AudioFlinger.h revision 5ba4440c11eb975ec0e104e0af1981838f42f57c 
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                                sp<IMemory>& cblk,
124                                sp<IMemory>& buffers,
125                                status_t *status /*non-NULL*/);
126
127    virtual     uint32_t    sampleRate(audio_io_handle_t output) const;
128    virtual     audio_format_t format(audio_io_handle_t output) const;
129    virtual     size_t      frameCount(audio_io_handle_t output) const;
130    virtual     uint32_t    latency(audio_io_handle_t output) const;
131
132    virtual     status_t    setMasterVolume(float value);
133    virtual     status_t    setMasterMute(bool muted);
134
135    virtual     float       masterVolume() const;
136    virtual     bool        masterMute() const;
137
138    virtual     status_t    setStreamVolume(audio_stream_type_t stream, float value,
139                                            audio_io_handle_t output);
140    virtual     status_t    setStreamMute(audio_stream_type_t stream, bool muted);
141
142    virtual     float       streamVolume(audio_stream_type_t stream,
143                                         audio_io_handle_t output) const;
144    virtual     bool        streamMute(audio_stream_type_t stream) const;
145
146    virtual     status_t    setMode(audio_mode_t mode);
147
148    virtual     status_t    setMicMute(bool state);
149    virtual     bool        getMicMute() const;
150
151    virtual     status_t    setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
152    virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
153
154    virtual     void        registerClient(const sp<IAudioFlingerClient>& client);
155
156    virtual     size_t      getInputBufferSize(uint32_t sampleRate, audio_format_t format,
157                                               audio_channel_mask_t channelMask) const;
158
159    virtual audio_io_handle_t openOutput(audio_module_handle_t module,
160                                         audio_devices_t *pDevices,
161                                         uint32_t *pSamplingRate,
162                                         audio_format_t *pFormat,
163                                         audio_channel_mask_t *pChannelMask,
164                                         uint32_t *pLatencyMs,
165                                         audio_output_flags_t flags,
166                                         const audio_offload_info_t *offloadInfo);
167
168    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
169                                                  audio_io_handle_t output2);
170
171    virtual status_t closeOutput(audio_io_handle_t output);
172
173    virtual status_t suspendOutput(audio_io_handle_t output);
174
175    virtual status_t restoreOutput(audio_io_handle_t output);
176
177    virtual audio_io_handle_t openInput(audio_module_handle_t module,
178                                        audio_devices_t *pDevices,
179                                        uint32_t *pSamplingRate,
180                                        audio_format_t *pFormat,
181                                        audio_channel_mask_t *pChannelMask);
182
183    virtual status_t closeInput(audio_io_handle_t input);
184
185    virtual status_t invalidateStream(audio_stream_type_t stream);
186
187    virtual status_t setVoiceVolume(float volume);
188
189    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
190                                       audio_io_handle_t output) const;
191
192    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
193
194    virtual int newAudioSessionId();
195
196    virtual void acquireAudioSessionId(int audioSession, pid_t pid);
197
198    virtual void releaseAudioSessionId(int audioSession, pid_t pid);
199
200    virtual status_t queryNumberEffects(uint32_t *numEffects) const;
201
202    virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
203
204    virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
205                                         effect_descriptor_t *descriptor) const;
206
207    virtual sp<IEffect> createEffect(
208                        effect_descriptor_t *pDesc,
209                        const sp<IEffectClient>& effectClient,
210                        int32_t priority,
211                        audio_io_handle_t io,
212                        int sessionId,
213                        status_t *status /*non-NULL*/,
214                        int *id,
215                        int *enabled);
216
217    virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
218                        audio_io_handle_t dstOutput);
219
220    virtual audio_module_handle_t loadHwModule(const char *name);
221
222    virtual uint32_t getPrimaryOutputSamplingRate();
223    virtual size_t getPrimaryOutputFrameCount();
224
225    virtual status_t setLowRamDevice(bool isLowRamDevice);
226
227    /* List available audio ports and their attributes */
228    virtual status_t listAudioPorts(unsigned int *num_ports,
229                                    struct audio_port *ports);
230
231    /* Get attributes for a given audio port */
232    virtual status_t getAudioPort(struct audio_port *port);
233
234    /* Create an audio patch between several source and sink ports */
235    virtual status_t createAudioPatch(const struct audio_patch *patch,
236                                       audio_patch_handle_t *handle);
237
238    /* Release an audio patch */
239    virtual status_t releaseAudioPatch(audio_patch_handle_t handle);
240
241    /* List existing audio patches */
242    virtual status_t listAudioPatches(unsigned int *num_patches,
243                                      struct audio_patch *patches);
244
245    /* Set audio port configuration */
246    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
247
248    virtual     status_t    onTransact(
249                                uint32_t code,
250                                const Parcel& data,
251                                Parcel* reply,
252                                uint32_t flags);
253
254    // end of IAudioFlinger interface
255
256    sp<NBLog::Writer>   newWriter_l(size_t size, const char *name);
257    void                unregisterWriter(const sp<NBLog::Writer>& writer);
258private:
259    static const size_t kLogMemorySize = 40 * 1024;
260    sp<MemoryDealer>    mLogMemoryDealer;   // == 0 when NBLog is disabled
261    // When a log writer is unregistered, it is done lazily so that media.log can continue to see it
262    // for as long as possible.  The memory is only freed when it is needed for another log writer.
263    Vector< sp<NBLog::Writer> > mUnregisteredWriters;
264    Mutex               mUnregisteredWritersLock;
265public:
266
267    class SyncEvent;
268
269    typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ;
270
271    class SyncEvent : public RefBase {
272    public:
273        SyncEvent(AudioSystem::sync_event_t type,
274                  int triggerSession,
275                  int listenerSession,
276                  sync_event_callback_t callBack,
277                  wp<RefBase> cookie)
278        : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession),
279          mCallback(callBack), mCookie(cookie)
280        {}
281
282        virtual ~SyncEvent() {}
283
284        void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); }
285        bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); }
286        void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; }
287        AudioSystem::sync_event_t type() const { return mType; }
288        int triggerSession() const { return mTriggerSession; }
289        int listenerSession() const { return mListenerSession; }
290        wp<RefBase> cookie() const { return mCookie; }
291
292    private:
293          const AudioSystem::sync_event_t mType;
294          const int mTriggerSession;
295          const int mListenerSession;
296          sync_event_callback_t mCallback;
297          const wp<RefBase> mCookie;
298          mutable Mutex mLock;
299    };
300
301    sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
302                                        int triggerSession,
303                                        int listenerSession,
304                                        sync_event_callback_t callBack,
305                                        wp<RefBase> cookie);
306
307private:
308    class AudioHwDevice;    // fwd declaration for findSuitableHwDev_l
309
310               audio_mode_t getMode() const { return mMode; }
311
312                bool        btNrecIsOff() const { return mBtNrecIsOff; }
313
314                            AudioFlinger() ANDROID_API;
315    virtual                 ~AudioFlinger();
316
317    // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
318    status_t                initCheck() const { return mPrimaryHardwareDev == NULL ?
319                                                        NO_INIT : NO_ERROR; }
320
321    // RefBase
322    virtual     void        onFirstRef();
323
324    AudioHwDevice*          findSuitableHwDev_l(audio_module_handle_t module,
325                                                audio_devices_t devices);
326    void                    purgeStaleEffects_l();
327
328    // Set kEnableExtendedPrecision to true to use extended precision in MixerThread
329    static const bool kEnableExtendedPrecision = false;
330
331    // Returns true if format is permitted for the PCM sink in the MixerThread
332    static inline bool isValidPcmSinkFormat(audio_format_t format) {
333        switch (format) {
334        case AUDIO_FORMAT_PCM_16_BIT:
335            return true;
336        case AUDIO_FORMAT_PCM_FLOAT:
337        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
338        case AUDIO_FORMAT_PCM_32_BIT:
339        case AUDIO_FORMAT_PCM_8_24_BIT:
340            return kEnableExtendedPrecision;
341        default:
342            return false;
343        }
344    }
345
346    // standby delay for MIXER and DUPLICATING playback threads is read from property
347    // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
348    static nsecs_t          mStandbyTimeInNsecs;
349
350    // incremented by 2 when screen state changes, bit 0 == 1 means "off"
351    // AudioFlinger::setParameters() updates, other threads read w/o lock
352    static uint32_t         mScreenState;
353
354    // Internal dump utilities.
355    static const int kDumpLockRetries = 50;
356    static const int kDumpLockSleepUs = 20000;
357    static bool dumpTryLock(Mutex& mutex);
358    void dumpPermissionDenial(int fd, const Vector<String16>& args);
359    void dumpClients(int fd, const Vector<String16>& args);
360    void dumpInternals(int fd, const Vector<String16>& args);
361
362    // --- Client ---
363    class Client : public RefBase {
364    public:
365                            Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
366        virtual             ~Client();
367        sp<MemoryDealer>    heap() const;
368        pid_t               pid() const { return mPid; }
369        sp<AudioFlinger>    audioFlinger() const { return mAudioFlinger; }
370
371        bool reserveTimedTrack();
372        void releaseTimedTrack();
373
374    private:
375                            Client(const Client&);
376                            Client& operator = (const Client&);
377        const sp<AudioFlinger> mAudioFlinger;
378        const sp<MemoryDealer> mMemoryDealer;
379        const pid_t         mPid;
380
381        Mutex               mTimedTrackLock;
382        int                 mTimedTrackCount;
383    };
384
385    // --- Notification Client ---
386    class NotificationClient : public IBinder::DeathRecipient {
387    public:
388                            NotificationClient(const sp<AudioFlinger>& audioFlinger,
389                                                const sp<IAudioFlingerClient>& client,
390                                                pid_t pid);
391        virtual             ~NotificationClient();
392
393                sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
394
395                // IBinder::DeathRecipient
396                virtual     void        binderDied(const wp<IBinder>& who);
397
398    private:
399                            NotificationClient(const NotificationClient&);
400                            NotificationClient& operator = (const NotificationClient&);
401
402        const sp<AudioFlinger>  mAudioFlinger;
403        const pid_t             mPid;
404        const sp<IAudioFlingerClient> mAudioFlingerClient;
405    };
406
407    class TrackHandle;
408    class RecordHandle;
409    class RecordThread;
410    class PlaybackThread;
411    class MixerThread;
412    class DirectOutputThread;
413    class OffloadThread;
414    class DuplicatingThread;
415    class AsyncCallbackThread;
416    class Track;
417    class RecordTrack;
418    class EffectModule;
419    class EffectHandle;
420    class EffectChain;
421    struct AudioStreamOut;
422    struct AudioStreamIn;
423
424    struct  stream_type_t {
425        stream_type_t()
426            :   volume(1.0f),
427                mute(false)
428        {
429        }
430        float       volume;
431        bool        mute;
432    };
433
434    // --- PlaybackThread ---
435
436#include "Threads.h"
437
438#include "Effects.h"
439
440#include "PatchPanel.h"
441
442    // server side of the client's IAudioTrack
443    class TrackHandle : public android::BnAudioTrack {
444    public:
445                            TrackHandle(const sp<PlaybackThread::Track>& track);
446        virtual             ~TrackHandle();
447        virtual sp<IMemory> getCblk() const;
448        virtual status_t    start();
449        virtual void        stop();
450        virtual void        flush();
451        virtual void        pause();
452        virtual status_t    attachAuxEffect(int effectId);
453        virtual status_t    allocateTimedBuffer(size_t size,
454                                                sp<IMemory>* buffer);
455        virtual status_t    queueTimedBuffer(const sp<IMemory>& buffer,
456                                             int64_t pts);
457        virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
458                                                  int target);
459        virtual status_t    setParameters(const String8& keyValuePairs);
460        virtual status_t    getTimestamp(AudioTimestamp& timestamp);
461        virtual void        signal(); // signal playback thread for a change in control block
462
463        virtual status_t onTransact(
464            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
465
466    private:
467        const sp<PlaybackThread::Track> mTrack;
468    };
469
470    // server side of the client's IAudioRecord
471    class RecordHandle : public android::BnAudioRecord {
472    public:
473        RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
474        virtual             ~RecordHandle();
475        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
476        virtual void        stop();
477        virtual status_t onTransact(
478            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
479    private:
480        const sp<RecordThread::RecordTrack> mRecordTrack;
481
482        // for use from destructor
483        void                stop_nonvirtual();
484    };
485
486
487              PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
488              MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
489              RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
490              // no range check, AudioFlinger::mLock held
491              bool streamMute_l(audio_stream_type_t stream) const
492                                { return mStreamTypes[stream].mute; }
493              // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held
494              float streamVolume_l(audio_stream_type_t stream) const
495                                { return mStreamTypes[stream].volume; }
496              void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2);
497
498              // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t.
499              // They all share the same ID space, but the namespaces are actually independent
500              // because there are separate KeyedVectors for each kind of ID.
501              // The return value is uint32_t, but is cast to signed for some IDs.
502              // FIXME This API does not handle rollover to zero (for unsigned IDs),
503              //       or from positive to negative (for signed IDs).
504              //       Thus it may fail by returning an ID of the wrong sign,
505              //       or by returning a non-unique ID.
506              uint32_t nextUniqueId();
507
508              status_t moveEffectChain_l(int sessionId,
509                                     PlaybackThread *srcThread,
510                                     PlaybackThread *dstThread,
511                                     bool reRegister);
512              // return thread associated with primary hardware device, or NULL
513              PlaybackThread *primaryPlaybackThread_l() const;
514              audio_devices_t primaryOutputDevice_l() const;
515
516              sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
517
518
519                void        removeClient_l(pid_t pid);
520                void        removeNotificationClient(pid_t pid);
521                bool isNonOffloadableGlobalEffectEnabled_l();
522                void onNonOffloadableGlobalEffectEnable();
523
524    class AudioHwDevice {
525    public:
526        enum Flags {
527            AHWD_CAN_SET_MASTER_VOLUME  = 0x1,
528            AHWD_CAN_SET_MASTER_MUTE    = 0x2,
529        };
530
531        AudioHwDevice(const char *moduleName,
532                      audio_hw_device_t *hwDevice,
533                      Flags flags)
534            : mModuleName(strdup(moduleName))
535            , mHwDevice(hwDevice)
536            , mFlags(flags) { }
537        /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); }
538
539        bool canSetMasterVolume() const {
540            return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME));
541        }
542
543        bool canSetMasterMute() const {
544            return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE));
545        }
546
547        const char *moduleName() const { return mModuleName; }
548        audio_hw_device_t *hwDevice() const { return mHwDevice; }
549        uint32_t version() const { return mHwDevice->common.version; }
550
551    private:
552        const char * const mModuleName;
553        audio_hw_device_t * const mHwDevice;
554        const Flags mFlags;
555    };
556
557    // AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
558    // For emphasis, we could also make all pointers to them be "const *",
559    // but that would clutter the code unnecessarily.
560
561    struct AudioStreamOut {
562        AudioHwDevice* const audioHwDev;
563        audio_stream_out_t* const stream;
564        const audio_output_flags_t flags;
565
566        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
567
568        AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) :
569            audioHwDev(dev), stream(out), flags(flags) {}
570    };
571
572    struct AudioStreamIn {
573        AudioHwDevice* const audioHwDev;
574        audio_stream_in_t* const stream;
575
576        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
577
578        AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) :
579            audioHwDev(dev), stream(in) {}
580    };
581
582    // for mAudioSessionRefs only
583    struct AudioSessionRef {
584        AudioSessionRef(int sessionid, pid_t pid) :
585            mSessionid(sessionid), mPid(pid), mCnt(1) {}
586        const int   mSessionid;
587        const pid_t mPid;
588        int         mCnt;
589    };
590
591    mutable     Mutex                               mLock;
592                // protects mClients and mNotificationClients.
593                // must be locked after mLock and ThreadBase::mLock if both must be locked
594                // avoids acquiring AudioFlinger::mLock from inside thread loop.
595    mutable     Mutex                               mClientLock;
596                // protected by mClientLock
597                DefaultKeyedVector< pid_t, wp<Client> >     mClients;   // see ~Client()
598
599                mutable     Mutex                   mHardwareLock;
600                // NOTE: If both mLock and mHardwareLock mutexes must be held,
601                // always take mLock before mHardwareLock
602
603                // These two fields are immutable after onFirstRef(), so no lock needed to access
604                AudioHwDevice*                      mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
605                DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*>  mAudioHwDevs;
606
607    // for dump, indicates which hardware operation is currently in progress (but not stream ops)
608    enum hardware_call_state {
609        AUDIO_HW_IDLE = 0,              // no operation in progress
610        AUDIO_HW_INIT,                  // init_check
611        AUDIO_HW_OUTPUT_OPEN,           // open_output_stream
612        AUDIO_HW_OUTPUT_CLOSE,          // unused
613        AUDIO_HW_INPUT_OPEN,            // unused
614        AUDIO_HW_INPUT_CLOSE,           // unused
615        AUDIO_HW_STANDBY,               // unused
616        AUDIO_HW_SET_MASTER_VOLUME,     // set_master_volume
617        AUDIO_HW_GET_ROUTING,           // unused
618        AUDIO_HW_SET_ROUTING,           // unused
619        AUDIO_HW_GET_MODE,              // unused
620        AUDIO_HW_SET_MODE,              // set_mode
621        AUDIO_HW_GET_MIC_MUTE,          // get_mic_mute
622        AUDIO_HW_SET_MIC_MUTE,          // set_mic_mute
623        AUDIO_HW_SET_VOICE_VOLUME,      // set_voice_volume
624        AUDIO_HW_SET_PARAMETER,         // set_parameters
625        AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size
626        AUDIO_HW_GET_MASTER_VOLUME,     // get_master_volume
627        AUDIO_HW_GET_PARAMETER,         // get_parameters
628        AUDIO_HW_SET_MASTER_MUTE,       // set_master_mute
629        AUDIO_HW_GET_MASTER_MUTE,       // get_master_mute
630    };
631
632    mutable     hardware_call_state                 mHardwareStatus;    // for dump only
633
634
635                DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> >  mPlaybackThreads;
636                stream_type_t                       mStreamTypes[AUDIO_STREAM_CNT];
637
638                // member variables below are protected by mLock
639                float                               mMasterVolume;
640                bool                                mMasterMute;
641                // end of variables protected by mLock
642
643                DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> >    mRecordThreads;
644
645                // protected by mClientLock
646                DefaultKeyedVector< pid_t, sp<NotificationClient> >    mNotificationClients;
647
648                volatile int32_t                    mNextUniqueId;  // updated by android_atomic_inc
649                // nextUniqueId() returns uint32_t, but this is declared int32_t
650                // because the atomic operations require an int32_t
651
652                audio_mode_t                        mMode;
653                bool                                mBtNrecIsOff;
654
655                // protected by mLock
656                Vector<AudioSessionRef*> mAudioSessionRefs;
657
658                float       masterVolume_l() const;
659                bool        masterMute_l() const;
660                audio_module_handle_t loadHwModule_l(const char *name);
661
662                Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session
663                                                             // to be created
664
665private:
666    sp<Client>  registerPid(pid_t pid);    // always returns non-0
667
668    // for use from destructor
669    status_t    closeOutput_nonvirtual(audio_io_handle_t output);
670    status_t    closeInput_nonvirtual(audio_io_handle_t input);
671
672#ifdef TEE_SINK
673    // all record threads serially share a common tee sink, which is re-created on format change
674    sp<NBAIO_Sink>   mRecordTeeSink;
675    sp<NBAIO_Source> mRecordTeeSource;
676#endif
677
678public:
679
680#ifdef TEE_SINK
681    // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file
682    static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0);
683
684    // whether tee sink is enabled by property
685    static bool mTeeSinkInputEnabled;
686    static bool mTeeSinkOutputEnabled;
687    static bool mTeeSinkTrackEnabled;
688
689    // runtime configured size of each tee sink pipe, in frames
690    static size_t mTeeSinkInputFrames;
691    static size_t mTeeSinkOutputFrames;
692    static size_t mTeeSinkTrackFrames;
693
694    // compile-time default size of tee sink pipes, in frames
695    // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes
696    static const size_t kTeeSinkInputFramesDefault = 0x200000;
697    static const size_t kTeeSinkOutputFramesDefault = 0x200000;
698    static const size_t kTeeSinkTrackFramesDefault = 0x200000;
699#endif
700
701    // This method reads from a variable without mLock, but the variable is updated under mLock.  So
702    // we might read a stale value, or a value that's inconsistent with respect to other variables.
703    // In this case, it's safe because the return value isn't used for making an important decision.
704    // The reason we don't want to take mLock is because it could block the caller for a long time.
705    bool    isLowRamDevice() const { return mIsLowRamDevice; }
706
707private:
708    bool    mIsLowRamDevice;
709    bool    mIsDeviceTypeKnown;
710    nsecs_t mGlobalEffectEnableTime;  // when a global effect was last enabled
711
712    sp<PatchPanel> mPatchPanel;
713
714    uint32_t    mPrimaryOutputSampleRate;   // sample rate of the primary output, or zero if none
715                                            // protected by mHardwareLock
716};
717
718#undef INCLUDING_FROM_AUDIOFLINGER_H
719
720const char *formatToString(audio_format_t format);
721
722// ----------------------------------------------------------------------------
723
724}; // namespace android
725
726#endif // ANDROID_AUDIO_FLINGER_H
727