AudioFlinger.h revision 1035194cee4fbd57e35ea15c56e66cd09b63d56e 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18#ifndef ANDROID_AUDIO_FLINGER_H
19#define ANDROID_AUDIO_FLINGER_H
20
21#include <stdint.h>
22#include <sys/types.h>
23#include <limits.h>
24
25#include <common_time/cc_helper.h>
26
27#include <cutils/compiler.h>
28
29#include <media/IAudioFlinger.h>
30#include <media/IAudioFlingerClient.h>
31#include <media/IAudioTrack.h>
32#include <media/IAudioRecord.h>
33#include <media/AudioSystem.h>
34#include <media/AudioTrack.h>
35
36#include <utils/Atomic.h>
37#include <utils/Errors.h>
38#include <utils/threads.h>
39#include <utils/SortedVector.h>
40#include <utils/TypeHelpers.h>
41#include <utils/Vector.h>
42
43#include <binder/BinderService.h>
44#include <binder/MemoryDealer.h>
45
46#include <system/audio.h>
47#include <hardware/audio.h>
48#include <hardware/audio_policy.h>
49
50#include <media/AudioBufferProvider.h>
51#include <media/ExtendedAudioBufferProvider.h>
52#include "FastMixer.h"
53#include <media/nbaio/NBAIO.h>
54#include "AudioWatchdog.h"
55
56#include <powermanager/IPowerManager.h>
57
58#include <media/nbaio/NBLog.h>
59#include <private/media/AudioTrackShared.h>
60
61namespace android {
62
63struct audio_track_cblk_t;
64struct effect_param_cblk_t;
65class AudioMixer;
66class AudioBuffer;
67class AudioResampler;
68class FastMixer;
69class ServerProxy;
70
71// ----------------------------------------------------------------------------
72
73// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback.
74// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect.
75// Adding full support for > 2 channel capture or playback would require more than simply changing
76// this #define.  There is an independent hard-coded upper limit in AudioMixer;
77// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels.
78// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions.
79// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc.
80#define FCC_2 2     // FCC_2 = Fixed Channel Count 2
81
82static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
83
84#define MAX_GAIN 4096.0f
85#define MAX_GAIN_INT 0x1000
86
87#define INCLUDING_FROM_AUDIOFLINGER_H
88
89class AudioFlinger :
90    public BinderService<AudioFlinger>,
91    public BnAudioFlinger
92{
93    friend class BinderService<AudioFlinger>;   // for AudioFlinger()
94public:
95    static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; }
96
97    virtual     status_t    dump(int fd, const Vector<String16>& args);
98
99    // IAudioFlinger interface, in binder opcode order
100    virtual sp<IAudioTrack> createTrack(
101                                audio_stream_type_t streamType,
102                                uint32_t sampleRate,
103                                audio_format_t format,
104                                audio_channel_mask_t channelMask,
105                                size_t *pFrameCount,
106                                IAudioFlinger::track_flags_t *flags,
107                                const sp<IMemory>& sharedBuffer,
108                                audio_io_handle_t output,
109                                pid_t tid,
110                                int *sessionId,
111                                int clientUid,
112                                status_t *status /*non-NULL*/);
113
114    virtual sp<IAudioRecord> openRecord(
115                                audio_io_handle_t input,
116                                uint32_t sampleRate,
117                                audio_format_t format,
118                                audio_channel_mask_t channelMask,
119                                size_t *pFrameCount,
120                                IAudioFlinger::track_flags_t *flags,
121                                pid_t tid,
122                                int *sessionId,
123                                status_t *status /*non-NULL*/);
124
125    virtual     uint32_t    sampleRate(audio_io_handle_t output) const;
126    virtual     int         channelCount(audio_io_handle_t output) const;
127    virtual     audio_format_t format(audio_io_handle_t output) const;
128    virtual     size_t      frameCount(audio_io_handle_t output) const;
129    virtual     uint32_t    latency(audio_io_handle_t output) const;
130
131    virtual     status_t    setMasterVolume(float value);
132    virtual     status_t    setMasterMute(bool muted);
133
134    virtual     float       masterVolume() const;
135    virtual     bool        masterMute() const;
136
137    virtual     status_t    setStreamVolume(audio_stream_type_t stream, float value,
138                                            audio_io_handle_t output);
139    virtual     status_t    setStreamMute(audio_stream_type_t stream, bool muted);
140
141    virtual     float       streamVolume(audio_stream_type_t stream,
142                                         audio_io_handle_t output) const;
143    virtual     bool        streamMute(audio_stream_type_t stream) const;
144
145    virtual     status_t    setMode(audio_mode_t mode);
146
147    virtual     status_t    setMicMute(bool state);
148    virtual     bool        getMicMute() const;
149
150    virtual     status_t    setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
151    virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
152
153    virtual     void        registerClient(const sp<IAudioFlingerClient>& client);
154
155    virtual     size_t      getInputBufferSize(uint32_t sampleRate, audio_format_t format,
156                                               audio_channel_mask_t channelMask) const;
157
158    virtual audio_io_handle_t openOutput(audio_module_handle_t module,
159                                         audio_devices_t *pDevices,
160                                         uint32_t *pSamplingRate,
161                                         audio_format_t *pFormat,
162                                         audio_channel_mask_t *pChannelMask,
163                                         uint32_t *pLatencyMs,
164                                         audio_output_flags_t flags,
165                                         const audio_offload_info_t *offloadInfo);
166
167    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
168                                                  audio_io_handle_t output2);
169
170    virtual status_t closeOutput(audio_io_handle_t output);
171
172    virtual status_t suspendOutput(audio_io_handle_t output);
173
174    virtual status_t restoreOutput(audio_io_handle_t output);
175
176    virtual audio_io_handle_t openInput(audio_module_handle_t module,
177                                        audio_devices_t *pDevices,
178                                        uint32_t *pSamplingRate,
179                                        audio_format_t *pFormat,
180                                        audio_channel_mask_t *pChannelMask);
181
182    virtual status_t closeInput(audio_io_handle_t input);
183
184    virtual status_t invalidateStream(audio_stream_type_t stream);
185
186    virtual status_t setVoiceVolume(float volume);
187
188    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
189                                       audio_io_handle_t output) const;
190
191    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
192
193    virtual int newAudioSessionId();
194
195    virtual void acquireAudioSessionId(int audioSession, pid_t pid);
196
197    virtual void releaseAudioSessionId(int audioSession, pid_t pid);
198
199    virtual status_t queryNumberEffects(uint32_t *numEffects) const;
200
201    virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
202
203    virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
204                                         effect_descriptor_t *descriptor) const;
205
206    virtual sp<IEffect> createEffect(
207                        effect_descriptor_t *pDesc,
208                        const sp<IEffectClient>& effectClient,
209                        int32_t priority,
210                        audio_io_handle_t io,
211                        int sessionId,
212                        status_t *status /*non-NULL*/,
213                        int *id,
214                        int *enabled);
215
216    virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
217                        audio_io_handle_t dstOutput);
218
219    virtual audio_module_handle_t loadHwModule(const char *name);
220
221    virtual uint32_t getPrimaryOutputSamplingRate();
222    virtual size_t getPrimaryOutputFrameCount();
223
224    virtual status_t setLowRamDevice(bool isLowRamDevice);
225
226    virtual     status_t    onTransact(
227                                uint32_t code,
228                                const Parcel& data,
229                                Parcel* reply,
230                                uint32_t flags);
231
232    // end of IAudioFlinger interface
233
234    sp<NBLog::Writer>   newWriter_l(size_t size, const char *name);
235    void                unregisterWriter(const sp<NBLog::Writer>& writer);
236private:
237    static const size_t kLogMemorySize = 40 * 1024;
238    sp<MemoryDealer>    mLogMemoryDealer;   // == 0 when NBLog is disabled
239    // When a log writer is unregistered, it is done lazily so that media.log can continue to see it
240    // for as long as possible.  The memory is only freed when it is needed for another log writer.
241    Vector< sp<NBLog::Writer> > mUnregisteredWriters;
242    Mutex               mUnregisteredWritersLock;
243public:
244
245    class SyncEvent;
246
247    typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ;
248
249    class SyncEvent : public RefBase {
250    public:
251        SyncEvent(AudioSystem::sync_event_t type,
252                  int triggerSession,
253                  int listenerSession,
254                  sync_event_callback_t callBack,
255                  wp<RefBase> cookie)
256        : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession),
257          mCallback(callBack), mCookie(cookie)
258        {}
259
260        virtual ~SyncEvent() {}
261
262        void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); }
263        bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); }
264        void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; }
265        AudioSystem::sync_event_t type() const { return mType; }
266        int triggerSession() const { return mTriggerSession; }
267        int listenerSession() const { return mListenerSession; }
268        wp<RefBase> cookie() const { return mCookie; }
269
270    private:
271          const AudioSystem::sync_event_t mType;
272          const int mTriggerSession;
273          const int mListenerSession;
274          sync_event_callback_t mCallback;
275          const wp<RefBase> mCookie;
276          mutable Mutex mLock;
277    };
278
279    sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
280                                        int triggerSession,
281                                        int listenerSession,
282                                        sync_event_callback_t callBack,
283                                        wp<RefBase> cookie);
284
285private:
286    class AudioHwDevice;    // fwd declaration for findSuitableHwDev_l
287
288               audio_mode_t getMode() const { return mMode; }
289
290                bool        btNrecIsOff() const { return mBtNrecIsOff; }
291
292                            AudioFlinger() ANDROID_API;
293    virtual                 ~AudioFlinger();
294
295    // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
296    status_t                initCheck() const { return mPrimaryHardwareDev == NULL ?
297                                                        NO_INIT : NO_ERROR; }
298
299    // RefBase
300    virtual     void        onFirstRef();
301
302    AudioHwDevice*          findSuitableHwDev_l(audio_module_handle_t module,
303                                                audio_devices_t devices);
304    void                    purgeStaleEffects_l();
305
306    // standby delay for MIXER and DUPLICATING playback threads is read from property
307    // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
308    static nsecs_t          mStandbyTimeInNsecs;
309
310    // incremented by 2 when screen state changes, bit 0 == 1 means "off"
311    // AudioFlinger::setParameters() updates, other threads read w/o lock
312    static uint32_t         mScreenState;
313
314    // Internal dump utilities.
315    static const int kDumpLockRetries = 50;
316    static const int kDumpLockSleepUs = 20000;
317    static bool dumpTryLock(Mutex& mutex);
318    void dumpPermissionDenial(int fd, const Vector<String16>& args);
319    void dumpClients(int fd, const Vector<String16>& args);
320    void dumpInternals(int fd, const Vector<String16>& args);
321
322    // --- Client ---
323    class Client : public RefBase {
324    public:
325                            Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
326        virtual             ~Client();
327        sp<MemoryDealer>    heap() const;
328        pid_t               pid() const { return mPid; }
329        sp<AudioFlinger>    audioFlinger() const { return mAudioFlinger; }
330
331        bool reserveTimedTrack();
332        void releaseTimedTrack();
333
334    private:
335                            Client(const Client&);
336                            Client& operator = (const Client&);
337        const sp<AudioFlinger> mAudioFlinger;
338        const sp<MemoryDealer> mMemoryDealer;
339        const pid_t         mPid;
340
341        Mutex               mTimedTrackLock;
342        int                 mTimedTrackCount;
343    };
344
345    // --- Notification Client ---
346    class NotificationClient : public IBinder::DeathRecipient {
347    public:
348                            NotificationClient(const sp<AudioFlinger>& audioFlinger,
349                                                const sp<IAudioFlingerClient>& client,
350                                                pid_t pid);
351        virtual             ~NotificationClient();
352
353                sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
354
355                // IBinder::DeathRecipient
356                virtual     void        binderDied(const wp<IBinder>& who);
357
358    private:
359                            NotificationClient(const NotificationClient&);
360                            NotificationClient& operator = (const NotificationClient&);
361
362        const sp<AudioFlinger>  mAudioFlinger;
363        const pid_t             mPid;
364        const sp<IAudioFlingerClient> mAudioFlingerClient;
365    };
366
367    class TrackHandle;
368    class RecordHandle;
369    class RecordThread;
370    class PlaybackThread;
371    class MixerThread;
372    class DirectOutputThread;
373    class OffloadThread;
374    class DuplicatingThread;
375    class AsyncCallbackThread;
376    class Track;
377    class RecordTrack;
378    class EffectModule;
379    class EffectHandle;
380    class EffectChain;
381    struct AudioStreamOut;
382    struct AudioStreamIn;
383
384    struct  stream_type_t {
385        stream_type_t()
386            :   volume(1.0f),
387                mute(false)
388        {
389        }
390        float       volume;
391        bool        mute;
392    };
393
394    // --- PlaybackThread ---
395
396#include "Threads.h"
397
398#include "Effects.h"
399
400    // server side of the client's IAudioTrack
401    class TrackHandle : public android::BnAudioTrack {
402    public:
403                            TrackHandle(const sp<PlaybackThread::Track>& track);
404        virtual             ~TrackHandle();
405        virtual sp<IMemory> getCblk() const;
406        virtual status_t    start();
407        virtual void        stop();
408        virtual void        flush();
409        virtual void        pause();
410        virtual status_t    attachAuxEffect(int effectId);
411        virtual status_t    allocateTimedBuffer(size_t size,
412                                                sp<IMemory>* buffer);
413        virtual status_t    queueTimedBuffer(const sp<IMemory>& buffer,
414                                             int64_t pts);
415        virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
416                                                  int target);
417        virtual status_t    setParameters(const String8& keyValuePairs);
418        virtual status_t    getTimestamp(AudioTimestamp& timestamp);
419        virtual void        signal(); // signal playback thread for a change in control block
420
421        virtual status_t onTransact(
422            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
423
424    private:
425        const sp<PlaybackThread::Track> mTrack;
426    };
427
428    // server side of the client's IAudioRecord
429    class RecordHandle : public android::BnAudioRecord {
430    public:
431        RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
432        virtual             ~RecordHandle();
433        virtual sp<IMemory> getCblk() const;
434        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
435        virtual void        stop();
436        virtual status_t onTransact(
437            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
438    private:
439        const sp<RecordThread::RecordTrack> mRecordTrack;
440
441        // for use from destructor
442        void                stop_nonvirtual();
443    };
444
445
446              PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
447              MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
448              RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
449              // no range check, AudioFlinger::mLock held
450              bool streamMute_l(audio_stream_type_t stream) const
451                                { return mStreamTypes[stream].mute; }
452              // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held
453              float streamVolume_l(audio_stream_type_t stream) const
454                                { return mStreamTypes[stream].volume; }
455              void audioConfigChanged_l(const DefaultKeyedVector< pid_t,sp<NotificationClient> >&
456                                           notificationClients,
457                                        int event,
458                                        audio_io_handle_t ioHandle,
459                                        const void *param2);
460
461              // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t.
462              // They all share the same ID space, but the namespaces are actually independent
463              // because there are separate KeyedVectors for each kind of ID.
464              // The return value is uint32_t, but is cast to signed for some IDs.
465              // FIXME This API does not handle rollover to zero (for unsigned IDs),
466              //       or from positive to negative (for signed IDs).
467              //       Thus it may fail by returning an ID of the wrong sign,
468              //       or by returning a non-unique ID.
469              uint32_t nextUniqueId();
470
471              status_t moveEffectChain_l(int sessionId,
472                                     PlaybackThread *srcThread,
473                                     PlaybackThread *dstThread,
474                                     bool reRegister);
475              // return thread associated with primary hardware device, or NULL
476              PlaybackThread *primaryPlaybackThread_l() const;
477              audio_devices_t primaryOutputDevice_l() const;
478
479              sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
480
481
482                void        removeClient_l(pid_t pid);
483                void        removeNotificationClient(pid_t pid);
484                DefaultKeyedVector< pid_t,sp<NotificationClient> > notificationClients() {
485                                        Mutex::Autolock _l(mLock); return mNotificationClients; }
486                bool isNonOffloadableGlobalEffectEnabled_l();
487                void onNonOffloadableGlobalEffectEnable();
488
489    class AudioHwDevice {
490    public:
491        enum Flags {
492            AHWD_CAN_SET_MASTER_VOLUME  = 0x1,
493            AHWD_CAN_SET_MASTER_MUTE    = 0x2,
494        };
495
496        AudioHwDevice(const char *moduleName,
497                      audio_hw_device_t *hwDevice,
498                      Flags flags)
499            : mModuleName(strdup(moduleName))
500            , mHwDevice(hwDevice)
501            , mFlags(flags) { }
502        /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); }
503
504        bool canSetMasterVolume() const {
505            return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME));
506        }
507
508        bool canSetMasterMute() const {
509            return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE));
510        }
511
512        const char *moduleName() const { return mModuleName; }
513        audio_hw_device_t *hwDevice() const { return mHwDevice; }
514    private:
515        const char * const mModuleName;
516        audio_hw_device_t * const mHwDevice;
517        const Flags mFlags;
518    };
519
520    // AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
521    // For emphasis, we could also make all pointers to them be "const *",
522    // but that would clutter the code unnecessarily.
523
524    struct AudioStreamOut {
525        AudioHwDevice* const audioHwDev;
526        audio_stream_out_t* const stream;
527        const audio_output_flags_t flags;
528
529        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
530
531        AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) :
532            audioHwDev(dev), stream(out), flags(flags) {}
533    };
534
535    struct AudioStreamIn {
536        AudioHwDevice* const audioHwDev;
537        audio_stream_in_t* const stream;
538
539        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
540
541        AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) :
542            audioHwDev(dev), stream(in) {}
543    };
544
545    // for mAudioSessionRefs only
546    struct AudioSessionRef {
547        AudioSessionRef(int sessionid, pid_t pid) :
548            mSessionid(sessionid), mPid(pid), mCnt(1) {}
549        const int   mSessionid;
550        const pid_t mPid;
551        int         mCnt;
552    };
553
554    mutable     Mutex                               mLock;
555
556                DefaultKeyedVector< pid_t, wp<Client> >     mClients;   // see ~Client()
557
558                mutable     Mutex                   mHardwareLock;
559                // NOTE: If both mLock and mHardwareLock mutexes must be held,
560                // always take mLock before mHardwareLock
561
562                // These two fields are immutable after onFirstRef(), so no lock needed to access
563                AudioHwDevice*                      mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
564                DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*>  mAudioHwDevs;
565
566    // for dump, indicates which hardware operation is currently in progress (but not stream ops)
567    enum hardware_call_state {
568        AUDIO_HW_IDLE = 0,              // no operation in progress
569        AUDIO_HW_INIT,                  // init_check
570        AUDIO_HW_OUTPUT_OPEN,           // open_output_stream
571        AUDIO_HW_OUTPUT_CLOSE,          // unused
572        AUDIO_HW_INPUT_OPEN,            // unused
573        AUDIO_HW_INPUT_CLOSE,           // unused
574        AUDIO_HW_STANDBY,               // unused
575        AUDIO_HW_SET_MASTER_VOLUME,     // set_master_volume
576        AUDIO_HW_GET_ROUTING,           // unused
577        AUDIO_HW_SET_ROUTING,           // unused
578        AUDIO_HW_GET_MODE,              // unused
579        AUDIO_HW_SET_MODE,              // set_mode
580        AUDIO_HW_GET_MIC_MUTE,          // get_mic_mute
581        AUDIO_HW_SET_MIC_MUTE,          // set_mic_mute
582        AUDIO_HW_SET_VOICE_VOLUME,      // set_voice_volume
583        AUDIO_HW_SET_PARAMETER,         // set_parameters
584        AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size
585        AUDIO_HW_GET_MASTER_VOLUME,     // get_master_volume
586        AUDIO_HW_GET_PARAMETER,         // get_parameters
587        AUDIO_HW_SET_MASTER_MUTE,       // set_master_mute
588        AUDIO_HW_GET_MASTER_MUTE,       // get_master_mute
589    };
590
591    mutable     hardware_call_state                 mHardwareStatus;    // for dump only
592
593
594                DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> >  mPlaybackThreads;
595                stream_type_t                       mStreamTypes[AUDIO_STREAM_CNT];
596
597                // member variables below are protected by mLock
598                float                               mMasterVolume;
599                bool                                mMasterMute;
600                // end of variables protected by mLock
601
602                DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> >    mRecordThreads;
603
604                DefaultKeyedVector< pid_t, sp<NotificationClient> >    mNotificationClients;
605
606                volatile int32_t                    mNextUniqueId;  // updated by android_atomic_inc
607                // nextUniqueId() returns uint32_t, but this is declared int32_t
608                // because the atomic operations require an int32_t
609
610                audio_mode_t                        mMode;
611                bool                                mBtNrecIsOff;
612
613                // protected by mLock
614                Vector<AudioSessionRef*> mAudioSessionRefs;
615
616                float       masterVolume_l() const;
617                bool        masterMute_l() const;
618                audio_module_handle_t loadHwModule_l(const char *name);
619
620                Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session
621                                                             // to be created
622
623private:
624    sp<Client>  registerPid_l(pid_t pid);    // always returns non-0
625
626    // for use from destructor
627    status_t    closeOutput_nonvirtual(audio_io_handle_t output);
628    status_t    closeInput_nonvirtual(audio_io_handle_t input);
629
630#ifdef TEE_SINK
631    // all record threads serially share a common tee sink, which is re-created on format change
632    sp<NBAIO_Sink>   mRecordTeeSink;
633    sp<NBAIO_Source> mRecordTeeSource;
634#endif
635
636public:
637
638#ifdef TEE_SINK
639    // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file
640    static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0);
641
642    // whether tee sink is enabled by property
643    static bool mTeeSinkInputEnabled;
644    static bool mTeeSinkOutputEnabled;
645    static bool mTeeSinkTrackEnabled;
646
647    // runtime configured size of each tee sink pipe, in frames
648    static size_t mTeeSinkInputFrames;
649    static size_t mTeeSinkOutputFrames;
650    static size_t mTeeSinkTrackFrames;
651
652    // compile-time default size of tee sink pipes, in frames
653    // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes
654    static const size_t kTeeSinkInputFramesDefault = 0x200000;
655    static const size_t kTeeSinkOutputFramesDefault = 0x200000;
656    static const size_t kTeeSinkTrackFramesDefault = 0x200000;
657#endif
658
659    // This method reads from a variable without mLock, but the variable is updated under mLock.  So
660    // we might read a stale value, or a value that's inconsistent with respect to other variables.
661    // In this case, it's safe because the return value isn't used for making an important decision.
662    // The reason we don't want to take mLock is because it could block the caller for a long time.
663    bool    isLowRamDevice() const { return mIsLowRamDevice; }
664
665private:
666    bool    mIsLowRamDevice;
667    bool    mIsDeviceTypeKnown;
668    nsecs_t mGlobalEffectEnableTime;  // when a global effect was last enabled
669};
670
671#undef INCLUDING_FROM_AUDIOFLINGER_H
672
673const char *formatToString(audio_format_t format);
674
675// ----------------------------------------------------------------------------
676
677}; // namespace android
678
679#endif // ANDROID_AUDIO_FLINGER_H
680