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ara-mdk / platform / frameworks / av / 7da35f231725aa6e0b2ac8714a7d66471368df77 / . / services / audioflinger / AudioFlinger.h
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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 <hardware/audio_policy.h>
#include <media/AudioBufferProvider.h>
#include <media/ExtendedAudioBufferProvider.h>
#include "FastMixer.h"
#include <media/nbaio/NBAIO.h>
#include "AudioWatchdog.h"
#include <powermanager/IPowerManager.h>
namespace android {
class audio_track_cblk_t;
class effect_param_cblk_t;
class AudioMixer;
class AudioBuffer;
class AudioResampler;
class FastMixer;
// ----------------------------------------------------------------------------
// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback.
// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect.
// Adding full support for > 2 channel capture or playback would require more than simply changing
// this #define. There is an independent hard-coded upper limit in AudioMixer;
// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels.
// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions.
// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc.
#define FCC_2 2 // FCC_2 = Fixed Channel Count 2
static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
class AudioFlinger :
public BinderService<AudioFlinger>,
public BnAudioFlinger
{
friend class BinderService<AudioFlinger>; // for 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,
audio_channel_mask_t channelMask,
size_t frameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
pid_t tid,
int *sessionId,
status_t *status);
virtual sp<IAudioRecord> openRecord(
pid_t pid,
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
IAudioFlinger::track_flags_t flags,
pid_t tid,
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 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,
audio_channel_mask_t channelMask) const;
virtual audio_io_handle_t openOutput(audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
uint32_t *pLatencyMs,
audio_output_flags_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(audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask);
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 audio_module_handle_t loadHwModule(const char *name);
virtual uint32_t getPrimaryOutputSamplingRate();
virtual size_t getPrimaryOutputFrameCount();
virtual status_t onTransact(
uint32_t code,
const Parcel& data,
Parcel* reply,
uint32_t flags);
// end of IAudioFlinger interface
class SyncEvent;
typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ;
class SyncEvent : public RefBase {
public:
SyncEvent(AudioSystem::sync_event_t type,
int triggerSession,
int listenerSession,
sync_event_callback_t callBack,
void *cookie)
: mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession),
mCallback(callBack), mCookie(cookie)
{}
virtual ~SyncEvent() {}
void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); }
bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); }
void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; }
AudioSystem::sync_event_t type() const { return mType; }
int triggerSession() const { return mTriggerSession; }
int listenerSession() const { return mListenerSession; }
void *cookie() const { return mCookie; }
private:
const AudioSystem::sync_event_t mType;
const int mTriggerSession;
const int mListenerSession;
sync_event_callback_t mCallback;
void * const mCookie;
mutable Mutex mLock;
};
sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
int triggerSession,
int listenerSession,
sync_event_callback_t callBack,
void *cookie);
private:
class AudioHwDevice; // fwd declaration for findSuitableHwDev_l
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; }
// RefBase
virtual void onFirstRef();
AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module,
audio_devices_t devices);
void purgeStaleEffects_l();
// standby delay for MIXER and DUPLICATING playback threads is read from property
// ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
static nsecs_t mStandbyTimeInNsecs;
// Internal dump utilities.
void dumpPermissionDenial(int fd, const Vector<String16>& args);
void dumpClients(int fd, const Vector<String16>& args);
void 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,
audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
virtual ~ThreadBase();
void dumpBase(int fd, const Vector<String16>& args);
void dumpEffectChains(int fd, const Vector<String16>& args);
void clearPowerManager();
// base for record and playback
class TrackBase : public ExtendedAudioBufferProvider, public RefBase {
public:
enum track_state {
IDLE,
TERMINATED,
FLUSHED,
STOPPED,
// next 2 states are currently used for fast tracks only
STOPPING_1, // waiting for first underrun
STOPPING_2, // waiting for presentation complete
RESUMING,
ACTIVE,
PAUSING,
PAUSED
};
TrackBase(ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
virtual ~TrackBase();
virtual status_t start(AudioSystem::sync_event_t event,
int triggerSession) = 0;
virtual void stop() = 0;
sp<IMemory> getCblk() const { return mCblkMemory; }
audio_track_cblk_t* cblk() const { return mCblk; }
int sessionId() const { return mSessionId; }
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
protected:
TrackBase(const TrackBase&);
TrackBase& operator = (const TrackBase&);
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
// ExtendedAudioBufferProvider interface is only needed for Track,
// but putting it in TrackBase avoids the complexity of virtual inheritance
virtual size_t framesReady() const { return SIZE_MAX; }
audio_format_t format() const {
return mFormat;
}
int channelCount() const { return mChannelCount; }
audio_channel_mask_t channelMask() const { return mChannelMask; }
uint32_t sampleRate() const; // FIXME inline after cblk sr moved
// Return a pointer to the start of a contiguous slice of the track buffer.
// Parameter 'offset' is the requested start position, expressed in
// monotonically increasing frame units relative to the track epoch.
// Parameter 'frames' is the requested length, also in frame units.
// Always returns non-NULL. It is the caller's responsibility to
// verify that this will be successful; the result of calling this
// function with invalid 'offset' or 'frames' is undefined.
void* getBuffer(uint32_t offset, uint32_t frames) const;
bool isStopped() const {
return (mState == STOPPED || mState == FLUSHED);
}
// for fast tracks only
bool isStopping() const {
return mState == STOPPING_1 || mState == STOPPING_2;
}
bool isStopping_1() const {
return mState == STOPPING_1;
}
bool isStopping_2() const {
return mState == STOPPING_2;
}
bool isTerminated() const {
return mState == TERMINATED;
}
bool step(); // mStepCount is an implicit input
void reset();
virtual bool isOut() const = 0; // true for Track and TimedTrack, false for RecordTrack,
// this could be a track type if needed later
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; // start of track buffer, typically in shared memory
void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize
// is based on mChannelCount and 16-bit samples
uint32_t mStepCount; // saves AudioBufferProvider::Buffer::frameCount as of
// time of releaseBuffer() for later use by step()
// we don't really need a lock for these
track_state mState;
const uint32_t mSampleRate; // initial sample rate only; for tracks which
// support dynamic rates, the current value is in control block
const audio_format_t mFormat;
const audio_channel_mask_t mChannelMask;
const uint8_t mChannelCount;
const size_t mFrameSize; // AudioFlinger's view of frame size in shared memory,
// where for AudioTrack (but not AudioRecord),
// 8-bit PCM samples are stored as 16-bit
bool mStepServerFailed;
const int mSessionId;
Vector < sp<SyncEvent> >mSyncEvents;
};
enum {
CFG_EVENT_IO,
CFG_EVENT_PRIO
};
class ConfigEvent {
public:
ConfigEvent(int type) : mType(type) {}
virtual ~ConfigEvent() {}
int type() const { return mType; }
virtual void dump(char *buffer, size_t size) = 0;
private:
const int mType;
};
class IoConfigEvent : public ConfigEvent {
public:
IoConfigEvent(int event, int param) :
ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {}
virtual ~IoConfigEvent() {}
int event() const { return mEvent; }
int param() const { return mParam; }
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "IO event: event %d, param %d\n", mEvent, mParam);
}
private:
const int mEvent;
const int mParam;
};
class PrioConfigEvent : public ConfigEvent {
public:
PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) :
ConfigEvent(CFG_EVENT_PRIO), mPid(pid), mTid(tid), mPrio(prio) {}
virtual ~PrioConfigEvent() {}
pid_t pid() const { return mPid; }
pid_t tid() const { return mTid; }
int32_t prio() const { return mPrio; }
virtual void dump(char *buffer, size_t size) {
snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d\n", mPid, mTid, mPrio);
}
private:
const pid_t mPid;
const pid_t mTid;
const int32_t mPrio;
};
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;
// static externally-visible
type_t type() const { return mType; }
audio_io_handle_t id() const { return mId;}
// dynamic externally-visible
uint32_t sampleRate() const { return mSampleRate; }
int channelCount() const { return mChannelCount; }
audio_channel_mask_t channelMask() const { return mChannelMask; }
audio_format_t format() const { return mFormat; }
// Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
// and returns the normal mix buffer's frame count.
size_t frameCount() const { return mNormalFrameCount; }
// Return's the HAL's frame count i.e. fast mixer buffer size.
size_t frameCountHAL() const { return mFrameCount; }
// 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 sendIoConfigEvent(int event, int param = 0);
void sendIoConfigEvent_l(int event, int param = 0);
void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio);
void processConfigEvents();
// see note at declaration of mStandby, mOutDevice and mInDevice
bool standby() const { return mStandby; }
audio_devices_t outDevice() const { return mOutDevice; }
audio_devices_t inDevice() const { return mInDevice; }
virtual audio_stream_t* stream() const = 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,
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) const;
// add an effect chain to the chain list (mEffectChains)
virtual status_t addEffectChain_l(const sp<EffectChain>& chain) = 0;
// remove an effect chain from the chain list (mEffectChains)
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain) = 0;
// lock all effect chains Mutexes. Must be called before releasing the
// ThreadBase mutex before processing the mixer and effects. This guarantees the
// integrity of the chains during the process.
// Also sets the parameter 'effectChains' to current value of mEffectChains.
void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains);
// unlock effect chains after process
void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains);
// 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(int sessionId, int effectId);
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) const = 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);
virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0;
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0;
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);
// 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);
virtual void preExit() { }
friend class AudioFlinger; // for mEffectChains
const type_t mType;
// Used by parameters, config events, addTrack_l, exit
Condition mWaitWorkCV;
const sp<AudioFlinger> mAudioFlinger;
uint32_t mSampleRate;
size_t mFrameCount; // output HAL, direct output, record
size_t mNormalFrameCount; // normal mixer and effects
audio_channel_mask_t mChannelMask;
uint16_t mChannelCount;
size_t mFrameSize;
audio_format_t mFormat;
// Parameter sequence by client: binder thread calling setParameters():
// 1. Lock mLock
// 2. Append to mNewParameters
// 3. mWaitWorkCV.signal
// 4. mParamCond.waitRelative with timeout
// 5. read mParamStatus
// 6. mWaitWorkCV.signal
// 7. Unlock
//
// Parameter sequence by server: threadLoop calling checkForNewParameters_l():
// 1. Lock mLock
// 2. If there is an entry in mNewParameters proceed ...
// 2. Read first entry in mNewParameters
// 3. Process
// 4. Remove first entry from mNewParameters
// 5. Set mParamStatus
// 6. mParamCond.signal
// 7. mWaitWorkCV.wait with timeout (this is to avoid overwriting mParamStatus)
// 8. Unlock
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
Vector<ConfigEvent *> mConfigEvents;
// These fields are written and read by thread itself without lock or barrier,
// and read by other threads without lock or barrier via standby() , outDevice()
// and inDevice().
// Because of the absence of a lock or barrier, any other thread that reads
// these fields must use the information in isolation, or be prepared to deal
// with possibility that it might be inconsistent with other information.
bool mStandby; // Whether thread is currently in standby.
audio_devices_t mOutDevice; // output device
audio_devices_t mInDevice; // input device
audio_source_t mAudioSource; // (see audio.h, audio_source_t)
const audio_io_handle_t mId;
Vector< sp<EffectChain> > mEffectChains;
static const int kNameLength = 16; // prctl(PR_SET_NAME) limit
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)
{
}
float volume;
bool mute;
};
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase {
public:
enum mixer_state {
MIXER_IDLE, // no active tracks
MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready
MIXER_TRACKS_READY // at least one active track, and at least one track has data
// standby mode does not have an enum value
// suspend by audio policy manager is orthogonal to mixer state
};
// playback track
class Track : public TrackBase, public VolumeProvider {
public:
Track( PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t flags);
virtual ~Track();
static void appendDumpHeader(String8& result);
void dump(char* buffer, size_t size);
virtual status_t start(AudioSystem::sync_event_t event =
AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
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; }
// implement FastMixerState::VolumeProvider interface
virtual uint32_t getVolumeLR();
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
protected:
// for numerous
friend class PlaybackThread;
friend class MixerThread;
friend class DirectOutputThread;
Track(const Track&);
Track& operator = (const Track&);
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts = kInvalidPTS);
// releaseBuffer() not overridden
virtual size_t framesReady() const;
bool isMuted() const { return mMute; }
bool isPausing() const {
return mState == PAUSING;
}
bool isPaused() const {
return mState == PAUSED;
}
bool isResuming() const {
return mState == RESUMING;
}
bool isReady() const;
void setPaused() { mState = PAUSED; }
void reset();
bool isOutputTrack() const {
return (mStreamType == AUDIO_STREAM_CNT);
}
sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
// framesWritten is cumulative, never reset, and is shared all tracks
// audioHalFrames is derived from output latency
// FIXME parameters not needed, could get them from the thread
bool presentationComplete(size_t framesWritten, size_t audioHalFrames);
public:
void triggerEvents(AudioSystem::sync_event_t type);
virtual bool isTimedTrack() const { return false; }
bool isFastTrack() const { return (mFlags & IAudioFlinger::TRACK_FAST) != 0; }
virtual bool isOut() const;
protected:
// written by Track::mute() called by binder thread(s), without a mutex or barrier.
// read by Track::isMuted() called by playback thread, also without a mutex or barrier.
// The lack of mutex or barrier is safe because the mute status is only used by itself.
bool mMute;
// FILLED state is used for suppressing volume ramp at begin of playing
enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE};
mutable uint8_t mFillingUpStatus;
int8_t mRetryCount;
const sp<IMemory> mSharedBuffer;
bool mResetDone;
const audio_stream_type_t mStreamType;
int mName; // track name on the normal mixer,
// allocated statically at track creation time,
// and is even allocated (though unused) for fast tracks
// FIXME don't allocate track name for fast tracks
int16_t *mMainBuffer;
int32_t *mAuxBuffer;
int mAuxEffectId;
bool mHasVolumeController;
size_t mPresentationCompleteFrames; // number of frames written to the
// audio HAL when this track will be fully rendered
// zero means not monitoring
private:
IAudioFlinger::track_flags_t mFlags;
// 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
uint32_t mUnderrunCount; // Counter of total number of underruns, never reset
volatile float mCachedVolume; // combined master volume and stream type volume;
// 'volatile' means accessed without lock or
// barrier, but is read/written atomically
}; // 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,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
virtual ~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; }
uint32_t position() const { return mPosition; }
void setPosition(uint32_t pos) { mPosition = pos; }
private:
sp<IMemory> mBuffer;
int64_t mPTS;
uint32_t mPosition;
};
// Mixer facing methods.
virtual bool isTimedTrack() const { return true; }
virtual size_t framesReady() const;
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
// Client/App facing methods.
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);
private:
TimedTrack(PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer);
void timedYieldSilence_l(uint32_t numFrames,
AudioBufferProvider::Buffer* buffer);
void trimTimedBufferQueue_l();
void trimTimedBufferQueueHead_l(const char* logTag);
void updateFramesPendingAfterTrim_l(const TimedBuffer& buf,
const char* logTag);
uint64_t mLocalTimeFreq;
LinearTransform mLocalTimeToSampleTransform;
LinearTransform mMediaTimeToSampleTransform;
sp<MemoryDealer> mTimedMemoryDealer;
Vector<TimedBuffer> mTimedBufferQueue;
bool mQueueHeadInFlight;
bool mTrimQueueHeadOnRelease;
uint32_t mFramesPendingInQueue;
uint8_t* mTimedSilenceBuffer;
uint32_t mTimedSilenceBufferSize;
mutable Mutex mTimedBufferQueueLock;
bool mTimedAudioOutputOnTime;
CCHelper mCCHelper;
Mutex mMediaTimeTransformLock;
LinearTransform mMediaTimeTransform;
bool mMediaTimeTransformValid;
TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
};
// playback track, used by DuplicatingThread
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,
audio_channel_mask_t channelMask,
size_t frameCount);
virtual ~OutputTrack();
virtual status_t start(AudioSystem::sync_event_t event =
AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
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()
void* mBuffers; // starting address of buffers in plain memory
}; // end of OutputTrack
PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, audio_devices_t device, type_t type);
virtual ~PlaybackThread();
void dump(int fd, const Vector<String16>& args);
// Thread virtuals
virtual status_t readyToRun();
virtual bool threadLoop();
// RefBase
virtual void onFirstRef();
protected:
// Code snippets that were lifted up out of threadLoop()
virtual void threadLoop_mix() = 0;
virtual void threadLoop_sleepTime() = 0;
virtual void threadLoop_write();
virtual void threadLoop_standby();
virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
// prepareTracks_l reads and writes mActiveTracks, and returns
// the pending set of tracks to remove via Vector 'tracksToRemove'. The caller
// is responsible for clearing or destroying this Vector later on, when it
// is safe to do so. That will drop the final ref count and destroy the tracks.
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0;
// ThreadBase virtuals
virtual void preExit();
public:
virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
// return estimated latency in milliseconds, as reported by HAL
uint32_t latency() const;
// same, but lock must already be held
uint32_t latency_l() const;
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,
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
status_t *status);
AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
virtual audio_stream_t* stream() const;
// a very large number of suspend() will eventually wraparound, but unlikely
void suspend() { (void) android_atomic_inc(&mSuspended); }
void restore()
{
// if restore() is done without suspend(), get back into
// range so that the next suspend() will operate correctly
if (android_atomic_dec(&mSuspended) <= 0) {
android_atomic_release_store(0, &mSuspended);
}
}
bool isSuspended() const
{ return android_atomic_acquire_load(&mSuspended) > 0; }
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
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) const;
virtual uint32_t getStrategyForSession_l(int sessionId);
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const;
void invalidateTracks(audio_stream_type_t streamType);
protected:
int16_t* mMixBuffer;
// suspend count, > 0 means suspended. While suspended, the thread continues to pull from
// tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle
// concurrent use of both of them, so Audio Policy Service suspends one of the threads to
// workaround that restriction.
// 'volatile' means accessed via atomic operations and no lock.
volatile int32_t mSuspended;
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; // FIXME check if this could be sp<>
// Allocate a track name for a given channel mask.
// Returns name >= 0 if successful, -1 on failure.
virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId) = 0;
virtual void deleteTrackName_l(int name) = 0;
// Time to sleep between cycles when:
virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED
virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE
virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us
// No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write()
// No sleep in standby mode; waits on a condition
// Code snippets that are temporarily lifted up out of threadLoop() until the merge
void checkSilentMode_l();
// Non-trivial for DUPLICATING only
virtual void saveOutputTracks() { }
virtual void clearOutputTracks() { }
// Cache various calculated values, at threadLoop() entry and after a parameter change
virtual void cacheParameters_l();
virtual uint32_t correctLatency(uint32_t latency) const;
private:
friend class AudioFlinger; // for numerous
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 void dumpInternals(int fd, const Vector<String16>& args);
void 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;
// FIXME rename these former local variables of threadLoop to standard "m" names
nsecs_t standbyTime;
size_t mixBufferSize;
// cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l()
uint32_t activeSleepTime;
uint32_t idleSleepTime;
uint32_t sleepTime;
// mixer status returned by prepareTracks_l()
mixer_state mMixerStatus; // current cycle
// previous cycle when in prepareTracks_l()
mixer_state mMixerStatusIgnoringFastTracks;
// FIXME or a separate ready state per track
// FIXME move these declarations into the specific sub-class that needs them
// MIXER only
uint32_t sleepTimeShift;
// same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value
nsecs_t standbyDelay;
// MIXER only
nsecs_t maxPeriod;
// DUPLICATING only
uint32_t writeFrames;
private:
// The HAL output sink is treated as non-blocking, but current implementation is blocking
sp<NBAIO_Sink> mOutputSink;
// If a fast mixer is present, the blocking pipe sink, otherwise clear
sp<NBAIO_Sink> mPipeSink;
// The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink
sp<NBAIO_Sink> mNormalSink;
// For dumpsys
sp<NBAIO_Sink> mTeeSink;
sp<NBAIO_Source> mTeeSource;
uint32_t mScreenState; // cached copy of gScreenState
public:
virtual bool hasFastMixer() const = 0;
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
{ FastTrackUnderruns dummy; return dummy; }
protected:
// accessed by both binder threads and within threadLoop(), lock on mutex needed
unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available
};
class MixerThread : public PlaybackThread {
public:
MixerThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
audio_devices_t device,
type_t type = MIXER);
virtual ~MixerThread();
// Thread virtuals
virtual bool checkForNewParameters_l();
virtual void dumpInternals(int fd, const Vector<String16>& args);
protected:
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
virtual void deleteTrackName_l(int name);
virtual uint32_t idleSleepTimeUs() const;
virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l();
// threadLoop snippets
virtual void threadLoop_write();
virtual void threadLoop_standby();
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
virtual void threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove);
virtual uint32_t correctLatency(uint32_t latency) const;
AudioMixer* mAudioMixer; // normal mixer
private:
// one-time initialization, no locks required
FastMixer* mFastMixer; // non-NULL if there is also a fast mixer
sp<AudioWatchdog> mAudioWatchdog; // non-0 if there is an audio watchdog thread
// contents are not guaranteed to be consistent, no locks required
FastMixerDumpState mFastMixerDumpState;
#ifdef STATE_QUEUE_DUMP
StateQueueObserverDump mStateQueueObserverDump;
StateQueueMutatorDump mStateQueueMutatorDump;
#endif
AudioWatchdogDump mAudioWatchdogDump;
// accessible only within the threadLoop(), no locks required
// mFastMixer->sq() // for mutating and pushing state
int32_t mFastMixerFutex; // for cold idle
public:
virtual bool hasFastMixer() const { return mFastMixer != NULL; }
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const {
ALOG_ASSERT(fastIndex < FastMixerState::kMaxFastTracks);
return mFastMixerDumpState.mTracks[fastIndex].mUnderruns;
}
};
class DirectOutputThread : public PlaybackThread {
public:
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, audio_devices_t device);
virtual ~DirectOutputThread();
// Thread virtuals
virtual bool checkForNewParameters_l();
protected:
virtual int getTrackName_l(audio_channel_mask_t channelMask, int sessionId);
virtual void deleteTrackName_l(int name);
virtual uint32_t activeSleepTimeUs() const;
virtual uint32_t idleSleepTimeUs() const;
virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l();
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
private:
// volumes last sent to audio HAL with stream->set_volume()
float mLeftVolFloat;
float mRightVolFloat;
// prepareTracks_l() tells threadLoop_mix() the name of the single active track
sp<Track> mActiveTrack;
public:
virtual bool hasFastMixer() const { return false; }
};
class DuplicatingThread : public MixerThread {
public:
DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
audio_io_handle_t id);
virtual ~DuplicatingThread();
// Thread virtuals
void addOutputTrack(MixerThread* thread);
void removeOutputTrack(MixerThread* thread);
uint32_t waitTimeMs() const { return mWaitTimeMs; }
protected:
virtual uint32_t activeSleepTimeUs() const;
private:
bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
protected:
// threadLoop snippets
virtual void threadLoop_mix();
virtual void threadLoop_sleepTime();
virtual void threadLoop_write();
virtual void threadLoop_standby();
virtual void cacheParameters_l();
private:
// called from threadLoop, addOutputTrack, removeOutputTrack
virtual void updateWaitTime_l();
protected:
virtual void saveOutputTracks();
virtual void clearOutputTracks();
private:
uint32_t mWaitTimeMs;
SortedVector < sp<OutputTrack> > outputTracks;
SortedVector < sp<OutputTrack> > mOutputTracks;
public:
virtual bool hasFastMixer() const { return false; }
};
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, const 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);
// return thread associated with primary hardware device, or NULL
PlaybackThread *primaryPlaybackThread_l() const;
audio_devices_t primaryOutputDevice_l() const;
sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
// 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();
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;
};
void removeClient_l(pid_t pid);
void removeNotificationClient(pid_t pid);
// record thread
class RecordThread : public ThreadBase, public AudioBufferProvider
// derives from AudioBufferProvider interface for use by resampler
{
public:
// record track
class RecordTrack : public TrackBase {
public:
RecordTrack(RecordThread *thread,
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
int sessionId);
virtual ~RecordTrack();
virtual status_t start(AudioSystem::sync_event_t event, int triggerSession);
virtual void stop();
void destroy();
// clear the buffer overflow flag
void clearOverflow() { mOverflow = false; }
// set the buffer overflow flag and return previous value
bool setOverflow() { bool tmp = mOverflow; mOverflow = true;
return tmp; }
static void appendDumpHeader(String8& result);
void dump(char* buffer, size_t size);
virtual bool isOut() const;
private:
friend class AudioFlinger; // for mState
RecordTrack(const RecordTrack&);
RecordTrack& operator = (const RecordTrack&);
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
int64_t pts = kInvalidPTS);
// releaseBuffer() not overridden
bool mOverflow; // overflow on most recent attempt to fill client buffer
};
RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_io_handle_t id,
audio_devices_t device,
const sp<NBAIO_Sink>& teeSink);
virtual ~RecordThread();
// no addTrack_l ?
void destroyTrack_l(const sp<RecordTrack>& track);
void removeTrack_l(const sp<RecordTrack>& track);
void dumpInternals(int fd, const Vector<String16>& args);
void dumpTracks(int fd, const Vector<String16>& args);
// Thread virtuals
virtual bool threadLoop();
virtual status_t readyToRun();
// RefBase
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,
audio_channel_mask_t channelMask,
size_t frameCount,
int sessionId,
IAudioFlinger::track_flags_t flags,
pid_t tid,
status_t *status);
status_t start(RecordTrack* recordTrack,
AudioSystem::sync_event_t event,
int triggerSession);
// ask the thread to stop the specified track, and
// return true if the caller should then do it's part of the stopping process
bool stop_l(RecordTrack* recordTrack);
void dump(int fd, const Vector<String16>& args);
AudioStreamIn* clearInput();
virtual audio_stream_t* stream() const;
// AudioBufferProvider interface
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) const;
// 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<int, bool> sessionIds() const;
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const;
static void syncStartEventCallback(const wp<SyncEvent>& event);
void handleSyncStartEvent(const sp<SyncEvent>& event);
private:
void clearSyncStartEvent();
// Enter standby if not already in standby, and set mStandby flag
void standby();
// Call the HAL standby method unconditionally, and don't change mStandby flag
void inputStandBy();
AudioStreamIn *mInput;
SortedVector < sp<RecordTrack> > mTracks;
// mActiveTrack has dual roles: it indicates the current active track, and
// is used together with mStartStopCond to indicate start()/stop() progress
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;
// sync event triggering actual audio capture. Frames read before this event will
// be dropped and therefore not read by the application.
sp<SyncEvent> mSyncStartEvent;
// number of captured frames to drop after the start sync event has been received.
// when < 0, maximum frames to drop before starting capture even if sync event is
// not received
ssize_t mFramestoDrop;
// For dumpsys
const sp<NBAIO_Sink> mTeeSink;
};
// 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(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
virtual void stop();
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
const sp<RecordThread::RecordTrack> mRecordTrack;
// for use from destructor
void stop_nonvirtual();
};
//--- 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);
status_t setEnabled_l(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(EffectHandle *handle);
size_t disconnect(EffectHandle *handle, bool unpinIfLast);
size_t removeHandle(EffectHandle *handle);
const effect_descriptor_t& desc() const { return mDescriptor; }
wp<EffectChain>& chain() { return mChain; }
status_t setDevice(audio_devices_t device);
status_t setVolume(uint32_t *left, uint32_t *right, bool controller);
status_t setMode(audio_mode_t mode);
status_t setAudioSource(audio_source_t source);
status_t start();
status_t stop();
void setSuspended(bool suspended);
bool suspended() const;
EffectHandle* controlHandle_l();
bool isPinned() const { return mPinned; }
void unPin() { mPinned = false; }
bool purgeHandles();
void lock() { mLock.lock(); }
void unlock() { mLock.unlock(); }
void dump(int fd, const Vector<String16>& args);
protected:
friend class AudioFlinger; // for mHandles
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
const int mId; // this instance unique ID
const int mSessionId; // audio session ID
const 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<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; }
// destroyed_l() must be called with the associated EffectModule mLock held
bool destroyed_l() const { return mDestroyed; }
void dump(char* buffer, size_t size);
protected:
friend class AudioFlinger; // for mEffect, mHasControl, mEnabled
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
bool mDestroyed; // Set to true by destructor. Access with EffectModule
// mLock held
};
// 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(audio_devices_t device);
void setMode_l(audio_mode_t mode);
void setAudioSource_l(audio_source_t source);
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 android_atomic_acquire_load(&mTrackCnt); }
void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt);
mTailBufferCount = mMaxTailBuffers; }
void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); }
int32_t activeTrackCnt() const { return android_atomic_acquire_load(&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);
void clearInputBuffer();
void dump(int fd, const Vector<String16>& args);
protected:
friend class AudioFlinger; // for mThread, mEffects
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);
void clearInputBuffer_l(sp<ThreadBase> thread);
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' here means these are accessed with atomic operations instead of mutex
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 effects 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.
// Updated by updateSuspendedSessions_l() only.
KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects;
};
class AudioHwDevice {
public:
enum Flags {
AHWD_CAN_SET_MASTER_VOLUME = 0x1,
AHWD_CAN_SET_MASTER_MUTE = 0x2,
};
AudioHwDevice(const char *moduleName,
audio_hw_device_t *hwDevice,
Flags flags)
: mModuleName(strdup(moduleName))
, mHwDevice(hwDevice)
, mFlags(flags) { }
/*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); }
bool canSetMasterVolume() const {
return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME));
}
bool canSetMasterMute() const {
return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE));
}
const char *moduleName() const { return mModuleName; }
audio_hw_device_t *hwDevice() const { return mHwDevice; }
private:
const char * const mModuleName;
audio_hw_device_t * const mHwDevice;
Flags mFlags;
};
// 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 {
AudioHwDevice* const audioHwDev;
audio_stream_out_t* const stream;
audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) :
audioHwDev(dev), stream(out) {}
};
struct AudioStreamIn {
AudioHwDevice* const audioHwDev;
audio_stream_in_t* const stream;
audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) :
audioHwDev(dev), stream(in) {}
};
// for mAudioSessionRefs only
struct AudioSessionRef {
AudioSessionRef(int sessionid, pid_t pid) :
mSessionid(sessionid), mPid(pid), mCnt(1) {}
const int mSessionid;
const pid_t mPid;
int mCnt;
};
mutable Mutex mLock;
DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client()
mutable Mutex mHardwareLock;
// NOTE: If both mLock and mHardwareLock mutexes must be held,
// always take mLock before mHardwareLock
// These two fields are immutable after onFirstRef(), so no lock needed to access
AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs;
// for dump, indicates which hardware operation is currently in progress (but not stream ops)
enum hardware_call_state {
AUDIO_HW_IDLE = 0, // no operation in progress
AUDIO_HW_INIT, // init_check
AUDIO_HW_OUTPUT_OPEN, // open_output_stream
AUDIO_HW_OUTPUT_CLOSE, // unused
AUDIO_HW_INPUT_OPEN, // unused
AUDIO_HW_INPUT_CLOSE, // unused
AUDIO_HW_STANDBY, // unused
AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume
AUDIO_HW_GET_ROUTING, // unused
AUDIO_HW_SET_ROUTING, // unused
AUDIO_HW_GET_MODE, // unused
AUDIO_HW_SET_MODE, // set_mode
AUDIO_HW_GET_MIC_MUTE, // get_mic_mute
AUDIO_HW_SET_MIC_MUTE, // set_mic_mute
AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume
AUDIO_HW_SET_PARAMETER, // set_parameters
AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size
AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume
AUDIO_HW_GET_PARAMETER, // get_parameters
AUDIO_HW_SET_MASTER_MUTE, // set_master_mute
AUDIO_HW_GET_MASTER_MUTE, // get_master_mute
};
mutable hardware_call_state mHardwareStatus; // for dump only
DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads;
stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
// member variables below are protected by mLock
float mMasterVolume;
bool mMasterMute;
// end of variables protected by mLock
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;
bool masterMute_l() const;
audio_module_handle_t loadHwModule_l(const char *name);
Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session
// to be created
private:
sp<Client> registerPid_l(pid_t pid); // always returns non-0
// for use from destructor
status_t closeOutput_nonvirtual(audio_io_handle_t output);
status_t closeInput_nonvirtual(audio_io_handle_t input);
// all record threads serially share a common tee sink, which is re-created on format change
sp<NBAIO_Sink> mRecordTeeSink;
sp<NBAIO_Source> mRecordTeeSource;
public:
static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0);
};
// ----------------------------------------------------------------------------
}; // namespace android
#endif // ANDROID_AUDIO_FLINGER_H