AudioFlinger.cpp revision ec40d284218466d8f0e832e7eb88e6ea6c479c88 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <dirent.h>
24#include <math.h>
25#include <signal.h>
26#include <sys/time.h>
27#include <sys/resource.h>
28
29#include <binder/IPCThreadState.h>
30#include <binder/IServiceManager.h>
31#include <utils/Log.h>
32#include <utils/Trace.h>
33#include <binder/Parcel.h>
34#include <utils/String16.h>
35#include <utils/threads.h>
36#include <utils/Atomic.h>
37
38#include <cutils/bitops.h>
39#include <cutils/properties.h>
40
41#include <system/audio.h>
42#include <hardware/audio.h>
43
44#include "AudioMixer.h"
45#include "AudioFlinger.h"
46#include "ServiceUtilities.h"
47
48#include <media/EffectsFactoryApi.h>
49#include <audio_effects/effect_visualizer.h>
50#include <audio_effects/effect_ns.h>
51#include <audio_effects/effect_aec.h>
52
53#include <audio_utils/primitives.h>
54
55#include <powermanager/PowerManager.h>
56
57#include <common_time/cc_helper.h>
58
59#include <media/IMediaLogService.h>
60
61#include <media/nbaio/Pipe.h>
62#include <media/nbaio/PipeReader.h>
63#include <media/AudioParameter.h>
64#include <private/android_filesystem_config.h>
65
66// ----------------------------------------------------------------------------
67
68// Note: the following macro is used for extremely verbose logging message.  In
69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
70// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
71// are so verbose that we want to suppress them even when we have ALOG_ASSERT
72// turned on.  Do not uncomment the #def below unless you really know what you
73// are doing and want to see all of the extremely verbose messages.
74//#define VERY_VERY_VERBOSE_LOGGING
75#ifdef VERY_VERY_VERBOSE_LOGGING
76#define ALOGVV ALOGV
77#else
78#define ALOGVV(a...) do { } while(0)
79#endif
80
81namespace android {
82
83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
84static const char kHardwareLockedString[] = "Hardware lock is taken\n";
85static const char kClientLockedString[] = "Client lock is taken\n";
86
87
88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
89
90uint32_t AudioFlinger::mScreenState;
91
92#ifdef TEE_SINK
93bool AudioFlinger::mTeeSinkInputEnabled = false;
94bool AudioFlinger::mTeeSinkOutputEnabled = false;
95bool AudioFlinger::mTeeSinkTrackEnabled = false;
96
97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
100#endif
101
102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
103// we define a minimum time during which a global effect is considered enabled.
104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
105
106// ----------------------------------------------------------------------------
107
108const char *formatToString(audio_format_t format) {
109    switch (format & AUDIO_FORMAT_MAIN_MASK) {
110    case AUDIO_FORMAT_PCM:
111        switch (format) {
112        case AUDIO_FORMAT_PCM_16_BIT: return "pcm16";
113        case AUDIO_FORMAT_PCM_8_BIT: return "pcm8";
114        case AUDIO_FORMAT_PCM_32_BIT: return "pcm32";
115        case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24";
116        case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat";
117        case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24";
118        default:
119            break;
120        }
121        break;
122    case AUDIO_FORMAT_MP3: return "mp3";
123    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
124    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
125    case AUDIO_FORMAT_AAC: return "aac";
126    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
127    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
128    case AUDIO_FORMAT_VORBIS: return "vorbis";
129    case AUDIO_FORMAT_OPUS: return "opus";
130    case AUDIO_FORMAT_AC3: return "ac-3";
131    case AUDIO_FORMAT_E_AC3: return "e-ac-3";
132    default:
133        break;
134    }
135    return "unknown";
136}
137
138static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
139{
140    const hw_module_t *mod;
141    int rc;
142
143    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
144    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
145                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
146    if (rc) {
147        goto out;
148    }
149    rc = audio_hw_device_open(mod, dev);
150    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
151                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
152    if (rc) {
153        goto out;
154    }
155    if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
156        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
157        rc = BAD_VALUE;
158        goto out;
159    }
160    return 0;
161
162out:
163    *dev = NULL;
164    return rc;
165}
166
167// ----------------------------------------------------------------------------
168
169AudioFlinger::AudioFlinger()
170    : BnAudioFlinger(),
171      mPrimaryHardwareDev(NULL),
172      mAudioHwDevs(NULL),
173      mHardwareStatus(AUDIO_HW_IDLE),
174      mMasterVolume(1.0f),
175      mMasterMute(false),
176      mNextUniqueId(1),
177      mMode(AUDIO_MODE_INVALID),
178      mBtNrecIsOff(false),
179      mIsLowRamDevice(true),
180      mIsDeviceTypeKnown(false),
181      mGlobalEffectEnableTime(0),
182      mPrimaryOutputSampleRate(0)
183{
184    getpid_cached = getpid();
185    char value[PROPERTY_VALUE_MAX];
186    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
187    if (doLog) {
188        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY);
189    }
190
191#ifdef TEE_SINK
192    (void) property_get("ro.debuggable", value, "0");
193    int debuggable = atoi(value);
194    int teeEnabled = 0;
195    if (debuggable) {
196        (void) property_get("af.tee", value, "0");
197        teeEnabled = atoi(value);
198    }
199    // FIXME symbolic constants here
200    if (teeEnabled & 1) {
201        mTeeSinkInputEnabled = true;
202    }
203    if (teeEnabled & 2) {
204        mTeeSinkOutputEnabled = true;
205    }
206    if (teeEnabled & 4) {
207        mTeeSinkTrackEnabled = true;
208    }
209#endif
210}
211
212void AudioFlinger::onFirstRef()
213{
214    int rc = 0;
215
216    Mutex::Autolock _l(mLock);
217
218    /* TODO: move all this work into an Init() function */
219    char val_str[PROPERTY_VALUE_MAX] = { 0 };
220    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
221        uint32_t int_val;
222        if (1 == sscanf(val_str, "%u", &int_val)) {
223            mStandbyTimeInNsecs = milliseconds(int_val);
224            ALOGI("Using %u mSec as standby time.", int_val);
225        } else {
226            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
227            ALOGI("Using default %u mSec as standby time.",
228                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
229        }
230    }
231
232    mPatchPanel = new PatchPanel(this);
233
234    mMode = AUDIO_MODE_NORMAL;
235}
236
237AudioFlinger::~AudioFlinger()
238{
239    while (!mRecordThreads.isEmpty()) {
240        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
241        closeInput_nonvirtual(mRecordThreads.keyAt(0));
242    }
243    while (!mPlaybackThreads.isEmpty()) {
244        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
245        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
246    }
247
248    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
249        // no mHardwareLock needed, as there are no other references to this
250        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
251        delete mAudioHwDevs.valueAt(i);
252    }
253
254    // Tell media.log service about any old writers that still need to be unregistered
255    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
256    if (binder != 0) {
257        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
258        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
259            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
260            mUnregisteredWriters.pop();
261            mediaLogService->unregisterWriter(iMemory);
262        }
263    }
264
265}
266
267static const char * const audio_interfaces[] = {
268    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
269    AUDIO_HARDWARE_MODULE_ID_A2DP,
270    AUDIO_HARDWARE_MODULE_ID_USB,
271};
272#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
273
274AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
275        audio_module_handle_t module,
276        audio_devices_t devices)
277{
278    // if module is 0, the request comes from an old policy manager and we should load
279    // well known modules
280    if (module == 0) {
281        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
282        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
283            loadHwModule_l(audio_interfaces[i]);
284        }
285        // then try to find a module supporting the requested device.
286        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
287            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
288            audio_hw_device_t *dev = audioHwDevice->hwDevice();
289            if ((dev->get_supported_devices != NULL) &&
290                    (dev->get_supported_devices(dev) & devices) == devices)
291                return audioHwDevice;
292        }
293    } else {
294        // check a match for the requested module handle
295        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
296        if (audioHwDevice != NULL) {
297            return audioHwDevice;
298        }
299    }
300
301    return NULL;
302}
303
304void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
305{
306    const size_t SIZE = 256;
307    char buffer[SIZE];
308    String8 result;
309
310    result.append("Clients:\n");
311    for (size_t i = 0; i < mClients.size(); ++i) {
312        sp<Client> client = mClients.valueAt(i).promote();
313        if (client != 0) {
314            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
315            result.append(buffer);
316        }
317    }
318
319    result.append("Notification Clients:\n");
320    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
321        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
322        result.append(buffer);
323    }
324
325    result.append("Global session refs:\n");
326    result.append("  session   pid count\n");
327    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
328        AudioSessionRef *r = mAudioSessionRefs[i];
329        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
330        result.append(buffer);
331    }
332    write(fd, result.string(), result.size());
333}
334
335
336void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
337{
338    const size_t SIZE = 256;
339    char buffer[SIZE];
340    String8 result;
341    hardware_call_state hardwareStatus = mHardwareStatus;
342
343    snprintf(buffer, SIZE, "Hardware status: %d\n"
344                           "Standby Time mSec: %u\n",
345                            hardwareStatus,
346                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
347    result.append(buffer);
348    write(fd, result.string(), result.size());
349}
350
351void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
352{
353    const size_t SIZE = 256;
354    char buffer[SIZE];
355    String8 result;
356    snprintf(buffer, SIZE, "Permission Denial: "
357            "can't dump AudioFlinger from pid=%d, uid=%d\n",
358            IPCThreadState::self()->getCallingPid(),
359            IPCThreadState::self()->getCallingUid());
360    result.append(buffer);
361    write(fd, result.string(), result.size());
362}
363
364bool AudioFlinger::dumpTryLock(Mutex& mutex)
365{
366    bool locked = false;
367    for (int i = 0; i < kDumpLockRetries; ++i) {
368        if (mutex.tryLock() == NO_ERROR) {
369            locked = true;
370            break;
371        }
372        usleep(kDumpLockSleepUs);
373    }
374    return locked;
375}
376
377status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
378{
379    if (!dumpAllowed()) {
380        dumpPermissionDenial(fd, args);
381    } else {
382        // get state of hardware lock
383        bool hardwareLocked = dumpTryLock(mHardwareLock);
384        if (!hardwareLocked) {
385            String8 result(kHardwareLockedString);
386            write(fd, result.string(), result.size());
387        } else {
388            mHardwareLock.unlock();
389        }
390
391        bool locked = dumpTryLock(mLock);
392
393        // failed to lock - AudioFlinger is probably deadlocked
394        if (!locked) {
395            String8 result(kDeadlockedString);
396            write(fd, result.string(), result.size());
397        }
398
399        bool clientLocked = dumpTryLock(mClientLock);
400        if (!clientLocked) {
401            String8 result(kClientLockedString);
402            write(fd, result.string(), result.size());
403        }
404        dumpClients(fd, args);
405        if (clientLocked) {
406            mClientLock.unlock();
407        }
408
409        dumpInternals(fd, args);
410
411        // dump playback threads
412        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
413            mPlaybackThreads.valueAt(i)->dump(fd, args);
414        }
415
416        // dump record threads
417        for (size_t i = 0; i < mRecordThreads.size(); i++) {
418            mRecordThreads.valueAt(i)->dump(fd, args);
419        }
420
421        // dump all hardware devs
422        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
423            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
424            dev->dump(dev, fd);
425        }
426
427#ifdef TEE_SINK
428        // dump the serially shared record tee sink
429        if (mRecordTeeSource != 0) {
430            dumpTee(fd, mRecordTeeSource);
431        }
432#endif
433
434        if (locked) {
435            mLock.unlock();
436        }
437
438        // append a copy of media.log here by forwarding fd to it, but don't attempt
439        // to lookup the service if it's not running, as it will block for a second
440        if (mLogMemoryDealer != 0) {
441            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
442            if (binder != 0) {
443                dprintf(fd, "\nmedia.log:\n");
444                Vector<String16> args;
445                binder->dump(fd, args);
446            }
447        }
448    }
449    return NO_ERROR;
450}
451
452sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
453{
454    Mutex::Autolock _cl(mClientLock);
455    // If pid is already in the mClients wp<> map, then use that entry
456    // (for which promote() is always != 0), otherwise create a new entry and Client.
457    sp<Client> client = mClients.valueFor(pid).promote();
458    if (client == 0) {
459        client = new Client(this, pid);
460        mClients.add(pid, client);
461    }
462
463    return client;
464}
465
466sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
467{
468    // If there is no memory allocated for logs, return a dummy writer that does nothing
469    if (mLogMemoryDealer == 0) {
470        return new NBLog::Writer();
471    }
472    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
473    // Similarly if we can't contact the media.log service, also return a dummy writer
474    if (binder == 0) {
475        return new NBLog::Writer();
476    }
477    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
478    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
479    // If allocation fails, consult the vector of previously unregistered writers
480    // and garbage-collect one or more them until an allocation succeeds
481    if (shared == 0) {
482        Mutex::Autolock _l(mUnregisteredWritersLock);
483        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
484            {
485                // Pick the oldest stale writer to garbage-collect
486                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
487                mUnregisteredWriters.removeAt(0);
488                mediaLogService->unregisterWriter(iMemory);
489                // Now the media.log remote reference to IMemory is gone.  When our last local
490                // reference to IMemory also drops to zero at end of this block,
491                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
492            }
493            // Re-attempt the allocation
494            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
495            if (shared != 0) {
496                goto success;
497            }
498        }
499        // Even after garbage-collecting all old writers, there is still not enough memory,
500        // so return a dummy writer
501        return new NBLog::Writer();
502    }
503success:
504    mediaLogService->registerWriter(shared, size, name);
505    return new NBLog::Writer(size, shared);
506}
507
508void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
509{
510    if (writer == 0) {
511        return;
512    }
513    sp<IMemory> iMemory(writer->getIMemory());
514    if (iMemory == 0) {
515        return;
516    }
517    // Rather than removing the writer immediately, append it to a queue of old writers to
518    // be garbage-collected later.  This allows us to continue to view old logs for a while.
519    Mutex::Autolock _l(mUnregisteredWritersLock);
520    mUnregisteredWriters.push(writer);
521}
522
523// IAudioFlinger interface
524
525
526sp<IAudioTrack> AudioFlinger::createTrack(
527        audio_stream_type_t streamType,
528        uint32_t sampleRate,
529        audio_format_t format,
530        audio_channel_mask_t channelMask,
531        size_t *frameCount,
532        IAudioFlinger::track_flags_t *flags,
533        const sp<IMemory>& sharedBuffer,
534        audio_io_handle_t output,
535        pid_t tid,
536        int *sessionId,
537        int clientUid,
538        status_t *status)
539{
540    sp<PlaybackThread::Track> track;
541    sp<TrackHandle> trackHandle;
542    sp<Client> client;
543    status_t lStatus;
544    int lSessionId;
545
546    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
547    // but if someone uses binder directly they could bypass that and cause us to crash
548    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
549        ALOGE("createTrack() invalid stream type %d", streamType);
550        lStatus = BAD_VALUE;
551        goto Exit;
552    }
553
554    // further sample rate checks are performed by createTrack_l() depending on the thread type
555    if (sampleRate == 0) {
556        ALOGE("createTrack() invalid sample rate %u", sampleRate);
557        lStatus = BAD_VALUE;
558        goto Exit;
559    }
560
561    // further channel mask checks are performed by createTrack_l() depending on the thread type
562    if (!audio_is_output_channel(channelMask)) {
563        ALOGE("createTrack() invalid channel mask %#x", channelMask);
564        lStatus = BAD_VALUE;
565        goto Exit;
566    }
567
568    // further format checks are performed by createTrack_l() depending on the thread type
569    if (!audio_is_valid_format(format)) {
570        ALOGE("createTrack() invalid format %#x", format);
571        lStatus = BAD_VALUE;
572        goto Exit;
573    }
574
575    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
576        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
577        lStatus = BAD_VALUE;
578        goto Exit;
579    }
580
581    {
582        Mutex::Autolock _l(mLock);
583        PlaybackThread *thread = checkPlaybackThread_l(output);
584        if (thread == NULL) {
585            ALOGE("no playback thread found for output handle %d", output);
586            lStatus = BAD_VALUE;
587            goto Exit;
588        }
589
590        pid_t pid = IPCThreadState::self()->getCallingPid();
591        client = registerPid(pid);
592
593        PlaybackThread *effectThread = NULL;
594        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
595            lSessionId = *sessionId;
596            // check if an effect chain with the same session ID is present on another
597            // output thread and move it here.
598            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
599                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
600                if (mPlaybackThreads.keyAt(i) != output) {
601                    uint32_t sessions = t->hasAudioSession(lSessionId);
602                    if (sessions & PlaybackThread::EFFECT_SESSION) {
603                        effectThread = t.get();
604                        break;
605                    }
606                }
607            }
608        } else {
609            // if no audio session id is provided, create one here
610            lSessionId = nextUniqueId();
611            if (sessionId != NULL) {
612                *sessionId = lSessionId;
613            }
614        }
615        ALOGV("createTrack() lSessionId: %d", lSessionId);
616
617        track = thread->createTrack_l(client, streamType, sampleRate, format,
618                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
619        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
620        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
621
622        // move effect chain to this output thread if an effect on same session was waiting
623        // for a track to be created
624        if (lStatus == NO_ERROR && effectThread != NULL) {
625            // no risk of deadlock because AudioFlinger::mLock is held
626            Mutex::Autolock _dl(thread->mLock);
627            Mutex::Autolock _sl(effectThread->mLock);
628            moveEffectChain_l(lSessionId, effectThread, thread, true);
629        }
630
631        // Look for sync events awaiting for a session to be used.
632        for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
633            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
634                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
635                    if (lStatus == NO_ERROR) {
636                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
637                    } else {
638                        mPendingSyncEvents[i]->cancel();
639                    }
640                    mPendingSyncEvents.removeAt(i);
641                    i--;
642                }
643            }
644        }
645
646    }
647
648    if (lStatus != NO_ERROR) {
649        // remove local strong reference to Client before deleting the Track so that the
650        // Client destructor is called by the TrackBase destructor with mClientLock held
651        // Don't hold mClientLock when releasing the reference on the track as the
652        // destructor will acquire it.
653        {
654            Mutex::Autolock _cl(mClientLock);
655            client.clear();
656        }
657        track.clear();
658        goto Exit;
659    }
660
661    // return handle to client
662    trackHandle = new TrackHandle(track);
663
664Exit:
665    *status = lStatus;
666    return trackHandle;
667}
668
669uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
670{
671    Mutex::Autolock _l(mLock);
672    PlaybackThread *thread = checkPlaybackThread_l(output);
673    if (thread == NULL) {
674        ALOGW("sampleRate() unknown thread %d", output);
675        return 0;
676    }
677    return thread->sampleRate();
678}
679
680audio_format_t AudioFlinger::format(audio_io_handle_t output) const
681{
682    Mutex::Autolock _l(mLock);
683    PlaybackThread *thread = checkPlaybackThread_l(output);
684    if (thread == NULL) {
685        ALOGW("format() unknown thread %d", output);
686        return AUDIO_FORMAT_INVALID;
687    }
688    return thread->format();
689}
690
691size_t AudioFlinger::frameCount(audio_io_handle_t output) const
692{
693    Mutex::Autolock _l(mLock);
694    PlaybackThread *thread = checkPlaybackThread_l(output);
695    if (thread == NULL) {
696        ALOGW("frameCount() unknown thread %d", output);
697        return 0;
698    }
699    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
700    //       should examine all callers and fix them to handle smaller counts
701    return thread->frameCount();
702}
703
704uint32_t AudioFlinger::latency(audio_io_handle_t output) const
705{
706    Mutex::Autolock _l(mLock);
707    PlaybackThread *thread = checkPlaybackThread_l(output);
708    if (thread == NULL) {
709        ALOGW("latency(): no playback thread found for output handle %d", output);
710        return 0;
711    }
712    return thread->latency();
713}
714
715status_t AudioFlinger::setMasterVolume(float value)
716{
717    status_t ret = initCheck();
718    if (ret != NO_ERROR) {
719        return ret;
720    }
721
722    // check calling permissions
723    if (!settingsAllowed()) {
724        return PERMISSION_DENIED;
725    }
726
727    Mutex::Autolock _l(mLock);
728    mMasterVolume = value;
729
730    // Set master volume in the HALs which support it.
731    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
732        AutoMutex lock(mHardwareLock);
733        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
734
735        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
736        if (dev->canSetMasterVolume()) {
737            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
738        }
739        mHardwareStatus = AUDIO_HW_IDLE;
740    }
741
742    // Now set the master volume in each playback thread.  Playback threads
743    // assigned to HALs which do not have master volume support will apply
744    // master volume during the mix operation.  Threads with HALs which do
745    // support master volume will simply ignore the setting.
746    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
747        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
748
749    return NO_ERROR;
750}
751
752status_t AudioFlinger::setMode(audio_mode_t mode)
753{
754    status_t ret = initCheck();
755    if (ret != NO_ERROR) {
756        return ret;
757    }
758
759    // check calling permissions
760    if (!settingsAllowed()) {
761        return PERMISSION_DENIED;
762    }
763    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
764        ALOGW("Illegal value: setMode(%d)", mode);
765        return BAD_VALUE;
766    }
767
768    { // scope for the lock
769        AutoMutex lock(mHardwareLock);
770        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
771        mHardwareStatus = AUDIO_HW_SET_MODE;
772        ret = dev->set_mode(dev, mode);
773        mHardwareStatus = AUDIO_HW_IDLE;
774    }
775
776    if (NO_ERROR == ret) {
777        Mutex::Autolock _l(mLock);
778        mMode = mode;
779        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
780            mPlaybackThreads.valueAt(i)->setMode(mode);
781    }
782
783    return ret;
784}
785
786status_t AudioFlinger::setMicMute(bool state)
787{
788    status_t ret = initCheck();
789    if (ret != NO_ERROR) {
790        return ret;
791    }
792
793    // check calling permissions
794    if (!settingsAllowed()) {
795        return PERMISSION_DENIED;
796    }
797
798    AutoMutex lock(mHardwareLock);
799    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
800    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
801    ret = dev->set_mic_mute(dev, state);
802    mHardwareStatus = AUDIO_HW_IDLE;
803    return ret;
804}
805
806bool AudioFlinger::getMicMute() const
807{
808    status_t ret = initCheck();
809    if (ret != NO_ERROR) {
810        return false;
811    }
812
813    bool state = AUDIO_MODE_INVALID;
814    AutoMutex lock(mHardwareLock);
815    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
816    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
817    dev->get_mic_mute(dev, &state);
818    mHardwareStatus = AUDIO_HW_IDLE;
819    return state;
820}
821
822status_t AudioFlinger::setMasterMute(bool muted)
823{
824    status_t ret = initCheck();
825    if (ret != NO_ERROR) {
826        return ret;
827    }
828
829    // check calling permissions
830    if (!settingsAllowed()) {
831        return PERMISSION_DENIED;
832    }
833
834    Mutex::Autolock _l(mLock);
835    mMasterMute = muted;
836
837    // Set master mute in the HALs which support it.
838    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
839        AutoMutex lock(mHardwareLock);
840        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
841
842        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
843        if (dev->canSetMasterMute()) {
844            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
845        }
846        mHardwareStatus = AUDIO_HW_IDLE;
847    }
848
849    // Now set the master mute in each playback thread.  Playback threads
850    // assigned to HALs which do not have master mute support will apply master
851    // mute during the mix operation.  Threads with HALs which do support master
852    // mute will simply ignore the setting.
853    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
854        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
855
856    return NO_ERROR;
857}
858
859float AudioFlinger::masterVolume() const
860{
861    Mutex::Autolock _l(mLock);
862    return masterVolume_l();
863}
864
865bool AudioFlinger::masterMute() const
866{
867    Mutex::Autolock _l(mLock);
868    return masterMute_l();
869}
870
871float AudioFlinger::masterVolume_l() const
872{
873    return mMasterVolume;
874}
875
876bool AudioFlinger::masterMute_l() const
877{
878    return mMasterMute;
879}
880
881status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
882        audio_io_handle_t output)
883{
884    // check calling permissions
885    if (!settingsAllowed()) {
886        return PERMISSION_DENIED;
887    }
888
889    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
890        ALOGE("setStreamVolume() invalid stream %d", stream);
891        return BAD_VALUE;
892    }
893
894    AutoMutex lock(mLock);
895    PlaybackThread *thread = NULL;
896    if (output != AUDIO_IO_HANDLE_NONE) {
897        thread = checkPlaybackThread_l(output);
898        if (thread == NULL) {
899            return BAD_VALUE;
900        }
901    }
902
903    mStreamTypes[stream].volume = value;
904
905    if (thread == NULL) {
906        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
907            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
908        }
909    } else {
910        thread->setStreamVolume(stream, value);
911    }
912
913    return NO_ERROR;
914}
915
916status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
917{
918    // check calling permissions
919    if (!settingsAllowed()) {
920        return PERMISSION_DENIED;
921    }
922
923    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
924        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
925        ALOGE("setStreamMute() invalid stream %d", stream);
926        return BAD_VALUE;
927    }
928
929    AutoMutex lock(mLock);
930    mStreamTypes[stream].mute = muted;
931    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
932        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
933
934    return NO_ERROR;
935}
936
937float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
938{
939    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
940        return 0.0f;
941    }
942
943    AutoMutex lock(mLock);
944    float volume;
945    if (output != AUDIO_IO_HANDLE_NONE) {
946        PlaybackThread *thread = checkPlaybackThread_l(output);
947        if (thread == NULL) {
948            return 0.0f;
949        }
950        volume = thread->streamVolume(stream);
951    } else {
952        volume = streamVolume_l(stream);
953    }
954
955    return volume;
956}
957
958bool AudioFlinger::streamMute(audio_stream_type_t stream) const
959{
960    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
961        return true;
962    }
963
964    AutoMutex lock(mLock);
965    return streamMute_l(stream);
966}
967
968status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
969{
970    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
971            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
972
973    // check calling permissions
974    if (!settingsAllowed()) {
975        return PERMISSION_DENIED;
976    }
977
978    // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
979    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
980        Mutex::Autolock _l(mLock);
981        status_t final_result = NO_ERROR;
982        {
983            AutoMutex lock(mHardwareLock);
984            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
985            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
986                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
987                status_t result = dev->set_parameters(dev, keyValuePairs.string());
988                final_result = result ?: final_result;
989            }
990            mHardwareStatus = AUDIO_HW_IDLE;
991        }
992        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
993        AudioParameter param = AudioParameter(keyValuePairs);
994        String8 value;
995        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
996            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
997            if (mBtNrecIsOff != btNrecIsOff) {
998                for (size_t i = 0; i < mRecordThreads.size(); i++) {
999                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
1000                    audio_devices_t device = thread->inDevice();
1001                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
1002                    // collect all of the thread's session IDs
1003                    KeyedVector<int, bool> ids = thread->sessionIds();
1004                    // suspend effects associated with those session IDs
1005                    for (size_t j = 0; j < ids.size(); ++j) {
1006                        int sessionId = ids.keyAt(j);
1007                        thread->setEffectSuspended(FX_IID_AEC,
1008                                                   suspend,
1009                                                   sessionId);
1010                        thread->setEffectSuspended(FX_IID_NS,
1011                                                   suspend,
1012                                                   sessionId);
1013                    }
1014                }
1015                mBtNrecIsOff = btNrecIsOff;
1016            }
1017        }
1018        String8 screenState;
1019        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1020            bool isOff = screenState == "off";
1021            if (isOff != (AudioFlinger::mScreenState & 1)) {
1022                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1023            }
1024        }
1025        return final_result;
1026    }
1027
1028    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1029    // and the thread is exited once the lock is released
1030    sp<ThreadBase> thread;
1031    {
1032        Mutex::Autolock _l(mLock);
1033        thread = checkPlaybackThread_l(ioHandle);
1034        if (thread == 0) {
1035            thread = checkRecordThread_l(ioHandle);
1036        } else if (thread == primaryPlaybackThread_l()) {
1037            // indicate output device change to all input threads for pre processing
1038            AudioParameter param = AudioParameter(keyValuePairs);
1039            int value;
1040            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1041                    (value != 0)) {
1042                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1043                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1044                }
1045            }
1046        }
1047    }
1048    if (thread != 0) {
1049        return thread->setParameters(keyValuePairs);
1050    }
1051    return BAD_VALUE;
1052}
1053
1054String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1055{
1056    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1057            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1058
1059    Mutex::Autolock _l(mLock);
1060
1061    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1062        String8 out_s8;
1063
1064        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1065            char *s;
1066            {
1067            AutoMutex lock(mHardwareLock);
1068            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1069            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1070            s = dev->get_parameters(dev, keys.string());
1071            mHardwareStatus = AUDIO_HW_IDLE;
1072            }
1073            out_s8 += String8(s ? s : "");
1074            free(s);
1075        }
1076        return out_s8;
1077    }
1078
1079    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1080    if (playbackThread != NULL) {
1081        return playbackThread->getParameters(keys);
1082    }
1083    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1084    if (recordThread != NULL) {
1085        return recordThread->getParameters(keys);
1086    }
1087    return String8("");
1088}
1089
1090size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1091        audio_channel_mask_t channelMask) const
1092{
1093    status_t ret = initCheck();
1094    if (ret != NO_ERROR) {
1095        return 0;
1096    }
1097
1098    AutoMutex lock(mHardwareLock);
1099    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1100    struct audio_config config;
1101    memset(&config, 0, sizeof(config));
1102    config.sample_rate = sampleRate;
1103    config.channel_mask = channelMask;
1104    config.format = format;
1105
1106    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1107    size_t size = dev->get_input_buffer_size(dev, &config);
1108    mHardwareStatus = AUDIO_HW_IDLE;
1109    return size;
1110}
1111
1112uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1113{
1114    Mutex::Autolock _l(mLock);
1115
1116    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1117    if (recordThread != NULL) {
1118        return recordThread->getInputFramesLost();
1119    }
1120    return 0;
1121}
1122
1123status_t AudioFlinger::setVoiceVolume(float value)
1124{
1125    status_t ret = initCheck();
1126    if (ret != NO_ERROR) {
1127        return ret;
1128    }
1129
1130    // check calling permissions
1131    if (!settingsAllowed()) {
1132        return PERMISSION_DENIED;
1133    }
1134
1135    AutoMutex lock(mHardwareLock);
1136    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1137    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1138    ret = dev->set_voice_volume(dev, value);
1139    mHardwareStatus = AUDIO_HW_IDLE;
1140
1141    return ret;
1142}
1143
1144status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1145        audio_io_handle_t output) const
1146{
1147    status_t status;
1148
1149    Mutex::Autolock _l(mLock);
1150
1151    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1152    if (playbackThread != NULL) {
1153        return playbackThread->getRenderPosition(halFrames, dspFrames);
1154    }
1155
1156    return BAD_VALUE;
1157}
1158
1159void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1160{
1161    Mutex::Autolock _l(mLock);
1162    bool clientAdded = false;
1163    {
1164        Mutex::Autolock _cl(mClientLock);
1165
1166        pid_t pid = IPCThreadState::self()->getCallingPid();
1167        if (mNotificationClients.indexOfKey(pid) < 0) {
1168            sp<NotificationClient> notificationClient = new NotificationClient(this,
1169                                                                                client,
1170                                                                                pid);
1171            ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1172
1173            mNotificationClients.add(pid, notificationClient);
1174
1175            sp<IBinder> binder = client->asBinder();
1176            binder->linkToDeath(notificationClient);
1177            clientAdded = true;
1178        }
1179    }
1180
1181    // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
1182    // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock.
1183    if (clientAdded) {
1184        // the config change is always sent from playback or record threads to avoid deadlock
1185        // with AudioSystem::gLock
1186        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1187            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1188        }
1189
1190        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1191            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1192        }
1193    }
1194}
1195
1196void AudioFlinger::removeNotificationClient(pid_t pid)
1197{
1198    Mutex::Autolock _l(mLock);
1199    {
1200        Mutex::Autolock _cl(mClientLock);
1201        mNotificationClients.removeItem(pid);
1202    }
1203
1204    ALOGV("%d died, releasing its sessions", pid);
1205    size_t num = mAudioSessionRefs.size();
1206    bool removed = false;
1207    for (size_t i = 0; i< num; ) {
1208        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1209        ALOGV(" pid %d @ %d", ref->mPid, i);
1210        if (ref->mPid == pid) {
1211            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1212            mAudioSessionRefs.removeAt(i);
1213            delete ref;
1214            removed = true;
1215            num--;
1216        } else {
1217            i++;
1218        }
1219    }
1220    if (removed) {
1221        purgeStaleEffects_l();
1222    }
1223}
1224
1225void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2)
1226{
1227    Mutex::Autolock _l(mClientLock);
1228    size_t size = mNotificationClients.size();
1229    for (size_t i = 0; i < size; i++) {
1230        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event,
1231                                                                              ioHandle,
1232                                                                              param2);
1233    }
1234}
1235
1236// removeClient_l() must be called with AudioFlinger::mClientLock held
1237void AudioFlinger::removeClient_l(pid_t pid)
1238{
1239    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1240            IPCThreadState::self()->getCallingPid());
1241    mClients.removeItem(pid);
1242}
1243
1244// getEffectThread_l() must be called with AudioFlinger::mLock held
1245sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1246{
1247    sp<PlaybackThread> thread;
1248
1249    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1250        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1251            ALOG_ASSERT(thread == 0);
1252            thread = mPlaybackThreads.valueAt(i);
1253        }
1254    }
1255
1256    return thread;
1257}
1258
1259
1260
1261// ----------------------------------------------------------------------------
1262
1263AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1264    :   RefBase(),
1265        mAudioFlinger(audioFlinger),
1266        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1267        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1268        mPid(pid),
1269        mTimedTrackCount(0)
1270{
1271    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1272}
1273
1274// Client destructor must be called with AudioFlinger::mClientLock held
1275AudioFlinger::Client::~Client()
1276{
1277    mAudioFlinger->removeClient_l(mPid);
1278}
1279
1280sp<MemoryDealer> AudioFlinger::Client::heap() const
1281{
1282    return mMemoryDealer;
1283}
1284
1285// Reserve one of the limited slots for a timed audio track associated
1286// with this client
1287bool AudioFlinger::Client::reserveTimedTrack()
1288{
1289    const int kMaxTimedTracksPerClient = 4;
1290
1291    Mutex::Autolock _l(mTimedTrackLock);
1292
1293    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1294        ALOGW("can not create timed track - pid %d has exceeded the limit",
1295             mPid);
1296        return false;
1297    }
1298
1299    mTimedTrackCount++;
1300    return true;
1301}
1302
1303// Release a slot for a timed audio track
1304void AudioFlinger::Client::releaseTimedTrack()
1305{
1306    Mutex::Autolock _l(mTimedTrackLock);
1307    mTimedTrackCount--;
1308}
1309
1310// ----------------------------------------------------------------------------
1311
1312AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1313                                                     const sp<IAudioFlingerClient>& client,
1314                                                     pid_t pid)
1315    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1316{
1317}
1318
1319AudioFlinger::NotificationClient::~NotificationClient()
1320{
1321}
1322
1323void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1324{
1325    sp<NotificationClient> keep(this);
1326    mAudioFlinger->removeNotificationClient(mPid);
1327}
1328
1329
1330// ----------------------------------------------------------------------------
1331
1332static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1333    return audio_is_remote_submix_device(inDevice);
1334}
1335
1336sp<IAudioRecord> AudioFlinger::openRecord(
1337        audio_io_handle_t input,
1338        uint32_t sampleRate,
1339        audio_format_t format,
1340        audio_channel_mask_t channelMask,
1341        size_t *frameCount,
1342        IAudioFlinger::track_flags_t *flags,
1343        pid_t tid,
1344        int *sessionId,
1345        size_t *notificationFrames,
1346        sp<IMemory>& cblk,
1347        sp<IMemory>& buffers,
1348        status_t *status)
1349{
1350    sp<RecordThread::RecordTrack> recordTrack;
1351    sp<RecordHandle> recordHandle;
1352    sp<Client> client;
1353    status_t lStatus;
1354    int lSessionId;
1355
1356    cblk.clear();
1357    buffers.clear();
1358
1359    // check calling permissions
1360    if (!recordingAllowed()) {
1361        ALOGE("openRecord() permission denied: recording not allowed");
1362        lStatus = PERMISSION_DENIED;
1363        goto Exit;
1364    }
1365
1366    // further sample rate checks are performed by createRecordTrack_l()
1367    if (sampleRate == 0) {
1368        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1369        lStatus = BAD_VALUE;
1370        goto Exit;
1371    }
1372
1373    // we don't yet support anything other than 16-bit PCM
1374    if (!(audio_is_valid_format(format) &&
1375            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
1376        ALOGE("openRecord() invalid format %#x", format);
1377        lStatus = BAD_VALUE;
1378        goto Exit;
1379    }
1380
1381    // further channel mask checks are performed by createRecordTrack_l()
1382    if (!audio_is_input_channel(channelMask)) {
1383        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1384        lStatus = BAD_VALUE;
1385        goto Exit;
1386    }
1387
1388    {
1389        Mutex::Autolock _l(mLock);
1390        RecordThread *thread = checkRecordThread_l(input);
1391        if (thread == NULL) {
1392            ALOGE("openRecord() checkRecordThread_l failed");
1393            lStatus = BAD_VALUE;
1394            goto Exit;
1395        }
1396
1397        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1398                && !captureAudioOutputAllowed()) {
1399            ALOGE("openRecord() permission denied: capture not allowed");
1400            lStatus = PERMISSION_DENIED;
1401            goto Exit;
1402        }
1403
1404        pid_t pid = IPCThreadState::self()->getCallingPid();
1405        client = registerPid(pid);
1406
1407        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1408            lSessionId = *sessionId;
1409        } else {
1410            // if no audio session id is provided, create one here
1411            lSessionId = nextUniqueId();
1412            if (sessionId != NULL) {
1413                *sessionId = lSessionId;
1414            }
1415        }
1416        ALOGV("openRecord() lSessionId: %d", lSessionId);
1417
1418        // TODO: the uid should be passed in as a parameter to openRecord
1419        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1420                                                  frameCount, lSessionId, notificationFrames,
1421                                                  IPCThreadState::self()->getCallingUid(),
1422                                                  flags, tid, &lStatus);
1423        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1424    }
1425
1426    if (lStatus != NO_ERROR) {
1427        // remove local strong reference to Client before deleting the RecordTrack so that the
1428        // Client destructor is called by the TrackBase destructor with mClientLock held
1429        // Don't hold mClientLock when releasing the reference on the track as the
1430        // destructor will acquire it.
1431        {
1432            Mutex::Autolock _cl(mClientLock);
1433            client.clear();
1434        }
1435        recordTrack.clear();
1436        goto Exit;
1437    }
1438
1439    cblk = recordTrack->getCblk();
1440    buffers = recordTrack->getBuffers();
1441
1442    // return handle to client
1443    recordHandle = new RecordHandle(recordTrack);
1444
1445Exit:
1446    *status = lStatus;
1447    return recordHandle;
1448}
1449
1450
1451
1452// ----------------------------------------------------------------------------
1453
1454audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1455{
1456    if (!settingsAllowed()) {
1457        return 0;
1458    }
1459    Mutex::Autolock _l(mLock);
1460    return loadHwModule_l(name);
1461}
1462
1463// loadHwModule_l() must be called with AudioFlinger::mLock held
1464audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1465{
1466    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1467        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1468            ALOGW("loadHwModule() module %s already loaded", name);
1469            return mAudioHwDevs.keyAt(i);
1470        }
1471    }
1472
1473    audio_hw_device_t *dev;
1474
1475    int rc = load_audio_interface(name, &dev);
1476    if (rc) {
1477        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1478        return 0;
1479    }
1480
1481    mHardwareStatus = AUDIO_HW_INIT;
1482    rc = dev->init_check(dev);
1483    mHardwareStatus = AUDIO_HW_IDLE;
1484    if (rc) {
1485        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1486        return 0;
1487    }
1488
1489    // Check and cache this HAL's level of support for master mute and master
1490    // volume.  If this is the first HAL opened, and it supports the get
1491    // methods, use the initial values provided by the HAL as the current
1492    // master mute and volume settings.
1493
1494    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1495    {  // scope for auto-lock pattern
1496        AutoMutex lock(mHardwareLock);
1497
1498        if (0 == mAudioHwDevs.size()) {
1499            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1500            if (NULL != dev->get_master_volume) {
1501                float mv;
1502                if (OK == dev->get_master_volume(dev, &mv)) {
1503                    mMasterVolume = mv;
1504                }
1505            }
1506
1507            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1508            if (NULL != dev->get_master_mute) {
1509                bool mm;
1510                if (OK == dev->get_master_mute(dev, &mm)) {
1511                    mMasterMute = mm;
1512                }
1513            }
1514        }
1515
1516        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1517        if ((NULL != dev->set_master_volume) &&
1518            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1519            flags = static_cast<AudioHwDevice::Flags>(flags |
1520                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1521        }
1522
1523        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1524        if ((NULL != dev->set_master_mute) &&
1525            (OK == dev->set_master_mute(dev, mMasterMute))) {
1526            flags = static_cast<AudioHwDevice::Flags>(flags |
1527                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1528        }
1529
1530        mHardwareStatus = AUDIO_HW_IDLE;
1531    }
1532
1533    audio_module_handle_t handle = nextUniqueId();
1534    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1535
1536    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1537          name, dev->common.module->name, dev->common.module->id, handle);
1538
1539    return handle;
1540
1541}
1542
1543// ----------------------------------------------------------------------------
1544
1545uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1546{
1547    Mutex::Autolock _l(mLock);
1548    PlaybackThread *thread = primaryPlaybackThread_l();
1549    return thread != NULL ? thread->sampleRate() : 0;
1550}
1551
1552size_t AudioFlinger::getPrimaryOutputFrameCount()
1553{
1554    Mutex::Autolock _l(mLock);
1555    PlaybackThread *thread = primaryPlaybackThread_l();
1556    return thread != NULL ? thread->frameCountHAL() : 0;
1557}
1558
1559// ----------------------------------------------------------------------------
1560
1561status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1562{
1563    uid_t uid = IPCThreadState::self()->getCallingUid();
1564    if (uid != AID_SYSTEM) {
1565        return PERMISSION_DENIED;
1566    }
1567    Mutex::Autolock _l(mLock);
1568    if (mIsDeviceTypeKnown) {
1569        return INVALID_OPERATION;
1570    }
1571    mIsLowRamDevice = isLowRamDevice;
1572    mIsDeviceTypeKnown = true;
1573    return NO_ERROR;
1574}
1575
1576// ----------------------------------------------------------------------------
1577
1578audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1579                                           audio_devices_t *pDevices,
1580                                           uint32_t *pSamplingRate,
1581                                           audio_format_t *pFormat,
1582                                           audio_channel_mask_t *pChannelMask,
1583                                           uint32_t *pLatencyMs,
1584                                           audio_output_flags_t flags,
1585                                           const audio_offload_info_t *offloadInfo)
1586{
1587    struct audio_config config;
1588    memset(&config, 0, sizeof(config));
1589    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1590    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1591    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1592    if (offloadInfo != NULL) {
1593        config.offload_info = *offloadInfo;
1594    }
1595
1596    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1597              module,
1598              (pDevices != NULL) ? *pDevices : 0,
1599              config.sample_rate,
1600              config.format,
1601              config.channel_mask,
1602              flags);
1603    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1604          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1605
1606    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1607        return AUDIO_IO_HANDLE_NONE;
1608    }
1609
1610    Mutex::Autolock _l(mLock);
1611
1612    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1613    if (outHwDev == NULL) {
1614        return AUDIO_IO_HANDLE_NONE;
1615    }
1616
1617    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1618    audio_io_handle_t id = nextUniqueId();
1619
1620    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1621
1622    audio_stream_out_t *outStream = NULL;
1623
1624    // FOR TESTING ONLY:
1625    // Enable increased sink precision for mixing mode if kEnableExtendedPrecision is true.
1626    if (kEnableExtendedPrecision &&  // Check only for Normal Mixing mode
1627            !(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
1628        // Update format
1629        //config.format = AUDIO_FORMAT_PCM_FLOAT;
1630        //config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
1631        //config.format = AUDIO_FORMAT_PCM_32_BIT;
1632        //config.format = AUDIO_FORMAT_PCM_8_24_BIT;
1633        // ALOGV("openOutput() upgrading format to %#08x", config.format);
1634    }
1635
1636    status_t status = hwDevHal->open_output_stream(hwDevHal,
1637                                          id,
1638                                          *pDevices,
1639                                          (audio_output_flags_t)flags,
1640                                          &config,
1641                                          &outStream);
1642
1643    mHardwareStatus = AUDIO_HW_IDLE;
1644    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1645            "Channels %x, status %d",
1646            outStream,
1647            config.sample_rate,
1648            config.format,
1649            config.channel_mask,
1650            status);
1651
1652    if (status == NO_ERROR && outStream != NULL) {
1653        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1654
1655        PlaybackThread *thread;
1656        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1657            thread = new OffloadThread(this, output, id, *pDevices);
1658            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1659        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
1660                || !isValidPcmSinkFormat(config.format)
1661                || (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1662            thread = new DirectOutputThread(this, output, id, *pDevices);
1663            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1664        } else {
1665            thread = new MixerThread(this, output, id, *pDevices);
1666            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1667        }
1668        mPlaybackThreads.add(id, thread);
1669
1670        if (pSamplingRate != NULL) {
1671            *pSamplingRate = config.sample_rate;
1672        }
1673        if (pFormat != NULL) {
1674            *pFormat = config.format;
1675        }
1676        if (pChannelMask != NULL) {
1677            *pChannelMask = config.channel_mask;
1678        }
1679        if (pLatencyMs != NULL) {
1680            *pLatencyMs = thread->latency();
1681        }
1682
1683        // notify client processes of the new output creation
1684        thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED);
1685
1686        // the first primary output opened designates the primary hw device
1687        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1688            ALOGI("Using module %d has the primary audio interface", module);
1689            mPrimaryHardwareDev = outHwDev;
1690
1691            AutoMutex lock(mHardwareLock);
1692            mHardwareStatus = AUDIO_HW_SET_MODE;
1693            hwDevHal->set_mode(hwDevHal, mMode);
1694            mHardwareStatus = AUDIO_HW_IDLE;
1695
1696            mPrimaryOutputSampleRate = config.sample_rate;
1697        }
1698        return id;
1699    }
1700
1701    return AUDIO_IO_HANDLE_NONE;
1702}
1703
1704audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1705        audio_io_handle_t output2)
1706{
1707    Mutex::Autolock _l(mLock);
1708    MixerThread *thread1 = checkMixerThread_l(output1);
1709    MixerThread *thread2 = checkMixerThread_l(output2);
1710
1711    if (thread1 == NULL || thread2 == NULL) {
1712        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1713                output2);
1714        return AUDIO_IO_HANDLE_NONE;
1715    }
1716
1717    audio_io_handle_t id = nextUniqueId();
1718    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1719    thread->addOutputTrack(thread2);
1720    mPlaybackThreads.add(id, thread);
1721    // notify client processes of the new output creation
1722    thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED);
1723    return id;
1724}
1725
1726status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1727{
1728    return closeOutput_nonvirtual(output);
1729}
1730
1731status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1732{
1733    // keep strong reference on the playback thread so that
1734    // it is not destroyed while exit() is executed
1735    sp<PlaybackThread> thread;
1736    {
1737        Mutex::Autolock _l(mLock);
1738        thread = checkPlaybackThread_l(output);
1739        if (thread == NULL) {
1740            return BAD_VALUE;
1741        }
1742
1743        ALOGV("closeOutput() %d", output);
1744
1745        if (thread->type() == ThreadBase::MIXER) {
1746            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1747                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1748                    DuplicatingThread *dupThread =
1749                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1750                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1751
1752                }
1753            }
1754        }
1755
1756
1757        mPlaybackThreads.removeItem(output);
1758        // save all effects to the default thread
1759        if (mPlaybackThreads.size()) {
1760            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1761            if (dstThread != NULL) {
1762                // audioflinger lock is held here so the acquisition order of thread locks does not
1763                // matter
1764                Mutex::Autolock _dl(dstThread->mLock);
1765                Mutex::Autolock _sl(thread->mLock);
1766                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1767                for (size_t i = 0; i < effectChains.size(); i ++) {
1768                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1769                }
1770            }
1771        }
1772        audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL);
1773    }
1774    thread->exit();
1775    // The thread entity (active unit of execution) is no longer running here,
1776    // but the ThreadBase container still exists.
1777
1778    if (thread->type() != ThreadBase::DUPLICATING) {
1779        AudioStreamOut *out = thread->clearOutput();
1780        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1781        // from now on thread->mOutput is NULL
1782        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1783        delete out;
1784    }
1785    return NO_ERROR;
1786}
1787
1788status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1789{
1790    Mutex::Autolock _l(mLock);
1791    PlaybackThread *thread = checkPlaybackThread_l(output);
1792
1793    if (thread == NULL) {
1794        return BAD_VALUE;
1795    }
1796
1797    ALOGV("suspendOutput() %d", output);
1798    thread->suspend();
1799
1800    return NO_ERROR;
1801}
1802
1803status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1804{
1805    Mutex::Autolock _l(mLock);
1806    PlaybackThread *thread = checkPlaybackThread_l(output);
1807
1808    if (thread == NULL) {
1809        return BAD_VALUE;
1810    }
1811
1812    ALOGV("restoreOutput() %d", output);
1813
1814    thread->restore();
1815
1816    return NO_ERROR;
1817}
1818
1819audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1820                                          audio_devices_t *pDevices,
1821                                          uint32_t *pSamplingRate,
1822                                          audio_format_t *pFormat,
1823                                          audio_channel_mask_t *pChannelMask,
1824                                          audio_input_flags_t flags)
1825{
1826    struct audio_config config;
1827    memset(&config, 0, sizeof(config));
1828    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1829    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1830    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1831
1832    uint32_t reqSamplingRate = config.sample_rate;
1833    audio_format_t reqFormat = config.format;
1834    audio_channel_mask_t reqChannelMask = config.channel_mask;
1835
1836    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1837        return 0;
1838    }
1839
1840    Mutex::Autolock _l(mLock);
1841
1842    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1843    if (inHwDev == NULL) {
1844        return 0;
1845    }
1846
1847    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1848    audio_io_handle_t id = nextUniqueId();
1849
1850    audio_stream_in_t *inStream = NULL;
1851    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1852                                        &inStream, flags);
1853    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
1854            "flags %#x, status %d",
1855            inStream,
1856            config.sample_rate,
1857            config.format,
1858            config.channel_mask,
1859            flags,
1860            status);
1861
1862    // If the input could not be opened with the requested parameters and we can handle the
1863    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1864    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1865    if (status == BAD_VALUE &&
1866        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1867        (config.sample_rate <= 2 * reqSamplingRate) &&
1868        (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) &&
1869        (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) {
1870        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1871        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1872        inStream = NULL;
1873        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream, flags);
1874        // FIXME log this new status; HAL should not propose any further changes
1875    }
1876
1877    if (status == NO_ERROR && inStream != NULL) {
1878
1879#ifdef TEE_SINK
1880        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1881        // or (re-)create if current Pipe is idle and does not match the new format
1882        sp<NBAIO_Sink> teeSink;
1883        enum {
1884            TEE_SINK_NO,    // don't copy input
1885            TEE_SINK_NEW,   // copy input using a new pipe
1886            TEE_SINK_OLD,   // copy input using an existing pipe
1887        } kind;
1888        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1889                audio_channel_count_from_in_mask(
1890                        inStream->common.get_channels(&inStream->common)));
1891        if (!mTeeSinkInputEnabled) {
1892            kind = TEE_SINK_NO;
1893        } else if (!Format_isValid(format)) {
1894            kind = TEE_SINK_NO;
1895        } else if (mRecordTeeSink == 0) {
1896            kind = TEE_SINK_NEW;
1897        } else if (mRecordTeeSink->getStrongCount() != 1) {
1898            kind = TEE_SINK_NO;
1899        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
1900            kind = TEE_SINK_OLD;
1901        } else {
1902            kind = TEE_SINK_NEW;
1903        }
1904        switch (kind) {
1905        case TEE_SINK_NEW: {
1906            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1907            size_t numCounterOffers = 0;
1908            const NBAIO_Format offers[1] = {format};
1909            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1910            ALOG_ASSERT(index == 0);
1911            PipeReader *pipeReader = new PipeReader(*pipe);
1912            numCounterOffers = 0;
1913            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1914            ALOG_ASSERT(index == 0);
1915            mRecordTeeSink = pipe;
1916            mRecordTeeSource = pipeReader;
1917            teeSink = pipe;
1918            }
1919            break;
1920        case TEE_SINK_OLD:
1921            teeSink = mRecordTeeSink;
1922            break;
1923        case TEE_SINK_NO:
1924        default:
1925            break;
1926        }
1927#endif
1928
1929        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1930
1931        // Start record thread
1932        // RecordThread requires both input and output device indication to forward to audio
1933        // pre processing modules
1934        RecordThread *thread = new RecordThread(this,
1935                                  input,
1936                                  id,
1937                                  primaryOutputDevice_l(),
1938                                  *pDevices
1939#ifdef TEE_SINK
1940                                  , teeSink
1941#endif
1942                                  );
1943        mRecordThreads.add(id, thread);
1944        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1945        if (pSamplingRate != NULL) {
1946            *pSamplingRate = reqSamplingRate;
1947        }
1948        if (pFormat != NULL) {
1949            *pFormat = config.format;
1950        }
1951        if (pChannelMask != NULL) {
1952            *pChannelMask = reqChannelMask;
1953        }
1954
1955        // notify client processes of the new input creation
1956        thread->audioConfigChanged(AudioSystem::INPUT_OPENED);
1957        return id;
1958    }
1959
1960    return 0;
1961}
1962
1963status_t AudioFlinger::closeInput(audio_io_handle_t input)
1964{
1965    return closeInput_nonvirtual(input);
1966}
1967
1968status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1969{
1970    // keep strong reference on the record thread so that
1971    // it is not destroyed while exit() is executed
1972    sp<RecordThread> thread;
1973    {
1974        Mutex::Autolock _l(mLock);
1975        thread = checkRecordThread_l(input);
1976        if (thread == 0) {
1977            return BAD_VALUE;
1978        }
1979
1980        ALOGV("closeInput() %d", input);
1981        audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL);
1982        mRecordThreads.removeItem(input);
1983    }
1984    thread->exit();
1985    // The thread entity (active unit of execution) is no longer running here,
1986    // but the ThreadBase container still exists.
1987
1988    AudioStreamIn *in = thread->clearInput();
1989    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1990    // from now on thread->mInput is NULL
1991    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1992    delete in;
1993
1994    return NO_ERROR;
1995}
1996
1997status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
1998{
1999    Mutex::Autolock _l(mLock);
2000    ALOGV("invalidateStream() stream %d", stream);
2001
2002    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2003        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2004        thread->invalidateTracks(stream);
2005    }
2006
2007    return NO_ERROR;
2008}
2009
2010
2011int AudioFlinger::newAudioSessionId()
2012{
2013    return nextUniqueId();
2014}
2015
2016void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
2017{
2018    Mutex::Autolock _l(mLock);
2019    pid_t caller = IPCThreadState::self()->getCallingPid();
2020    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
2021    if (pid != -1 && (caller == getpid_cached)) {
2022        caller = pid;
2023    }
2024
2025    {
2026        Mutex::Autolock _cl(mClientLock);
2027        // Ignore requests received from processes not known as notification client. The request
2028        // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
2029        // called from a different pid leaving a stale session reference.  Also we don't know how
2030        // to clear this reference if the client process dies.
2031        if (mNotificationClients.indexOfKey(caller) < 0) {
2032            ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
2033            return;
2034        }
2035    }
2036
2037    size_t num = mAudioSessionRefs.size();
2038    for (size_t i = 0; i< num; i++) {
2039        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
2040        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2041            ref->mCnt++;
2042            ALOGV(" incremented refcount to %d", ref->mCnt);
2043            return;
2044        }
2045    }
2046    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
2047    ALOGV(" added new entry for %d", audioSession);
2048}
2049
2050void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
2051{
2052    Mutex::Autolock _l(mLock);
2053    pid_t caller = IPCThreadState::self()->getCallingPid();
2054    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
2055    if (pid != -1 && (caller == getpid_cached)) {
2056        caller = pid;
2057    }
2058    size_t num = mAudioSessionRefs.size();
2059    for (size_t i = 0; i< num; i++) {
2060        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2061        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2062            ref->mCnt--;
2063            ALOGV(" decremented refcount to %d", ref->mCnt);
2064            if (ref->mCnt == 0) {
2065                mAudioSessionRefs.removeAt(i);
2066                delete ref;
2067                purgeStaleEffects_l();
2068            }
2069            return;
2070        }
2071    }
2072    // If the caller is mediaserver it is likely that the session being released was acquired
2073    // on behalf of a process not in notification clients and we ignore the warning.
2074    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2075}
2076
2077void AudioFlinger::purgeStaleEffects_l() {
2078
2079    ALOGV("purging stale effects");
2080
2081    Vector< sp<EffectChain> > chains;
2082
2083    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2084        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2085        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2086            sp<EffectChain> ec = t->mEffectChains[j];
2087            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2088                chains.push(ec);
2089            }
2090        }
2091    }
2092    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2093        sp<RecordThread> t = mRecordThreads.valueAt(i);
2094        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2095            sp<EffectChain> ec = t->mEffectChains[j];
2096            chains.push(ec);
2097        }
2098    }
2099
2100    for (size_t i = 0; i < chains.size(); i++) {
2101        sp<EffectChain> ec = chains[i];
2102        int sessionid = ec->sessionId();
2103        sp<ThreadBase> t = ec->mThread.promote();
2104        if (t == 0) {
2105            continue;
2106        }
2107        size_t numsessionrefs = mAudioSessionRefs.size();
2108        bool found = false;
2109        for (size_t k = 0; k < numsessionrefs; k++) {
2110            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2111            if (ref->mSessionid == sessionid) {
2112                ALOGV(" session %d still exists for %d with %d refs",
2113                    sessionid, ref->mPid, ref->mCnt);
2114                found = true;
2115                break;
2116            }
2117        }
2118        if (!found) {
2119            Mutex::Autolock _l(t->mLock);
2120            // remove all effects from the chain
2121            while (ec->mEffects.size()) {
2122                sp<EffectModule> effect = ec->mEffects[0];
2123                effect->unPin();
2124                t->removeEffect_l(effect);
2125                if (effect->purgeHandles()) {
2126                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2127                }
2128                AudioSystem::unregisterEffect(effect->id());
2129            }
2130        }
2131    }
2132    return;
2133}
2134
2135// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2136AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2137{
2138    return mPlaybackThreads.valueFor(output).get();
2139}
2140
2141// checkMixerThread_l() must be called with AudioFlinger::mLock held
2142AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2143{
2144    PlaybackThread *thread = checkPlaybackThread_l(output);
2145    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2146}
2147
2148// checkRecordThread_l() must be called with AudioFlinger::mLock held
2149AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2150{
2151    return mRecordThreads.valueFor(input).get();
2152}
2153
2154uint32_t AudioFlinger::nextUniqueId()
2155{
2156    return (uint32_t) android_atomic_inc(&mNextUniqueId);
2157}
2158
2159AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2160{
2161    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2162        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2163        AudioStreamOut *output = thread->getOutput();
2164        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2165            return thread;
2166        }
2167    }
2168    return NULL;
2169}
2170
2171audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2172{
2173    PlaybackThread *thread = primaryPlaybackThread_l();
2174
2175    if (thread == NULL) {
2176        return 0;
2177    }
2178
2179    return thread->outDevice();
2180}
2181
2182sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2183                                    int triggerSession,
2184                                    int listenerSession,
2185                                    sync_event_callback_t callBack,
2186                                    wp<RefBase> cookie)
2187{
2188    Mutex::Autolock _l(mLock);
2189
2190    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2191    status_t playStatus = NAME_NOT_FOUND;
2192    status_t recStatus = NAME_NOT_FOUND;
2193    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2194        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2195        if (playStatus == NO_ERROR) {
2196            return event;
2197        }
2198    }
2199    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2200        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2201        if (recStatus == NO_ERROR) {
2202            return event;
2203        }
2204    }
2205    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2206        mPendingSyncEvents.add(event);
2207    } else {
2208        ALOGV("createSyncEvent() invalid event %d", event->type());
2209        event.clear();
2210    }
2211    return event;
2212}
2213
2214// ----------------------------------------------------------------------------
2215//  Effect management
2216// ----------------------------------------------------------------------------
2217
2218
2219status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2220{
2221    Mutex::Autolock _l(mLock);
2222    return EffectQueryNumberEffects(numEffects);
2223}
2224
2225status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2226{
2227    Mutex::Autolock _l(mLock);
2228    return EffectQueryEffect(index, descriptor);
2229}
2230
2231status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2232        effect_descriptor_t *descriptor) const
2233{
2234    Mutex::Autolock _l(mLock);
2235    return EffectGetDescriptor(pUuid, descriptor);
2236}
2237
2238
2239sp<IEffect> AudioFlinger::createEffect(
2240        effect_descriptor_t *pDesc,
2241        const sp<IEffectClient>& effectClient,
2242        int32_t priority,
2243        audio_io_handle_t io,
2244        int sessionId,
2245        status_t *status,
2246        int *id,
2247        int *enabled)
2248{
2249    status_t lStatus = NO_ERROR;
2250    sp<EffectHandle> handle;
2251    effect_descriptor_t desc;
2252
2253    pid_t pid = IPCThreadState::self()->getCallingPid();
2254    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2255            pid, effectClient.get(), priority, sessionId, io);
2256
2257    if (pDesc == NULL) {
2258        lStatus = BAD_VALUE;
2259        goto Exit;
2260    }
2261
2262    // check audio settings permission for global effects
2263    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2264        lStatus = PERMISSION_DENIED;
2265        goto Exit;
2266    }
2267
2268    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2269    // that can only be created by audio policy manager (running in same process)
2270    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2271        lStatus = PERMISSION_DENIED;
2272        goto Exit;
2273    }
2274
2275    {
2276        if (!EffectIsNullUuid(&pDesc->uuid)) {
2277            // if uuid is specified, request effect descriptor
2278            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2279            if (lStatus < 0) {
2280                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2281                goto Exit;
2282            }
2283        } else {
2284            // if uuid is not specified, look for an available implementation
2285            // of the required type in effect factory
2286            if (EffectIsNullUuid(&pDesc->type)) {
2287                ALOGW("createEffect() no effect type");
2288                lStatus = BAD_VALUE;
2289                goto Exit;
2290            }
2291            uint32_t numEffects = 0;
2292            effect_descriptor_t d;
2293            d.flags = 0; // prevent compiler warning
2294            bool found = false;
2295
2296            lStatus = EffectQueryNumberEffects(&numEffects);
2297            if (lStatus < 0) {
2298                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2299                goto Exit;
2300            }
2301            for (uint32_t i = 0; i < numEffects; i++) {
2302                lStatus = EffectQueryEffect(i, &desc);
2303                if (lStatus < 0) {
2304                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2305                    continue;
2306                }
2307                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2308                    // If matching type found save effect descriptor. If the session is
2309                    // 0 and the effect is not auxiliary, continue enumeration in case
2310                    // an auxiliary version of this effect type is available
2311                    found = true;
2312                    d = desc;
2313                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2314                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2315                        break;
2316                    }
2317                }
2318            }
2319            if (!found) {
2320                lStatus = BAD_VALUE;
2321                ALOGW("createEffect() effect not found");
2322                goto Exit;
2323            }
2324            // For same effect type, chose auxiliary version over insert version if
2325            // connect to output mix (Compliance to OpenSL ES)
2326            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2327                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2328                desc = d;
2329            }
2330        }
2331
2332        // Do not allow auxiliary effects on a session different from 0 (output mix)
2333        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2334             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2335            lStatus = INVALID_OPERATION;
2336            goto Exit;
2337        }
2338
2339        // check recording permission for visualizer
2340        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2341            !recordingAllowed()) {
2342            lStatus = PERMISSION_DENIED;
2343            goto Exit;
2344        }
2345
2346        // return effect descriptor
2347        *pDesc = desc;
2348        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2349            // if the output returned by getOutputForEffect() is removed before we lock the
2350            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2351            // and we will exit safely
2352            io = AudioSystem::getOutputForEffect(&desc);
2353            ALOGV("createEffect got output %d", io);
2354        }
2355
2356        Mutex::Autolock _l(mLock);
2357
2358        // If output is not specified try to find a matching audio session ID in one of the
2359        // output threads.
2360        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2361        // because of code checking output when entering the function.
2362        // Note: io is never 0 when creating an effect on an input
2363        if (io == AUDIO_IO_HANDLE_NONE) {
2364            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2365                // output must be specified by AudioPolicyManager when using session
2366                // AUDIO_SESSION_OUTPUT_STAGE
2367                lStatus = BAD_VALUE;
2368                goto Exit;
2369            }
2370            // look for the thread where the specified audio session is present
2371            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2372                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2373                    io = mPlaybackThreads.keyAt(i);
2374                    break;
2375                }
2376            }
2377            if (io == 0) {
2378                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2379                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2380                        io = mRecordThreads.keyAt(i);
2381                        break;
2382                    }
2383                }
2384            }
2385            // If no output thread contains the requested session ID, default to
2386            // first output. The effect chain will be moved to the correct output
2387            // thread when a track with the same session ID is created
2388            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2389                io = mPlaybackThreads.keyAt(0);
2390            }
2391            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2392        }
2393        ThreadBase *thread = checkRecordThread_l(io);
2394        if (thread == NULL) {
2395            thread = checkPlaybackThread_l(io);
2396            if (thread == NULL) {
2397                ALOGE("createEffect() unknown output thread");
2398                lStatus = BAD_VALUE;
2399                goto Exit;
2400            }
2401        }
2402
2403        sp<Client> client = registerPid(pid);
2404
2405        // create effect on selected output thread
2406        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2407                &desc, enabled, &lStatus);
2408        if (handle != 0 && id != NULL) {
2409            *id = handle->id();
2410        }
2411        if (handle == 0) {
2412            // remove local strong reference to Client with mClientLock held
2413            Mutex::Autolock _cl(mClientLock);
2414            client.clear();
2415        }
2416    }
2417
2418Exit:
2419    *status = lStatus;
2420    return handle;
2421}
2422
2423status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2424        audio_io_handle_t dstOutput)
2425{
2426    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2427            sessionId, srcOutput, dstOutput);
2428    Mutex::Autolock _l(mLock);
2429    if (srcOutput == dstOutput) {
2430        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2431        return NO_ERROR;
2432    }
2433    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2434    if (srcThread == NULL) {
2435        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2436        return BAD_VALUE;
2437    }
2438    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2439    if (dstThread == NULL) {
2440        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2441        return BAD_VALUE;
2442    }
2443
2444    Mutex::Autolock _dl(dstThread->mLock);
2445    Mutex::Autolock _sl(srcThread->mLock);
2446    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2447}
2448
2449// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2450status_t AudioFlinger::moveEffectChain_l(int sessionId,
2451                                   AudioFlinger::PlaybackThread *srcThread,
2452                                   AudioFlinger::PlaybackThread *dstThread,
2453                                   bool reRegister)
2454{
2455    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2456            sessionId, srcThread, dstThread);
2457
2458    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2459    if (chain == 0) {
2460        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2461                sessionId, srcThread);
2462        return INVALID_OPERATION;
2463    }
2464
2465    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2466    // so that a new chain is created with correct parameters when first effect is added. This is
2467    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2468    // removed.
2469    srcThread->removeEffectChain_l(chain);
2470
2471    // transfer all effects one by one so that new effect chain is created on new thread with
2472    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2473    sp<EffectChain> dstChain;
2474    uint32_t strategy = 0; // prevent compiler warning
2475    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2476    Vector< sp<EffectModule> > removed;
2477    status_t status = NO_ERROR;
2478    while (effect != 0) {
2479        srcThread->removeEffect_l(effect);
2480        removed.add(effect);
2481        status = dstThread->addEffect_l(effect);
2482        if (status != NO_ERROR) {
2483            break;
2484        }
2485        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2486        if (effect->state() == EffectModule::ACTIVE ||
2487                effect->state() == EffectModule::STOPPING) {
2488            effect->start();
2489        }
2490        // if the move request is not received from audio policy manager, the effect must be
2491        // re-registered with the new strategy and output
2492        if (dstChain == 0) {
2493            dstChain = effect->chain().promote();
2494            if (dstChain == 0) {
2495                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2496                status = NO_INIT;
2497                break;
2498            }
2499            strategy = dstChain->strategy();
2500        }
2501        if (reRegister) {
2502            AudioSystem::unregisterEffect(effect->id());
2503            AudioSystem::registerEffect(&effect->desc(),
2504                                        dstThread->id(),
2505                                        strategy,
2506                                        sessionId,
2507                                        effect->id());
2508            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2509        }
2510        effect = chain->getEffectFromId_l(0);
2511    }
2512
2513    if (status != NO_ERROR) {
2514        for (size_t i = 0; i < removed.size(); i++) {
2515            srcThread->addEffect_l(removed[i]);
2516            if (dstChain != 0 && reRegister) {
2517                AudioSystem::unregisterEffect(removed[i]->id());
2518                AudioSystem::registerEffect(&removed[i]->desc(),
2519                                            srcThread->id(),
2520                                            strategy,
2521                                            sessionId,
2522                                            removed[i]->id());
2523                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2524            }
2525        }
2526    }
2527
2528    return status;
2529}
2530
2531bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2532{
2533    if (mGlobalEffectEnableTime != 0 &&
2534            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2535        return true;
2536    }
2537
2538    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2539        sp<EffectChain> ec =
2540                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2541        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2542            return true;
2543        }
2544    }
2545    return false;
2546}
2547
2548void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2549{
2550    Mutex::Autolock _l(mLock);
2551
2552    mGlobalEffectEnableTime = systemTime();
2553
2554    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2555        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2556        if (t->mType == ThreadBase::OFFLOAD) {
2557            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2558        }
2559    }
2560
2561}
2562
2563struct Entry {
2564#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2565    char mName[MAX_NAME];
2566};
2567
2568int comparEntry(const void *p1, const void *p2)
2569{
2570    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2571}
2572
2573#ifdef TEE_SINK
2574void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2575{
2576    NBAIO_Source *teeSource = source.get();
2577    if (teeSource != NULL) {
2578        // .wav rotation
2579        // There is a benign race condition if 2 threads call this simultaneously.
2580        // They would both traverse the directory, but the result would simply be
2581        // failures at unlink() which are ignored.  It's also unlikely since
2582        // normally dumpsys is only done by bugreport or from the command line.
2583        char teePath[32+256];
2584        strcpy(teePath, "/data/misc/media");
2585        size_t teePathLen = strlen(teePath);
2586        DIR *dir = opendir(teePath);
2587        teePath[teePathLen++] = '/';
2588        if (dir != NULL) {
2589#define MAX_SORT 20 // number of entries to sort
2590#define MAX_KEEP 10 // number of entries to keep
2591            struct Entry entries[MAX_SORT];
2592            size_t entryCount = 0;
2593            while (entryCount < MAX_SORT) {
2594                struct dirent de;
2595                struct dirent *result = NULL;
2596                int rc = readdir_r(dir, &de, &result);
2597                if (rc != 0) {
2598                    ALOGW("readdir_r failed %d", rc);
2599                    break;
2600                }
2601                if (result == NULL) {
2602                    break;
2603                }
2604                if (result != &de) {
2605                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2606                    break;
2607                }
2608                // ignore non .wav file entries
2609                size_t nameLen = strlen(de.d_name);
2610                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2611                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2612                    continue;
2613                }
2614                strcpy(entries[entryCount++].mName, de.d_name);
2615            }
2616            (void) closedir(dir);
2617            if (entryCount > MAX_KEEP) {
2618                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2619                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2620                    strcpy(&teePath[teePathLen], entries[i].mName);
2621                    (void) unlink(teePath);
2622                }
2623            }
2624        } else {
2625            if (fd >= 0) {
2626                dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2627            }
2628        }
2629        char teeTime[16];
2630        struct timeval tv;
2631        gettimeofday(&tv, NULL);
2632        struct tm tm;
2633        localtime_r(&tv.tv_sec, &tm);
2634        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2635        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2636        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2637        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2638        if (teeFd >= 0) {
2639            char wavHeader[44];
2640            memcpy(wavHeader,
2641                "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0",
2642                sizeof(wavHeader));
2643            NBAIO_Format format = teeSource->format();
2644            unsigned channelCount = Format_channelCount(format);
2645            ALOG_ASSERT(channelCount <= FCC_2);
2646            uint32_t sampleRate = Format_sampleRate(format);
2647            wavHeader[22] = channelCount;       // number of channels
2648            wavHeader[24] = sampleRate;         // sample rate
2649            wavHeader[25] = sampleRate >> 8;
2650            wavHeader[32] = channelCount * 2;   // block alignment
2651            write(teeFd, wavHeader, sizeof(wavHeader));
2652            size_t total = 0;
2653            bool firstRead = true;
2654            for (;;) {
2655#define TEE_SINK_READ 1024
2656                short buffer[TEE_SINK_READ * FCC_2];
2657                size_t count = TEE_SINK_READ;
2658                ssize_t actual = teeSource->read(buffer, count,
2659                        AudioBufferProvider::kInvalidPTS);
2660                bool wasFirstRead = firstRead;
2661                firstRead = false;
2662                if (actual <= 0) {
2663                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2664                        continue;
2665                    }
2666                    break;
2667                }
2668                ALOG_ASSERT(actual <= (ssize_t)count);
2669                write(teeFd, buffer, actual * channelCount * sizeof(short));
2670                total += actual;
2671            }
2672            lseek(teeFd, (off_t) 4, SEEK_SET);
2673            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2674            write(teeFd, &temp, sizeof(temp));
2675            lseek(teeFd, (off_t) 40, SEEK_SET);
2676            temp =  total * channelCount * sizeof(short);
2677            write(teeFd, &temp, sizeof(temp));
2678            close(teeFd);
2679            if (fd >= 0) {
2680                dprintf(fd, "tee copied to %s\n", teePath);
2681            }
2682        } else {
2683            if (fd >= 0) {
2684                dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2685            }
2686        }
2687    }
2688}
2689#endif
2690
2691// ----------------------------------------------------------------------------
2692
2693status_t AudioFlinger::onTransact(
2694        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2695{
2696    return BnAudioFlinger::onTransact(code, data, reply, flags);
2697}
2698
2699}; // namespace android
2700