AudioFlinger.h revision 9a59276fb465e492138e0576523b54079671e8f4 
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#include "AudioMixer.h"
59
60#include <powermanager/IPowerManager.h>
61
62#include <media/nbaio/NBLog.h>
63#include <private/media/AudioTrackShared.h>
64
65namespace android {
66
67struct audio_track_cblk_t;
68struct effect_param_cblk_t;
69class AudioMixer;
70class AudioBuffer;
71class AudioResampler;
72class FastMixer;
73class ServerProxy;
74
75// ----------------------------------------------------------------------------
76
77// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback.
78// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect.
79// Adding full support for > 2 channel capture or playback would require more than simply changing
80// this #define.  There is an independent hard-coded upper limit in AudioMixer;
81// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels.
82// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions.
83// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc.
84#define FCC_2 2     // FCC_2 = Fixed Channel Count 2
85
86static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
87
88#define INCLUDING_FROM_AUDIOFLINGER_H
89
90class AudioFlinger :
91    public BinderService<AudioFlinger>,
92    public BnAudioFlinger
93{
94    friend class BinderService<AudioFlinger>;   // for AudioFlinger()
95public:
96    static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; }
97
98    virtual     status_t    dump(int fd, const Vector<String16>& args);
99
100    // IAudioFlinger interface, in binder opcode order
101    virtual sp<IAudioTrack> createTrack(
102                                audio_stream_type_t streamType,
103                                uint32_t sampleRate,
104                                audio_format_t format,
105                                audio_channel_mask_t channelMask,
106                                size_t *pFrameCount,
107                                IAudioFlinger::track_flags_t *flags,
108                                const sp<IMemory>& sharedBuffer,
109                                audio_io_handle_t output,
110                                pid_t tid,
111                                int *sessionId,
112                                int clientUid,
113                                status_t *status /*non-NULL*/);
114
115    virtual sp<IAudioRecord> openRecord(
116                                audio_io_handle_t input,
117                                uint32_t sampleRate,
118                                audio_format_t format,
119                                audio_channel_mask_t channelMask,
120                                size_t *pFrameCount,
121                                IAudioFlinger::track_flags_t *flags,
122                                pid_t tid,
123                                int *sessionId,
124                                size_t *notificationFrames,
125                                sp<IMemory>& cblk,
126                                sp<IMemory>& buffers,
127                                status_t *status /*non-NULL*/);
128
129    virtual     uint32_t    sampleRate(audio_io_handle_t output) const;
130    virtual     audio_format_t format(audio_io_handle_t output) const;
131    virtual     size_t      frameCount(audio_io_handle_t output) const;
132    virtual     uint32_t    latency(audio_io_handle_t output) const;
133
134    virtual     status_t    setMasterVolume(float value);
135    virtual     status_t    setMasterMute(bool muted);
136
137    virtual     float       masterVolume() const;
138    virtual     bool        masterMute() const;
139
140    virtual     status_t    setStreamVolume(audio_stream_type_t stream, float value,
141                                            audio_io_handle_t output);
142    virtual     status_t    setStreamMute(audio_stream_type_t stream, bool muted);
143
144    virtual     float       streamVolume(audio_stream_type_t stream,
145                                         audio_io_handle_t output) const;
146    virtual     bool        streamMute(audio_stream_type_t stream) const;
147
148    virtual     status_t    setMode(audio_mode_t mode);
149
150    virtual     status_t    setMicMute(bool state);
151    virtual     bool        getMicMute() const;
152
153    virtual     status_t    setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
154    virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
155
156    virtual     void        registerClient(const sp<IAudioFlingerClient>& client);
157
158    virtual     size_t      getInputBufferSize(uint32_t sampleRate, audio_format_t format,
159                                               audio_channel_mask_t channelMask) const;
160
161    virtual audio_io_handle_t openOutput(audio_module_handle_t module,
162                                         audio_devices_t *pDevices,
163                                         uint32_t *pSamplingRate,
164                                         audio_format_t *pFormat,
165                                         audio_channel_mask_t *pChannelMask,
166                                         uint32_t *pLatencyMs,
167                                         audio_output_flags_t flags,
168                                         const audio_offload_info_t *offloadInfo);
169
170    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
171                                                  audio_io_handle_t output2);
172
173    virtual status_t closeOutput(audio_io_handle_t output);
174
175    virtual status_t suspendOutput(audio_io_handle_t output);
176
177    virtual status_t restoreOutput(audio_io_handle_t output);
178
179    virtual audio_io_handle_t openInput(audio_module_handle_t module,
180                                        audio_devices_t *pDevices,
181                                        uint32_t *pSamplingRate,
182                                        audio_format_t *pFormat,
183                                        audio_channel_mask_t *pChannelMask,
184                                        audio_input_flags_t flags);
185
186    virtual status_t closeInput(audio_io_handle_t input);
187
188    virtual status_t invalidateStream(audio_stream_type_t stream);
189
190    virtual status_t setVoiceVolume(float volume);
191
192    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
193                                       audio_io_handle_t output) const;
194
195    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
196
197    virtual int newAudioSessionId();
198
199    virtual void acquireAudioSessionId(int audioSession, pid_t pid);
200
201    virtual void releaseAudioSessionId(int audioSession, pid_t pid);
202
203    virtual status_t queryNumberEffects(uint32_t *numEffects) const;
204
205    virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
206
207    virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
208                                         effect_descriptor_t *descriptor) const;
209
210    virtual sp<IEffect> createEffect(
211                        effect_descriptor_t *pDesc,
212                        const sp<IEffectClient>& effectClient,
213                        int32_t priority,
214                        audio_io_handle_t io,
215                        int sessionId,
216                        status_t *status /*non-NULL*/,
217                        int *id,
218                        int *enabled);
219
220    virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
221                        audio_io_handle_t dstOutput);
222
223    virtual audio_module_handle_t loadHwModule(const char *name);
224
225    virtual uint32_t getPrimaryOutputSamplingRate();
226    virtual size_t getPrimaryOutputFrameCount();
227
228    virtual status_t setLowRamDevice(bool isLowRamDevice);
229
230    /* List available audio ports and their attributes */
231    virtual status_t listAudioPorts(unsigned int *num_ports,
232                                    struct audio_port *ports);
233
234    /* Get attributes for a given audio port */
235    virtual status_t getAudioPort(struct audio_port *port);
236
237    /* Create an audio patch between several source and sink ports */
238    virtual status_t createAudioPatch(const struct audio_patch *patch,
239                                       audio_patch_handle_t *handle);
240
241    /* Release an audio patch */
242    virtual status_t releaseAudioPatch(audio_patch_handle_t handle);
243
244    /* List existing audio patches */
245    virtual status_t listAudioPatches(unsigned int *num_patches,
246                                      struct audio_patch *patches);
247
248    /* Set audio port configuration */
249    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
250
251    virtual     status_t    onTransact(
252                                uint32_t code,
253                                const Parcel& data,
254                                Parcel* reply,
255                                uint32_t flags);
256
257    // end of IAudioFlinger interface
258
259    sp<NBLog::Writer>   newWriter_l(size_t size, const char *name);
260    void                unregisterWriter(const sp<NBLog::Writer>& writer);
261private:
262    static const size_t kLogMemorySize = 40 * 1024;
263    sp<MemoryDealer>    mLogMemoryDealer;   // == 0 when NBLog is disabled
264    // When a log writer is unregistered, it is done lazily so that media.log can continue to see it
265    // for as long as possible.  The memory is only freed when it is needed for another log writer.
266    Vector< sp<NBLog::Writer> > mUnregisteredWriters;
267    Mutex               mUnregisteredWritersLock;
268public:
269
270    class SyncEvent;
271
272    typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ;
273
274    class SyncEvent : public RefBase {
275    public:
276        SyncEvent(AudioSystem::sync_event_t type,
277                  int triggerSession,
278                  int listenerSession,
279                  sync_event_callback_t callBack,
280                  wp<RefBase> cookie)
281        : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession),
282          mCallback(callBack), mCookie(cookie)
283        {}
284
285        virtual ~SyncEvent() {}
286
287        void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); }
288        bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); }
289        void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; }
290        AudioSystem::sync_event_t type() const { return mType; }
291        int triggerSession() const { return mTriggerSession; }
292        int listenerSession() const { return mListenerSession; }
293        wp<RefBase> cookie() const { return mCookie; }
294
295    private:
296          const AudioSystem::sync_event_t mType;
297          const int mTriggerSession;
298          const int mListenerSession;
299          sync_event_callback_t mCallback;
300          const wp<RefBase> mCookie;
301          mutable Mutex mLock;
302    };
303
304    sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
305                                        int triggerSession,
306                                        int listenerSession,
307                                        sync_event_callback_t callBack,
308                                        wp<RefBase> cookie);
309
310private:
311    class AudioHwDevice;    // fwd declaration for findSuitableHwDev_l
312
313               audio_mode_t getMode() const { return mMode; }
314
315                bool        btNrecIsOff() const { return mBtNrecIsOff; }
316
317                            AudioFlinger() ANDROID_API;
318    virtual                 ~AudioFlinger();
319
320    // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
321    status_t                initCheck() const { return mPrimaryHardwareDev == NULL ?
322                                                        NO_INIT : NO_ERROR; }
323
324    // RefBase
325    virtual     void        onFirstRef();
326
327    AudioHwDevice*          findSuitableHwDev_l(audio_module_handle_t module,
328                                                audio_devices_t devices);
329    void                    purgeStaleEffects_l();
330
331    // Set kEnableExtendedChannels to true to enable greater than stereo output
332    // for the MixerThread and device sink.  Number of channels allowed is
333    // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS.
334    static const bool kEnableExtendedChannels = false;
335
336    // Returns true if channel mask is permitted for the PCM sink in the MixerThread
337    static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) {
338        switch (audio_channel_mask_get_representation(channelMask)) {
339        case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
340            uint32_t channelCount = FCC_2; // stereo is default
341            if (kEnableExtendedChannels) {
342                channelCount = audio_channel_count_from_out_mask(channelMask);
343                if (channelCount > AudioMixer::MAX_NUM_CHANNELS) {
344                    return false;
345                }
346            }
347            // check that channelMask is the "canonical" one we expect for the channelCount.
348            return channelMask == audio_channel_out_mask_from_count(channelCount);
349            }
350        default:
351            return false;
352        }
353    }
354
355    // Set kEnableExtendedPrecision to true to use extended precision in MixerThread
356    static const bool kEnableExtendedPrecision = true;
357
358    // Returns true if format is permitted for the PCM sink in the MixerThread
359    static inline bool isValidPcmSinkFormat(audio_format_t format) {
360        switch (format) {
361        case AUDIO_FORMAT_PCM_16_BIT:
362            return true;
363        case AUDIO_FORMAT_PCM_FLOAT:
364        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
365        case AUDIO_FORMAT_PCM_32_BIT:
366        case AUDIO_FORMAT_PCM_8_24_BIT:
367            return kEnableExtendedPrecision;
368        default:
369            return false;
370        }
371    }
372
373    // standby delay for MIXER and DUPLICATING playback threads is read from property
374    // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
375    static nsecs_t          mStandbyTimeInNsecs;
376
377    // incremented by 2 when screen state changes, bit 0 == 1 means "off"
378    // AudioFlinger::setParameters() updates, other threads read w/o lock
379    static uint32_t         mScreenState;
380
381    // Internal dump utilities.
382    static const int kDumpLockRetries = 50;
383    static const int kDumpLockSleepUs = 20000;
384    static bool dumpTryLock(Mutex& mutex);
385    void dumpPermissionDenial(int fd, const Vector<String16>& args);
386    void dumpClients(int fd, const Vector<String16>& args);
387    void dumpInternals(int fd, const Vector<String16>& args);
388
389    // --- Client ---
390    class Client : public RefBase {
391    public:
392                            Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
393        virtual             ~Client();
394        sp<MemoryDealer>    heap() const;
395        pid_t               pid() const { return mPid; }
396        sp<AudioFlinger>    audioFlinger() const { return mAudioFlinger; }
397
398        bool reserveTimedTrack();
399        void releaseTimedTrack();
400
401    private:
402                            Client(const Client&);
403                            Client& operator = (const Client&);
404        const sp<AudioFlinger> mAudioFlinger;
405        const sp<MemoryDealer> mMemoryDealer;
406        const pid_t         mPid;
407
408        Mutex               mTimedTrackLock;
409        int                 mTimedTrackCount;
410    };
411
412    // --- Notification Client ---
413    class NotificationClient : public IBinder::DeathRecipient {
414    public:
415                            NotificationClient(const sp<AudioFlinger>& audioFlinger,
416                                                const sp<IAudioFlingerClient>& client,
417                                                pid_t pid);
418        virtual             ~NotificationClient();
419
420                sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
421
422                // IBinder::DeathRecipient
423                virtual     void        binderDied(const wp<IBinder>& who);
424
425    private:
426                            NotificationClient(const NotificationClient&);
427                            NotificationClient& operator = (const NotificationClient&);
428
429        const sp<AudioFlinger>  mAudioFlinger;
430        const pid_t             mPid;
431        const sp<IAudioFlingerClient> mAudioFlingerClient;
432    };
433
434    class TrackHandle;
435    class RecordHandle;
436    class RecordThread;
437    class PlaybackThread;
438    class MixerThread;
439    class DirectOutputThread;
440    class OffloadThread;
441    class DuplicatingThread;
442    class AsyncCallbackThread;
443    class Track;
444    class RecordTrack;
445    class EffectModule;
446    class EffectHandle;
447    class EffectChain;
448    struct AudioStreamOut;
449    struct AudioStreamIn;
450
451    struct  stream_type_t {
452        stream_type_t()
453            :   volume(1.0f),
454                mute(false)
455        {
456        }
457        float       volume;
458        bool        mute;
459    };
460
461    // --- PlaybackThread ---
462
463#include "Threads.h"
464
465#include "Effects.h"
466
467#include "PatchPanel.h"
468
469    // server side of the client's IAudioTrack
470    class TrackHandle : public android::BnAudioTrack {
471    public:
472                            TrackHandle(const sp<PlaybackThread::Track>& track);
473        virtual             ~TrackHandle();
474        virtual sp<IMemory> getCblk() const;
475        virtual status_t    start();
476        virtual void        stop();
477        virtual void        flush();
478        virtual void        pause();
479        virtual status_t    attachAuxEffect(int effectId);
480        virtual status_t    allocateTimedBuffer(size_t size,
481                                                sp<IMemory>* buffer);
482        virtual status_t    queueTimedBuffer(const sp<IMemory>& buffer,
483                                             int64_t pts);
484        virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
485                                                  int target);
486        virtual status_t    setParameters(const String8& keyValuePairs);
487        virtual status_t    getTimestamp(AudioTimestamp& timestamp);
488        virtual void        signal(); // signal playback thread for a change in control block
489
490        virtual status_t onTransact(
491            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
492
493    private:
494        const sp<PlaybackThread::Track> mTrack;
495    };
496
497    // server side of the client's IAudioRecord
498    class RecordHandle : public android::BnAudioRecord {
499    public:
500        RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
501        virtual             ~RecordHandle();
502        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
503        virtual void        stop();
504        virtual status_t onTransact(
505            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
506    private:
507        const sp<RecordThread::RecordTrack> mRecordTrack;
508
509        // for use from destructor
510        void                stop_nonvirtual();
511    };
512
513
514              PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
515              MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
516              RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
517              sp<RecordThread> openInput_l(audio_module_handle_t module,
518                                           audio_devices_t device,
519                                           struct audio_config *config,
520                                           audio_input_flags_t flags);
521              sp<PlaybackThread> openOutput_l(audio_module_handle_t module,
522                                              audio_devices_t device,
523                                              struct audio_config *config,
524                                              audio_output_flags_t flags);
525
526              void closeOutputFinish(sp<PlaybackThread> thread);
527              void closeInputFinish(sp<RecordThread> thread);
528
529              // no range check, AudioFlinger::mLock held
530              bool streamMute_l(audio_stream_type_t stream) const
531                                { return mStreamTypes[stream].mute; }
532              // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held
533              float streamVolume_l(audio_stream_type_t stream) const
534                                { return mStreamTypes[stream].volume; }
535              void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2);
536
537              // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t.
538              // They all share the same ID space, but the namespaces are actually independent
539              // because there are separate KeyedVectors for each kind of ID.
540              // The return value is uint32_t, but is cast to signed for some IDs.
541              // FIXME This API does not handle rollover to zero (for unsigned IDs),
542              //       or from positive to negative (for signed IDs).
543              //       Thus it may fail by returning an ID of the wrong sign,
544              //       or by returning a non-unique ID.
545              uint32_t nextUniqueId();
546
547              status_t moveEffectChain_l(int sessionId,
548                                     PlaybackThread *srcThread,
549                                     PlaybackThread *dstThread,
550                                     bool reRegister);
551              // return thread associated with primary hardware device, or NULL
552              PlaybackThread *primaryPlaybackThread_l() const;
553              audio_devices_t primaryOutputDevice_l() const;
554
555              sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
556
557
558                void        removeClient_l(pid_t pid);
559                void        removeNotificationClient(pid_t pid);
560                bool isNonOffloadableGlobalEffectEnabled_l();
561                void onNonOffloadableGlobalEffectEnable();
562
563    class AudioHwDevice {
564    public:
565        enum Flags {
566            AHWD_CAN_SET_MASTER_VOLUME  = 0x1,
567            AHWD_CAN_SET_MASTER_MUTE    = 0x2,
568        };
569
570        AudioHwDevice(audio_module_handle_t handle,
571                      const char *moduleName,
572                      audio_hw_device_t *hwDevice,
573                      Flags flags)
574            : mHandle(handle), mModuleName(strdup(moduleName))
575            , mHwDevice(hwDevice)
576            , mFlags(flags) { }
577        /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); }
578
579        bool canSetMasterVolume() const {
580            return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME));
581        }
582
583        bool canSetMasterMute() const {
584            return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE));
585        }
586
587        audio_module_handle_t handle() const { return mHandle; }
588        const char *moduleName() const { return mModuleName; }
589        audio_hw_device_t *hwDevice() const { return mHwDevice; }
590        uint32_t version() const { return mHwDevice->common.version; }
591
592    private:
593        const audio_module_handle_t mHandle;
594        const char * const mModuleName;
595        audio_hw_device_t * const mHwDevice;
596        const Flags mFlags;
597    };
598
599    // AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
600    // For emphasis, we could also make all pointers to them be "const *",
601    // but that would clutter the code unnecessarily.
602
603    struct AudioStreamOut {
604        AudioHwDevice* const audioHwDev;
605        audio_stream_out_t* const stream;
606        const audio_output_flags_t flags;
607
608        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
609
610        AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) :
611            audioHwDev(dev), stream(out), flags(flags) {}
612    };
613
614    struct AudioStreamIn {
615        AudioHwDevice* const audioHwDev;
616        audio_stream_in_t* const stream;
617
618        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
619
620        AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) :
621            audioHwDev(dev), stream(in) {}
622    };
623
624    // for mAudioSessionRefs only
625    struct AudioSessionRef {
626        AudioSessionRef(int sessionid, pid_t pid) :
627            mSessionid(sessionid), mPid(pid), mCnt(1) {}
628        const int   mSessionid;
629        const pid_t mPid;
630        int         mCnt;
631    };
632
633    mutable     Mutex                               mLock;
634                // protects mClients and mNotificationClients.
635                // must be locked after mLock and ThreadBase::mLock if both must be locked
636                // avoids acquiring AudioFlinger::mLock from inside thread loop.
637    mutable     Mutex                               mClientLock;
638                // protected by mClientLock
639                DefaultKeyedVector< pid_t, wp<Client> >     mClients;   // see ~Client()
640
641                mutable     Mutex                   mHardwareLock;
642                // NOTE: If both mLock and mHardwareLock mutexes must be held,
643                // always take mLock before mHardwareLock
644
645                // These two fields are immutable after onFirstRef(), so no lock needed to access
646                AudioHwDevice*                      mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
647                DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*>  mAudioHwDevs;
648
649    // for dump, indicates which hardware operation is currently in progress (but not stream ops)
650    enum hardware_call_state {
651        AUDIO_HW_IDLE = 0,              // no operation in progress
652        AUDIO_HW_INIT,                  // init_check
653        AUDIO_HW_OUTPUT_OPEN,           // open_output_stream
654        AUDIO_HW_OUTPUT_CLOSE,          // unused
655        AUDIO_HW_INPUT_OPEN,            // unused
656        AUDIO_HW_INPUT_CLOSE,           // unused
657        AUDIO_HW_STANDBY,               // unused
658        AUDIO_HW_SET_MASTER_VOLUME,     // set_master_volume
659        AUDIO_HW_GET_ROUTING,           // unused
660        AUDIO_HW_SET_ROUTING,           // unused
661        AUDIO_HW_GET_MODE,              // unused
662        AUDIO_HW_SET_MODE,              // set_mode
663        AUDIO_HW_GET_MIC_MUTE,          // get_mic_mute
664        AUDIO_HW_SET_MIC_MUTE,          // set_mic_mute
665        AUDIO_HW_SET_VOICE_VOLUME,      // set_voice_volume
666        AUDIO_HW_SET_PARAMETER,         // set_parameters
667        AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size
668        AUDIO_HW_GET_MASTER_VOLUME,     // get_master_volume
669        AUDIO_HW_GET_PARAMETER,         // get_parameters
670        AUDIO_HW_SET_MASTER_MUTE,       // set_master_mute
671        AUDIO_HW_GET_MASTER_MUTE,       // get_master_mute
672    };
673
674    mutable     hardware_call_state                 mHardwareStatus;    // for dump only
675
676
677                DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> >  mPlaybackThreads;
678                stream_type_t                       mStreamTypes[AUDIO_STREAM_CNT];
679
680                // member variables below are protected by mLock
681                float                               mMasterVolume;
682                bool                                mMasterMute;
683                // end of variables protected by mLock
684
685                DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> >    mRecordThreads;
686
687                // protected by mClientLock
688                DefaultKeyedVector< pid_t, sp<NotificationClient> >    mNotificationClients;
689
690                volatile int32_t                    mNextUniqueId;  // updated by android_atomic_inc
691                // nextUniqueId() returns uint32_t, but this is declared int32_t
692                // because the atomic operations require an int32_t
693
694                audio_mode_t                        mMode;
695                bool                                mBtNrecIsOff;
696
697                // protected by mLock
698                Vector<AudioSessionRef*> mAudioSessionRefs;
699
700                float       masterVolume_l() const;
701                bool        masterMute_l() const;
702                audio_module_handle_t loadHwModule_l(const char *name);
703
704                Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session
705                                                             // to be created
706
707private:
708    sp<Client>  registerPid(pid_t pid);    // always returns non-0
709
710    // for use from destructor
711    status_t    closeOutput_nonvirtual(audio_io_handle_t output);
712    void        closeOutputInternal_l(sp<PlaybackThread> thread);
713    status_t    closeInput_nonvirtual(audio_io_handle_t input);
714    void        closeInputInternal_l(sp<RecordThread> thread);
715
716#ifdef TEE_SINK
717    // all record threads serially share a common tee sink, which is re-created on format change
718    sp<NBAIO_Sink>   mRecordTeeSink;
719    sp<NBAIO_Source> mRecordTeeSource;
720#endif
721
722public:
723
724#ifdef TEE_SINK
725    // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file
726    static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0);
727
728    // whether tee sink is enabled by property
729    static bool mTeeSinkInputEnabled;
730    static bool mTeeSinkOutputEnabled;
731    static bool mTeeSinkTrackEnabled;
732
733    // runtime configured size of each tee sink pipe, in frames
734    static size_t mTeeSinkInputFrames;
735    static size_t mTeeSinkOutputFrames;
736    static size_t mTeeSinkTrackFrames;
737
738    // compile-time default size of tee sink pipes, in frames
739    // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes
740    static const size_t kTeeSinkInputFramesDefault = 0x200000;
741    static const size_t kTeeSinkOutputFramesDefault = 0x200000;
742    static const size_t kTeeSinkTrackFramesDefault = 0x200000;
743#endif
744
745    // This method reads from a variable without mLock, but the variable is updated under mLock.  So
746    // we might read a stale value, or a value that's inconsistent with respect to other variables.
747    // In this case, it's safe because the return value isn't used for making an important decision.
748    // The reason we don't want to take mLock is because it could block the caller for a long time.
749    bool    isLowRamDevice() const { return mIsLowRamDevice; }
750
751private:
752    bool    mIsLowRamDevice;
753    bool    mIsDeviceTypeKnown;
754    nsecs_t mGlobalEffectEnableTime;  // when a global effect was last enabled
755
756    sp<PatchPanel> mPatchPanel;
757
758    uint32_t    mPrimaryOutputSampleRate;   // sample rate of the primary output, or zero if none
759                                            // protected by mHardwareLock
760};
761
762#undef INCLUDING_FROM_AUDIOFLINGER_H
763
764const char *formatToString(audio_format_t format);
765
766// ----------------------------------------------------------------------------
767
768}; // namespace android
769
770#endif // ANDROID_AUDIO_FLINGER_H
771