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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
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** See the License for the specific language governing permissions and
** limitations under the License.
#include "Configuration.h"
#include <stdint.h>
#include <sys/types.h>
#include <limits.h>
#include <common_time/cc_helper.h>
#include <cutils/compiler.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 "FastCapture.h"
#include "FastMixer.h"
#include <media/nbaio/NBAIO.h>
#include "AudioWatchdog.h"
#include "AudioMixer.h"
#include "AudioStreamOut.h"
#include "SpdifStreamOut.h"
#include "AudioHwDevice.h"
#include <powermanager/IPowerManager.h>
#include <media/nbaio/NBLog.h>
#include <private/media/AudioTrackShared.h>
namespace android {
struct audio_track_cblk_t;
struct effect_param_cblk_t;
class AudioMixer;
class AudioBuffer;
class AudioResampler;
class FastMixer;
class PassthruBufferProvider;
class ServerProxy;
// ----------------------------------------------------------------------------
// The macro FCC_2 highlights some (but not all) places where there are are 2-channel assumptions.
// This is typically due to legacy implementation of stereo input or output.
// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc.
#define FCC_2 2 // FCC_2 = Fixed Channel Count 2
// The macro FCC_8 highlights places where there are 8-channel assumptions.
// This is typically due to audio mixer and resampler limitations.
#define FCC_8 8 // FCC_8 = Fixed Channel Count 8
static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
// Max shared memory size for audio tracks and audio records per client process
static const size_t kClientSharedHeapSizeBytes = 1024*1024;
// Shared memory size multiplier for non low ram devices
static const size_t kClientSharedHeapSizeMultiplier = 4;
class AudioFlinger :
public BinderService<AudioFlinger>,
public BnAudioFlinger
friend class BinderService<AudioFlinger>; // for AudioFlinger()
static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; }
virtual status_t dump(int fd, const Vector<String16>& args);
// IAudioFlinger interface, in binder opcode order
virtual sp<IAudioTrack> createTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *pFrameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
pid_t tid,
int *sessionId,
int clientUid,
status_t *status /*non-NULL*/);
virtual sp<IAudioRecord> openRecord(
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
const String16& opPackageName,
size_t *pFrameCount,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int clientUid,
int *sessionId,
size_t *notificationFrames,
sp<IMemory>& cblk,
sp<IMemory>& buffers,
status_t *status /*non-NULL*/);
virtual uint32_t sampleRate(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 status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t *devices,
const String8& address,
uint32_t *latencyMs,
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 status_t openInput(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t *device,
const String8& address,
audio_source_t source,
audio_input_flags_t flags);
virtual status_t closeInput(audio_io_handle_t input);
virtual status_t invalidateStream(audio_stream_type_t stream);
virtual status_t setVoiceVolume(float volume);
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
audio_io_handle_t output) const;
virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
virtual audio_unique_id_t newAudioUniqueId();
virtual void acquireAudioSessionId(int audioSession, pid_t pid);
virtual void releaseAudioSessionId(int audioSession, pid_t pid);
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(
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
audio_io_handle_t io,
int sessionId,
const String16& opPackageName,
status_t *status /*non-NULL*/,
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 setLowRamDevice(bool isLowRamDevice);
/* List available audio ports and their attributes */
virtual status_t listAudioPorts(unsigned int *num_ports,
struct audio_port *ports);
/* Get attributes for a given audio port */
virtual status_t getAudioPort(struct audio_port *port);
/* Create an audio patch between several source and sink ports */
virtual status_t createAudioPatch(const struct audio_patch *patch,
audio_patch_handle_t *handle);
/* Release an audio patch */
virtual status_t releaseAudioPatch(audio_patch_handle_t handle);
/* List existing audio patches */
virtual status_t listAudioPatches(unsigned int *num_patches,
struct audio_patch *patches);
/* Set audio port configuration */
virtual status_t setAudioPortConfig(const struct audio_port_config *config);
/* Get the HW synchronization source used for an audio session */
virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId);
/* Indicate JAVA services are ready (scheduling, power management ...) */
virtual status_t systemReady();
virtual status_t onTransact(
uint32_t code,
const Parcel& data,
Parcel* reply,
uint32_t flags);
// end of IAudioFlinger interface
sp<NBLog::Writer> newWriter_l(size_t size, const char *name);
void unregisterWriter(const sp<NBLog::Writer>& writer);
static const size_t kLogMemorySize = 40 * 1024;
sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled
// When a log writer is unregistered, it is done lazily so that media.log can continue to see it
// for as long as possible. The memory is only freed when it is needed for another log writer.
Vector< sp<NBLog::Writer> > mUnregisteredWriters;
Mutex mUnregisteredWritersLock;
class SyncEvent;
typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ;
class SyncEvent : public RefBase {
SyncEvent(AudioSystem::sync_event_t type,
int triggerSession,
int listenerSession,
sync_event_callback_t callBack,
wp<RefBase> 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; }
wp<RefBase> cookie() const { return mCookie; }
const AudioSystem::sync_event_t mType;
const int mTriggerSession;
const int mListenerSession;
sync_event_callback_t mCallback;
const wp<RefBase> mCookie;
mutable Mutex mLock;
sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
int triggerSession,
int listenerSession,
sync_event_callback_t callBack,
wp<RefBase> cookie);
audio_mode_t getMode() const { return mMode; }
bool btNrecIsOff() const { return mBtNrecIsOff; }
AudioFlinger() ANDROID_API;
virtual ~AudioFlinger();
// call in any IAudioFlinger method that accesses mPrimaryHardwareDev
status_t initCheck() const { return mPrimaryHardwareDev == NULL ?
// RefBase
virtual void onFirstRef();
AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module,
audio_devices_t devices);
void purgeStaleEffects_l();
// Set kEnableExtendedChannels to true to enable greater than stereo output
// for the MixerThread and device sink. Number of channels allowed is
// FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS.
static const bool kEnableExtendedChannels = true;
// Returns true if channel mask is permitted for the PCM sink in the MixerThread
static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) {
switch (audio_channel_mask_get_representation(channelMask)) {
uint32_t channelCount = FCC_2; // stereo is default
if (kEnableExtendedChannels) {
channelCount = audio_channel_count_from_out_mask(channelMask);
if (channelCount < FCC_2 // mono is not supported at this time
|| channelCount > AudioMixer::MAX_NUM_CHANNELS) {
return false;
// check that channelMask is the "canonical" one we expect for the channelCount.
return channelMask == audio_channel_out_mask_from_count(channelCount);
if (kEnableExtendedChannels) {
const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
if (channelCount >= FCC_2 // mono is not supported at this time
&& channelCount <= AudioMixer::MAX_NUM_CHANNELS) {
return true;
return false;
return false;
// Set kEnableExtendedPrecision to true to use extended precision in MixerThread
static const bool kEnableExtendedPrecision = true;
// Returns true if format is permitted for the PCM sink in the MixerThread
static inline bool isValidPcmSinkFormat(audio_format_t format) {
switch (format) {
return true;
return kEnableExtendedPrecision;
return false;
// standby delay for MIXER and DUPLICATING playback threads is read from property
// or defaults to kDefaultStandbyTimeInNsecs
static nsecs_t mStandbyTimeInNsecs;
// incremented by 2 when screen state changes, bit 0 == 1 means "off"
// AudioFlinger::setParameters() updates, other threads read w/o lock
static uint32_t mScreenState;
// Internal dump utilities.
static const int kDumpLockRetries = 50;
static const int kDumpLockSleepUs = 20000;
static bool dumpTryLock(Mutex& mutex);
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 {
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();
Client(const Client&);
Client& operator = (const Client&);
const sp<AudioFlinger> mAudioFlinger;
sp<MemoryDealer> mMemoryDealer;
const pid_t mPid;
Mutex mTimedTrackLock;
int mTimedTrackCount;
// --- Notification Client ---
class NotificationClient : public IBinder::DeathRecipient {
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);
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 OffloadThread;
class DuplicatingThread;
class AsyncCallbackThread;
class Track;
class RecordTrack;
class EffectModule;
class EffectHandle;
class EffectChain;
struct AudioStreamIn;
struct stream_type_t {
: volume(1.0f),
float volume;
bool mute;
// --- PlaybackThread ---
#include "Threads.h"
#include "Effects.h"
#include "PatchPanel.h"
// server side of the client's IAudioTrack
class TrackHandle : public android::BnAudioTrack {
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 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 setParameters(const String8& keyValuePairs);
virtual status_t getTimestamp(AudioTimestamp& timestamp);
virtual void signal(); // signal playback thread for a change in control block
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
const sp<PlaybackThread::Track> mTrack;
// server side of the client's IAudioRecord
class RecordHandle : public android::BnAudioRecord {
RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
virtual ~RecordHandle();
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);
const sp<RecordThread::RecordTrack> mRecordTrack;
// for use from destructor
void stop_nonvirtual();
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;
sp<RecordThread> openInput_l(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t device,
const String8& address,
audio_source_t source,
audio_input_flags_t flags);
sp<PlaybackThread> openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t devices,
const String8& address,
audio_output_flags_t flags);
void closeOutputFinish(sp<PlaybackThread> thread);
void closeInputFinish(sp<RecordThread> thread);
// 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 ioConfigChanged(audio_io_config_event event,
const sp<AudioIoDescriptor>& ioDesc,
pid_t pid = 0);
// Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t.
// They all share the same ID space, but the namespaces are actually independent
// because there are separate KeyedVectors for each kind of ID.
// The return value is uint32_t, but is cast to signed for some IDs.
// FIXME This API does not handle rollover to zero (for unsigned IDs),
// or from positive to negative (for signed IDs).
// Thus it may fail by returning an ID of the wrong sign,
// or by returning a non-unique 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);
void removeClient_l(pid_t pid);
void removeNotificationClient(pid_t pid);
bool isNonOffloadableGlobalEffectEnabled_l();
void onNonOffloadableGlobalEffectEnable();
// Store an effect chain to mOrphanEffectChains keyed vector.
// Called when a thread exits and effects are still attached to it.
// If effects are later created on the same session, they will reuse the same
// effect chain and same instances in the effect library.
// return ALREADY_EXISTS if a chain with the same session already exists in
// mOrphanEffectChains. Note that this should never happen as there is only one
// chain for a given session and it is attached to only one thread at a time.
status_t putOrphanEffectChain_l(const sp<EffectChain>& chain);
// Get an effect chain for the specified session in mOrphanEffectChains and remove
// it if found. Returns 0 if not found (this is the most common case).
sp<EffectChain> getOrphanEffectChain_l(audio_session_t session);
// Called when the last effect handle on an effect instance is removed. If this
// effect belongs to an effect chain in mOrphanEffectChains, the chain is updated
// and removed from mOrphanEffectChains if it does not contain any effect.
// Return true if the effect was found in mOrphanEffectChains, false otherwise.
bool updateOrphanEffectChains(const sp<EffectModule>& effect);
void broacastParametersToRecordThreads_l(const String8& keyValuePairs);
// AudioStreamIn is 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 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;
// protects mClients and mNotificationClients.
// must be locked after mLock and ThreadBase::mLock if both must be locked
// avoids acquiring AudioFlinger::mLock from inside thread loop.
mutable Mutex mClientLock;
// protected by mClientLock
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_SET_MASTER_VOLUME, // set_master_volume
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;
// protected by mClientLock
DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients;
volatile int32_t mNextUniqueId; // updated by android_atomic_inc
// nextUniqueId() returns uint32_t, but this is declared int32_t
// because the atomic operations require an int32_t
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
// Effect chains without a valid thread
DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains;
// list of sessions for which a valid HW A/V sync ID was retrieved from the HAL
DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds;
sp<Client> registerPid(pid_t pid); // always returns non-0
// for use from destructor
status_t closeOutput_nonvirtual(audio_io_handle_t output);
void closeOutputInternal_l(sp<PlaybackThread> thread);
status_t closeInput_nonvirtual(audio_io_handle_t input);
void closeInputInternal_l(sp<RecordThread> thread);
void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId);
status_t checkStreamType(audio_stream_type_t stream) const;
#ifdef TEE_SINK
// all record threads serially share a common tee sink, which is re-created on format change
sp<NBAIO_Sink> mRecordTeeSink;
sp<NBAIO_Source> mRecordTeeSource;
#ifdef TEE_SINK
// tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file
static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0);
// whether tee sink is enabled by property
static bool mTeeSinkInputEnabled;
static bool mTeeSinkOutputEnabled;
static bool mTeeSinkTrackEnabled;
// runtime configured size of each tee sink pipe, in frames
static size_t mTeeSinkInputFrames;
static size_t mTeeSinkOutputFrames;
static size_t mTeeSinkTrackFrames;
// compile-time default size of tee sink pipes, in frames
// 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes
static const size_t kTeeSinkInputFramesDefault = 0x200000;
static const size_t kTeeSinkOutputFramesDefault = 0x200000;
static const size_t kTeeSinkTrackFramesDefault = 0x200000;
// This method reads from a variable without mLock, but the variable is updated under mLock. So
// we might read a stale value, or a value that's inconsistent with respect to other variables.
// In this case, it's safe because the return value isn't used for making an important decision.
// The reason we don't want to take mLock is because it could block the caller for a long time.
bool isLowRamDevice() const { return mIsLowRamDevice; }
bool mIsLowRamDevice;
bool mIsDeviceTypeKnown;
nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled
sp<PatchPanel> mPatchPanel;
bool mSystemReady;
const char *formatToString(audio_format_t format);
String8 inputFlagsToString(audio_input_flags_t flags);
String8 outputFlagsToString(audio_output_flags_t flags);
String8 devicesToString(audio_devices_t devices);
const char *sourceToString(audio_source_t source);
// ----------------------------------------------------------------------------
} // namespace android