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android_frameworks_base/services/audioflinger/AudioFlinger.h
Glenn Kasten 685c9ce3bd Avoid wp<>::unsafe_get() with a few exceptions 
Avoid using wp<>::unsafe_get() except in a log, and other specific cases
when it's known to be safe.

Use more specific subclass types for parameters to avoid down-casts.

When a constructor or method parameter is "this" of an object that is
currently being constructed, it's better to use a raw pointer rather
than either sp<> or wp<>.

Using the raw pointer is safe, provided either:
 - it is "this" of an object being constructed (which has sp<> refcount of 0),
 - or the caller already holds an sp<>

The raw pointer is simpler and faster, and it avoids the problem of the
sp<> reference count being incremented and then decremented to zero on
scope exit, which would cause the object's destructor to run while the
object is still being constructed.

Also removed some dead code per a review comment.

Change-Id: I7375f64da3aec11b928c33cb01faff186252ef5e
2012-02-22 13:19:26 -08:00

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/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#ifndef ANDROID_AUDIO_FLINGER_H
#define ANDROID_AUDIO_FLINGER_H
#include <stdint.h>
#include <sys/types.h>
#include <limits.h>
#include <common_time/cc_helper.h>
#include <media/IAudioFlinger.h>
#include <media/IAudioFlingerClient.h>
#include <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
#include <media/AudioSystem.h>
#include <media/AudioTrack.h>
#include <utils/Atomic.h>
#include <utils/Errors.h>
#include <utils/threads.h>
#include <utils/SortedVector.h>
#include <utils/TypeHelpers.h>
#include <utils/Vector.h>
#include <binder/BinderService.h>
#include <binder/MemoryDealer.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include "AudioBufferProvider.h"
#include <powermanager/IPowerManager.h>
namespace android {
class audio_track_cblk_t;
class effect_param_cblk_t;
class AudioMixer;
class AudioBuffer;
class AudioResampler;
// ----------------------------------------------------------------------------
static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
class AudioFlinger :
public BinderService<AudioFlinger>,
public BnAudioFlinger
{
friend class BinderService<AudioFlinger>;
public:
static const char* getServiceName() { return "media.audio_flinger"; }
virtual status_t dump(int fd, const Vector<String16>& args);
// IAudioFlinger interface, in binder opcode order
virtual sp<IAudioTrack> createTrack(
pid_t pid,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
bool isTimed,
int *sessionId,
status_t *status);
virtual sp<IAudioRecord> openRecord(
pid_t pid,
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
int *sessionId,
status_t *status);
virtual uint32_t sampleRate(audio_io_handle_t output) const;
virtual int channelCount(audio_io_handle_t output) const;
virtual audio_format_t format(audio_io_handle_t output) const;
virtual size_t frameCount(audio_io_handle_t output) const;
virtual uint32_t latency(audio_io_handle_t output) const;
virtual status_t setMasterVolume(float value);
virtual status_t setMasterMute(bool muted);
virtual float masterVolume() const;
virtual float masterVolumeSW() const;
virtual bool masterMute() const;
virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
audio_io_handle_t output);
virtual status_t setStreamMute(audio_stream_type_t stream, bool muted);
virtual float streamVolume(audio_stream_type_t stream,
audio_io_handle_t output) const;
virtual bool streamMute(audio_stream_type_t stream) const;
virtual status_t setMode(audio_mode_t mode);
virtual status_t setMicMute(bool state);
virtual bool getMicMute() const;
virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
virtual void registerClient(const sp<IAudioFlingerClient>& client);
virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const;
virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags);
virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2);
virtual status_t closeOutput(audio_io_handle_t output);
virtual status_t suspendOutput(audio_io_handle_t output);
virtual status_t restoreOutput(audio_io_handle_t output);
virtual audio_io_handle_t openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
audio_in_acoustics_t acoustics);
virtual status_t closeInput(audio_io_handle_t input);
virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output);
virtual status_t setVoiceVolume(float volume);
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
audio_io_handle_t output) const;
virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const;
virtual int newAudioSessionId();
virtual void acquireAudioSessionId(int audioSession);
virtual void releaseAudioSessionId(int audioSession);
virtual status_t queryNumberEffects(uint32_t *numEffects) const;
virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
effect_descriptor_t *descriptor) const;
virtual sp<IEffect> createEffect(pid_t pid,
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
audio_io_handle_t io,
int sessionId,
status_t *status,
int *id,
int *enabled);
virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
audio_io_handle_t dstOutput);
virtual status_t onTransact(
uint32_t code,
const Parcel& data,
Parcel* reply,
uint32_t flags);
// end of IAudioFlinger interface
private:
audio_mode_t getMode() const { return mMode; }
bool btNrecIsOff() const { return mBtNrecIsOff; }
AudioFlinger();
virtual ~AudioFlinger();
// call in any IAudioFlinger method that accesses mPrimaryHardwareDev
status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; }
virtual void onFirstRef();
audio_hw_device_t* findSuitableHwDev_l(uint32_t devices);
void purgeStaleEffects_l();
static nsecs_t mStandbyTimeInNsecs;
// Internal dump utilites.
status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
status_t dumpClients(int fd, const Vector<String16>& args);
status_t dumpInternals(int fd, const Vector<String16>& args);
// --- Client ---
class Client : public RefBase {
public:
Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
virtual ~Client();
sp<MemoryDealer> heap() const;
pid_t pid() const { return mPid; }
sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; }
bool reserveTimedTrack();
void releaseTimedTrack();
private:
Client(const Client&);
Client& operator = (const Client&);
const sp<AudioFlinger> mAudioFlinger;
const sp<MemoryDealer> mMemoryDealer;
const pid_t mPid;
Mutex mTimedTrackLock;
int mTimedTrackCount;
};
// --- Notification Client ---
class NotificationClient : public IBinder::DeathRecipient {
public:
NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
pid_t pid);
virtual ~NotificationClient();
sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
private:
NotificationClient(const NotificationClient&);
NotificationClient& operator = (const NotificationClient&);
const sp<AudioFlinger> mAudioFlinger;
const pid_t mPid;
const sp<IAudioFlingerClient> mAudioFlingerClient;
};
class TrackHandle;
class RecordHandle;
class RecordThread;
class PlaybackThread;
class MixerThread;
class DirectOutputThread;
class DuplicatingThread;
class Track;
class RecordTrack;
class EffectModule;
class EffectHandle;
class EffectChain;
struct AudioStreamOut;
struct AudioStreamIn;
class ThreadBase : public Thread {
public:
enum type_t {
MIXER, // Thread class is MixerThread
DIRECT, // Thread class is DirectOutputThread
DUPLICATING, // Thread class is DuplicatingThread
RECORD // Thread class is RecordThread
};
ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type);
virtual ~ThreadBase();
status_t dumpBase(int fd, const Vector<String16>& args);
status_t dumpEffectChains(int fd, const Vector<String16>& args);
void clearPowerManager();
// base for record and playback
class TrackBase : public AudioBufferProvider, public RefBase {
public:
enum track_state {
IDLE,
TERMINATED,
// These are order-sensitive; do not change order without reviewing the impact.
// In particular there are assumptions about > STOPPED.
STOPPED,
RESUMING,
ACTIVE,
PAUSING,
PAUSED
};
enum track_flags {
STEPSERVER_FAILED = 0x01, // StepServer could not acquire cblk->lock mutex
SYSTEM_FLAGS_MASK = 0x0000ffffUL,
// The upper 16 bits are used for track-specific flags.
};
TrackBase(ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId);
virtual ~TrackBase();
virtual status_t start(pid_t tid) = 0;
virtual void stop() = 0;
sp<IMemory> getCblk() const { return mCblkMemory; }
audio_track_cblk_t* cblk() const { return mCblk; }
int sessionId() const { return mSessionId; }
protected:
friend class ThreadBase;
friend class RecordHandle;
friend class PlaybackThread;
friend class RecordThread;
friend class MixerThread;
friend class DirectOutputThread;
TrackBase(const TrackBase&);
TrackBase& operator = (const TrackBase&);
virtual status_t getNextBuffer(
AudioBufferProvider::Buffer* buffer,
int64_t pts) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
audio_format_t format() const {
return mFormat;
}
int channelCount() const { return mChannelCount; }
uint32_t channelMask() const { return mChannelMask; }
int sampleRate() const; // FIXME inline after cblk sr moved
void* getBuffer(uint32_t offset, uint32_t frames) const;
bool isStopped() const {
return mState == STOPPED;
}
bool isTerminated() const {
return mState == TERMINATED;
}
bool step();
void reset();
const wp<ThreadBase> mThread;
/*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk;
void* mBuffer;
void* mBufferEnd;
uint32_t mFrameCount;
// we don't really need a lock for these
track_state mState;
const audio_format_t mFormat;
uint32_t mFlags;
const int mSessionId;
uint8_t mChannelCount;
uint32_t mChannelMask;
};
class ConfigEvent {
public:
ConfigEvent() : mEvent(0), mParam(0) {}
int mEvent;
int mParam;
};
class PMDeathRecipient : public IBinder::DeathRecipient {
public:
PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
virtual ~PMDeathRecipient() {}
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
private:
PMDeathRecipient(const PMDeathRecipient&);
PMDeathRecipient& operator = (const PMDeathRecipient&);
wp<ThreadBase> mThread;
};
virtual status_t initCheck() const = 0;
type_t type() const { return mType; }
uint32_t sampleRate() const { return mSampleRate; }
int channelCount() const { return mChannelCount; }
audio_format_t format() const { return mFormat; }
size_t frameCount() const { return mFrameCount; }
void wakeUp() { mWaitWorkCV.broadcast(); }
// Should be "virtual status_t requestExitAndWait()" and override same
// method in Thread, but Thread::requestExitAndWait() is not yet virtual.
void exit();
virtual bool checkForNewParameters_l() = 0;
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys) = 0;
virtual void audioConfigChanged_l(int event, int param = 0) = 0;
void sendConfigEvent(int event, int param = 0);
void sendConfigEvent_l(int event, int param = 0);
void processConfigEvents();
audio_io_handle_t id() const { return mId;}
bool standby() { return mStandby; }
uint32_t device() { return mDevice; }
virtual audio_stream_t* stream() = 0;
sp<EffectHandle> createEffect_l(
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority,
int sessionId,
effect_descriptor_t *desc,
int *enabled,
status_t *status);
void disconnectEffect(const sp< EffectModule>& effect,
const wp<EffectHandle>& handle,
bool unpinIfLast);
// 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
};
// get effect chain corresponding to session Id.
sp<EffectChain> getEffectChain(int sessionId);
// same as getEffectChain() but must be called with ThreadBase mutex locked
sp<EffectChain> getEffectChain_l(int sessionId);
// add an effect chain to the chain list (mEffectChains)
virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0;
// remove an effect chain from the chain list (mEffectChains)
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0;
// lock mall 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.
void lockEffectChains_l(Vector<sp <EffectChain> >& effectChains);
// unlock effect chains after process
void unlockEffectChains(Vector<sp <EffectChain> >& effectChains);
// set audio mode to all effect chains
void setMode(audio_mode_t mode);
// get effect module with corresponding ID on specified audio session
sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId);
// add and effect module. Also creates the effect chain is none exists for
// the effects audio session
status_t addEffect_l(const sp< EffectModule>& effect);
// remove and effect module. Also removes the effect chain is this was the last
// effect
void removeEffect_l(const sp< EffectModule>& effect);
// detach all tracks connected to an auxiliary effect
virtual void detachAuxEffect_l(int effectId) {}
// returns either EFFECT_SESSION if effects on this audio session exist in one
// chain, or TRACK_SESSION if tracks on this audio session exist, or both
virtual uint32_t hasAudioSession(int sessionId) = 0;
// the value returned by default implementation is not important as the
// strategy is only meaningful for PlaybackThread which implements this method
virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
// suspend or restore effect according to the type of effect passed. a NULL
// type pointer means suspend all effects in the session
void setEffectSuspended(const effect_uuid_t *type,
bool suspend,
int sessionId = AUDIO_SESSION_OUTPUT_MIX);
// check if some effects must be suspended/restored when an effect is enabled
// or disabled
void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
bool enabled,
int sessionId = AUDIO_SESSION_OUTPUT_MIX);
void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
bool enabled,
int sessionId = AUDIO_SESSION_OUTPUT_MIX);
mutable Mutex mLock;
protected:
// entry describing an effect being suspended in mSuspendedSessions keyed vector
class SuspendedSessionDesc : public RefBase {
public:
SuspendedSessionDesc() : mRefCount(0) {}
int mRefCount; // number of active suspend requests
effect_uuid_t mType; // effect type UUID
};
void acquireWakeLock();
void acquireWakeLock_l();
void releaseWakeLock();
void releaseWakeLock_l();
void setEffectSuspended_l(const effect_uuid_t *type,
bool suspend,
int sessionId = AUDIO_SESSION_OUTPUT_MIX);
// updated mSuspendedSessions when an effect suspended or restored
void updateSuspendedSessions_l(const effect_uuid_t *type,
bool suspend,
int sessionId);
// check if some effects must be suspended when an effect chain is added
void checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain);
friend class AudioFlinger;
friend class Track;
friend class TrackBase;
friend class PlaybackThread;
friend class MixerThread;
friend class DirectOutputThread;
friend class DuplicatingThread;
friend class RecordThread;
friend class RecordTrack;
const type_t mType;
Condition mWaitWorkCV;
const sp<AudioFlinger> mAudioFlinger;
uint32_t mSampleRate;
size_t mFrameCount;
uint32_t mChannelMask;
uint16_t mChannelCount;
size_t mFrameSize;
audio_format_t mFormat;
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
Vector<ConfigEvent> mConfigEvents;
bool mStandby;
const audio_io_handle_t mId;
Vector< sp<EffectChain> > mEffectChains;
uint32_t mDevice; // output device for PlaybackThread
// input + output devices for RecordThread
static const int kNameLength = 32;
char mName[kNameLength];
sp<IPowerManager> mPowerManager;
sp<IBinder> mWakeLockToken;
const sp<PMDeathRecipient> mDeathRecipient;
// list of suspended effects per session and per type. The first vector is
// keyed by session ID, the second by type UUID timeLow field
KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions;
};
struct stream_type_t {
stream_type_t()
: volume(1.0f),
mute(false),
valid(true)
{
}
float volume;
bool mute;
bool valid;
};
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase {
public:
enum mixer_state {
MIXER_IDLE,
MIXER_TRACKS_ENABLED,
MIXER_TRACKS_READY
};
// playback track
class Track : public TrackBase {
public:
Track( PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
virtual ~Track();
void dump(char* buffer, size_t size);
virtual status_t start(pid_t tid);
virtual void stop();
void pause();
void flush();
void destroy();
void mute(bool);
int name() const {
return mName;
}
audio_stream_type_t streamType() const {
return mStreamType;
}
status_t attachAuxEffect(int EffectId);
void setAuxBuffer(int EffectId, int32_t *buffer);
int32_t *auxBuffer() const { return mAuxBuffer; }
void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
int16_t *mainBuffer() const { return mMainBuffer; }
int auxEffectId() const { return mAuxEffectId; }
protected:
friend class ThreadBase;
friend class TrackHandle;
friend class PlaybackThread;
friend class MixerThread;
friend class DirectOutputThread;
Track(const Track&);
Track& operator = (const Track&);
virtual status_t getNextBuffer(
AudioBufferProvider::Buffer* buffer,
int64_t pts);
virtual uint32_t framesReady() const;
bool isMuted() const { return mMute; }
bool isPausing() const {
return mState == PAUSING;
}
bool isPaused() const {
return mState == PAUSED;
}
bool isReady() const;
void setPaused() { mState = PAUSED; }
void reset();
bool isOutputTrack() const {
return (mStreamType == AUDIO_STREAM_CNT);
}
virtual bool isTimedTrack() const { return false; }
// we don't really need a lock for these
volatile bool mMute;
// FILLED state is used for suppressing volume ramp at begin of playing
enum {FS_FILLING, FS_FILLED, FS_ACTIVE};
mutable uint8_t mFillingUpStatus;
int8_t mRetryCount;
sp<IMemory> mSharedBuffer;
bool mResetDone;
audio_stream_type_t mStreamType;
int mName;
int16_t *mMainBuffer;
int32_t *mAuxBuffer;
int mAuxEffectId;
bool mHasVolumeController;
}; // end of Track
class TimedTrack : public Track {
public:
static sp<TimedTrack> create(PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
~TimedTrack();
class TimedBuffer {
public:
TimedBuffer();
TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
const sp<IMemory>& buffer() const { return mBuffer; }
int64_t pts() const { return mPTS; }
int position() const { return mPosition; }
void setPosition(int pos) { mPosition = pos; }
private:
sp<IMemory> mBuffer;
int64_t mPTS;
int mPosition;
};
virtual bool isTimedTrack() const { return true; }
virtual uint32_t framesReady() const;
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
void timedYieldSamples(AudioBufferProvider::Buffer* buffer);
void timedYieldSilence(uint32_t numFrames,
AudioBufferProvider::Buffer* buffer);
status_t allocateTimedBuffer(size_t size,
sp<IMemory>* buffer);
status_t queueTimedBuffer(const sp<IMemory>& buffer,
int64_t pts);
status_t setMediaTimeTransform(const LinearTransform& xform,
TimedAudioTrack::TargetTimeline target);
void trimTimedBufferQueue_l();
private:
TimedTrack(PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
uint64_t mLocalTimeFreq;
LinearTransform mLocalTimeToSampleTransform;
sp<MemoryDealer> mTimedMemoryDealer;
Vector<TimedBuffer> mTimedBufferQueue;
uint8_t* mTimedSilenceBuffer;
uint32_t mTimedSilenceBufferSize;
mutable Mutex mTimedBufferQueueLock;
bool mTimedAudioOutputOnTime;
CCHelper mCCHelper;
Mutex mMediaTimeTransformLock;
LinearTransform mMediaTimeTransform;
bool mMediaTimeTransformValid;
TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
};
// playback track
class OutputTrack : public Track {
public:
class Buffer: public AudioBufferProvider::Buffer {
public:
int16_t *mBuffer;
};
OutputTrack(PlaybackThread *thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount);
virtual ~OutputTrack();
virtual status_t start(pid_t tid);
virtual void stop();
bool write(int16_t* data, uint32_t frames);
bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
bool isActive() const { return mActive; }
const wp<ThreadBase>& thread() const { return mThread; }
private:
enum {
NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value
};
status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
void clearBufferQueue();
// Maximum number of pending buffers allocated by OutputTrack::write()
static const uint8_t kMaxOverFlowBuffers = 10;
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
bool mActive;
DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
}; // end of OutputTrack
PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, uint32_t device, type_t type);
virtual ~PlaybackThread();
virtual status_t dump(int fd, const Vector<String16>& args);
// Thread virtuals
virtual status_t readyToRun();
virtual void onFirstRef();
virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
virtual uint32_t latency() const;
void setMasterVolume(float value);
void setMasterMute(bool muted);
void setStreamVolume(audio_stream_type_t stream, float value);
void setStreamMute(audio_stream_type_t stream, bool muted);
float streamVolume(audio_stream_type_t stream) const;
sp<Track> createTrack_l(
const sp<AudioFlinger::Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
bool isTimed,
status_t *status);
AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
virtual audio_stream_t* stream();
void suspend() { mSuspended++; }
void restore() { if (mSuspended) mSuspended--; }
bool isSuspended() const { return (mSuspended != 0); }
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
int16_t *mixBuffer() const { return mMixBuffer; };
virtual void detachAuxEffect_l(int effectId);
status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
int EffectId);
status_t attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track,
int EffectId);
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
virtual uint32_t hasAudioSession(int sessionId);
virtual uint32_t getStrategyForSession_l(int sessionId);
void setStreamValid(audio_stream_type_t streamType, bool valid);
protected:
int16_t* mMixBuffer;
int mSuspended;
int mBytesWritten;
private:
// mMasterMute is in both PlaybackThread and in AudioFlinger. When a
// PlaybackThread needs to find out if master-muted, it checks it's local
// copy rather than the one in AudioFlinger. This optimization saves a lock.
bool mMasterMute;
void setMasterMute_l(bool muted) { mMasterMute = muted; }
protected:
SortedVector< wp<Track> > mActiveTracks;
virtual int getTrackName_l() = 0;
virtual void deleteTrackName_l(int name) = 0;
virtual uint32_t activeSleepTimeUs();
virtual uint32_t idleSleepTimeUs() = 0;
virtual uint32_t suspendSleepTimeUs() = 0;
private:
friend class AudioFlinger;
friend class OutputTrack;
friend class Track;
friend class TrackBase;
friend class MixerThread;
friend class DirectOutputThread;
friend class DuplicatingThread;
PlaybackThread(const Client&);
PlaybackThread& operator = (const PlaybackThread&);
status_t addTrack_l(const sp<Track>& track);
void destroyTrack_l(const sp<Track>& track);
void removeTrack_l(const sp<Track>& track);
void readOutputParameters();
virtual status_t dumpInternals(int fd, const Vector<String16>& args);
status_t dumpTracks(int fd, const Vector<String16>& args);
SortedVector< sp<Track> > mTracks;
// mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread
stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1];
AudioStreamOut *mOutput;
float mMasterVolume;
nsecs_t mLastWriteTime;
int mNumWrites;
int mNumDelayedWrites;
bool mInWrite;
};
class MixerThread : public PlaybackThread {
public:
MixerThread (const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
uint32_t device,
type_t type = MIXER);
virtual ~MixerThread();
// Thread virtuals
virtual bool threadLoop();
void invalidateTracks(audio_stream_type_t streamType);
virtual bool checkForNewParameters_l();
virtual status_t dumpInternals(int fd, const Vector<String16>& args);
protected:
mixer_state prepareTracks_l(const SortedVector< wp<Track> >& activeTracks,
Vector< sp<Track> > *tracksToRemove);
virtual int getTrackName_l();
virtual void deleteTrackName_l(int name);
virtual uint32_t idleSleepTimeUs();
virtual uint32_t suspendSleepTimeUs();
AudioMixer* mAudioMixer;
mixer_state mPrevMixerStatus; // previous status returned by prepareTracks_l()
};
class DirectOutputThread : public PlaybackThread {
public:
DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, uint32_t device);
virtual ~DirectOutputThread();
// Thread virtuals
virtual bool threadLoop();
virtual bool checkForNewParameters_l();
protected:
virtual int getTrackName_l();
virtual void deleteTrackName_l(int name);
virtual uint32_t activeSleepTimeUs();
virtual uint32_t idleSleepTimeUs();
virtual uint32_t suspendSleepTimeUs();
private:
void applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp);
// volumes last sent to audio HAL with stream->set_volume()
// FIXME use standard representation and names
float mLeftVolFloat;
float mRightVolFloat;
uint16_t mLeftVolShort;
uint16_t mRightVolShort;
};
class DuplicatingThread : public MixerThread {
public:
DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
audio_io_handle_t id);
virtual ~DuplicatingThread();
// Thread virtuals
virtual bool threadLoop();
void addOutputTrack(MixerThread* thread);
void removeOutputTrack(MixerThread* thread);
uint32_t waitTimeMs() { return mWaitTimeMs; }
protected:
virtual uint32_t activeSleepTimeUs();
private:
bool outputsReady(SortedVector< sp<OutputTrack> > &outputTracks);
void updateWaitTime();
SortedVector < sp<OutputTrack> > mOutputTracks;
uint32_t mWaitTimeMs;
};
PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
// no range check, AudioFlinger::mLock held
bool streamMute_l(audio_stream_type_t stream) const
{ return mStreamTypes[stream].mute; }
// no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held
float streamVolume_l(audio_stream_type_t stream) const
{ return mStreamTypes[stream].volume; }
void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2);
// allocate an audio_io_handle_t, session ID, or effect ID
uint32_t nextUniqueId();
status_t moveEffectChain_l(int sessionId,
PlaybackThread *srcThread,
PlaybackThread *dstThread,
bool reRegister);
PlaybackThread *primaryPlaybackThread_l();
uint32_t primaryOutputDevice_l();
friend class AudioBuffer;
// server side of the client's IAudioTrack
class TrackHandle : public android::BnAudioTrack {
public:
TrackHandle(const sp<PlaybackThread::Track>& track);
virtual ~TrackHandle();
virtual sp<IMemory> getCblk() const;
virtual status_t start(pid_t tid);
virtual void stop();
virtual void flush();
virtual void mute(bool);
virtual void pause();
virtual status_t attachAuxEffect(int effectId);
virtual status_t allocateTimedBuffer(size_t size,
sp<IMemory>* buffer);
virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
int64_t pts);
virtual status_t setMediaTimeTransform(const LinearTransform& xform,
int target);
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
const sp<PlaybackThread::Track> mTrack;
};
friend class Client;
friend class PlaybackThread::Track;
void removeClient_l(pid_t pid);
void removeNotificationClient(pid_t pid);
// record thread
class RecordThread : public ThreadBase, public AudioBufferProvider
{
public:
// record track
class RecordTrack : public TrackBase {
public:
RecordTrack(RecordThread *thread,
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
int sessionId);
virtual ~RecordTrack();
virtual status_t start(pid_t tid);
virtual void stop();
bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; }
void dump(char* buffer, size_t size);
private:
friend class AudioFlinger;
friend class RecordThread;
RecordTrack(const RecordTrack&);
RecordTrack& operator = (const RecordTrack&);
virtual status_t getNextBuffer(
AudioBufferProvider::Buffer* buffer,
int64_t pts);
bool mOverflow;
};
RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
uint32_t sampleRate,
uint32_t channels,
audio_io_handle_t id,
uint32_t device);
virtual ~RecordThread();
virtual bool threadLoop();
virtual status_t readyToRun();
virtual void onFirstRef();
virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
audio_format_t format,
int channelMask,
int frameCount,
uint32_t flags,
int sessionId,
status_t *status);
status_t start(RecordTrack* recordTrack);
status_t start(RecordTrack* recordTrack, pid_t tid);
void stop(RecordTrack* recordTrack);
status_t dump(int fd, const Vector<String16>& args);
AudioStreamIn* getInput() const;
AudioStreamIn* clearInput();
virtual audio_stream_t* stream();
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
virtual bool checkForNewParameters_l();
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
void readInputParameters();
virtual unsigned int getInputFramesLost();
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
virtual uint32_t hasAudioSession(int sessionId);
RecordTrack* track();
private:
RecordThread();
AudioStreamIn *mInput;
RecordTrack* mTrack;
sp<RecordTrack> mActiveTrack;
Condition mStartStopCond;
AudioResampler *mResampler;
int32_t *mRsmpOutBuffer;
int16_t *mRsmpInBuffer;
size_t mRsmpInIndex;
size_t mInputBytes;
const int mReqChannelCount;
const uint32_t mReqSampleRate;
ssize_t mBytesRead;
};
// server side of the client's IAudioRecord
class RecordHandle : public android::BnAudioRecord {
public:
RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
virtual ~RecordHandle();
virtual sp<IMemory> getCblk() const;
virtual status_t start(pid_t tid);
virtual void stop();
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
const sp<RecordThread::RecordTrack> mRecordTrack;
};
//--- Audio Effect Management
// EffectModule and EffectChain classes both have their own mutex to protect
// state changes or resource modifications. Always respect the following order
// if multiple mutexes must be acquired to avoid cross deadlock:
// AudioFlinger -> ThreadBase -> EffectChain -> EffectModule
// The EffectModule class is a wrapper object controlling the effect engine implementation
// in the effect library. It prevents concurrent calls to process() and command() functions
// from different client threads. It keeps a list of EffectHandle objects corresponding
// to all client applications using this effect and notifies applications of effect state,
// control or parameter changes. It manages the activation state machine to send appropriate
// reset, enable, disable commands to effect engine and provide volume
// ramping when effects are activated/deactivated.
// When controlling an auxiliary effect, the EffectModule also provides an input buffer used by
// the attached track(s) to accumulate their auxiliary channel.
class EffectModule: public RefBase {
public:
EffectModule(ThreadBase *thread,
const wp<AudioFlinger::EffectChain>& chain,
effect_descriptor_t *desc,
int id,
int sessionId);
virtual ~EffectModule();
enum effect_state {
IDLE,
RESTART,
STARTING,
ACTIVE,
STOPPING,
STOPPED,
DESTROYED
};
int id() const { return mId; }
void process();
void updateState();
status_t command(uint32_t cmdCode,
uint32_t cmdSize,
void *pCmdData,
uint32_t *replySize,
void *pReplyData);
void reset_l();
status_t configure();
status_t init();
effect_state state() const {
return mState;
}
uint32_t status() {
return mStatus;
}
int sessionId() const {
return mSessionId;
}
status_t setEnabled(bool enabled);
bool isEnabled() const;
bool isProcessEnabled() const;
void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; }
int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; }
void setOutBuffer(int16_t *buffer) { mConfig.outputCfg.buffer.s16 = buffer; }
int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; }
void setChain(const wp<EffectChain>& chain) { mChain = chain; }
void setThread(const wp<ThreadBase>& thread) { mThread = thread; }
const wp<ThreadBase>& thread() { return mThread; }
status_t addHandle(const sp<EffectHandle>& handle);
void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast);
size_t removeHandle (const wp<EffectHandle>& handle);
effect_descriptor_t& desc() { return mDescriptor; }
wp<EffectChain>& chain() { return mChain; }
status_t setDevice(uint32_t device);
status_t setVolume(uint32_t *left, uint32_t *right, bool controller);
status_t setMode(audio_mode_t mode);
status_t start();
status_t stop();
void setSuspended(bool suspended);
bool suspended() const;
sp<EffectHandle> controlHandle();
bool isPinned() const { return mPinned; }
void unPin() { mPinned = false; }
status_t dump(int fd, const Vector<String16>& args);
protected:
friend class EffectHandle;
friend class AudioFlinger;
bool mPinned;
// Maximum time allocated to effect engines to complete the turn off sequence
static const uint32_t MAX_DISABLE_TIME_MS = 10000;
EffectModule(const EffectModule&);
EffectModule& operator = (const EffectModule&);
status_t start_l();
status_t stop_l();
mutable Mutex mLock; // mutex for process, commands and handles list protection
wp<ThreadBase> mThread; // parent thread
wp<EffectChain> mChain; // parent effect chain
int mId; // this instance unique ID
int mSessionId; // audio session ID
effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
effect_config_t mConfig; // input and output audio configuration
effect_handle_t mEffectInterface; // Effect module C API
status_t mStatus; // initialization status
effect_state mState; // current activation state
Vector< wp<EffectHandle> > mHandles; // list of client handles
// First handle in mHandles has highest priority and controls the effect module
uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after
// sending disable command.
uint32_t mDisableWaitCnt; // current process() calls count during disable period.
bool mSuspended; // effect is suspended: temporarily disabled by framework
};
// The EffectHandle class implements the IEffect interface. It provides resources
// to receive parameter updates, keeps track of effect control
// ownership and state and has a pointer to the EffectModule object it is controlling.
// There is one EffectHandle object for each application controlling (or using)
// an effect module.
// The EffectHandle is obtained by calling AudioFlinger::createEffect().
class EffectHandle: public android::BnEffect {
public:
EffectHandle(const sp<EffectModule>& effect,
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority);
virtual ~EffectHandle();
// IEffect
virtual status_t enable();
virtual status_t disable();
virtual status_t command(uint32_t cmdCode,
uint32_t cmdSize,
void *pCmdData,
uint32_t *replySize,
void *pReplyData);
virtual void disconnect();
private:
void disconnect(bool unpinIfLast);
public:
virtual sp<IMemory> getCblk() const { return mCblkMemory; }
virtual status_t onTransact(uint32_t code, const Parcel& data,
Parcel* reply, uint32_t flags);
// Give or take control of effect module
// - hasControl: true if control is given, false if removed
// - signal: true client app should be signaled of change, false otherwise
// - enabled: state of the effect when control is passed
void setControl(bool hasControl, bool signal, bool enabled);
void commandExecuted(uint32_t cmdCode,
uint32_t cmdSize,
void *pCmdData,
uint32_t replySize,
void *pReplyData);
void setEnabled(bool enabled);
bool enabled() const { return mEnabled; }
// Getters
int id() const { return mEffect->id(); }
int priority() const { return mPriority; }
bool hasControl() const { return mHasControl; }
sp<EffectModule> effect() const { return mEffect; }
void dump(char* buffer, size_t size);
protected:
friend class AudioFlinger;
friend class EffectModule;
EffectHandle(const EffectHandle&);
EffectHandle& operator =(const EffectHandle&);
sp<EffectModule> mEffect; // pointer to controlled EffectModule
sp<IEffectClient> mEffectClient; // callback interface for client notifications
/*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect()
sp<IMemory> mCblkMemory; // shared memory for control block
effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory
uint8_t* mBuffer; // pointer to parameter area in shared memory
int mPriority; // client application priority to control the effect
bool mHasControl; // true if this handle is controlling the effect
bool mEnabled; // cached enable state: needed when the effect is
// restored after being suspended
};
// the EffectChain class represents a group of effects associated to one audio session.
// There can be any number of EffectChain objects per output mixer thread (PlaybackThread).
// The EffecChain with session ID 0 contains global effects applied to the output mix.
// Effects in this chain can be insert or auxiliary. Effects in other chains (attached to tracks)
// are insert only. The EffectChain maintains an ordered list of effect module, the order corresponding
// in the effect process order. When attached to a track (session ID != 0), it also provide it's own
// input buffer used by the track as accumulation buffer.
class EffectChain: public RefBase {
public:
EffectChain(const wp<ThreadBase>& wThread, int sessionId);
EffectChain(ThreadBase *thread, int sessionId);
virtual ~EffectChain();
// special key used for an entry in mSuspendedEffects keyed vector
// corresponding to a suspend all request.
static const int kKeyForSuspendAll = 0;
// minimum duration during which we force calling effect process when last track on
// a session is stopped or removed to allow effect tail to be rendered
static const int kProcessTailDurationMs = 1000;
void process_l();
void lock() {
mLock.lock();
}
void unlock() {
mLock.unlock();
}
status_t addEffect_l(const sp<EffectModule>& handle);
size_t removeEffect_l(const sp<EffectModule>& handle);
int sessionId() const { return mSessionId; }
void setSessionId(int sessionId) { mSessionId = sessionId; }
sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
sp<EffectModule> getEffectFromId_l(int id);
sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
bool setVolume_l(uint32_t *left, uint32_t *right);
void setDevice_l(uint32_t device);
void setMode_l(audio_mode_t mode);
void setInBuffer(int16_t *buffer, bool ownsBuffer = false) {
mInBuffer = buffer;
mOwnInBuffer = ownsBuffer;
}
int16_t *inBuffer() const {
return mInBuffer;
}
void setOutBuffer(int16_t *buffer) {
mOutBuffer = buffer;
}
int16_t *outBuffer() const {
return mOutBuffer;
}
void incTrackCnt() { android_atomic_inc(&mTrackCnt); }
void decTrackCnt() { android_atomic_dec(&mTrackCnt); }
int32_t trackCnt() const { return mTrackCnt;}
void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt);
mTailBufferCount = mMaxTailBuffers; }
void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); }
int32_t activeTrackCnt() const { return mActiveTrackCnt;}
uint32_t strategy() const { return mStrategy; }
void setStrategy(uint32_t strategy)
{ mStrategy = strategy; }
// suspend effect of the given type
void setEffectSuspended_l(const effect_uuid_t *type,
bool suspend);
// suspend all eligible effects
void setEffectSuspendedAll_l(bool suspend);
// check if effects should be suspend or restored when a given effect is enable or disabled
void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
bool enabled);
status_t dump(int fd, const Vector<String16>& args);
protected:
friend class AudioFlinger;
EffectChain(const EffectChain&);
EffectChain& operator =(const EffectChain&);
class SuspendedEffectDesc : public RefBase {
public:
SuspendedEffectDesc() : mRefCount(0) {}
int mRefCount;
effect_uuid_t mType;
wp<EffectModule> mEffect;
};
// get a list of effect modules to suspend when an effect of the type
// passed is enabled.
void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects);
// get an effect module if it is currently enable
sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type);
// true if the effect whose descriptor is passed can be suspended
// OEMs can modify the rules implemented in this method to exclude specific effect
// types or implementations from the suspend/restore mechanism.
bool isEffectEligibleForSuspend(const effect_descriptor_t& desc);
wp<ThreadBase> mThread; // parent mixer thread
Mutex mLock; // mutex protecting effect list
Vector<sp<EffectModule> > mEffects; // list of effect modules
int mSessionId; // audio session ID
int16_t *mInBuffer; // chain input buffer
int16_t *mOutBuffer; // chain output buffer
volatile int32_t mActiveTrackCnt; // number of active tracks connected
volatile int32_t mTrackCnt; // number of tracks connected
int32_t mTailBufferCount; // current effect tail buffer count
int32_t mMaxTailBuffers; // maximum effect tail buffers
bool mOwnInBuffer; // true if the chain owns its input buffer
int mVolumeCtrlIdx; // index of insert effect having control over volume
uint32_t mLeftVolume; // previous volume on left channel
uint32_t mRightVolume; // previous volume on right channel
uint32_t mNewLeftVolume; // new volume on left channel
uint32_t mNewRightVolume; // new volume on right channel
uint32_t mStrategy; // strategy for this effect chain
// mSuspendedEffects lists all effect currently suspended in the chain
// use effect type UUID timelow field as key. There is no real risk of identical
// timeLow fields among effect type UUIDs.
KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects;
};
// AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
// For emphasis, we could also make all pointers to them be "const *",
// but that would clutter the code unnecessarily.
struct AudioStreamOut {
audio_hw_device_t* const hwDev;
audio_stream_out_t* const stream;
AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) :
hwDev(dev), stream(out) {}
};
struct AudioStreamIn {
audio_hw_device_t* const hwDev;
audio_stream_in_t* const stream;
AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) :
hwDev(dev), stream(in) {}
};
// for mAudioSessionRefs only
struct AudioSessionRef {
// FIXME rename parameter names when fields get "m" prefix
AudioSessionRef(int sessionid_, pid_t pid_) :
sessionid(sessionid_), pid(pid_), cnt(1) {}
const int sessionid;
const pid_t pid;
int cnt;
};
friend class RecordThread;
friend class PlaybackThread;
enum master_volume_support {
// MVS_NONE:
// Audio HAL has no support for master volume, either setting or
// getting. All master volume control must be implemented in SW by the
// AudioFlinger mixing core.
MVS_NONE,
// MVS_SETONLY:
// Audio HAL has support for setting master volume, but not for getting
// master volume (original HAL design did not include a getter).
// AudioFlinger needs to keep track of the last set master volume in
// addition to needing to set an initial, default, master volume at HAL
// load time.
MVS_SETONLY,
// MVS_FULL:
// Audio HAL has support both for setting and getting master volume.
// AudioFlinger should send all set and get master volume requests
// directly to the HAL.
MVS_FULL,
};
mutable Mutex mLock;
DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client()
mutable Mutex mHardwareLock;
// These two fields are immutable after onFirstRef(), so no lock needed to access
audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
Vector<audio_hw_device_t*> mAudioHwDevs;
enum hardware_call_state {
AUDIO_HW_IDLE = 0,
AUDIO_HW_INIT,
AUDIO_HW_OUTPUT_OPEN,
AUDIO_HW_OUTPUT_CLOSE,
AUDIO_HW_INPUT_OPEN,
AUDIO_HW_INPUT_CLOSE,
AUDIO_HW_STANDBY,
AUDIO_HW_SET_MASTER_VOLUME,
AUDIO_HW_GET_ROUTING,
AUDIO_HW_SET_ROUTING,
AUDIO_HW_GET_MODE,
AUDIO_HW_SET_MODE,
AUDIO_HW_GET_MIC_MUTE,
AUDIO_HW_SET_MIC_MUTE,
AUDIO_SET_VOICE_VOLUME,
AUDIO_SET_PARAMETER,
AUDIO_HW_GET_INPUT_BUFFER_SIZE,
AUDIO_HW_GET_MASTER_VOLUME,
};
mutable hardware_call_state mHardwareStatus; // for dump only
DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads;
stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
// both are protected by mLock
float mMasterVolume;
float mMasterVolumeSW;
master_volume_support mMasterVolumeSupportLvl;
bool mMasterMute;
DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads;
DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients;
volatile int32_t mNextUniqueId; // updated by android_atomic_inc
audio_mode_t mMode;
bool mBtNrecIsOff;
// protected by mLock
Vector<AudioSessionRef*> mAudioSessionRefs;
float masterVolume_l() const;
float masterVolumeSW_l() const { return mMasterVolumeSW; }
bool masterMute_l() const { return mMasterMute; }
private:
sp<Client> registerPid_l(pid_t pid); // always returns non-0
};
// ----------------------------------------------------------------------------
}; // namespace android
#endif // ANDROID_AUDIO_FLINGER_H
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