AudioFlinger.h revision a494e82c3c73508b4d3cfe89e9134de94e12fd31 
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
184    virtual status_t closeInput(audio_io_handle_t input);
185
186    virtual status_t invalidateStream(audio_stream_type_t stream);
187
188    virtual status_t setVoiceVolume(float volume);
189
190    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
191                                       audio_io_handle_t output) const;
192
193    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
194
195    virtual int newAudioSessionId();
196
197    virtual void acquireAudioSessionId(int audioSession, pid_t pid);
198
199    virtual void releaseAudioSessionId(int audioSession, pid_t pid);
200
201    virtual status_t queryNumberEffects(uint32_t *numEffects) const;
202
203    virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
204
205    virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
206                                         effect_descriptor_t *descriptor) const;
207
208    virtual sp<IEffect> createEffect(
209                        effect_descriptor_t *pDesc,
210                        const sp<IEffectClient>& effectClient,
211                        int32_t priority,
212                        audio_io_handle_t io,
213                        int sessionId,
214                        status_t *status /*non-NULL*/,
215                        int *id,
216                        int *enabled);
217
218    virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
219                        audio_io_handle_t dstOutput);
220
221    virtual audio_module_handle_t loadHwModule(const char *name);
222
223    virtual uint32_t getPrimaryOutputSamplingRate();
224    virtual size_t getPrimaryOutputFrameCount();
225
226    virtual status_t setLowRamDevice(bool isLowRamDevice);
227
228    /* List available audio ports and their attributes */
229    virtual status_t listAudioPorts(unsigned int *num_ports,
230                                    struct audio_port *ports);
231
232    /* Get attributes for a given audio port */
233    virtual status_t getAudioPort(struct audio_port *port);
234
235    /* Create an audio patch between several source and sink ports */
236    virtual status_t createAudioPatch(const struct audio_patch *patch,
237                                       audio_patch_handle_t *handle);
238
239    /* Release an audio patch */
240    virtual status_t releaseAudioPatch(audio_patch_handle_t handle);
241
242    /* List existing audio patches */
243    virtual status_t listAudioPatches(unsigned int *num_patches,
244                                      struct audio_patch *patches);
245
246    /* Set audio port configuration */
247    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
248
249    virtual     status_t    onTransact(
250                                uint32_t code,
251                                const Parcel& data,
252                                Parcel* reply,
253                                uint32_t flags);
254
255    // end of IAudioFlinger interface
256
257    sp<NBLog::Writer>   newWriter_l(size_t size, const char *name);
258    void                unregisterWriter(const sp<NBLog::Writer>& writer);
259private:
260    static const size_t kLogMemorySize = 40 * 1024;
261    sp<MemoryDealer>    mLogMemoryDealer;   // == 0 when NBLog is disabled
262    // When a log writer is unregistered, it is done lazily so that media.log can continue to see it
263    // for as long as possible.  The memory is only freed when it is needed for another log writer.
264    Vector< sp<NBLog::Writer> > mUnregisteredWriters;
265    Mutex               mUnregisteredWritersLock;
266public:
267
268    class SyncEvent;
269
270    typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ;
271
272    class SyncEvent : public RefBase {
273    public:
274        SyncEvent(AudioSystem::sync_event_t type,
275                  int triggerSession,
276                  int listenerSession,
277                  sync_event_callback_t callBack,
278                  wp<RefBase> cookie)
279        : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession),
280          mCallback(callBack), mCookie(cookie)
281        {}
282
283        virtual ~SyncEvent() {}
284
285        void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); }
286        bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); }
287        void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; }
288        AudioSystem::sync_event_t type() const { return mType; }
289        int triggerSession() const { return mTriggerSession; }
290        int listenerSession() const { return mListenerSession; }
291        wp<RefBase> cookie() const { return mCookie; }
292
293    private:
294          const AudioSystem::sync_event_t mType;
295          const int mTriggerSession;
296          const int mListenerSession;
297          sync_event_callback_t mCallback;
298          const wp<RefBase> mCookie;
299          mutable Mutex mLock;
300    };
301
302    sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type,
303                                        int triggerSession,
304                                        int listenerSession,
305                                        sync_event_callback_t callBack,
306                                        wp<RefBase> cookie);
307
308private:
309    class AudioHwDevice;    // fwd declaration for findSuitableHwDev_l
310
311               audio_mode_t getMode() const { return mMode; }
312
313                bool        btNrecIsOff() const { return mBtNrecIsOff; }
314
315                            AudioFlinger() ANDROID_API;
316    virtual                 ~AudioFlinger();
317
318    // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
319    status_t                initCheck() const { return mPrimaryHardwareDev == NULL ?
320                                                        NO_INIT : NO_ERROR; }
321
322    // RefBase
323    virtual     void        onFirstRef();
324
325    AudioHwDevice*          findSuitableHwDev_l(audio_module_handle_t module,
326                                                audio_devices_t devices);
327    void                    purgeStaleEffects_l();
328
329    // Set kEnableExtendedPrecision to true to use extended precision in MixerThread
330    static const bool kEnableExtendedPrecision = true;
331
332    // Returns true if format is permitted for the PCM sink in the MixerThread
333    static inline bool isValidPcmSinkFormat(audio_format_t format) {
334        switch (format) {
335        case AUDIO_FORMAT_PCM_16_BIT:
336            return true;
337        case AUDIO_FORMAT_PCM_FLOAT:
338        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
339        case AUDIO_FORMAT_PCM_32_BIT:
340        case AUDIO_FORMAT_PCM_8_24_BIT:
341            return kEnableExtendedPrecision;
342        default:
343            return false;
344        }
345    }
346
347    // standby delay for MIXER and DUPLICATING playback threads is read from property
348    // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs
349    static nsecs_t          mStandbyTimeInNsecs;
350
351    // incremented by 2 when screen state changes, bit 0 == 1 means "off"
352    // AudioFlinger::setParameters() updates, other threads read w/o lock
353    static uint32_t         mScreenState;
354
355    // Internal dump utilities.
356    static const int kDumpLockRetries = 50;
357    static const int kDumpLockSleepUs = 20000;
358    static bool dumpTryLock(Mutex& mutex);
359    void dumpPermissionDenial(int fd, const Vector<String16>& args);
360    void dumpClients(int fd, const Vector<String16>& args);
361    void dumpInternals(int fd, const Vector<String16>& args);
362
363    // --- Client ---
364    class Client : public RefBase {
365    public:
366                            Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
367        virtual             ~Client();
368        sp<MemoryDealer>    heap() const;
369        pid_t               pid() const { return mPid; }
370        sp<AudioFlinger>    audioFlinger() const { return mAudioFlinger; }
371
372        bool reserveTimedTrack();
373        void releaseTimedTrack();
374
375    private:
376                            Client(const Client&);
377                            Client& operator = (const Client&);
378        const sp<AudioFlinger> mAudioFlinger;
379        const sp<MemoryDealer> mMemoryDealer;
380        const pid_t         mPid;
381
382        Mutex               mTimedTrackLock;
383        int                 mTimedTrackCount;
384    };
385
386    // --- Notification Client ---
387    class NotificationClient : public IBinder::DeathRecipient {
388    public:
389                            NotificationClient(const sp<AudioFlinger>& audioFlinger,
390                                                const sp<IAudioFlingerClient>& client,
391                                                pid_t pid);
392        virtual             ~NotificationClient();
393
394                sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
395
396                // IBinder::DeathRecipient
397                virtual     void        binderDied(const wp<IBinder>& who);
398
399    private:
400                            NotificationClient(const NotificationClient&);
401                            NotificationClient& operator = (const NotificationClient&);
402
403        const sp<AudioFlinger>  mAudioFlinger;
404        const pid_t             mPid;
405        const sp<IAudioFlingerClient> mAudioFlingerClient;
406    };
407
408    class TrackHandle;
409    class RecordHandle;
410    class RecordThread;
411    class PlaybackThread;
412    class MixerThread;
413    class DirectOutputThread;
414    class OffloadThread;
415    class DuplicatingThread;
416    class AsyncCallbackThread;
417    class Track;
418    class RecordTrack;
419    class EffectModule;
420    class EffectHandle;
421    class EffectChain;
422    struct AudioStreamOut;
423    struct AudioStreamIn;
424
425    struct  stream_type_t {
426        stream_type_t()
427            :   volume(1.0f),
428                mute(false)
429        {
430        }
431        float       volume;
432        bool        mute;
433    };
434
435    // --- PlaybackThread ---
436
437#include "Threads.h"
438
439#include "Effects.h"
440
441#include "PatchPanel.h"
442
443    // server side of the client's IAudioTrack
444    class TrackHandle : public android::BnAudioTrack {
445    public:
446                            TrackHandle(const sp<PlaybackThread::Track>& track);
447        virtual             ~TrackHandle();
448        virtual sp<IMemory> getCblk() const;
449        virtual status_t    start();
450        virtual void        stop();
451        virtual void        flush();
452        virtual void        pause();
453        virtual status_t    attachAuxEffect(int effectId);
454        virtual status_t    allocateTimedBuffer(size_t size,
455                                                sp<IMemory>* buffer);
456        virtual status_t    queueTimedBuffer(const sp<IMemory>& buffer,
457                                             int64_t pts);
458        virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
459                                                  int target);
460        virtual status_t    setParameters(const String8& keyValuePairs);
461        virtual status_t    getTimestamp(AudioTimestamp& timestamp);
462        virtual void        signal(); // signal playback thread for a change in control block
463
464        virtual status_t onTransact(
465            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
466
467    private:
468        const sp<PlaybackThread::Track> mTrack;
469    };
470
471    // server side of the client's IAudioRecord
472    class RecordHandle : public android::BnAudioRecord {
473    public:
474        RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
475        virtual             ~RecordHandle();
476        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
477        virtual void        stop();
478        virtual status_t onTransact(
479            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
480    private:
481        const sp<RecordThread::RecordTrack> mRecordTrack;
482
483        // for use from destructor
484        void                stop_nonvirtual();
485    };
486
487
488              PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
489              MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
490              RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
491              // no range check, AudioFlinger::mLock held
492              bool streamMute_l(audio_stream_type_t stream) const
493                                { return mStreamTypes[stream].mute; }
494              // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held
495              float streamVolume_l(audio_stream_type_t stream) const
496                                { return mStreamTypes[stream].volume; }
497              void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2);
498
499              // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t.
500              // They all share the same ID space, but the namespaces are actually independent
501              // because there are separate KeyedVectors for each kind of ID.
502              // The return value is uint32_t, but is cast to signed for some IDs.
503              // FIXME This API does not handle rollover to zero (for unsigned IDs),
504              //       or from positive to negative (for signed IDs).
505              //       Thus it may fail by returning an ID of the wrong sign,
506              //       or by returning a non-unique ID.
507              uint32_t nextUniqueId();
508
509              status_t moveEffectChain_l(int sessionId,
510                                     PlaybackThread *srcThread,
511                                     PlaybackThread *dstThread,
512                                     bool reRegister);
513              // return thread associated with primary hardware device, or NULL
514              PlaybackThread *primaryPlaybackThread_l() const;
515              audio_devices_t primaryOutputDevice_l() const;
516
517              sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
518
519
520                void        removeClient_l(pid_t pid);
521                void        removeNotificationClient(pid_t pid);
522                bool isNonOffloadableGlobalEffectEnabled_l();
523                void onNonOffloadableGlobalEffectEnable();
524
525    class AudioHwDevice {
526    public:
527        enum Flags {
528            AHWD_CAN_SET_MASTER_VOLUME  = 0x1,
529            AHWD_CAN_SET_MASTER_MUTE    = 0x2,
530        };
531
532        AudioHwDevice(const char *moduleName,
533                      audio_hw_device_t *hwDevice,
534                      Flags flags)
535            : mModuleName(strdup(moduleName))
536            , mHwDevice(hwDevice)
537            , mFlags(flags) { }
538        /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); }
539
540        bool canSetMasterVolume() const {
541            return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME));
542        }
543
544        bool canSetMasterMute() const {
545            return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE));
546        }
547
548        const char *moduleName() const { return mModuleName; }
549        audio_hw_device_t *hwDevice() const { return mHwDevice; }
550        uint32_t version() const { return mHwDevice->common.version; }
551
552    private:
553        const char * const mModuleName;
554        audio_hw_device_t * const mHwDevice;
555        const Flags mFlags;
556    };
557
558    // AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
559    // For emphasis, we could also make all pointers to them be "const *",
560    // but that would clutter the code unnecessarily.
561
562    struct AudioStreamOut {
563        AudioHwDevice* const audioHwDev;
564        audio_stream_out_t* const stream;
565        const audio_output_flags_t flags;
566
567        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
568
569        AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) :
570            audioHwDev(dev), stream(out), flags(flags) {}
571    };
572
573    struct AudioStreamIn {
574        AudioHwDevice* const audioHwDev;
575        audio_stream_in_t* const stream;
576
577        audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); }
578
579        AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) :
580            audioHwDev(dev), stream(in) {}
581    };
582
583    // for mAudioSessionRefs only
584    struct AudioSessionRef {
585        AudioSessionRef(int sessionid, pid_t pid) :
586            mSessionid(sessionid), mPid(pid), mCnt(1) {}
587        const int   mSessionid;
588        const pid_t mPid;
589        int         mCnt;
590    };
591
592    mutable     Mutex                               mLock;
593                // protects mClients and mNotificationClients.
594                // must be locked after mLock and ThreadBase::mLock if both must be locked
595                // avoids acquiring AudioFlinger::mLock from inside thread loop.
596    mutable     Mutex                               mClientLock;
597                // protected by mClientLock
598                DefaultKeyedVector< pid_t, wp<Client> >     mClients;   // see ~Client()
599
600                mutable     Mutex                   mHardwareLock;
601                // NOTE: If both mLock and mHardwareLock mutexes must be held,
602                // always take mLock before mHardwareLock
603
604                // These two fields are immutable after onFirstRef(), so no lock needed to access
605                AudioHwDevice*                      mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
606                DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*>  mAudioHwDevs;
607
608    // for dump, indicates which hardware operation is currently in progress (but not stream ops)
609    enum hardware_call_state {
610        AUDIO_HW_IDLE = 0,              // no operation in progress
611        AUDIO_HW_INIT,                  // init_check
612        AUDIO_HW_OUTPUT_OPEN,           // open_output_stream
613        AUDIO_HW_OUTPUT_CLOSE,          // unused
614        AUDIO_HW_INPUT_OPEN,            // unused
615        AUDIO_HW_INPUT_CLOSE,           // unused
616        AUDIO_HW_STANDBY,               // unused
617        AUDIO_HW_SET_MASTER_VOLUME,     // set_master_volume
618        AUDIO_HW_GET_ROUTING,           // unused
619        AUDIO_HW_SET_ROUTING,           // unused
620        AUDIO_HW_GET_MODE,              // unused
621        AUDIO_HW_SET_MODE,              // set_mode
622        AUDIO_HW_GET_MIC_MUTE,          // get_mic_mute
623        AUDIO_HW_SET_MIC_MUTE,          // set_mic_mute
624        AUDIO_HW_SET_VOICE_VOLUME,      // set_voice_volume
625        AUDIO_HW_SET_PARAMETER,         // set_parameters
626        AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size
627        AUDIO_HW_GET_MASTER_VOLUME,     // get_master_volume
628        AUDIO_HW_GET_PARAMETER,         // get_parameters
629        AUDIO_HW_SET_MASTER_MUTE,       // set_master_mute
630        AUDIO_HW_GET_MASTER_MUTE,       // get_master_mute
631    };
632
633    mutable     hardware_call_state                 mHardwareStatus;    // for dump only
634
635
636                DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> >  mPlaybackThreads;
637                stream_type_t                       mStreamTypes[AUDIO_STREAM_CNT];
638
639                // member variables below are protected by mLock
640                float                               mMasterVolume;
641                bool                                mMasterMute;
642                // end of variables protected by mLock
643
644                DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> >    mRecordThreads;
645
646                // protected by mClientLock
647                DefaultKeyedVector< pid_t, sp<NotificationClient> >    mNotificationClients;
648
649                volatile int32_t                    mNextUniqueId;  // updated by android_atomic_inc
650                // nextUniqueId() returns uint32_t, but this is declared int32_t
651                // because the atomic operations require an int32_t
652
653                audio_mode_t                        mMode;
654                bool                                mBtNrecIsOff;
655
656                // protected by mLock
657                Vector<AudioSessionRef*> mAudioSessionRefs;
658
659                float       masterVolume_l() const;
660                bool        masterMute_l() const;
661                audio_module_handle_t loadHwModule_l(const char *name);
662
663                Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session
664                                                             // to be created
665
666private:
667    sp<Client>  registerPid(pid_t pid);    // always returns non-0
668
669    // for use from destructor
670    status_t    closeOutput_nonvirtual(audio_io_handle_t output);
671    status_t    closeInput_nonvirtual(audio_io_handle_t input);
672
673#ifdef TEE_SINK
674    // all record threads serially share a common tee sink, which is re-created on format change
675    sp<NBAIO_Sink>   mRecordTeeSink;
676    sp<NBAIO_Source> mRecordTeeSource;
677#endif
678
679public:
680
681#ifdef TEE_SINK
682    // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file
683    static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0);
684
685    // whether tee sink is enabled by property
686    static bool mTeeSinkInputEnabled;
687    static bool mTeeSinkOutputEnabled;
688    static bool mTeeSinkTrackEnabled;
689
690    // runtime configured size of each tee sink pipe, in frames
691    static size_t mTeeSinkInputFrames;
692    static size_t mTeeSinkOutputFrames;
693    static size_t mTeeSinkTrackFrames;
694
695    // compile-time default size of tee sink pipes, in frames
696    // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes
697    static const size_t kTeeSinkInputFramesDefault = 0x200000;
698    static const size_t kTeeSinkOutputFramesDefault = 0x200000;
699    static const size_t kTeeSinkTrackFramesDefault = 0x200000;
700#endif
701
702    // This method reads from a variable without mLock, but the variable is updated under mLock.  So
703    // we might read a stale value, or a value that's inconsistent with respect to other variables.
704    // In this case, it's safe because the return value isn't used for making an important decision.
705    // The reason we don't want to take mLock is because it could block the caller for a long time.
706    bool    isLowRamDevice() const { return mIsLowRamDevice; }
707
708private:
709    bool    mIsLowRamDevice;
710    bool    mIsDeviceTypeKnown;
711    nsecs_t mGlobalEffectEnableTime;  // when a global effect was last enabled
712
713    sp<PatchPanel> mPatchPanel;
714
715    uint32_t    mPrimaryOutputSampleRate;   // sample rate of the primary output, or zero if none
716                                            // protected by mHardwareLock
717};
718
719#undef INCLUDING_FROM_AUDIOFLINGER_H
720
721const char *formatToString(audio_format_t format);
722
723// ----------------------------------------------------------------------------
724
725}; // namespace android
726
727#endif // ANDROID_AUDIO_FLINGER_H
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