct audio_port_config* config) override;
sp<StreamHalInterface> stream() const final { return mHalStream; }
status_t addEffectChain_l(const sp<IAfEffectChain>& chain) final REQUIRES(mutex());
size_t removeEffectChain_l(const sp<IAfEffectChain>& chain) final REQUIRES(mutex());
status_t checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId) final REQUIRES(mutex());
uint32_t hasAudioSession_l(audio_session_t sessionId) const override REQUIRES(mutex()) {
// Note: using mActiveTracks as no mTracks here.
return ThreadBase::hasAudioSession_l(sessionId, mActiveTracks);
}
status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final;
bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& event) const final;
virtual void checkSilentMode_l() REQUIRES(mutex()) {} // cannot be const (RecordThread)
virtual void processVolume_l() REQUIRES(mutex()) {}
void checkInvalidTracks_l() REQUIRES(mutex());
// Not in ThreadBase
virtual audio_stream_type_t streamType_l() const REQUIRES(mutex()) {
return AUDIO_STREAM_DEFAULT;
}
virtual void invalidateTracks(audio_stream_type_t /* streamType */)
EXCLUDES_ThreadBase_Mutex {}
void invalidateTracks(std::set<audio_port_handle_t>& /* portIds */) override
EXCLUDES_ThreadBase_Mutex {}
// Sets the UID records silence
void setRecordSilenced(
audio_port_handle_t /* portId */, bool /* silenced */) override
EXCLUDES_ThreadBase_Mutex {}
bool isStreamInitialized() const override { return false; }
void setClientSilencedState_l(audio_port_handle_t portId, bool silenced) REQUIRES(mutex()) {
mClientSilencedStates[portId] = silenced;
}
size_t eraseClientSilencedState_l(audio_port_handle_t portId) REQUIRES(mutex()) {
return mClientSilencedStates.erase(portId);
}
bool isClientSilenced_l(audio_port_handle_t portId) const REQUIRES(mutex()) {
const auto it = mClientSilencedStates.find(portId);
return it != mClientSilencedStates.end() ? it->second : false;
}
void setClientSilencedIfExists_l(audio_port_handle_t portId, bool silenced)
REQUIRES(mutex()) {
const auto it = mClientSilencedStates.find(portId);
if (it != mClientSilencedStates.end()) {
it->second = silenced;
}
}
protected:
void dumpInternals_l(int fd, const Vector<String16>& args) override REQUIRES(mutex());
void dumpTracks_l(int fd, const Vector<String16>& args) final REQUIRES(mutex());
/**
* @brief mDeviceId current device port unique identifier
*/
audio_port_handle_t mDeviceId GUARDED_BY(mutex()) = AUDIO_PORT_HANDLE_NONE;
audio_attributes_t mAttr GUARDED_BY(mutex());
audio_session_t mSessionId GUARDED_BY(mutex());
audio_port_handle_t mPortId GUARDED_BY(mutex());
wp<MmapStreamCallback> mCallback GUARDED_BY(mutex());
sp<StreamHalInterface> mHalStream; // NO_THREAD_SAFETY_ANALYSIS
sp<DeviceHalInterface> mHalDevice GUARDED_BY(mutex());
AudioHwDevice* const mAudioHwDev GUARDED_BY(mutex());
ActiveTracks<IAfMmapTrack> mActiveTracks GUARDED_BY(mutex());
float mHalVolFloat GUARDED_BY(mutex());
std::map<audio_port_handle_t, bool> mClientSilencedStates GUARDED_BY(mutex());
int32_t mNoCallbackWarningCount GUARDED_BY(mutex());
static constexpr int32_t kMaxNoCallbackWarnings = 5;
};
class MmapPlaybackThread : public MmapThread, public IAfMmapPlaybackThread,
public virtual VolumeInterface {
public:
MmapPlaybackThread(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamOut *output, bool systemReady);
sp<IAfMmapPlaybackThread> asIAfMmapPlaybackThread() final {
return sp<IAfMmapPlaybackThread>::fromExisting(this);
}
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) final EXCLUDES_ThreadBase_Mutex;
AudioStreamOut* clearOutput() final EXCLUDES_ThreadBase_Mutex;
// VolumeInterface
void setMasterVolume(float value) final;
// Needs implementation?
void setMasterBalance(float /* value */) final EXCLUDES_ThreadBase_Mutex {}
void setMasterMute(bool muted) final EXCLUDES_ThreadBase_Mutex;
void setStreamVolume(audio_stream_type_t stream, float value) final EXCLUDES_ThreadBase_Mutex;
void setStreamMute(audio_stream_type_t stream, bool muted) final EXCLUDES_ThreadBase_Mutex;
float streamVolume(audio_stream_type_t stream) const final EXCLUDES_ThreadBase_Mutex;
void setMasterMute_l(bool muted) REQUIRES(mutex()) { mMasterMute = muted; }
void invalidateTracks(audio_stream_type_t streamType) final EXCLUDES_ThreadBase_Mutex;
void invalidateTracks(std::set<audio_port_handle_t>& portIds) final EXCLUDES_ThreadBase_Mutex;
audio_stream_type_t streamType_l() const final REQUIRES(mutex()) {
return mStreamType;
}
void checkSilentMode_l() final REQUIRES(mutex());
void processVolume_l() final REQUIRES(mutex());
MetadataUpdate updateMetadata_l() final REQUIRES(mutex());
void toAudioPortConfig(struct audio_port_config* config) final;
status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const final;
bool isStreamInitialized() const final {
return !(mOutput == nullptr || mOutput->stream == nullptr);
}
status_t reportData(const void* buffer, size_t frameCount) final;
void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) final
REQUIRES(audio_utils::AudioFlinger_Mutex);
void stopMelComputation_l() final
REQUIRES(audio_utils::AudioFlinger_Mutex);
protected:
void dumpInternals_l(int fd, const Vector<String16>& args) final REQUIRES(mutex());
float streamVolume_l() const REQUIRES(mutex()) {
return mStreamTypes[mStreamType].volume;
}
bool streamMuted_l() const REQUIRES(mutex()) {
return mStreamTypes[mStreamType].mute;
}
stream_type_t mStreamTypes[AUDIO_STREAM_CNT] GUARDED_BY(mutex());
audio_stream_type_t mStreamType GUARDED_BY(mutex());
float mMasterVolume GUARDED_BY(mutex());
bool mMasterMute GUARDED_BY(mutex());
AudioStreamOut* mOutput; // NO_THREAD_SAFETY_ANALYSIS
mediautils::atomic_sp<audio_utils::MelProcessor> mMelProcessor; // locked internally
};
class MmapCaptureThread : public MmapThread, public IAfMmapCaptureThread
{
public:
MmapCaptureThread(const sp<IAfThreadCallback>& afThreadCallback, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamIn *input, bool systemReady);
sp<IAfMmapCaptureThread> asIAfMmapCaptureThread() final {
return sp<IAfMmapCaptureThread>::fromExisting(this);
}
AudioStreamIn* clearInput() final EXCLUDES_ThreadBase_Mutex;
status_t exitStandby_l() REQUIRES(mutex()) final;
MetadataUpdate updateMetadata_l() final REQUIRES(mutex());
void processVolume_l() final REQUIRES(mutex());
void setRecordSilenced(audio_port_handle_t portId, bool silenced) final
EXCLUDES_ThreadBase_Mutex;
void toAudioPortConfig(struct audio_port_config* config) final;
status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const final;
bool isStreamInitialized() const final {
return !(mInput == nullptr || mInput->stream == nullptr);
}
protected:
AudioStreamIn* mInput; // NO_THREAD_SAFETY_ANALYSIS
};
class BitPerfectThread : public MixerThread {
public:
BitPerfectThread(const sp<IAfThreadCallback>& afThreadCallback, AudioStreamOut *output,
audio_io_handle_t id, bool systemReady);
protected:
mixer_state prepareTracks_l(Vector<sp<IAfTrack>>* tracksToRemove) final
REQUIRES(mutex(), ThreadBase_ThreadLoop);
void threadLoop_mix() final REQUIRES(ThreadBase_ThreadLoop);
private:
// These variables are only accessed on the threadLoop; hence need no mutex.
bool mIsBitPerfect GUARDED_BY(ThreadBase_ThreadLoop) = false;
float mVolumeLeft GUARDED_BY(ThreadBase_ThreadLoop) = 0.f;
float mVolumeRight GUARDED_BY(ThreadBase_ThreadLoop) = 0.f;
};
} // namespace android