AudioFlinger.h revision ec40d284218466d8f0e832e7eb88e6ea6c479c88 
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 "Configuration.h"
22#include <stdint.h>
23#include <sys/types.h>
24#include <limits.h>
25
26#include <common_time/cc_helper.h>
27
28#include <cutils/compiler.h>
29
30#include <media/IAudioFlinger.h>
31#include <media/IAudioFlingerClient.h>
32#include <media/IAudioTrack.h>
33#include <media/IAudioRecord.h>
34#include <media/AudioSystem.h>
35#include <media/AudioTrack.h>
36
37#include <utils/Atomic.h>
38#include <utils/Errors.h>
39#include <utils/threads.h>
40#include <utils/SortedVector.h>
41#include <utils/TypeHelpers.h>
42#include <utils/Vector.h>
43
44#include <binder/BinderService.h>
45#include <binder/MemoryDealer.h>
46
47#include <system/audio.h>
48#include <hardware/audio.h>
49#include <hardware/audio_policy.h>
50
51#include <media/AudioBufferProvider.h>
52#include <media/ExtendedAudioBufferProvider.h>
53
54#include "FastCapture.h"
55#include "FastMixer.h"
56#include <media/nbaio/NBAIO.h>
57#include "AudioWatchdog.h"
58
59#include <powermanager/IPowerManager.h>
60
61#include <media/nbaio/NBLog.h>
62#include <private/media/AudioTrackShared.h>
63
64namespace android {
65
66struct audio_track_cblk_t;
67struct effect_param_cblk_t;
68class AudioMixer;
69class AudioBuffer;
70class AudioResampler;
71class FastMixer;
72class ServerProxy;
73
74// ----------------------------------------------------------------------------
75
76// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback.
77// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect.
78// Adding full support for > 2 channel capture or playback would require more than simply changing
79// this #define.  There is an independent hard-coded upper limit in AudioMixer;
80// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels.
81// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions.
82// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc.
83#define FCC_2 2     // FCC_2 = Fixed Channel Count 2
84
85static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
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                                size_t *notificationFrames,
124                                sp<IMemory>& cblk,
125                                sp<IMemory>& buffers,
126                                status_t *status /*non-NULL*/);
127
128    virtual     uint32_t    sampleRate(audio_io_handle_t output) const;
129    virtual     audio_format_t format(audio_io_handle_t output) const;
130    virtual     size_t      frameCount(audio_io_handle_t output) const;
131    virtual     uint32_t    latency(audio_io_handle_t output) const;
132
133    virtual     status_t    setMasterVolume(float value);
134    virtual     status_t    setMasterMute(bool muted);
135
136    virtual     float       masterVolume() const;
137    virtual     bool        masterMute() const;
138
139    virtual     status_t    setStreamVolume(audio_stream_type_t stream, float value,
140                                            audio_io_handle_t output);
141    virtual     status_t    setStreamMute(audio_stream_type_t stream, bool muted);
142
143    virtual     float       streamVolume(audio_stream_type_t stream,
144                                         audio_io_handle_t output) const;
145    virtual     bool        streamMute(audio_stream_type_t stream) const;
146
147    virtual     status_t    setMode(audio_mode_t mode);
148
149    virtual     status_t    setMicMute(bool state);
150    virtual     bool        getMicMute() const;
151
152    virtual     status_t    setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
153    virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
154
155    virtual     void        registerClient(const sp<IAudioFlingerClient>& client);
156
157    virtual     size_t      getInputBufferSize(uint32_t sampleRate, audio_format_t format,
158                                               audio_channel_mask_t channelMask) const;
159
160    virtual audio_io_handle_t openOutput(audio_module_handle_t module,
161                                         audio_devices_t *pDevices,
162                                         uint32_t *pSamplingRate,
163                                         audio_format_t *pFormat,
164                                         audio_channel_mask_t *pChannelMask,
165                                         uint32_t *pLatencyMs,
166                                         audio_output_flags_t flags,
167                                         const audio_offload_info_t *offloadInfo);
168
169    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
170                                                  audio_io_handle_t output2);
171
172    virtual status_t closeOutput(audio_io_handle_t output);
173
174    virtual status_t suspendOutput(audio_io_handle_t output);
175
176    virtual status_t restoreOutput(audio_io_handle_t output);
177
178    virtual audio_io_handle_t openInput(audio_module_handle_t module,
179                                        audio_devices_t *pDevices,
180                                        uint32_t *pSamplingRate,
181                                        audio_format_t *pFormat,
182                                        audio_channel_mask_t *pChannelMask,
183                                        audio_input_flags_t flags);
184
185    virtual status_t closeInput(audio_io_handle_t input);
186
187    virtual status_t invalidateStream(audio_stream_type_t stream);
188
189    virtual status_t setVoiceVolume(float volume);
190
191    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
192                                       audio_io_handle_t output) const;
193
194    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
195
196    virtual int newAudioSessionId();
197
198    virtual void acquireAudioSessionId(int audioSession, pid_t pid);
199
200    virtual void releaseAudioSessionId(int audioSession, pid_t pid);
201
202    virtual status_t queryNumberEffects(uint32_t *numEffects) const;
203
204    virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
205
206    virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
207                                         effect_descriptor_t *descriptor) const;
208
209    virtual sp<IEffect> createEffect(
210                        effect_descriptor_t *pDesc,
211                        const sp<IEffectClient>& effectClient,
212                        int32_t priority,
213                        audio_io_handle_t io,
214                        int sessionId,
215                        status_t *status /*non-NULL*/,
216                        int *id,
217                        int *enabled);
218
219    virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
220                        audio_io_handle_t dstOutput);
221
222    virtual audio_module_handle_t loadHwModule(const char *name);
223
224    virtual uint32_t getPrimaryOutputSamplingRate();
225    virtual size_t getPrimaryOutputFrameCount();
226
227    virtual status_t setLowRamDevice(bool isLowRamDevice);
228
229    /* List available audio ports and their attributes */
230    virtual status_t listAudioPorts(unsigned int *num_ports,
231                                    struct audio_port *ports);
232
233    /* Get attributes for a given audio port */
234    virtual status_t getAudioPort(struct audio_port *port);
235
236    /* Create an audio patch between several source and sink ports */
237    virtual status_t createAudioPatch(const struct audio_patch *patch,
238                                       audio_patch_handle_t *handle);
239
240    /* Release an audio patch */
241    virtual status_t releaseAudioPatch(audio_patch_handle_t handle);
242
243    /* List existing audio patches */
244    virtual status_t listAudioPatches(unsigned int *num_patches,
245                                      struct audio_patch *patches);
246
247    /* Set audio port configuration */
248    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
249
250    virtual     status_t    onTransact(
251                                uint32_t code,
252                                const Parcel& data,
253                                Parcel* reply,
254                                uint32_t flags);
255
256    // end of IAudioFlinger interface
257
258    sp<NBLog::Writer>   newWriter_l(size_t size, const char *name);
259    void                unregisterWriter(const sp<NBLog::Writer>& writer);
260private:
261    static const size_t kLogMemorySize = 40 * 1024;
262    sp<MemoryDealer>    mLogMemoryDealer;   // == 0 when NBLog is disabled
263    // When a log writer is unregistered, it is done lazily so that media.log can continue to see it
264    // for as long as possible.  The memory is only freed when it is needed for another log writer.
265    Vector< sp<NBLog::Writer> > mUnregisteredWriters;
266    Mutex               mUnregisteredWritersLock;
267public:
268
269    class SyncEvent;
270
271    typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ;
272
273    class SyncEvent : public RefBase {
274    public:
275        SyncEvent(AudioSystem::sync_event_t type,
276                  int triggerSession,
277                  int listenerSession,
278                  sync_event_callback_t callBack,
279                  wp<RefBase> cookie)
280        : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession),
281          mCallback(callBack), mCookie(cookie)
282        {}
283
284        virtual ~SyncEvent() {}
285
286        void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); }
287        bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); }
288        void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; }
289        AudioSystem::sync_event_t type() const { return mType; }
290        int triggerSession() const { return mTriggerSession; }
291        int listenerSession() const { return mListenerSession; }
292        wp<RefBase> cookie() const { return mCookie; }
293
294    private:
295          const AudioSystem::sync_event_t mType;
296          const int mTriggerSession;
297          const int mListenerSession;
298          sync_event_callback_t mCallback;
299          const wp<RefBase> mCookie;
300          mutable Mutex mLock;
301    };
302
303    sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
304                                        int triggerSession,
305                                        int listenerSession,
306                                        sync_event_callback_t callBack,
307                                        wp<RefBase> cookie);
308
309private:
310    class AudioHwDevice;    // fwd declaration for findSuitableHwDev_l
311
312               audio_mode_t getMode() const { return mMode; }
313
314                bool        btNrecIsOff() const { return mBtNrecIsOff; }
315
316                            AudioFlinger() ANDROID_API;
317    virtual                 ~AudioFlinger();
318
319    // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
320    status_t                initCheck() const { return mPrimaryHardwareDev == NULL ?
321                                                        NO_INIT : NO_ERROR; }
322
323    // RefBase
324    virtual     void        onFirstRef();
325
326    AudioHwDevice*          findSuitableHwDev_l(audio_module_handle_t module,
327                                                audio_devices_t devices);
328    void                    purgeStaleEffects_l();
329
330    // Set kEnableExtendedPrecision to true to use extended precision in MixerThread
331    static const bool kEnableExtendedPrecision = true;
332
333    // Returns true if format is permitted for the PCM sink in the MixerThread
334    static inline bool isValidPcmSinkFormat(audio_format_t format) {
335        switch (format) {
336        case AUDIO_FORMAT_PCM_16_BIT:
337            return true;
338        case AUDIO_FORMAT_PCM_FLOAT:
339        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
340        case AUDIO_FORMAT_PCM_32_BIT:
341        case AUDIO_FORMAT_PCM_8_24_BIT:
342            return kEnableExtendedPrecision;
343        default:
344            return false;
345        }
346    }
347
348    // standby delay for MIXER and DUPLICATING playback threads is read from property
349    // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
350    static nsecs_t          mStandbyTimeInNsecs;
351
352    // incremented by 2 when screen state changes, bit 0 == 1 means "off"
353    // AudioFlinger::setParameters() updates, other threads read w/o lock
354    static uint32_t         mScreenState;
355
356    // Internal dump utilities.
357    static const int kDumpLockRetries = 50;
358    static const int kDumpLockSleepUs = 20000;
359    static bool dumpTryLock(Mutex& mutex);
360    void dumpPermissionDenial(int fd, const Vector<String16>& args);
361    void dumpClients(int fd, const Vector<String16>& args);
362    void dumpInternals(int fd, const Vector<String16>& args);
363
364    // --- Client ---
365    class Client : public RefBase {
366    public:
367                            Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
368        virtual             ~Client();
369        sp<MemoryDealer>    heap() const;
370        pid_t               pid() const { return mPid; }
371        sp<AudioFlinger>    audioFlinger() const { return mAudioFlinger; }
372
373        bool reserveTimedTrack();
374        void releaseTimedTrack();
375
376    private:
377                            Client(const Client&);
378                            Client& operator = (const Client&);
379        const sp<AudioFlinger> mAudioFlinger;
380        const sp<MemoryDealer> mMemoryDealer;
381        const pid_t         mPid;
382
383        Mutex               mTimedTrackLock;
384        int                 mTimedTrackCount;
385    };
386
387    // --- Notification Client ---
388    class NotificationClient : public IBinder::DeathRecipient {
389    public:
390                            NotificationClient(const sp<AudioFlinger>& audioFlinger,
391                                                const sp<IAudioFlingerClient>& client,
392                                                pid_t pid);
393        virtual             ~NotificationClient();
394
395                sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
396
397                // IBinder::DeathRecipient
398                virtual     void        binderDied(const wp<IBinder>& who);
399
400    private:
401                            NotificationClient(const NotificationClient&);
402                            NotificationClient& operator = (const NotificationClient&);
403
404        const sp<AudioFlinger>  mAudioFlinger;
405        const pid_t             mPid;
406        const sp<IAudioFlingerClient> mAudioFlingerClient;
407    };
408
409    class TrackHandle;
410    class RecordHandle;
411    class RecordThread;
412    class PlaybackThread;
413    class MixerThread;
414    class DirectOutputThread;
415    class OffloadThread;
416    class DuplicatingThread;
417    class AsyncCallbackThread;
418    class Track;
419    class RecordTrack;
420    class EffectModule;
421    class EffectHandle;
422    class EffectChain;
423    struct AudioStreamOut;
424    struct AudioStreamIn;
425
426    struct  stream_type_t {
427        stream_type_t()
428            :   volume(1.0f),
429                mute(false)
430        {
431        }
432        float       volume;
433        bool        mute;
434    };
435
436    // --- PlaybackThread ---
437
438#include "Threads.h"
439
440#include "Effects.h"
441
442#include "PatchPanel.h"
443
444    // server side of the client's IAudioTrack
445    class TrackHandle : public android::BnAudioTrack {
446    public:
447                            TrackHandle(const sp<PlaybackThread::Track>& track);
448        virtual             ~TrackHandle();
449        virtual sp<IMemory> getCblk() const;
450        virtual status_t    start();
451        virtual void        stop();
452        virtual void        flush();
453        virtual void        pause();
454        virtual status_t    attachAuxEffect(int effectId);
455        virtual status_t    allocateTimedBuffer(size_t size,
456                                                sp<IMemory>* buffer);
457        virtual status_t    queueTimedBuffer(const sp<IMemory>& buffer,
458                                             int64_t pts);
459        virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
460                                                  int target);
461        virtual status_t    setParameters(const String8& keyValuePairs);
462        virtual status_t    getTimestamp(AudioTimestamp& timestamp);
463        virtual void        signal(); // signal playback thread for a change in control block
464
465        virtual status_t onTransact(
466            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
467
468    private:
469        const sp<PlaybackThread::Track> mTrack;
470    };
471
472    // server side of the client's IAudioRecord
473    class RecordHandle : public android::BnAudioRecord {
474    public:
475        RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
476        virtual             ~RecordHandle();
477        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
478        virtual void        stop();
479        virtual status_t onTransact(
480            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
481    private:
482        const sp<RecordThread::RecordTrack> mRecordTrack;
483
484        // for use from destructor
485        void                stop_nonvirtual();
486    };
487
488
489              PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
490              MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
491              RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
492              // no range check, AudioFlinger::mLock held
493              bool streamMute_l(audio_stream_type_t stream) const
494                                { return mStreamTypes[stream].mute; }
495              // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held
496              float streamVolume_l(audio_stream_type_t stream) const
497                                { return mStreamTypes[stream].volume; }
498              void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2);
499
500              // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t.
501              // They all share the same ID space, but the namespaces are actually independent
502              // because there are separate KeyedVectors for each kind of ID.
503              // The return value is uint32_t, but is cast to signed for some IDs.
504              // FIXME This API does not handle rollover to zero (for unsigned IDs),
505              //       or from positive to negative (for signed IDs).
506              //       Thus it may fail by returning an ID of the wrong sign,
507              //       or by returning a non-unique ID.
508              uint32_t nextUniqueId();
509
510              status_t moveEffectChain_l(int sessionId,
511                                     PlaybackThread *srcThread,
512                                     PlaybackThread *dstThread,
513                                     bool reRegister);
514              // return thread associated with primary hardware device, or NULL
515              PlaybackThread *primaryPlaybackThread_l() const;
516              audio_devices_t primaryOutputDevice_l() const;
517
518              sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
519
520
521                void        removeClient_l(pid_t pid);
522                void        removeNotificationClient(pid_t pid);
523                bool isNonOffloadableGlobalEffectEnabled_l();
524                void onNonOffloadableGlobalEffectEnable();
525
526    class AudioHwDevice {
527    public:
528        enum Flags {
529            AHWD_CAN_SET_MASTER_VOLUME  = 0x1,
530            AHWD_CAN_SET_MASTER_MUTE    = 0x2,
531        };
532
533        AudioHwDevice(const char *moduleName,
534                      audio_hw_device_t *hwDevice,
535                      Flags flags)
536            : mModuleName(strdup(moduleName))
537            , mHwDevice(hwDevice)
538            , mFlags(flags) { }
539        /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); }
540
541        bool canSetMasterVolume() const {
542            return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME));
543        }
544
545        bool canSetMasterMute() const {
546            return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE));
547        }
548
549        const char *moduleName() const { return mModuleName; }
550        audio_hw_device_t *hwDevice() const { return mHwDevice; }
551        uint32_t version() const { return mHwDevice->common.version; }
552
553    private:
554        const char * const mModuleName;
555        audio_hw_device_t * const mHwDevice;
556        const Flags mFlags;
557    };
558
559    // AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
560    // For emphasis, we could also make all pointers to them be "const *",
561    // but that would clutter the code unnecessarily.
562
563    struct AudioStreamOut {
564        AudioHwDevice* const audioHwDev;
565        audio_stream_out_t* const stream;
566        const audio_output_flags_t flags;
567
568        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
569
570        AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) :
571            audioHwDev(dev), stream(out), flags(flags) {}
572    };
573
574    struct AudioStreamIn {
575        AudioHwDevice* const audioHwDev;
576        audio_stream_in_t* const stream;
577
578        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
579
580        AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) :
581            audioHwDev(dev), stream(in) {}
582    };
583
584    // for mAudioSessionRefs only
585    struct AudioSessionRef {
586        AudioSessionRef(int sessionid, pid_t pid) :
587            mSessionid(sessionid), mPid(pid), mCnt(1) {}
588        const int   mSessionid;
589        const pid_t mPid;
590        int         mCnt;
591    };
592
593    mutable     Mutex                               mLock;
594                // protects mClients and mNotificationClients.
595                // must be locked after mLock and ThreadBase::mLock if both must be locked
596                // avoids acquiring AudioFlinger::mLock from inside thread loop.
597    mutable     Mutex                               mClientLock;
598                // protected by mClientLock
599                DefaultKeyedVector< pid_t, wp<Client> >     mClients;   // see ~Client()
600
601                mutable     Mutex                   mHardwareLock;
602                // NOTE: If both mLock and mHardwareLock mutexes must be held,
603                // always take mLock before mHardwareLock
604
605                // These two fields are immutable after onFirstRef(), so no lock needed to access
606                AudioHwDevice*                      mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
607                DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*>  mAudioHwDevs;
608
609    // for dump, indicates which hardware operation is currently in progress (but not stream ops)
610    enum hardware_call_state {
611        AUDIO_HW_IDLE = 0,              // no operation in progress
612        AUDIO_HW_INIT,                  // init_check
613        AUDIO_HW_OUTPUT_OPEN,           // open_output_stream
614        AUDIO_HW_OUTPUT_CLOSE,          // unused
615        AUDIO_HW_INPUT_OPEN,            // unused
616        AUDIO_HW_INPUT_CLOSE,           // unused
617        AUDIO_HW_STANDBY,               // unused
618        AUDIO_HW_SET_MASTER_VOLUME,     // set_master_volume
619        AUDIO_HW_GET_ROUTING,           // unused
620        AUDIO_HW_SET_ROUTING,           // unused
621        AUDIO_HW_GET_MODE,              // unused
622        AUDIO_HW_SET_MODE,              // set_mode
623        AUDIO_HW_GET_MIC_MUTE,          // get_mic_mute
624        AUDIO_HW_SET_MIC_MUTE,          // set_mic_mute
625        AUDIO_HW_SET_VOICE_VOLUME,      // set_voice_volume
626        AUDIO_HW_SET_PARAMETER,         // set_parameters
627        AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size
628        AUDIO_HW_GET_MASTER_VOLUME,     // get_master_volume
629        AUDIO_HW_GET_PARAMETER,         // get_parameters
630        AUDIO_HW_SET_MASTER_MUTE,       // set_master_mute
631        AUDIO_HW_GET_MASTER_MUTE,       // get_master_mute
632    };
633
634    mutable     hardware_call_state                 mHardwareStatus;    // for dump only
635
636
637                DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> >  mPlaybackThreads;
638                stream_type_t                       mStreamTypes[AUDIO_STREAM_CNT];
639
640                // member variables below are protected by mLock
641                float                               mMasterVolume;
642                bool                                mMasterMute;
643                // end of variables protected by mLock
644
645                DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> >    mRecordThreads;
646
647                // protected by mClientLock
648                DefaultKeyedVector< pid_t, sp<NotificationClient> >    mNotificationClients;
649
650                volatile int32_t                    mNextUniqueId;  // updated by android_atomic_inc
651                // nextUniqueId() returns uint32_t, but this is declared int32_t
652                // because the atomic operations require an int32_t
653
654                audio_mode_t                        mMode;
655                bool                                mBtNrecIsOff;
656
657                // protected by mLock
658                Vector<AudioSessionRef*> mAudioSessionRefs;
659
660                float       masterVolume_l() const;
661                bool        masterMute_l() const;
662                audio_module_handle_t loadHwModule_l(const char *name);
663
664                Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session
665                                                             // to be created
666
667private:
668    sp<Client>  registerPid(pid_t pid);    // always returns non-0
669
670    // for use from destructor
671    status_t    closeOutput_nonvirtual(audio_io_handle_t output);
672    status_t    closeInput_nonvirtual(audio_io_handle_t input);
673
674#ifdef TEE_SINK
675    // all record threads serially share a common tee sink, which is re-created on format change
676    sp<NBAIO_Sink>   mRecordTeeSink;
677    sp<NBAIO_Source> mRecordTeeSource;
678#endif
679
680public:
681
682#ifdef TEE_SINK
683    // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file
684    static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0);
685
686    // whether tee sink is enabled by property
687    static bool mTeeSinkInputEnabled;
688    static bool mTeeSinkOutputEnabled;
689    static bool mTeeSinkTrackEnabled;
690
691    // runtime configured size of each tee sink pipe, in frames
692    static size_t mTeeSinkInputFrames;
693    static size_t mTeeSinkOutputFrames;
694    static size_t mTeeSinkTrackFrames;
695
696    // compile-time default size of tee sink pipes, in frames
697    // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes
698    static const size_t kTeeSinkInputFramesDefault = 0x200000;
699    static const size_t kTeeSinkOutputFramesDefault = 0x200000;
700    static const size_t kTeeSinkTrackFramesDefault = 0x200000;
701#endif
702
703    // This method reads from a variable without mLock, but the variable is updated under mLock.  So
704    // we might read a stale value, or a value that's inconsistent with respect to other variables.
705    // In this case, it's safe because the return value isn't used for making an important decision.
706    // The reason we don't want to take mLock is because it could block the caller for a long time.
707    bool    isLowRamDevice() const { return mIsLowRamDevice; }
708
709private:
710    bool    mIsLowRamDevice;
711    bool    mIsDeviceTypeKnown;
712    nsecs_t mGlobalEffectEnableTime;  // when a global effect was last enabled
713
714    sp<PatchPanel> mPatchPanel;
715
716    uint32_t    mPrimaryOutputSampleRate;   // sample rate of the primary output, or zero if none
717                                            // protected by mHardwareLock
718};
719
720#undef INCLUDING_FROM_AUDIOFLINGER_H
721
722const char *formatToString(audio_format_t format);
723
724// ----------------------------------------------------------------------------
725
726}; // namespace android
727
728#endif // ANDROID_AUDIO_FLINGER_H
729