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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.
*/
#pragma once
#include "TrackBase.h"
#include <android/os/BnExternalVibrationController.h>
#include <audio_utils/mutex.h>
#include <audio_utils/LinearMap.h>
#include <binder/AppOpsManager.h>
namespace android {
// Checks and monitors OP_PLAY_AUDIO
class OpPlayAudioMonitor : public RefBase {
friend class sp<OpPlayAudioMonitor>;
public:
~OpPlayAudioMonitor() override;
bool hasOpPlayAudio() const;
static sp<OpPlayAudioMonitor> createIfNeeded(
IAfThreadBase* thread,
const AttributionSourceState& attributionSource,
const audio_attributes_t& attr, int id,
audio_stream_type_t streamType);
private:
OpPlayAudioMonitor(IAfThreadBase* thread,
const AttributionSourceState& attributionSource,
audio_usage_t usage, int id, uid_t uid);
void onFirstRef() override;
static void getPackagesForUid(uid_t uid, Vector<String16>& packages);
AppOpsManager mAppOpsManager;
class PlayAudioOpCallback : public BnAppOpsCallback {
public:
explicit PlayAudioOpCallback(const wp<OpPlayAudioMonitor>& monitor);
void opChanged(int32_t op, const String16& packageName) override;
private:
const wp<OpPlayAudioMonitor> mMonitor;
};
sp<PlayAudioOpCallback> mOpCallback;
// called by PlayAudioOpCallback when OP_PLAY_AUDIO is updated in AppOp callback
void checkPlayAudioForUsage(bool doBroadcast);
wp<IAfThreadBase> mThread;
std::atomic_bool mHasOpPlayAudio;
const AttributionSourceState mAttributionSource;
const int32_t mUsage; // on purpose not audio_usage_t because always checked in appOps as int32_t
const int mId; // for logging purposes only
const uid_t mUid;
const String16 mPackageName;
};
// playback track
class Track : public TrackBase, public virtual IAfTrack, public VolumeProvider {
public:
Track(IAfPlaybackThread* thread,
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
const sp<IMemory>& sharedBuffer,
audio_session_t sessionId,
pid_t creatorPid,
const AttributionSourceState& attributionSource,
audio_output_flags_t flags,
track_type type,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
/** default behaviour is to start when there are as many frames
* ready as possible (aka. Buffer is full). */
size_t frameCountToBeReady = SIZE_MAX,
float speed = 1.0f,
bool isSpatialized = false,
bool isBitPerfect = false);
~Track() override;
status_t initCheck() const final;
void appendDumpHeader(String8& result) const final;
void appendDump(String8& result, bool active) const final;
status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
audio_session_t triggerSession = AUDIO_SESSION_NONE) override;
void stop() override;
void pause() final;
void flush() final;
void destroy() final;
uint32_t sampleRate() const final;
audio_stream_type_t streamType() const final {
return mStreamType;
}
bool isOffloaded() const final
{ return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
bool isDirect() const final
{ return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; }
bool isOffloadedOrDirect() const final { return (mFlags
& (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD
| AUDIO_OUTPUT_FLAG_DIRECT)) != 0; }
bool isStatic() const final { return mSharedBuffer.get() != nullptr; }
status_t setParameters(const String8& keyValuePairs) final;
status_t selectPresentation(int presentationId, int programId) final;
status_t attachAuxEffect(int EffectId) final;
void setAuxBuffer(int EffectId, int32_t* buffer) final;
int32_t* auxBuffer() const final { return mAuxBuffer; }
void setMainBuffer(float* buffer) final { mMainBuffer = buffer; }
float* mainBuffer() const final { return mMainBuffer; }
int auxEffectId() const final { return mAuxEffectId; }
status_t getTimestamp(AudioTimestamp& timestamp) final;
void signal() final;
status_t getDualMonoMode(audio_dual_mono_mode_t* mode) const final;
status_t setDualMonoMode(audio_dual_mono_mode_t mode) final;
status_t getAudioDescriptionMixLevel(float* leveldB) const final;
status_t setAudioDescriptionMixLevel(float leveldB) final;
status_t getPlaybackRateParameters(audio_playback_rate_t* playbackRate) const final;
status_t setPlaybackRateParameters(const audio_playback_rate_t& playbackRate) final;
// implement FastMixerState::VolumeProvider interface
gain_minifloat_packed_t getVolumeLR() const final;
status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) final;
bool isFastTrack() const final { return (mFlags & AUDIO_OUTPUT_FLAG_FAST) != 0; }
double bufferLatencyMs() const final {
return isStatic() ? 0. : TrackBase::bufferLatencyMs();
}
// implement volume handling.
media::VolumeShaper::Status applyVolumeShaper(
const sp<media::VolumeShaper::Configuration>& configuration,
const sp<media::VolumeShaper::Operation>& operation);
sp<media::VolumeShaper::State> getVolumeShaperState(int id) const final;
sp<media::VolumeHandler> getVolumeHandler() const final{ return mVolumeHandler; }
/** Set the computed normalized final volume of the track.
* !masterMute * masterVolume * streamVolume * averageLRVolume */
void setFinalVolume(float volumeLeft, float volumeRight) final;
float getFinalVolume() const final { return mFinalVolume; }
void getFinalVolume(float* left, float* right) const final {
*left = mFinalVolumeLeft;
*right = mFinalVolumeRight;
}
using SourceMetadatas = std::vector<playback_track_metadata_v7_t>;
using MetadataInserter = std::back_insert_iterator<SourceMetadatas>;
/** Copy the track metadata in the provided iterator. Thread safe. */
void copyMetadataTo(MetadataInserter& backInserter) const override;
/** Return haptic playback of the track is enabled or not, used in mixer. */
bool getHapticPlaybackEnabled() const final { return mHapticPlaybackEnabled; }
/** Set haptic playback of the track is enabled or not, should be
* set after query or get callback from vibrator service */
void setHapticPlaybackEnabled(bool hapticPlaybackEnabled) final {
mHapticPlaybackEnabled = hapticPlaybackEnabled;
}
/** Return at what intensity to play haptics, used in mixer. */
os::HapticScale getHapticIntensity() const final { return mHapticIntensity; }
/** Return the maximum amplitude allowed for haptics data, used in mixer. */
float getHapticMaxAmplitude() const final { return mHapticMaxAmplitude; }
/** Set intensity of haptic playback, should be set after querying vibrator service. */
void setHapticIntensity(os::HapticScale hapticIntensity) final {
if (os::isValidHapticScale(hapticIntensity)) {
mHapticIntensity = hapticIntensity;
setHapticPlaybackEnabled(mHapticIntensity != os::HapticScale::MUTE);
}
}
/** Set maximum amplitude allowed for haptic data, should be set after querying
* vibrator service.
*/
void setHapticMaxAmplitude(float maxAmplitude) final {
mHapticMaxAmplitude = maxAmplitude;
}
sp<os::ExternalVibration> getExternalVibration() const final { return mExternalVibration; }
// This function should be called with holding thread lock.
void updateTeePatches_l() final;
void setTeePatchesToUpdate_l(TeePatches teePatchesToUpdate) final;
void tallyUnderrunFrames(size_t frames) final {
if (isOut()) { // we expect this from output tracks only
mAudioTrackServerProxy->tallyUnderrunFrames(frames);
// Fetch absolute numbers from AudioTrackShared as it counts
// contiguous underruns as a one -- we want a consistent number.
// TODO: isolate this counting into a class.
mTrackMetrics.logUnderruns(mAudioTrackServerProxy->getUnderrunCount(),
mAudioTrackServerProxy->getUnderrunFrames());
}
}
audio_output_flags_t getOutputFlags() const final { return mFlags; }
float getSpeed() const final { return mSpeed; }
bool isSpatialized() const final { return mIsSpatialized; }
bool isBitPerfect() const final { return mIsBitPerfect; }
/**
* Updates the mute state and notifies the audio service. Call this only when holding player
* thread lock.
*/
void processMuteEvent_l(const sp<IAudioManager>& audioManager, mute_state_t muteState) final;
protected:
DISALLOW_COPY_AND_ASSIGN(Track);
// AudioBufferProvider interface
status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override;
void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
// ExtendedAudioBufferProvider interface
size_t framesReady() const override;
int64_t framesReleased() const override;
void onTimestamp(const ExtendedTimestamp &timestamp) override;
// Used by thread
bool isPausing() const final { return mState == PAUSING; }
bool isPaused() const final { return mState == PAUSED; }
bool isResuming() const final { return mState == RESUMING; }
bool isReady() const final;
void setPaused() final { mState = PAUSED; }
void reset() final;
bool isFlushPending() const final { return mFlushHwPending; }
void flushAck() final;
bool isResumePending() const final;
void resumeAck() final;
// For direct or offloaded tracks ensure that the pause state is acknowledged
// by the playback thread in case of an immediate flush.
bool isPausePending() const final { return mPauseHwPending; }
void pauseAck() final;
void updateTrackFrameInfo(int64_t trackFramesReleased, int64_t sinkFramesWritten,
uint32_t halSampleRate, const ExtendedTimestamp& timeStamp) final;
sp<IMemory> sharedBuffer() const final { return mSharedBuffer; }
// presentationComplete checked by frames. (Mixed Tracks).
// framesWritten is cumulative, never reset, and is shared all tracks
// audioHalFrames is derived from output latency
bool presentationComplete(int64_t framesWritten, size_t audioHalFrames) final;
// presentationComplete checked by time. (Direct Tracks).
bool presentationComplete(uint32_t latencyMs) final;
void resetPresentationComplete() final {
mPresentationCompleteFrames = 0;
mPresentationCompleteTimeNs = 0;
}
// notifyPresentationComplete is called when presentationComplete() detects
// that the track is finished stopping.
void notifyPresentationComplete();
void signalClientFlag(int32_t flag);
void triggerEvents(AudioSystem::sync_event_t type) final;
void invalidate() final;
void disable() final;
int& fastIndex() final { return mFastIndex; }
bool isPlaybackRestricted() const final {
// The monitor is only created for tracks that can be silenced.
return mOpPlayAudioMonitor ? !mOpPlayAudioMonitor->hasOpPlayAudio() : false; }
const sp<AudioTrackServerProxy>& audioTrackServerProxy() const final {
return mAudioTrackServerProxy;
}
bool hasVolumeController() const final { return mHasVolumeController; }
void setHasVolumeController(bool hasVolumeController) final {
mHasVolumeController = hasVolumeController;
}
void setCachedVolume(float volume) final {
mCachedVolume = volume;
}
void setResetDone(bool resetDone) final {
mResetDone = resetDone;
}
ExtendedAudioBufferProvider* asExtendedAudioBufferProvider() final {
return this;
}
VolumeProvider* asVolumeProvider() final {
return this;
}
FillingStatus& fillingStatus() final { return mFillingStatus; }
int8_t& retryCount() final { return mRetryCount; }
FastTrackUnderruns& fastTrackUnderruns() final { return mObservedUnderruns; }
protected:
mutable FillingStatus mFillingStatus;
int8_t mRetryCount;
// see comment at ~Track for why this can't be const
sp<IMemory> mSharedBuffer;
bool mResetDone;
const audio_stream_type_t mStreamType;
float *mMainBuffer;
int32_t *mAuxBuffer;
int mAuxEffectId;
bool mHasVolumeController;
// access these three variables only when holding thread lock.
LinearMap<int64_t> mFrameMap; // track frame to server frame mapping
ExtendedTimestamp mSinkTimestamp;
sp<media::VolumeHandler> mVolumeHandler; // handles multiple VolumeShaper configs and operations
sp<OpPlayAudioMonitor> mOpPlayAudioMonitor;
bool mHapticPlaybackEnabled = false; // indicates haptic playback enabled or not
// intensity to play haptic data
os::HapticScale mHapticIntensity = os::HapticScale::MUTE;
// max amplitude allowed for haptic data
float mHapticMaxAmplitude = NAN;
class AudioVibrationController : public os::BnExternalVibrationController {
public:
explicit AudioVibrationController(Track* track) : mTrack(track) {}
binder::Status mute(/*out*/ bool *ret) override;
binder::Status unmute(/*out*/ bool *ret) override;
private:
Track* const mTrack;
bool setMute(bool muted);
};
sp<AudioVibrationController> mAudioVibrationController;
sp<os::ExternalVibration> mExternalVibration;
audio_dual_mono_mode_t mDualMonoMode = AUDIO_DUAL_MONO_MODE_OFF;
float mAudioDescriptionMixLevel = -std::numeric_limits<float>::infinity();
audio_playback_rate_t mPlaybackRateParameters = AUDIO_PLAYBACK_RATE_INITIALIZER;
private:
void interceptBuffer(const AudioBufferProvider::Buffer& buffer);
// Must hold thread lock to access tee patches
template <class F>
void forEachTeePatchTrack_l(F f) {
for (auto& tp : mTeePatches) { f(tp.patchTrack); }
};
size_t mPresentationCompleteFrames = 0; // (Used for Mixed tracks)
// The number of frames written to the
// audio HAL when this track is considered fully rendered.
// Zero means not monitoring.
int64_t mPresentationCompleteTimeNs = 0; // (Used for Direct tracks)
// The time when this track is considered fully rendered.
// Zero means not monitoring.
// The following fields are only for fast tracks, and should be in a subclass
int mFastIndex; // index within FastMixerState::mFastTracks[];
// either mFastIndex == -1 if not isFastTrack()
// or 0 < mFastIndex < FastMixerState::kMaxFast because
// index 0 is reserved for normal mixer's submix;
// index is allocated statically at track creation time
// but the slot is only used if track is active
FastTrackUnderruns mObservedUnderruns; // Most recently observed value of
// mFastMixerDumpState.mTracks[mFastIndex].mUnderruns
volatile float mCachedVolume; // combined master volume and stream type volume;
// 'volatile' means accessed without lock or
// barrier, but is read/written atomically
float mFinalVolume; // combine master volume, stream type volume and track volume
float mFinalVolumeLeft; // combine master volume, stream type volume and track
// volume
float mFinalVolumeRight; // combine master volume, stream type volume and track
// volume
sp<AudioTrackServerProxy> mAudioTrackServerProxy;
bool mResumeToStopping; // track was paused in stopping state.
bool mFlushHwPending; // track requests for thread flush
bool mPauseHwPending = false; // direct/offload track request for thread pause
audio_output_flags_t mFlags;
TeePatches mTeePatches;
std::optional<TeePatches> mTeePatchesToUpdate;
const float mSpeed;
const bool mIsSpatialized;
const bool mIsBitPerfect;
// TODO: replace PersistableBundle with own struct
// access these two variables only when holding player thread lock.
std::unique_ptr<os::PersistableBundle> mMuteEventExtras;
mute_state_t mMuteState;
}; // end of Track
// playback track, used by DuplicatingThread
class OutputTrack : public Track, public IAfOutputTrack {
public:
class Buffer : public AudioBufferProvider::Buffer {
public:
void *mBuffer;
};
OutputTrack(IAfPlaybackThread* thread,
IAfDuplicatingThread* sourceThread,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const AttributionSourceState& attributionSource);
~OutputTrack() override;
status_t start(AudioSystem::sync_event_t event =
AudioSystem::SYNC_EVENT_NONE,
audio_session_t triggerSession = AUDIO_SESSION_NONE) final;
void stop() final;
ssize_t write(void* data, uint32_t frames) final;
bool bufferQueueEmpty() const final { return mBufferQueue.size() == 0; }
bool isActive() const final { return mActive; }
void copyMetadataTo(MetadataInserter& backInserter) const final;
/** Set the metadatas of the upstream tracks. Thread safe. */
void setMetadatas(const SourceMetadatas& metadatas) final;
/** returns client timestamp to the upstream duplicating thread. */
ExtendedTimestamp getClientProxyTimestamp() const final {
// server - kernel difference is not true latency when drained
// i.e. mServerProxy->isDrained().
ExtendedTimestamp timestamp;
(void) mClientProxy->getTimestamp(&timestamp);
// On success, the timestamp LOCATION_SERVER and LOCATION_KERNEL
// entries will be properly filled. If getTimestamp()
// is unsuccessful, then a default initialized timestamp
// (with mTimeNs[] filled with -1's) is returned.
return timestamp;
}
private:
status_t obtainBuffer(AudioBufferProvider::Buffer* buffer,
uint32_t waitTimeMs);
void queueBuffer(Buffer& inBuffer);
void clearBufferQueue();
void restartIfDisabled();
// Maximum number of pending buffers allocated by OutputTrack::write()
static const uint8_t kMaxOverFlowBuffers = 10;
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
bool mActive;
IAfDuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
sp<AudioTrackClientProxy> mClientProxy;
/** Attributes of the source tracks.
*
* This member must be accessed with mTrackMetadatasMutex taken.
* There is one writer (duplicating thread) and one reader (downstream mixer).
*
* That means that the duplicating thread can block the downstream mixer
* thread and vice versa for the time of the copy.
* If this becomes an issue, the metadata could be stored in an atomic raw pointer,
* and a exchange with nullptr and delete can be used.
* Alternatively a read-copy-update might be implemented.
*/
SourceMetadatas mTrackMetadatas;
/** Protects mTrackMetadatas against concurrent access. */
audio_utils::mutex& trackMetadataMutex() const { return mTrackMetadataMutex; }
mutable audio_utils::mutex mTrackMetadataMutex;
}; // end of OutputTrack
// playback track, used by PatchPanel
class PatchTrack : public Track, public PatchTrackBase, public IAfPatchTrack {
public:
PatchTrack(IAfPlaybackThread* playbackThread,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_output_flags_t flags,
const Timeout& timeout = {},
size_t frameCountToBeReady = 1 /** Default behaviour is to start
* as soon as possible to have
* the lowest possible latency
* even if it might glitch. */);
~PatchTrack() override;
size_t framesReady() const final;
status_t start(AudioSystem::sync_event_t event =
AudioSystem::SYNC_EVENT_NONE,
audio_session_t triggerSession = AUDIO_SESSION_NONE) final;
// AudioBufferProvider interface
status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) final;
void releaseBuffer(AudioBufferProvider::Buffer* buffer) final;
// PatchProxyBufferProvider interface
status_t obtainBuffer(Proxy::Buffer* buffer, const struct timespec* timeOut = nullptr) final;
void releaseBuffer(Proxy::Buffer* buffer) final;
private:
void restartIfDisabled();
}; // end of PatchTrack
} // namespace android