AudioFlinger.cpp revision 5fb71f62474c92206dd068b725dab0572c9ede95 
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 <memunreachable/memunreachable.h>
35#include <utils/String16.h>
36#include <utils/threads.h>
37#include <utils/Atomic.h>
38
39#include <cutils/bitops.h>
40#include <cutils/properties.h>
41
42#include <system/audio.h>
43#include <hardware/audio.h>
44
45#include "AudioMixer.h"
46#include "AudioFlinger.h"
47#include "ServiceUtilities.h"
48
49#include <media/AudioResamplerPublic.h>
50
51#include <media/EffectsFactoryApi.h>
52#include <audio_effects/effect_visualizer.h>
53#include <audio_effects/effect_ns.h>
54#include <audio_effects/effect_aec.h>
55
56#include <audio_utils/primitives.h>
57
58#include <powermanager/PowerManager.h>
59
60#include <media/IMediaLogService.h>
61
62#include <media/nbaio/Pipe.h>
63#include <media/nbaio/PipeReader.h>
64#include <media/AudioParameter.h>
65#include <mediautils/BatteryNotifier.h>
66#include <private/android_filesystem_config.h>
67
68// ----------------------------------------------------------------------------
69
70// Note: the following macro is used for extremely verbose logging message.  In
71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
72// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
73// are so verbose that we want to suppress them even when we have ALOG_ASSERT
74// turned on.  Do not uncomment the #def below unless you really know what you
75// are doing and want to see all of the extremely verbose messages.
76//#define VERY_VERY_VERBOSE_LOGGING
77#ifdef VERY_VERY_VERBOSE_LOGGING
78#define ALOGVV ALOGV
79#else
80#define ALOGVV(a...) do { } while(0)
81#endif
82
83namespace android {
84
85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
86static const char kHardwareLockedString[] = "Hardware lock is taken\n";
87static const char kClientLockedString[] = "Client lock is taken\n";
88
89
90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
91
92uint32_t AudioFlinger::mScreenState;
93
94#ifdef TEE_SINK
95bool AudioFlinger::mTeeSinkInputEnabled = false;
96bool AudioFlinger::mTeeSinkOutputEnabled = false;
97bool AudioFlinger::mTeeSinkTrackEnabled = false;
98
99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
102#endif
103
104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
105// we define a minimum time during which a global effect is considered enabled.
106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
107
108// ----------------------------------------------------------------------------
109
110const char *formatToString(audio_format_t format) {
111    switch (audio_get_main_format(format)) {
112    case AUDIO_FORMAT_PCM:
113        switch (format) {
114        case AUDIO_FORMAT_PCM_16_BIT: return "pcm16";
115        case AUDIO_FORMAT_PCM_8_BIT: return "pcm8";
116        case AUDIO_FORMAT_PCM_32_BIT: return "pcm32";
117        case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24";
118        case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat";
119        case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24";
120        default:
121            break;
122        }
123        break;
124    case AUDIO_FORMAT_MP3: return "mp3";
125    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
126    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
127    case AUDIO_FORMAT_AAC: return "aac";
128    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
129    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
130    case AUDIO_FORMAT_VORBIS: return "vorbis";
131    case AUDIO_FORMAT_OPUS: return "opus";
132    case AUDIO_FORMAT_AC3: return "ac-3";
133    case AUDIO_FORMAT_E_AC3: return "e-ac-3";
134    case AUDIO_FORMAT_IEC61937: return "iec61937";
135    default:
136        break;
137    }
138    return "unknown";
139}
140
141static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
142{
143    const hw_module_t *mod;
144    int rc;
145
146    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
147    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
148                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
149    if (rc) {
150        goto out;
151    }
152    rc = audio_hw_device_open(mod, dev);
153    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
154                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
155    if (rc) {
156        goto out;
157    }
158    if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
159        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
160        rc = BAD_VALUE;
161        goto out;
162    }
163    return 0;
164
165out:
166    *dev = NULL;
167    return rc;
168}
169
170// ----------------------------------------------------------------------------
171
172AudioFlinger::AudioFlinger()
173    : BnAudioFlinger(),
174      mPrimaryHardwareDev(NULL),
175      mAudioHwDevs(NULL),
176      mHardwareStatus(AUDIO_HW_IDLE),
177      mMasterVolume(1.0f),
178      mMasterMute(false),
179      // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX),
180      mMode(AUDIO_MODE_INVALID),
181      mBtNrecIsOff(false),
182      mIsLowRamDevice(true),
183      mIsDeviceTypeKnown(false),
184      mGlobalEffectEnableTime(0),
185      mSystemReady(false)
186{
187    // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum
188    for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) {
189        // zero ID has a special meaning, so unavailable
190        mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX;
191    }
192
193    getpid_cached = getpid();
194    const bool doLog = property_get_bool("ro.test_harness", false);
195    if (doLog) {
196        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
197                MemoryHeapBase::READ_ONLY);
198    }
199
200    // reset battery stats.
201    // if the audio service has crashed, battery stats could be left
202    // in bad state, reset the state upon service start.
203    BatteryNotifier::getInstance().noteResetAudio();
204
205#ifdef TEE_SINK
206    char value[PROPERTY_VALUE_MAX];
207    (void) property_get("ro.debuggable", value, "0");
208    int debuggable = atoi(value);
209    int teeEnabled = 0;
210    if (debuggable) {
211        (void) property_get("af.tee", value, "0");
212        teeEnabled = atoi(value);
213    }
214    // FIXME symbolic constants here
215    if (teeEnabled & 1) {
216        mTeeSinkInputEnabled = true;
217    }
218    if (teeEnabled & 2) {
219        mTeeSinkOutputEnabled = true;
220    }
221    if (teeEnabled & 4) {
222        mTeeSinkTrackEnabled = true;
223    }
224#endif
225}
226
227void AudioFlinger::onFirstRef()
228{
229    Mutex::Autolock _l(mLock);
230
231    /* TODO: move all this work into an Init() function */
232    char val_str[PROPERTY_VALUE_MAX] = { 0 };
233    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
234        uint32_t int_val;
235        if (1 == sscanf(val_str, "%u", &int_val)) {
236            mStandbyTimeInNsecs = milliseconds(int_val);
237            ALOGI("Using %u mSec as standby time.", int_val);
238        } else {
239            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
240            ALOGI("Using default %u mSec as standby time.",
241                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
242        }
243    }
244
245    mPatchPanel = new PatchPanel(this);
246
247    mMode = AUDIO_MODE_NORMAL;
248}
249
250AudioFlinger::~AudioFlinger()
251{
252    while (!mRecordThreads.isEmpty()) {
253        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
254        closeInput_nonvirtual(mRecordThreads.keyAt(0));
255    }
256    while (!mPlaybackThreads.isEmpty()) {
257        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
258        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
259    }
260
261    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
262        // no mHardwareLock needed, as there are no other references to this
263        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
264        delete mAudioHwDevs.valueAt(i);
265    }
266
267    // Tell media.log service about any old writers that still need to be unregistered
268    if (mLogMemoryDealer != 0) {
269        sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
270        if (binder != 0) {
271            sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
272            for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
273                sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
274                mUnregisteredWriters.pop();
275                mediaLogService->unregisterWriter(iMemory);
276            }
277        }
278    }
279}
280
281static const char * const audio_interfaces[] = {
282    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
283    AUDIO_HARDWARE_MODULE_ID_A2DP,
284    AUDIO_HARDWARE_MODULE_ID_USB,
285};
286#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
287
288AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
289        audio_module_handle_t module,
290        audio_devices_t devices)
291{
292    // if module is 0, the request comes from an old policy manager and we should load
293    // well known modules
294    if (module == 0) {
295        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
296        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
297            loadHwModule_l(audio_interfaces[i]);
298        }
299        // then try to find a module supporting the requested device.
300        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
301            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
302            audio_hw_device_t *dev = audioHwDevice->hwDevice();
303            if ((dev->get_supported_devices != NULL) &&
304                    (dev->get_supported_devices(dev) & devices) == devices)
305                return audioHwDevice;
306        }
307    } else {
308        // check a match for the requested module handle
309        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
310        if (audioHwDevice != NULL) {
311            return audioHwDevice;
312        }
313    }
314
315    return NULL;
316}
317
318void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
319{
320    const size_t SIZE = 256;
321    char buffer[SIZE];
322    String8 result;
323
324    result.append("Clients:\n");
325    for (size_t i = 0; i < mClients.size(); ++i) {
326        sp<Client> client = mClients.valueAt(i).promote();
327        if (client != 0) {
328            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
329            result.append(buffer);
330        }
331    }
332
333    result.append("Notification Clients:\n");
334    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
335        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
336        result.append(buffer);
337    }
338
339    result.append("Global session refs:\n");
340    result.append("  session   pid count\n");
341    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
342        AudioSessionRef *r = mAudioSessionRefs[i];
343        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
344        result.append(buffer);
345    }
346    write(fd, result.string(), result.size());
347}
348
349
350void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
351{
352    const size_t SIZE = 256;
353    char buffer[SIZE];
354    String8 result;
355    hardware_call_state hardwareStatus = mHardwareStatus;
356
357    snprintf(buffer, SIZE, "Hardware status: %d\n"
358                           "Standby Time mSec: %u\n",
359                            hardwareStatus,
360                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
361    result.append(buffer);
362    write(fd, result.string(), result.size());
363}
364
365void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
366{
367    const size_t SIZE = 256;
368    char buffer[SIZE];
369    String8 result;
370    snprintf(buffer, SIZE, "Permission Denial: "
371            "can't dump AudioFlinger from pid=%d, uid=%d\n",
372            IPCThreadState::self()->getCallingPid(),
373            IPCThreadState::self()->getCallingUid());
374    result.append(buffer);
375    write(fd, result.string(), result.size());
376}
377
378bool AudioFlinger::dumpTryLock(Mutex& mutex)
379{
380    bool locked = false;
381    for (int i = 0; i < kDumpLockRetries; ++i) {
382        if (mutex.tryLock() == NO_ERROR) {
383            locked = true;
384            break;
385        }
386        usleep(kDumpLockSleepUs);
387    }
388    return locked;
389}
390
391status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
392{
393    if (!dumpAllowed()) {
394        dumpPermissionDenial(fd, args);
395    } else {
396        // get state of hardware lock
397        bool hardwareLocked = dumpTryLock(mHardwareLock);
398        if (!hardwareLocked) {
399            String8 result(kHardwareLockedString);
400            write(fd, result.string(), result.size());
401        } else {
402            mHardwareLock.unlock();
403        }
404
405        bool locked = dumpTryLock(mLock);
406
407        // failed to lock - AudioFlinger is probably deadlocked
408        if (!locked) {
409            String8 result(kDeadlockedString);
410            write(fd, result.string(), result.size());
411        }
412
413        bool clientLocked = dumpTryLock(mClientLock);
414        if (!clientLocked) {
415            String8 result(kClientLockedString);
416            write(fd, result.string(), result.size());
417        }
418
419        EffectDumpEffects(fd);
420
421        dumpClients(fd, args);
422        if (clientLocked) {
423            mClientLock.unlock();
424        }
425
426        dumpInternals(fd, args);
427
428        // dump playback threads
429        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
430            mPlaybackThreads.valueAt(i)->dump(fd, args);
431        }
432
433        // dump record threads
434        for (size_t i = 0; i < mRecordThreads.size(); i++) {
435            mRecordThreads.valueAt(i)->dump(fd, args);
436        }
437
438        // dump orphan effect chains
439        if (mOrphanEffectChains.size() != 0) {
440            write(fd, "  Orphan Effect Chains\n", strlen("  Orphan Effect Chains\n"));
441            for (size_t i = 0; i < mOrphanEffectChains.size(); i++) {
442                mOrphanEffectChains.valueAt(i)->dump(fd, args);
443            }
444        }
445        // dump all hardware devs
446        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
447            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
448            dev->dump(dev, fd);
449        }
450
451#ifdef TEE_SINK
452        // dump the serially shared record tee sink
453        if (mRecordTeeSource != 0) {
454            dumpTee(fd, mRecordTeeSource);
455        }
456#endif
457
458        if (locked) {
459            mLock.unlock();
460        }
461
462        // append a copy of media.log here by forwarding fd to it, but don't attempt
463        // to lookup the service if it's not running, as it will block for a second
464        if (mLogMemoryDealer != 0) {
465            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
466            if (binder != 0) {
467                dprintf(fd, "\nmedia.log:\n");
468                Vector<String16> args;
469                binder->dump(fd, args);
470            }
471        }
472
473        // check for optional arguments
474        bool unreachableMemory = false;
475        for (const auto &arg : args) {
476            if (arg == String16("--unreachable")) {
477                unreachableMemory = true;
478            }
479        }
480
481        if (unreachableMemory) {
482            dprintf(fd, "\nDumping unreachable memory:\n");
483            // TODO - should limit be an argument parameter?
484            std::string s = GetUnreachableMemoryString(true /* contents */, 10000 /* limit */);
485            write(fd, s.c_str(), s.size());
486        }
487    }
488    return NO_ERROR;
489}
490
491sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
492{
493    Mutex::Autolock _cl(mClientLock);
494    // If pid is already in the mClients wp<> map, then use that entry
495    // (for which promote() is always != 0), otherwise create a new entry and Client.
496    sp<Client> client = mClients.valueFor(pid).promote();
497    if (client == 0) {
498        client = new Client(this, pid);
499        mClients.add(pid, client);
500    }
501
502    return client;
503}
504
505sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
506{
507    // If there is no memory allocated for logs, return a dummy writer that does nothing
508    if (mLogMemoryDealer == 0) {
509        return new NBLog::Writer();
510    }
511    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
512    // Similarly if we can't contact the media.log service, also return a dummy writer
513    if (binder == 0) {
514        return new NBLog::Writer();
515    }
516    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
517    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
518    // If allocation fails, consult the vector of previously unregistered writers
519    // and garbage-collect one or more them until an allocation succeeds
520    if (shared == 0) {
521        Mutex::Autolock _l(mUnregisteredWritersLock);
522        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
523            {
524                // Pick the oldest stale writer to garbage-collect
525                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
526                mUnregisteredWriters.removeAt(0);
527                mediaLogService->unregisterWriter(iMemory);
528                // Now the media.log remote reference to IMemory is gone.  When our last local
529                // reference to IMemory also drops to zero at end of this block,
530                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
531            }
532            // Re-attempt the allocation
533            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
534            if (shared != 0) {
535                goto success;
536            }
537        }
538        // Even after garbage-collecting all old writers, there is still not enough memory,
539        // so return a dummy writer
540        return new NBLog::Writer();
541    }
542success:
543    mediaLogService->registerWriter(shared, size, name);
544    return new NBLog::Writer(size, shared);
545}
546
547void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
548{
549    if (writer == 0) {
550        return;
551    }
552    sp<IMemory> iMemory(writer->getIMemory());
553    if (iMemory == 0) {
554        return;
555    }
556    // Rather than removing the writer immediately, append it to a queue of old writers to
557    // be garbage-collected later.  This allows us to continue to view old logs for a while.
558    Mutex::Autolock _l(mUnregisteredWritersLock);
559    mUnregisteredWriters.push(writer);
560}
561
562// IAudioFlinger interface
563
564
565sp<IAudioTrack> AudioFlinger::createTrack(
566        audio_stream_type_t streamType,
567        uint32_t sampleRate,
568        audio_format_t format,
569        audio_channel_mask_t channelMask,
570        size_t *frameCount,
571        IAudioFlinger::track_flags_t *flags,
572        const sp<IMemory>& sharedBuffer,
573        audio_io_handle_t output,
574        pid_t tid,
575        audio_session_t *sessionId,
576        int clientUid,
577        status_t *status)
578{
579    sp<PlaybackThread::Track> track;
580    sp<TrackHandle> trackHandle;
581    sp<Client> client;
582    status_t lStatus;
583    audio_session_t lSessionId;
584
585    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
586    // but if someone uses binder directly they could bypass that and cause us to crash
587    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
588        ALOGE("createTrack() invalid stream type %d", streamType);
589        lStatus = BAD_VALUE;
590        goto Exit;
591    }
592
593    // further sample rate checks are performed by createTrack_l() depending on the thread type
594    if (sampleRate == 0) {
595        ALOGE("createTrack() invalid sample rate %u", sampleRate);
596        lStatus = BAD_VALUE;
597        goto Exit;
598    }
599
600    // further channel mask checks are performed by createTrack_l() depending on the thread type
601    if (!audio_is_output_channel(channelMask)) {
602        ALOGE("createTrack() invalid channel mask %#x", channelMask);
603        lStatus = BAD_VALUE;
604        goto Exit;
605    }
606
607    // further format checks are performed by createTrack_l() depending on the thread type
608    if (!audio_is_valid_format(format)) {
609        ALOGE("createTrack() invalid format %#x", format);
610        lStatus = BAD_VALUE;
611        goto Exit;
612    }
613
614    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
615        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
616        lStatus = BAD_VALUE;
617        goto Exit;
618    }
619
620    {
621        Mutex::Autolock _l(mLock);
622        PlaybackThread *thread = checkPlaybackThread_l(output);
623        if (thread == NULL) {
624            ALOGE("no playback thread found for output handle %d", output);
625            lStatus = BAD_VALUE;
626            goto Exit;
627        }
628
629        pid_t pid = IPCThreadState::self()->getCallingPid();
630        client = registerPid(pid);
631
632        PlaybackThread *effectThread = NULL;
633        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
634            if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
635                ALOGE("createTrack() invalid session ID %d", *sessionId);
636                lStatus = BAD_VALUE;
637                goto Exit;
638            }
639            lSessionId = *sessionId;
640            // check if an effect chain with the same session ID is present on another
641            // output thread and move it here.
642            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
643                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
644                if (mPlaybackThreads.keyAt(i) != output) {
645                    uint32_t sessions = t->hasAudioSession(lSessionId);
646                    if (sessions & PlaybackThread::EFFECT_SESSION) {
647                        effectThread = t.get();
648                        break;
649                    }
650                }
651            }
652        } else {
653            // if no audio session id is provided, create one here
654            lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
655            if (sessionId != NULL) {
656                *sessionId = lSessionId;
657            }
658        }
659        ALOGV("createTrack() lSessionId: %d", lSessionId);
660
661        track = thread->createTrack_l(client, streamType, sampleRate, format,
662                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
663        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
664        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
665
666        // move effect chain to this output thread if an effect on same session was waiting
667        // for a track to be created
668        if (lStatus == NO_ERROR && effectThread != NULL) {
669            // no risk of deadlock because AudioFlinger::mLock is held
670            Mutex::Autolock _dl(thread->mLock);
671            Mutex::Autolock _sl(effectThread->mLock);
672            moveEffectChain_l(lSessionId, effectThread, thread, true);
673        }
674
675        // Look for sync events awaiting for a session to be used.
676        for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
677            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
678                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
679                    if (lStatus == NO_ERROR) {
680                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
681                    } else {
682                        mPendingSyncEvents[i]->cancel();
683                    }
684                    mPendingSyncEvents.removeAt(i);
685                    i--;
686                }
687            }
688        }
689
690        setAudioHwSyncForSession_l(thread, lSessionId);
691    }
692
693    if (lStatus != NO_ERROR) {
694        // remove local strong reference to Client before deleting the Track so that the
695        // Client destructor is called by the TrackBase destructor with mClientLock held
696        // Don't hold mClientLock when releasing the reference on the track as the
697        // destructor will acquire it.
698        {
699            Mutex::Autolock _cl(mClientLock);
700            client.clear();
701        }
702        track.clear();
703        goto Exit;
704    }
705
706    // return handle to client
707    trackHandle = new TrackHandle(track);
708
709Exit:
710    *status = lStatus;
711    return trackHandle;
712}
713
714uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const
715{
716    Mutex::Autolock _l(mLock);
717    ThreadBase *thread = checkThread_l(ioHandle);
718    if (thread == NULL) {
719        ALOGW("sampleRate() unknown thread %d", ioHandle);
720        return 0;
721    }
722    return thread->sampleRate();
723}
724
725audio_format_t AudioFlinger::format(audio_io_handle_t output) const
726{
727    Mutex::Autolock _l(mLock);
728    PlaybackThread *thread = checkPlaybackThread_l(output);
729    if (thread == NULL) {
730        ALOGW("format() unknown thread %d", output);
731        return AUDIO_FORMAT_INVALID;
732    }
733    return thread->format();
734}
735
736size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const
737{
738    Mutex::Autolock _l(mLock);
739    ThreadBase *thread = checkThread_l(ioHandle);
740    if (thread == NULL) {
741        ALOGW("frameCount() unknown thread %d", ioHandle);
742        return 0;
743    }
744    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
745    //       should examine all callers and fix them to handle smaller counts
746    return thread->frameCount();
747}
748
749size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const
750{
751    Mutex::Autolock _l(mLock);
752    ThreadBase *thread = checkThread_l(ioHandle);
753    if (thread == NULL) {
754        ALOGW("frameCountHAL() unknown thread %d", ioHandle);
755        return 0;
756    }
757    return thread->frameCountHAL();
758}
759
760uint32_t AudioFlinger::latency(audio_io_handle_t output) const
761{
762    Mutex::Autolock _l(mLock);
763    PlaybackThread *thread = checkPlaybackThread_l(output);
764    if (thread == NULL) {
765        ALOGW("latency(): no playback thread found for output handle %d", output);
766        return 0;
767    }
768    return thread->latency();
769}
770
771status_t AudioFlinger::setMasterVolume(float value)
772{
773    status_t ret = initCheck();
774    if (ret != NO_ERROR) {
775        return ret;
776    }
777
778    // check calling permissions
779    if (!settingsAllowed()) {
780        return PERMISSION_DENIED;
781    }
782
783    Mutex::Autolock _l(mLock);
784    mMasterVolume = value;
785
786    // Set master volume in the HALs which support it.
787    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
788        AutoMutex lock(mHardwareLock);
789        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
790
791        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
792        if (dev->canSetMasterVolume()) {
793            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
794        }
795        mHardwareStatus = AUDIO_HW_IDLE;
796    }
797
798    // Now set the master volume in each playback thread.  Playback threads
799    // assigned to HALs which do not have master volume support will apply
800    // master volume during the mix operation.  Threads with HALs which do
801    // support master volume will simply ignore the setting.
802    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
803        if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
804            continue;
805        }
806        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
807    }
808
809    return NO_ERROR;
810}
811
812status_t AudioFlinger::setMode(audio_mode_t mode)
813{
814    status_t ret = initCheck();
815    if (ret != NO_ERROR) {
816        return ret;
817    }
818
819    // check calling permissions
820    if (!settingsAllowed()) {
821        return PERMISSION_DENIED;
822    }
823    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
824        ALOGW("Illegal value: setMode(%d)", mode);
825        return BAD_VALUE;
826    }
827
828    { // scope for the lock
829        AutoMutex lock(mHardwareLock);
830        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
831        mHardwareStatus = AUDIO_HW_SET_MODE;
832        ret = dev->set_mode(dev, mode);
833        mHardwareStatus = AUDIO_HW_IDLE;
834    }
835
836    if (NO_ERROR == ret) {
837        Mutex::Autolock _l(mLock);
838        mMode = mode;
839        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
840            mPlaybackThreads.valueAt(i)->setMode(mode);
841    }
842
843    return ret;
844}
845
846status_t AudioFlinger::setMicMute(bool state)
847{
848    status_t ret = initCheck();
849    if (ret != NO_ERROR) {
850        return ret;
851    }
852
853    // check calling permissions
854    if (!settingsAllowed()) {
855        return PERMISSION_DENIED;
856    }
857
858    AutoMutex lock(mHardwareLock);
859    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
860    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
861        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
862        status_t result = dev->set_mic_mute(dev, state);
863        if (result != NO_ERROR) {
864            ret = result;
865        }
866    }
867    mHardwareStatus = AUDIO_HW_IDLE;
868    return ret;
869}
870
871bool AudioFlinger::getMicMute() const
872{
873    status_t ret = initCheck();
874    if (ret != NO_ERROR) {
875        return false;
876    }
877    bool mute = true;
878    bool state = AUDIO_MODE_INVALID;
879    AutoMutex lock(mHardwareLock);
880    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
881    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
882        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
883        status_t result = dev->get_mic_mute(dev, &state);
884        if (result == NO_ERROR) {
885            mute = mute && state;
886        }
887    }
888    mHardwareStatus = AUDIO_HW_IDLE;
889
890    return mute;
891}
892
893status_t AudioFlinger::setMasterMute(bool muted)
894{
895    status_t ret = initCheck();
896    if (ret != NO_ERROR) {
897        return ret;
898    }
899
900    // check calling permissions
901    if (!settingsAllowed()) {
902        return PERMISSION_DENIED;
903    }
904
905    Mutex::Autolock _l(mLock);
906    mMasterMute = muted;
907
908    // Set master mute in the HALs which support it.
909    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
910        AutoMutex lock(mHardwareLock);
911        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
912
913        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
914        if (dev->canSetMasterMute()) {
915            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
916        }
917        mHardwareStatus = AUDIO_HW_IDLE;
918    }
919
920    // Now set the master mute in each playback thread.  Playback threads
921    // assigned to HALs which do not have master mute support will apply master
922    // mute during the mix operation.  Threads with HALs which do support master
923    // mute will simply ignore the setting.
924    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
925        if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
926            continue;
927        }
928        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
929    }
930
931    return NO_ERROR;
932}
933
934float AudioFlinger::masterVolume() const
935{
936    Mutex::Autolock _l(mLock);
937    return masterVolume_l();
938}
939
940bool AudioFlinger::masterMute() const
941{
942    Mutex::Autolock _l(mLock);
943    return masterMute_l();
944}
945
946float AudioFlinger::masterVolume_l() const
947{
948    return mMasterVolume;
949}
950
951bool AudioFlinger::masterMute_l() const
952{
953    return mMasterMute;
954}
955
956status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const
957{
958    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
959        ALOGW("setStreamVolume() invalid stream %d", stream);
960        return BAD_VALUE;
961    }
962    pid_t caller = IPCThreadState::self()->getCallingPid();
963    if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) {
964        ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream);
965        return PERMISSION_DENIED;
966    }
967
968    return NO_ERROR;
969}
970
971status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
972        audio_io_handle_t output)
973{
974    // check calling permissions
975    if (!settingsAllowed()) {
976        return PERMISSION_DENIED;
977    }
978
979    status_t status = checkStreamType(stream);
980    if (status != NO_ERROR) {
981        return status;
982    }
983    ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume");
984
985    AutoMutex lock(mLock);
986    PlaybackThread *thread = NULL;
987    if (output != AUDIO_IO_HANDLE_NONE) {
988        thread = checkPlaybackThread_l(output);
989        if (thread == NULL) {
990            return BAD_VALUE;
991        }
992    }
993
994    mStreamTypes[stream].volume = value;
995
996    if (thread == NULL) {
997        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
998            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
999        }
1000    } else {
1001        thread->setStreamVolume(stream, value);
1002    }
1003
1004    return NO_ERROR;
1005}
1006
1007status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
1008{
1009    // check calling permissions
1010    if (!settingsAllowed()) {
1011        return PERMISSION_DENIED;
1012    }
1013
1014    status_t status = checkStreamType(stream);
1015    if (status != NO_ERROR) {
1016        return status;
1017    }
1018    ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH");
1019
1020    if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
1021        ALOGE("setStreamMute() invalid stream %d", stream);
1022        return BAD_VALUE;
1023    }
1024
1025    AutoMutex lock(mLock);
1026    mStreamTypes[stream].mute = muted;
1027    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
1028        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
1029
1030    return NO_ERROR;
1031}
1032
1033float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
1034{
1035    status_t status = checkStreamType(stream);
1036    if (status != NO_ERROR) {
1037        return 0.0f;
1038    }
1039
1040    AutoMutex lock(mLock);
1041    float volume;
1042    if (output != AUDIO_IO_HANDLE_NONE) {
1043        PlaybackThread *thread = checkPlaybackThread_l(output);
1044        if (thread == NULL) {
1045            return 0.0f;
1046        }
1047        volume = thread->streamVolume(stream);
1048    } else {
1049        volume = streamVolume_l(stream);
1050    }
1051
1052    return volume;
1053}
1054
1055bool AudioFlinger::streamMute(audio_stream_type_t stream) const
1056{
1057    status_t status = checkStreamType(stream);
1058    if (status != NO_ERROR) {
1059        return true;
1060    }
1061
1062    AutoMutex lock(mLock);
1063    return streamMute_l(stream);
1064}
1065
1066
1067void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs)
1068{
1069    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1070        mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1071    }
1072}
1073
1074status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
1075{
1076    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
1077            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
1078
1079    // check calling permissions
1080    if (!settingsAllowed()) {
1081        return PERMISSION_DENIED;
1082    }
1083
1084    // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
1085    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1086        Mutex::Autolock _l(mLock);
1087        status_t final_result = NO_ERROR;
1088        {
1089            AutoMutex lock(mHardwareLock);
1090            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
1091            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1092                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1093                status_t result = dev->set_parameters(dev, keyValuePairs.string());
1094                final_result = result ?: final_result;
1095            }
1096            mHardwareStatus = AUDIO_HW_IDLE;
1097        }
1098        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
1099        AudioParameter param = AudioParameter(keyValuePairs);
1100        String8 value;
1101        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
1102            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
1103            if (mBtNrecIsOff != btNrecIsOff) {
1104                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1105                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
1106                    audio_devices_t device = thread->inDevice();
1107                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
1108                    // collect all of the thread's session IDs
1109                    KeyedVector<audio_session_t, bool> ids = thread->sessionIds();
1110                    // suspend effects associated with those session IDs
1111                    for (size_t j = 0; j < ids.size(); ++j) {
1112                        audio_session_t sessionId = ids.keyAt(j);
1113                        thread->setEffectSuspended(FX_IID_AEC,
1114                                                   suspend,
1115                                                   sessionId);
1116                        thread->setEffectSuspended(FX_IID_NS,
1117                                                   suspend,
1118                                                   sessionId);
1119                    }
1120                }
1121                mBtNrecIsOff = btNrecIsOff;
1122            }
1123        }
1124        String8 screenState;
1125        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1126            bool isOff = screenState == "off";
1127            if (isOff != (AudioFlinger::mScreenState & 1)) {
1128                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1129            }
1130        }
1131        return final_result;
1132    }
1133
1134    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1135    // and the thread is exited once the lock is released
1136    sp<ThreadBase> thread;
1137    {
1138        Mutex::Autolock _l(mLock);
1139        thread = checkPlaybackThread_l(ioHandle);
1140        if (thread == 0) {
1141            thread = checkRecordThread_l(ioHandle);
1142        } else if (thread == primaryPlaybackThread_l()) {
1143            // indicate output device change to all input threads for pre processing
1144            AudioParameter param = AudioParameter(keyValuePairs);
1145            int value;
1146            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1147                    (value != 0)) {
1148                broacastParametersToRecordThreads_l(keyValuePairs);
1149            }
1150        }
1151    }
1152    if (thread != 0) {
1153        return thread->setParameters(keyValuePairs);
1154    }
1155    return BAD_VALUE;
1156}
1157
1158String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1159{
1160    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1161            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1162
1163    Mutex::Autolock _l(mLock);
1164
1165    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1166        String8 out_s8;
1167
1168        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1169            char *s;
1170            {
1171            AutoMutex lock(mHardwareLock);
1172            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1173            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1174            s = dev->get_parameters(dev, keys.string());
1175            mHardwareStatus = AUDIO_HW_IDLE;
1176            }
1177            out_s8 += String8(s ? s : "");
1178            free(s);
1179        }
1180        return out_s8;
1181    }
1182
1183    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1184    if (playbackThread != NULL) {
1185        return playbackThread->getParameters(keys);
1186    }
1187    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1188    if (recordThread != NULL) {
1189        return recordThread->getParameters(keys);
1190    }
1191    return String8("");
1192}
1193
1194size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1195        audio_channel_mask_t channelMask) const
1196{
1197    status_t ret = initCheck();
1198    if (ret != NO_ERROR) {
1199        return 0;
1200    }
1201    if ((sampleRate == 0) ||
1202            !audio_is_valid_format(format) || !audio_has_proportional_frames(format) ||
1203            !audio_is_input_channel(channelMask)) {
1204        return 0;
1205    }
1206
1207    AutoMutex lock(mHardwareLock);
1208    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1209    audio_config_t config, proposed;
1210    memset(&proposed, 0, sizeof(proposed));
1211    proposed.sample_rate = sampleRate;
1212    proposed.channel_mask = channelMask;
1213    proposed.format = format;
1214
1215    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1216    size_t frames;
1217    for (;;) {
1218        // Note: config is currently a const parameter for get_input_buffer_size()
1219        // but we use a copy from proposed in case config changes from the call.
1220        config = proposed;
1221        frames = dev->get_input_buffer_size(dev, &config);
1222        if (frames != 0) {
1223            break; // hal success, config is the result
1224        }
1225        // change one parameter of the configuration each iteration to a more "common" value
1226        // to see if the device will support it.
1227        if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
1228            proposed.format = AUDIO_FORMAT_PCM_16_BIT;
1229        } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
1230            proposed.sample_rate = 44100;           // legacy AudioRecord.java. TODO: Query hw?
1231        } else {
1232            ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
1233                    "format %#x, channelMask 0x%X",
1234                    sampleRate, format, channelMask);
1235            break; // retries failed, break out of loop with frames == 0.
1236        }
1237    }
1238    mHardwareStatus = AUDIO_HW_IDLE;
1239    if (frames > 0 && config.sample_rate != sampleRate) {
1240        frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
1241    }
1242    return frames; // may be converted to bytes at the Java level.
1243}
1244
1245uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1246{
1247    Mutex::Autolock _l(mLock);
1248
1249    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1250    if (recordThread != NULL) {
1251        return recordThread->getInputFramesLost();
1252    }
1253    return 0;
1254}
1255
1256status_t AudioFlinger::setVoiceVolume(float value)
1257{
1258    status_t ret = initCheck();
1259    if (ret != NO_ERROR) {
1260        return ret;
1261    }
1262
1263    // check calling permissions
1264    if (!settingsAllowed()) {
1265        return PERMISSION_DENIED;
1266    }
1267
1268    AutoMutex lock(mHardwareLock);
1269    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1270    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1271    ret = dev->set_voice_volume(dev, value);
1272    mHardwareStatus = AUDIO_HW_IDLE;
1273
1274    return ret;
1275}
1276
1277status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1278        audio_io_handle_t output) const
1279{
1280    Mutex::Autolock _l(mLock);
1281
1282    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1283    if (playbackThread != NULL) {
1284        return playbackThread->getRenderPosition(halFrames, dspFrames);
1285    }
1286
1287    return BAD_VALUE;
1288}
1289
1290void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1291{
1292    Mutex::Autolock _l(mLock);
1293    if (client == 0) {
1294        return;
1295    }
1296    pid_t pid = IPCThreadState::self()->getCallingPid();
1297    {
1298        Mutex::Autolock _cl(mClientLock);
1299        if (mNotificationClients.indexOfKey(pid) < 0) {
1300            sp<NotificationClient> notificationClient = new NotificationClient(this,
1301                                                                                client,
1302                                                                                pid);
1303            ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1304
1305            mNotificationClients.add(pid, notificationClient);
1306
1307            sp<IBinder> binder = IInterface::asBinder(client);
1308            binder->linkToDeath(notificationClient);
1309        }
1310    }
1311
1312    // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
1313    // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock.
1314    // the config change is always sent from playback or record threads to avoid deadlock
1315    // with AudioSystem::gLock
1316    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1317        mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid);
1318    }
1319
1320    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1321        mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid);
1322    }
1323}
1324
1325void AudioFlinger::removeNotificationClient(pid_t pid)
1326{
1327    Mutex::Autolock _l(mLock);
1328    {
1329        Mutex::Autolock _cl(mClientLock);
1330        mNotificationClients.removeItem(pid);
1331    }
1332
1333    ALOGV("%d died, releasing its sessions", pid);
1334    size_t num = mAudioSessionRefs.size();
1335    bool removed = false;
1336    for (size_t i = 0; i< num; ) {
1337        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1338        ALOGV(" pid %d @ %zu", ref->mPid, i);
1339        if (ref->mPid == pid) {
1340            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1341            mAudioSessionRefs.removeAt(i);
1342            delete ref;
1343            removed = true;
1344            num--;
1345        } else {
1346            i++;
1347        }
1348    }
1349    if (removed) {
1350        purgeStaleEffects_l();
1351    }
1352}
1353
1354void AudioFlinger::ioConfigChanged(audio_io_config_event event,
1355                                   const sp<AudioIoDescriptor>& ioDesc,
1356                                   pid_t pid)
1357{
1358    Mutex::Autolock _l(mClientLock);
1359    size_t size = mNotificationClients.size();
1360    for (size_t i = 0; i < size; i++) {
1361        if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) {
1362            mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc);
1363        }
1364    }
1365}
1366
1367// removeClient_l() must be called with AudioFlinger::mClientLock held
1368void AudioFlinger::removeClient_l(pid_t pid)
1369{
1370    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1371            IPCThreadState::self()->getCallingPid());
1372    mClients.removeItem(pid);
1373}
1374
1375// getEffectThread_l() must be called with AudioFlinger::mLock held
1376sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
1377        int EffectId)
1378{
1379    sp<PlaybackThread> thread;
1380
1381    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1382        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1383            ALOG_ASSERT(thread == 0);
1384            thread = mPlaybackThreads.valueAt(i);
1385        }
1386    }
1387
1388    return thread;
1389}
1390
1391
1392
1393// ----------------------------------------------------------------------------
1394
1395AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1396    :   RefBase(),
1397        mAudioFlinger(audioFlinger),
1398        mPid(pid)
1399{
1400    size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0);
1401    heapSize *= 1024;
1402    if (!heapSize) {
1403        heapSize = kClientSharedHeapSizeBytes;
1404        // Increase heap size on non low ram devices to limit risk of reconnection failure for
1405        // invalidated tracks
1406        if (!audioFlinger->isLowRamDevice()) {
1407            heapSize *= kClientSharedHeapSizeMultiplier;
1408        }
1409    }
1410    mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client");
1411}
1412
1413// Client destructor must be called with AudioFlinger::mClientLock held
1414AudioFlinger::Client::~Client()
1415{
1416    mAudioFlinger->removeClient_l(mPid);
1417}
1418
1419sp<MemoryDealer> AudioFlinger::Client::heap() const
1420{
1421    return mMemoryDealer;
1422}
1423
1424// ----------------------------------------------------------------------------
1425
1426AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1427                                                     const sp<IAudioFlingerClient>& client,
1428                                                     pid_t pid)
1429    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1430{
1431}
1432
1433AudioFlinger::NotificationClient::~NotificationClient()
1434{
1435}
1436
1437void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1438{
1439    sp<NotificationClient> keep(this);
1440    mAudioFlinger->removeNotificationClient(mPid);
1441}
1442
1443
1444// ----------------------------------------------------------------------------
1445
1446sp<IAudioRecord> AudioFlinger::openRecord(
1447        audio_io_handle_t input,
1448        uint32_t sampleRate,
1449        audio_format_t format,
1450        audio_channel_mask_t channelMask,
1451        const String16& opPackageName,
1452        size_t *frameCount,
1453        IAudioFlinger::track_flags_t *flags,
1454        pid_t tid,
1455        int clientUid,
1456        audio_session_t *sessionId,
1457        size_t *notificationFrames,
1458        sp<IMemory>& cblk,
1459        sp<IMemory>& buffers,
1460        status_t *status)
1461{
1462    sp<RecordThread::RecordTrack> recordTrack;
1463    sp<RecordHandle> recordHandle;
1464    sp<Client> client;
1465    status_t lStatus;
1466    audio_session_t lSessionId;
1467
1468    cblk.clear();
1469    buffers.clear();
1470
1471    const uid_t callingUid = IPCThreadState::self()->getCallingUid();
1472    if (!isTrustedCallingUid(callingUid)) {
1473        ALOGW_IF((uid_t)clientUid != callingUid,
1474                "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid);
1475        clientUid = callingUid;
1476    }
1477
1478    // check calling permissions
1479    if (!recordingAllowed(opPackageName, tid, clientUid)) {
1480        ALOGE("openRecord() permission denied: recording not allowed");
1481        lStatus = PERMISSION_DENIED;
1482        goto Exit;
1483    }
1484
1485    // further sample rate checks are performed by createRecordTrack_l()
1486    if (sampleRate == 0) {
1487        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1488        lStatus = BAD_VALUE;
1489        goto Exit;
1490    }
1491
1492    // we don't yet support anything other than linear PCM
1493    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
1494        ALOGE("openRecord() invalid format %#x", format);
1495        lStatus = BAD_VALUE;
1496        goto Exit;
1497    }
1498
1499    // further channel mask checks are performed by createRecordTrack_l()
1500    if (!audio_is_input_channel(channelMask)) {
1501        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1502        lStatus = BAD_VALUE;
1503        goto Exit;
1504    }
1505
1506    {
1507        Mutex::Autolock _l(mLock);
1508        RecordThread *thread = checkRecordThread_l(input);
1509        if (thread == NULL) {
1510            ALOGE("openRecord() checkRecordThread_l failed");
1511            lStatus = BAD_VALUE;
1512            goto Exit;
1513        }
1514
1515        pid_t pid = IPCThreadState::self()->getCallingPid();
1516        client = registerPid(pid);
1517
1518        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1519            if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
1520                lStatus = BAD_VALUE;
1521                goto Exit;
1522            }
1523            lSessionId = *sessionId;
1524        } else {
1525            // if no audio session id is provided, create one here
1526            lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
1527            if (sessionId != NULL) {
1528                *sessionId = lSessionId;
1529            }
1530        }
1531        ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input);
1532
1533        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1534                                                  frameCount, lSessionId, notificationFrames,
1535                                                  clientUid, flags, tid, &lStatus);
1536        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1537
1538        if (lStatus == NO_ERROR) {
1539            // Check if one effect chain was awaiting for an AudioRecord to be created on this
1540            // session and move it to this thread.
1541            sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId);
1542            if (chain != 0) {
1543                Mutex::Autolock _l(thread->mLock);
1544                thread->addEffectChain_l(chain);
1545            }
1546        }
1547    }
1548
1549    if (lStatus != NO_ERROR) {
1550        // remove local strong reference to Client before deleting the RecordTrack so that the
1551        // Client destructor is called by the TrackBase destructor with mClientLock held
1552        // Don't hold mClientLock when releasing the reference on the track as the
1553        // destructor will acquire it.
1554        {
1555            Mutex::Autolock _cl(mClientLock);
1556            client.clear();
1557        }
1558        recordTrack.clear();
1559        goto Exit;
1560    }
1561
1562    cblk = recordTrack->getCblk();
1563    buffers = recordTrack->getBuffers();
1564
1565    // return handle to client
1566    recordHandle = new RecordHandle(recordTrack);
1567
1568Exit:
1569    *status = lStatus;
1570    return recordHandle;
1571}
1572
1573
1574
1575// ----------------------------------------------------------------------------
1576
1577audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1578{
1579    if (name == NULL) {
1580        return AUDIO_MODULE_HANDLE_NONE;
1581    }
1582    if (!settingsAllowed()) {
1583        return AUDIO_MODULE_HANDLE_NONE;
1584    }
1585    Mutex::Autolock _l(mLock);
1586    return loadHwModule_l(name);
1587}
1588
1589// loadHwModule_l() must be called with AudioFlinger::mLock held
1590audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1591{
1592    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1593        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1594            ALOGW("loadHwModule() module %s already loaded", name);
1595            return mAudioHwDevs.keyAt(i);
1596        }
1597    }
1598
1599    audio_hw_device_t *dev;
1600
1601    int rc = load_audio_interface(name, &dev);
1602    if (rc) {
1603        ALOGE("loadHwModule() error %d loading module %s", rc, name);
1604        return AUDIO_MODULE_HANDLE_NONE;
1605    }
1606
1607    mHardwareStatus = AUDIO_HW_INIT;
1608    rc = dev->init_check(dev);
1609    mHardwareStatus = AUDIO_HW_IDLE;
1610    if (rc) {
1611        ALOGE("loadHwModule() init check error %d for module %s", rc, name);
1612        return AUDIO_MODULE_HANDLE_NONE;
1613    }
1614
1615    // Check and cache this HAL's level of support for master mute and master
1616    // volume.  If this is the first HAL opened, and it supports the get
1617    // methods, use the initial values provided by the HAL as the current
1618    // master mute and volume settings.
1619
1620    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1621    {  // scope for auto-lock pattern
1622        AutoMutex lock(mHardwareLock);
1623
1624        if (0 == mAudioHwDevs.size()) {
1625            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1626            if (NULL != dev->get_master_volume) {
1627                float mv;
1628                if (OK == dev->get_master_volume(dev, &mv)) {
1629                    mMasterVolume = mv;
1630                }
1631            }
1632
1633            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1634            if (NULL != dev->get_master_mute) {
1635                bool mm;
1636                if (OK == dev->get_master_mute(dev, &mm)) {
1637                    mMasterMute = mm;
1638                }
1639            }
1640        }
1641
1642        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1643        if ((NULL != dev->set_master_volume) &&
1644            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1645            flags = static_cast<AudioHwDevice::Flags>(flags |
1646                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1647        }
1648
1649        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1650        if ((NULL != dev->set_master_mute) &&
1651            (OK == dev->set_master_mute(dev, mMasterMute))) {
1652            flags = static_cast<AudioHwDevice::Flags>(flags |
1653                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1654        }
1655
1656        mHardwareStatus = AUDIO_HW_IDLE;
1657    }
1658
1659    audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE);
1660    mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags));
1661
1662    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1663          name, dev->common.module->name, dev->common.module->id, handle);
1664
1665    return handle;
1666
1667}
1668
1669// ----------------------------------------------------------------------------
1670
1671uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1672{
1673    Mutex::Autolock _l(mLock);
1674    PlaybackThread *thread = primaryPlaybackThread_l();
1675    return thread != NULL ? thread->sampleRate() : 0;
1676}
1677
1678size_t AudioFlinger::getPrimaryOutputFrameCount()
1679{
1680    Mutex::Autolock _l(mLock);
1681    PlaybackThread *thread = primaryPlaybackThread_l();
1682    return thread != NULL ? thread->frameCountHAL() : 0;
1683}
1684
1685// ----------------------------------------------------------------------------
1686
1687status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1688{
1689    uid_t uid = IPCThreadState::self()->getCallingUid();
1690    if (uid != AID_SYSTEM) {
1691        return PERMISSION_DENIED;
1692    }
1693    Mutex::Autolock _l(mLock);
1694    if (mIsDeviceTypeKnown) {
1695        return INVALID_OPERATION;
1696    }
1697    mIsLowRamDevice = isLowRamDevice;
1698    mIsDeviceTypeKnown = true;
1699    return NO_ERROR;
1700}
1701
1702audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId)
1703{
1704    Mutex::Autolock _l(mLock);
1705
1706    ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
1707    if (index >= 0) {
1708        ALOGV("getAudioHwSyncForSession found ID %d for session %d",
1709              mHwAvSyncIds.valueAt(index), sessionId);
1710        return mHwAvSyncIds.valueAt(index);
1711    }
1712
1713    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1714    if (dev == NULL) {
1715        return AUDIO_HW_SYNC_INVALID;
1716    }
1717    char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC);
1718    AudioParameter param = AudioParameter(String8(reply));
1719    free(reply);
1720
1721    int value;
1722    if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) {
1723        ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId);
1724        return AUDIO_HW_SYNC_INVALID;
1725    }
1726
1727    // allow only one session for a given HW A/V sync ID.
1728    for (size_t i = 0; i < mHwAvSyncIds.size(); i++) {
1729        if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) {
1730            ALOGV("getAudioHwSyncForSession removing ID %d for session %d",
1731                  value, mHwAvSyncIds.keyAt(i));
1732            mHwAvSyncIds.removeItemsAt(i);
1733            break;
1734        }
1735    }
1736
1737    mHwAvSyncIds.add(sessionId, value);
1738
1739    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1740        sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i);
1741        uint32_t sessions = thread->hasAudioSession(sessionId);
1742        if (sessions & PlaybackThread::TRACK_SESSION) {
1743            AudioParameter param = AudioParameter();
1744            param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value);
1745            thread->setParameters(param.toString());
1746            break;
1747        }
1748    }
1749
1750    ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId);
1751    return (audio_hw_sync_t)value;
1752}
1753
1754status_t AudioFlinger::systemReady()
1755{
1756    Mutex::Autolock _l(mLock);
1757    ALOGI("%s", __FUNCTION__);
1758    if (mSystemReady) {
1759        ALOGW("%s called twice", __FUNCTION__);
1760        return NO_ERROR;
1761    }
1762    mSystemReady = true;
1763    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1764        ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get();
1765        thread->systemReady();
1766    }
1767    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1768        ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
1769        thread->systemReady();
1770    }
1771    return NO_ERROR;
1772}
1773
1774// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held
1775void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId)
1776{
1777    ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
1778    if (index >= 0) {
1779        audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index);
1780        ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId);
1781        AudioParameter param = AudioParameter();
1782        param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId);
1783        thread->setParameters(param.toString());
1784    }
1785}
1786
1787
1788// ----------------------------------------------------------------------------
1789
1790
1791sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module,
1792                                                            audio_io_handle_t *output,
1793                                                            audio_config_t *config,
1794                                                            audio_devices_t devices,
1795                                                            const String8& address,
1796                                                            audio_output_flags_t flags)
1797{
1798    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices);
1799    if (outHwDev == NULL) {
1800        return 0;
1801    }
1802
1803    if (*output == AUDIO_IO_HANDLE_NONE) {
1804        *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
1805    } else {
1806        // Audio Policy does not currently request a specific output handle.
1807        // If this is ever needed, see openInput_l() for example code.
1808        ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output);
1809        return 0;
1810    }
1811
1812    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1813
1814    // FOR TESTING ONLY:
1815    // This if statement allows overriding the audio policy settings
1816    // and forcing a specific format or channel mask to the HAL/Sink device for testing.
1817    if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
1818        // Check only for Normal Mixing mode
1819        if (kEnableExtendedPrecision) {
1820            // Specify format (uncomment one below to choose)
1821            //config->format = AUDIO_FORMAT_PCM_FLOAT;
1822            //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
1823            //config->format = AUDIO_FORMAT_PCM_32_BIT;
1824            //config->format = AUDIO_FORMAT_PCM_8_24_BIT;
1825            // ALOGV("openOutput_l() upgrading format to %#08x", config->format);
1826        }
1827        if (kEnableExtendedChannels) {
1828            // Specify channel mask (uncomment one below to choose)
1829            //config->channel_mask = audio_channel_out_mask_from_count(4);  // for USB 4ch
1830            //config->channel_mask = audio_channel_mask_from_representation_and_bits(
1831            //        AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1);  // another 4ch example
1832        }
1833    }
1834
1835    AudioStreamOut *outputStream = NULL;
1836    status_t status = outHwDev->openOutputStream(
1837            &outputStream,
1838            *output,
1839            devices,
1840            flags,
1841            config,
1842            address.string());
1843
1844    mHardwareStatus = AUDIO_HW_IDLE;
1845
1846    if (status == NO_ERROR) {
1847
1848        PlaybackThread *thread;
1849        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1850            thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady,
1851                                       config->offload_info.bit_rate);
1852            ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread);
1853        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
1854                || !isValidPcmSinkFormat(config->format)
1855                || !isValidPcmSinkChannelMask(config->channel_mask)) {
1856            thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
1857            ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread);
1858        } else {
1859            thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
1860            ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread);
1861        }
1862        mPlaybackThreads.add(*output, thread);
1863        return thread;
1864    }
1865
1866    return 0;
1867}
1868
1869status_t AudioFlinger::openOutput(audio_module_handle_t module,
1870                                  audio_io_handle_t *output,
1871                                  audio_config_t *config,
1872                                  audio_devices_t *devices,
1873                                  const String8& address,
1874                                  uint32_t *latencyMs,
1875                                  audio_output_flags_t flags)
1876{
1877    ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1878              module,
1879              (devices != NULL) ? *devices : 0,
1880              config->sample_rate,
1881              config->format,
1882              config->channel_mask,
1883              flags);
1884
1885    if (*devices == AUDIO_DEVICE_NONE) {
1886        return BAD_VALUE;
1887    }
1888
1889    Mutex::Autolock _l(mLock);
1890
1891    sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags);
1892    if (thread != 0) {
1893        *latencyMs = thread->latency();
1894
1895        // notify client processes of the new output creation
1896        thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
1897
1898        // the first primary output opened designates the primary hw device
1899        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1900            ALOGI("Using module %d has the primary audio interface", module);
1901            mPrimaryHardwareDev = thread->getOutput()->audioHwDev;
1902
1903            AutoMutex lock(mHardwareLock);
1904            mHardwareStatus = AUDIO_HW_SET_MODE;
1905            mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode);
1906            mHardwareStatus = AUDIO_HW_IDLE;
1907        }
1908        return NO_ERROR;
1909    }
1910
1911    return NO_INIT;
1912}
1913
1914audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1915        audio_io_handle_t output2)
1916{
1917    Mutex::Autolock _l(mLock);
1918    MixerThread *thread1 = checkMixerThread_l(output1);
1919    MixerThread *thread2 = checkMixerThread_l(output2);
1920
1921    if (thread1 == NULL || thread2 == NULL) {
1922        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1923                output2);
1924        return AUDIO_IO_HANDLE_NONE;
1925    }
1926
1927    audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
1928    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady);
1929    thread->addOutputTrack(thread2);
1930    mPlaybackThreads.add(id, thread);
1931    // notify client processes of the new output creation
1932    thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
1933    return id;
1934}
1935
1936status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1937{
1938    return closeOutput_nonvirtual(output);
1939}
1940
1941status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1942{
1943    // keep strong reference on the playback thread so that
1944    // it is not destroyed while exit() is executed
1945    sp<PlaybackThread> thread;
1946    {
1947        Mutex::Autolock _l(mLock);
1948        thread = checkPlaybackThread_l(output);
1949        if (thread == NULL) {
1950            return BAD_VALUE;
1951        }
1952
1953        ALOGV("closeOutput() %d", output);
1954
1955        if (thread->type() == ThreadBase::MIXER) {
1956            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1957                if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
1958                    DuplicatingThread *dupThread =
1959                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1960                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1961                }
1962            }
1963        }
1964
1965
1966        mPlaybackThreads.removeItem(output);
1967        // save all effects to the default thread
1968        if (mPlaybackThreads.size()) {
1969            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1970            if (dstThread != NULL) {
1971                // audioflinger lock is held here so the acquisition order of thread locks does not
1972                // matter
1973                Mutex::Autolock _dl(dstThread->mLock);
1974                Mutex::Autolock _sl(thread->mLock);
1975                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1976                for (size_t i = 0; i < effectChains.size(); i ++) {
1977                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1978                }
1979            }
1980        }
1981        const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
1982        ioDesc->mIoHandle = output;
1983        ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc);
1984    }
1985    thread->exit();
1986    // The thread entity (active unit of execution) is no longer running here,
1987    // but the ThreadBase container still exists.
1988
1989    if (!thread->isDuplicating()) {
1990        closeOutputFinish(thread);
1991    }
1992
1993    return NO_ERROR;
1994}
1995
1996void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread)
1997{
1998    AudioStreamOut *out = thread->clearOutput();
1999    ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
2000    // from now on thread->mOutput is NULL
2001    out->hwDev()->close_output_stream(out->hwDev(), out->stream);
2002    delete out;
2003}
2004
2005void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread)
2006{
2007    mPlaybackThreads.removeItem(thread->mId);
2008    thread->exit();
2009    closeOutputFinish(thread);
2010}
2011
2012status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
2013{
2014    Mutex::Autolock _l(mLock);
2015    PlaybackThread *thread = checkPlaybackThread_l(output);
2016
2017    if (thread == NULL) {
2018        return BAD_VALUE;
2019    }
2020
2021    ALOGV("suspendOutput() %d", output);
2022    thread->suspend();
2023
2024    return NO_ERROR;
2025}
2026
2027status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
2028{
2029    Mutex::Autolock _l(mLock);
2030    PlaybackThread *thread = checkPlaybackThread_l(output);
2031
2032    if (thread == NULL) {
2033        return BAD_VALUE;
2034    }
2035
2036    ALOGV("restoreOutput() %d", output);
2037
2038    thread->restore();
2039
2040    return NO_ERROR;
2041}
2042
2043status_t AudioFlinger::openInput(audio_module_handle_t module,
2044                                          audio_io_handle_t *input,
2045                                          audio_config_t *config,
2046                                          audio_devices_t *devices,
2047                                          const String8& address,
2048                                          audio_source_t source,
2049                                          audio_input_flags_t flags)
2050{
2051    Mutex::Autolock _l(mLock);
2052
2053    if (*devices == AUDIO_DEVICE_NONE) {
2054        return BAD_VALUE;
2055    }
2056
2057    sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags);
2058
2059    if (thread != 0) {
2060        // notify client processes of the new input creation
2061        thread->ioConfigChanged(AUDIO_INPUT_OPENED);
2062        return NO_ERROR;
2063    }
2064    return NO_INIT;
2065}
2066
2067sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module,
2068                                                         audio_io_handle_t *input,
2069                                                         audio_config_t *config,
2070                                                         audio_devices_t devices,
2071                                                         const String8& address,
2072                                                         audio_source_t source,
2073                                                         audio_input_flags_t flags)
2074{
2075    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
2076    if (inHwDev == NULL) {
2077        *input = AUDIO_IO_HANDLE_NONE;
2078        return 0;
2079    }
2080
2081    // Audio Policy can request a specific handle for hardware hotword.
2082    // The goal here is not to re-open an already opened input.
2083    // It is to use a pre-assigned I/O handle.
2084    if (*input == AUDIO_IO_HANDLE_NONE) {
2085        *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT);
2086    } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) {
2087        ALOGE("openInput_l() requested input handle %d is invalid", *input);
2088        return 0;
2089    } else if (mRecordThreads.indexOfKey(*input) >= 0) {
2090        // This should not happen in a transient state with current design.
2091        ALOGE("openInput_l() requested input handle %d is already assigned", *input);
2092        return 0;
2093    }
2094
2095    audio_config_t halconfig = *config;
2096    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
2097    audio_stream_in_t *inStream = NULL;
2098    status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
2099                                        &inStream, flags, address.string(), source);
2100    ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d"
2101           ", Format %#x, Channels %x, flags %#x, status %d addr %s",
2102            inStream,
2103            halconfig.sample_rate,
2104            halconfig.format,
2105            halconfig.channel_mask,
2106            flags,
2107            status, address.string());
2108
2109    // If the input could not be opened with the requested parameters and we can handle the
2110    // conversion internally, try to open again with the proposed parameters.
2111    if (status == BAD_VALUE &&
2112        audio_is_linear_pcm(config->format) &&
2113        audio_is_linear_pcm(halconfig.format) &&
2114        (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
2115        (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) &&
2116        (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) {
2117        // FIXME describe the change proposed by HAL (save old values so we can log them here)
2118        ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
2119        inStream = NULL;
2120        status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
2121                                            &inStream, flags, address.string(), source);
2122        // FIXME log this new status; HAL should not propose any further changes
2123    }
2124
2125    if (status == NO_ERROR && inStream != NULL) {
2126
2127#ifdef TEE_SINK
2128        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
2129        // or (re-)create if current Pipe is idle and does not match the new format
2130        sp<NBAIO_Sink> teeSink;
2131        enum {
2132            TEE_SINK_NO,    // don't copy input
2133            TEE_SINK_NEW,   // copy input using a new pipe
2134            TEE_SINK_OLD,   // copy input using an existing pipe
2135        } kind;
2136        NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate,
2137                audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format);
2138        if (!mTeeSinkInputEnabled) {
2139            kind = TEE_SINK_NO;
2140        } else if (!Format_isValid(format)) {
2141            kind = TEE_SINK_NO;
2142        } else if (mRecordTeeSink == 0) {
2143            kind = TEE_SINK_NEW;
2144        } else if (mRecordTeeSink->getStrongCount() != 1) {
2145            kind = TEE_SINK_NO;
2146        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
2147            kind = TEE_SINK_OLD;
2148        } else {
2149            kind = TEE_SINK_NEW;
2150        }
2151        switch (kind) {
2152        case TEE_SINK_NEW: {
2153            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
2154            size_t numCounterOffers = 0;
2155            const NBAIO_Format offers[1] = {format};
2156            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
2157            ALOG_ASSERT(index == 0);
2158            PipeReader *pipeReader = new PipeReader(*pipe);
2159            numCounterOffers = 0;
2160            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
2161            ALOG_ASSERT(index == 0);
2162            mRecordTeeSink = pipe;
2163            mRecordTeeSource = pipeReader;
2164            teeSink = pipe;
2165            }
2166            break;
2167        case TEE_SINK_OLD:
2168            teeSink = mRecordTeeSink;
2169            break;
2170        case TEE_SINK_NO:
2171        default:
2172            break;
2173        }
2174#endif
2175
2176        AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream);
2177
2178        // Start record thread
2179        // RecordThread requires both input and output device indication to forward to audio
2180        // pre processing modules
2181        sp<RecordThread> thread = new RecordThread(this,
2182                                  inputStream,
2183                                  *input,
2184                                  primaryOutputDevice_l(),
2185                                  devices,
2186                                  mSystemReady
2187#ifdef TEE_SINK
2188                                  , teeSink
2189#endif
2190                                  );
2191        mRecordThreads.add(*input, thread);
2192        ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
2193        return thread;
2194    }
2195
2196    *input = AUDIO_IO_HANDLE_NONE;
2197    return 0;
2198}
2199
2200status_t AudioFlinger::closeInput(audio_io_handle_t input)
2201{
2202    return closeInput_nonvirtual(input);
2203}
2204
2205status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
2206{
2207    // keep strong reference on the record thread so that
2208    // it is not destroyed while exit() is executed
2209    sp<RecordThread> thread;
2210    {
2211        Mutex::Autolock _l(mLock);
2212        thread = checkRecordThread_l(input);
2213        if (thread == 0) {
2214            return BAD_VALUE;
2215        }
2216
2217        ALOGV("closeInput() %d", input);
2218
2219        // If we still have effect chains, it means that a client still holds a handle
2220        // on at least one effect. We must either move the chain to an existing thread with the
2221        // same session ID or put it aside in case a new record thread is opened for a
2222        // new capture on the same session
2223        sp<EffectChain> chain;
2224        {
2225            Mutex::Autolock _sl(thread->mLock);
2226            Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
2227            // Note: maximum one chain per record thread
2228            if (effectChains.size() != 0) {
2229                chain = effectChains[0];
2230            }
2231        }
2232        if (chain != 0) {
2233            // first check if a record thread is already opened with a client on the same session.
2234            // This should only happen in case of overlap between one thread tear down and the
2235            // creation of its replacement
2236            size_t i;
2237            for (i = 0; i < mRecordThreads.size(); i++) {
2238                sp<RecordThread> t = mRecordThreads.valueAt(i);
2239                if (t == thread) {
2240                    continue;
2241                }
2242                if (t->hasAudioSession(chain->sessionId()) != 0) {
2243                    Mutex::Autolock _l(t->mLock);
2244                    ALOGV("closeInput() found thread %d for effect session %d",
2245                          t->id(), chain->sessionId());
2246                    t->addEffectChain_l(chain);
2247                    break;
2248                }
2249            }
2250            // put the chain aside if we could not find a record thread with the same session id.
2251            if (i == mRecordThreads.size()) {
2252                putOrphanEffectChain_l(chain);
2253            }
2254        }
2255        const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
2256        ioDesc->mIoHandle = input;
2257        ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc);
2258        mRecordThreads.removeItem(input);
2259    }
2260    // FIXME: calling thread->exit() without mLock held should not be needed anymore now that
2261    // we have a different lock for notification client
2262    closeInputFinish(thread);
2263    return NO_ERROR;
2264}
2265
2266void AudioFlinger::closeInputFinish(sp<RecordThread> thread)
2267{
2268    thread->exit();
2269    AudioStreamIn *in = thread->clearInput();
2270    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
2271    // from now on thread->mInput is NULL
2272    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
2273    delete in;
2274}
2275
2276void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread)
2277{
2278    mRecordThreads.removeItem(thread->mId);
2279    closeInputFinish(thread);
2280}
2281
2282status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
2283{
2284    Mutex::Autolock _l(mLock);
2285    ALOGV("invalidateStream() stream %d", stream);
2286
2287    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2288        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2289        thread->invalidateTracks(stream);
2290    }
2291
2292    return NO_ERROR;
2293}
2294
2295
2296audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use)
2297{
2298    // This is a binder API, so a malicious client could pass in a bad parameter.
2299    // Check for that before calling the internal API nextUniqueId().
2300    if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) {
2301        ALOGE("newAudioUniqueId invalid use %d", use);
2302        return AUDIO_UNIQUE_ID_ALLOCATE;
2303    }
2304    return nextUniqueId(use);
2305}
2306
2307void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid)
2308{
2309    Mutex::Autolock _l(mLock);
2310    pid_t caller = IPCThreadState::self()->getCallingPid();
2311    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
2312    if (pid != -1 && (caller == getpid_cached)) {
2313        caller = pid;
2314    }
2315
2316    {
2317        Mutex::Autolock _cl(mClientLock);
2318        // Ignore requests received from processes not known as notification client. The request
2319        // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
2320        // called from a different pid leaving a stale session reference.  Also we don't know how
2321        // to clear this reference if the client process dies.
2322        if (mNotificationClients.indexOfKey(caller) < 0) {
2323            ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
2324            return;
2325        }
2326    }
2327
2328    size_t num = mAudioSessionRefs.size();
2329    for (size_t i = 0; i< num; i++) {
2330        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
2331        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2332            ref->mCnt++;
2333            ALOGV(" incremented refcount to %d", ref->mCnt);
2334            return;
2335        }
2336    }
2337    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
2338    ALOGV(" added new entry for %d", audioSession);
2339}
2340
2341void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid)
2342{
2343    Mutex::Autolock _l(mLock);
2344    pid_t caller = IPCThreadState::self()->getCallingPid();
2345    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
2346    if (pid != -1 && (caller == getpid_cached)) {
2347        caller = pid;
2348    }
2349    size_t num = mAudioSessionRefs.size();
2350    for (size_t i = 0; i< num; i++) {
2351        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2352        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2353            ref->mCnt--;
2354            ALOGV(" decremented refcount to %d", ref->mCnt);
2355            if (ref->mCnt == 0) {
2356                mAudioSessionRefs.removeAt(i);
2357                delete ref;
2358                purgeStaleEffects_l();
2359            }
2360            return;
2361        }
2362    }
2363    // If the caller is mediaserver it is likely that the session being released was acquired
2364    // on behalf of a process not in notification clients and we ignore the warning.
2365    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2366}
2367
2368void AudioFlinger::purgeStaleEffects_l() {
2369
2370    ALOGV("purging stale effects");
2371
2372    Vector< sp<EffectChain> > chains;
2373
2374    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2375        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2376        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2377            sp<EffectChain> ec = t->mEffectChains[j];
2378            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2379                chains.push(ec);
2380            }
2381        }
2382    }
2383    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2384        sp<RecordThread> t = mRecordThreads.valueAt(i);
2385        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2386            sp<EffectChain> ec = t->mEffectChains[j];
2387            chains.push(ec);
2388        }
2389    }
2390
2391    for (size_t i = 0; i < chains.size(); i++) {
2392        sp<EffectChain> ec = chains[i];
2393        int sessionid = ec->sessionId();
2394        sp<ThreadBase> t = ec->mThread.promote();
2395        if (t == 0) {
2396            continue;
2397        }
2398        size_t numsessionrefs = mAudioSessionRefs.size();
2399        bool found = false;
2400        for (size_t k = 0; k < numsessionrefs; k++) {
2401            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2402            if (ref->mSessionid == sessionid) {
2403                ALOGV(" session %d still exists for %d with %d refs",
2404                    sessionid, ref->mPid, ref->mCnt);
2405                found = true;
2406                break;
2407            }
2408        }
2409        if (!found) {
2410            Mutex::Autolock _l(t->mLock);
2411            // remove all effects from the chain
2412            while (ec->mEffects.size()) {
2413                sp<EffectModule> effect = ec->mEffects[0];
2414                effect->unPin();
2415                t->removeEffect_l(effect);
2416                if (effect->purgeHandles()) {
2417                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2418                }
2419                AudioSystem::unregisterEffect(effect->id());
2420            }
2421        }
2422    }
2423    return;
2424}
2425
2426// checkThread_l() must be called with AudioFlinger::mLock held
2427AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const
2428{
2429    ThreadBase *thread = NULL;
2430    switch (audio_unique_id_get_use(ioHandle)) {
2431    case AUDIO_UNIQUE_ID_USE_OUTPUT:
2432        thread = checkPlaybackThread_l(ioHandle);
2433        break;
2434    case AUDIO_UNIQUE_ID_USE_INPUT:
2435        thread = checkRecordThread_l(ioHandle);
2436        break;
2437    default:
2438        break;
2439    }
2440    return thread;
2441}
2442
2443// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2444AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2445{
2446    return mPlaybackThreads.valueFor(output).get();
2447}
2448
2449// checkMixerThread_l() must be called with AudioFlinger::mLock held
2450AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2451{
2452    PlaybackThread *thread = checkPlaybackThread_l(output);
2453    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2454}
2455
2456// checkRecordThread_l() must be called with AudioFlinger::mLock held
2457AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2458{
2459    return mRecordThreads.valueFor(input).get();
2460}
2461
2462audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use)
2463{
2464    // This is the internal API, so it is OK to assert on bad parameter.
2465    LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX);
2466    const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1;
2467    for (int retry = 0; retry < maxRetries; retry++) {
2468        // The cast allows wraparound from max positive to min negative instead of abort
2469        uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use],
2470                (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel);
2471        ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED);
2472        // allow wrap by skipping 0 and -1 for session ids
2473        if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) {
2474            ALOGW_IF(retry != 0, "unique ID overflow for use %d", use);
2475            return (audio_unique_id_t) (base | use);
2476        }
2477    }
2478    // We have no way of recovering from wraparound
2479    LOG_ALWAYS_FATAL("unique ID overflow for use %d", use);
2480    // TODO Use a floor after wraparound.  This may need a mutex.
2481}
2482
2483AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2484{
2485    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2486        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2487        if(thread->isDuplicating()) {
2488            continue;
2489        }
2490        AudioStreamOut *output = thread->getOutput();
2491        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2492            return thread;
2493        }
2494    }
2495    return NULL;
2496}
2497
2498audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2499{
2500    PlaybackThread *thread = primaryPlaybackThread_l();
2501
2502    if (thread == NULL) {
2503        return 0;
2504    }
2505
2506    return thread->outDevice();
2507}
2508
2509sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2510                                    audio_session_t triggerSession,
2511                                    audio_session_t listenerSession,
2512                                    sync_event_callback_t callBack,
2513                                    wp<RefBase> cookie)
2514{
2515    Mutex::Autolock _l(mLock);
2516
2517    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2518    status_t playStatus = NAME_NOT_FOUND;
2519    status_t recStatus = NAME_NOT_FOUND;
2520    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2521        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2522        if (playStatus == NO_ERROR) {
2523            return event;
2524        }
2525    }
2526    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2527        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2528        if (recStatus == NO_ERROR) {
2529            return event;
2530        }
2531    }
2532    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2533        mPendingSyncEvents.add(event);
2534    } else {
2535        ALOGV("createSyncEvent() invalid event %d", event->type());
2536        event.clear();
2537    }
2538    return event;
2539}
2540
2541// ----------------------------------------------------------------------------
2542//  Effect management
2543// ----------------------------------------------------------------------------
2544
2545
2546status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2547{
2548    Mutex::Autolock _l(mLock);
2549    return EffectQueryNumberEffects(numEffects);
2550}
2551
2552status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2553{
2554    Mutex::Autolock _l(mLock);
2555    return EffectQueryEffect(index, descriptor);
2556}
2557
2558status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2559        effect_descriptor_t *descriptor) const
2560{
2561    Mutex::Autolock _l(mLock);
2562    return EffectGetDescriptor(pUuid, descriptor);
2563}
2564
2565
2566sp<IEffect> AudioFlinger::createEffect(
2567        effect_descriptor_t *pDesc,
2568        const sp<IEffectClient>& effectClient,
2569        int32_t priority,
2570        audio_io_handle_t io,
2571        audio_session_t sessionId,
2572        const String16& opPackageName,
2573        status_t *status,
2574        int *id,
2575        int *enabled)
2576{
2577    status_t lStatus = NO_ERROR;
2578    sp<EffectHandle> handle;
2579    effect_descriptor_t desc;
2580
2581    pid_t pid = IPCThreadState::self()->getCallingPid();
2582    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2583            pid, effectClient.get(), priority, sessionId, io);
2584
2585    if (pDesc == NULL) {
2586        lStatus = BAD_VALUE;
2587        goto Exit;
2588    }
2589
2590    // check audio settings permission for global effects
2591    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2592        lStatus = PERMISSION_DENIED;
2593        goto Exit;
2594    }
2595
2596    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2597    // that can only be created by audio policy manager (running in same process)
2598    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2599        lStatus = PERMISSION_DENIED;
2600        goto Exit;
2601    }
2602
2603    {
2604        if (!EffectIsNullUuid(&pDesc->uuid)) {
2605            // if uuid is specified, request effect descriptor
2606            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2607            if (lStatus < 0) {
2608                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2609                goto Exit;
2610            }
2611        } else {
2612            // if uuid is not specified, look for an available implementation
2613            // of the required type in effect factory
2614            if (EffectIsNullUuid(&pDesc->type)) {
2615                ALOGW("createEffect() no effect type");
2616                lStatus = BAD_VALUE;
2617                goto Exit;
2618            }
2619            uint32_t numEffects = 0;
2620            effect_descriptor_t d;
2621            d.flags = 0; // prevent compiler warning
2622            bool found = false;
2623
2624            lStatus = EffectQueryNumberEffects(&numEffects);
2625            if (lStatus < 0) {
2626                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2627                goto Exit;
2628            }
2629            for (uint32_t i = 0; i < numEffects; i++) {
2630                lStatus = EffectQueryEffect(i, &desc);
2631                if (lStatus < 0) {
2632                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2633                    continue;
2634                }
2635                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2636                    // If matching type found save effect descriptor. If the session is
2637                    // 0 and the effect is not auxiliary, continue enumeration in case
2638                    // an auxiliary version of this effect type is available
2639                    found = true;
2640                    d = desc;
2641                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2642                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2643                        break;
2644                    }
2645                }
2646            }
2647            if (!found) {
2648                lStatus = BAD_VALUE;
2649                ALOGW("createEffect() effect not found");
2650                goto Exit;
2651            }
2652            // For same effect type, chose auxiliary version over insert version if
2653            // connect to output mix (Compliance to OpenSL ES)
2654            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2655                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2656                desc = d;
2657            }
2658        }
2659
2660        // Do not allow auxiliary effects on a session different from 0 (output mix)
2661        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2662             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2663            lStatus = INVALID_OPERATION;
2664            goto Exit;
2665        }
2666
2667        // check recording permission for visualizer
2668        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2669            !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) {
2670            lStatus = PERMISSION_DENIED;
2671            goto Exit;
2672        }
2673
2674        // return effect descriptor
2675        *pDesc = desc;
2676        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2677            // if the output returned by getOutputForEffect() is removed before we lock the
2678            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2679            // and we will exit safely
2680            io = AudioSystem::getOutputForEffect(&desc);
2681            ALOGV("createEffect got output %d", io);
2682        }
2683
2684        Mutex::Autolock _l(mLock);
2685
2686        // If output is not specified try to find a matching audio session ID in one of the
2687        // output threads.
2688        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2689        // because of code checking output when entering the function.
2690        // Note: io is never 0 when creating an effect on an input
2691        if (io == AUDIO_IO_HANDLE_NONE) {
2692            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2693                // output must be specified by AudioPolicyManager when using session
2694                // AUDIO_SESSION_OUTPUT_STAGE
2695                lStatus = BAD_VALUE;
2696                goto Exit;
2697            }
2698            // look for the thread where the specified audio session is present
2699            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2700                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2701                    io = mPlaybackThreads.keyAt(i);
2702                    break;
2703                }
2704            }
2705            if (io == 0) {
2706                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2707                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2708                        io = mRecordThreads.keyAt(i);
2709                        break;
2710                    }
2711                }
2712            }
2713            // If no output thread contains the requested session ID, default to
2714            // first output. The effect chain will be moved to the correct output
2715            // thread when a track with the same session ID is created
2716            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2717                io = mPlaybackThreads.keyAt(0);
2718            }
2719            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2720        }
2721        ThreadBase *thread = checkRecordThread_l(io);
2722        if (thread == NULL) {
2723            thread = checkPlaybackThread_l(io);
2724            if (thread == NULL) {
2725                ALOGE("createEffect() unknown output thread");
2726                lStatus = BAD_VALUE;
2727                goto Exit;
2728            }
2729        } else {
2730            // Check if one effect chain was awaiting for an effect to be created on this
2731            // session and used it instead of creating a new one.
2732            sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
2733            if (chain != 0) {
2734                Mutex::Autolock _l(thread->mLock);
2735                thread->addEffectChain_l(chain);
2736            }
2737        }
2738
2739        sp<Client> client = registerPid(pid);
2740
2741        // create effect on selected output thread
2742        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2743                &desc, enabled, &lStatus);
2744        if (handle != 0 && id != NULL) {
2745            *id = handle->id();
2746        }
2747        if (handle == 0) {
2748            // remove local strong reference to Client with mClientLock held
2749            Mutex::Autolock _cl(mClientLock);
2750            client.clear();
2751        }
2752    }
2753
2754Exit:
2755    *status = lStatus;
2756    return handle;
2757}
2758
2759status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
2760        audio_io_handle_t dstOutput)
2761{
2762    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2763            sessionId, srcOutput, dstOutput);
2764    Mutex::Autolock _l(mLock);
2765    if (srcOutput == dstOutput) {
2766        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2767        return NO_ERROR;
2768    }
2769    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2770    if (srcThread == NULL) {
2771        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2772        return BAD_VALUE;
2773    }
2774    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2775    if (dstThread == NULL) {
2776        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2777        return BAD_VALUE;
2778    }
2779
2780    Mutex::Autolock _dl(dstThread->mLock);
2781    Mutex::Autolock _sl(srcThread->mLock);
2782    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2783}
2784
2785// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2786status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId,
2787                                   AudioFlinger::PlaybackThread *srcThread,
2788                                   AudioFlinger::PlaybackThread *dstThread,
2789                                   bool reRegister)
2790{
2791    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2792            sessionId, srcThread, dstThread);
2793
2794    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2795    if (chain == 0) {
2796        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2797                sessionId, srcThread);
2798        return INVALID_OPERATION;
2799    }
2800
2801    // Check whether the destination thread has a channel count of FCC_2, which is
2802    // currently required for (most) effects. Prevent moving the effect chain here rather
2803    // than disabling the addEffect_l() call in dstThread below.
2804    if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) &&
2805            dstThread->mChannelCount != FCC_2) {
2806        ALOGW("moveEffectChain_l() effect chain failed because"
2807                " destination thread %p channel count(%u) != %u",
2808                dstThread, dstThread->mChannelCount, FCC_2);
2809        return INVALID_OPERATION;
2810    }
2811
2812    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2813    // so that a new chain is created with correct parameters when first effect is added. This is
2814    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2815    // removed.
2816    srcThread->removeEffectChain_l(chain);
2817
2818    // transfer all effects one by one so that new effect chain is created on new thread with
2819    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2820    sp<EffectChain> dstChain;
2821    uint32_t strategy = 0; // prevent compiler warning
2822    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2823    Vector< sp<EffectModule> > removed;
2824    status_t status = NO_ERROR;
2825    while (effect != 0) {
2826        srcThread->removeEffect_l(effect);
2827        removed.add(effect);
2828        status = dstThread->addEffect_l(effect);
2829        if (status != NO_ERROR) {
2830            break;
2831        }
2832        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2833        if (effect->state() == EffectModule::ACTIVE ||
2834                effect->state() == EffectModule::STOPPING) {
2835            effect->start();
2836        }
2837        // if the move request is not received from audio policy manager, the effect must be
2838        // re-registered with the new strategy and output
2839        if (dstChain == 0) {
2840            dstChain = effect->chain().promote();
2841            if (dstChain == 0) {
2842                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2843                status = NO_INIT;
2844                break;
2845            }
2846            strategy = dstChain->strategy();
2847        }
2848        if (reRegister) {
2849            AudioSystem::unregisterEffect(effect->id());
2850            AudioSystem::registerEffect(&effect->desc(),
2851                                        dstThread->id(),
2852                                        strategy,
2853                                        sessionId,
2854                                        effect->id());
2855            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2856        }
2857        effect = chain->getEffectFromId_l(0);
2858    }
2859
2860    if (status != NO_ERROR) {
2861        for (size_t i = 0; i < removed.size(); i++) {
2862            srcThread->addEffect_l(removed[i]);
2863            if (dstChain != 0 && reRegister) {
2864                AudioSystem::unregisterEffect(removed[i]->id());
2865                AudioSystem::registerEffect(&removed[i]->desc(),
2866                                            srcThread->id(),
2867                                            strategy,
2868                                            sessionId,
2869                                            removed[i]->id());
2870                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2871            }
2872        }
2873    }
2874
2875    return status;
2876}
2877
2878bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2879{
2880    if (mGlobalEffectEnableTime != 0 &&
2881            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2882        return true;
2883    }
2884
2885    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2886        sp<EffectChain> ec =
2887                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2888        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2889            return true;
2890        }
2891    }
2892    return false;
2893}
2894
2895void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2896{
2897    Mutex::Autolock _l(mLock);
2898
2899    mGlobalEffectEnableTime = systemTime();
2900
2901    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2902        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2903        if (t->mType == ThreadBase::OFFLOAD) {
2904            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2905        }
2906    }
2907
2908}
2909
2910status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain)
2911{
2912    audio_session_t session = chain->sessionId();
2913    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2914    ALOGV("putOrphanEffectChain_l session %d index %zd", session, index);
2915    if (index >= 0) {
2916        ALOGW("putOrphanEffectChain_l chain for session %d already present", session);
2917        return ALREADY_EXISTS;
2918    }
2919    mOrphanEffectChains.add(session, chain);
2920    return NO_ERROR;
2921}
2922
2923sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session)
2924{
2925    sp<EffectChain> chain;
2926    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2927    ALOGV("getOrphanEffectChain_l session %d index %zd", session, index);
2928    if (index >= 0) {
2929        chain = mOrphanEffectChains.valueAt(index);
2930        mOrphanEffectChains.removeItemsAt(index);
2931    }
2932    return chain;
2933}
2934
2935bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect)
2936{
2937    Mutex::Autolock _l(mLock);
2938    audio_session_t session = effect->sessionId();
2939    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2940    ALOGV("updateOrphanEffectChains session %d index %zd", session, index);
2941    if (index >= 0) {
2942        sp<EffectChain> chain = mOrphanEffectChains.valueAt(index);
2943        if (chain->removeEffect_l(effect) == 0) {
2944            ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index);
2945            mOrphanEffectChains.removeItemsAt(index);
2946        }
2947        return true;
2948    }
2949    return false;
2950}
2951
2952
2953struct Entry {
2954#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21
2955    char mFileName[TEE_MAX_FILENAME];
2956};
2957
2958int comparEntry(const void *p1, const void *p2)
2959{
2960    return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName);
2961}
2962
2963#ifdef TEE_SINK
2964void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2965{
2966    NBAIO_Source *teeSource = source.get();
2967    if (teeSource != NULL) {
2968        // .wav rotation
2969        // There is a benign race condition if 2 threads call this simultaneously.
2970        // They would both traverse the directory, but the result would simply be
2971        // failures at unlink() which are ignored.  It's also unlikely since
2972        // normally dumpsys is only done by bugreport or from the command line.
2973        char teePath[32+256];
2974        strcpy(teePath, "/data/misc/audioserver");
2975        size_t teePathLen = strlen(teePath);
2976        DIR *dir = opendir(teePath);
2977        teePath[teePathLen++] = '/';
2978        if (dir != NULL) {
2979#define TEE_MAX_SORT 20 // number of entries to sort
2980#define TEE_MAX_KEEP 10 // number of entries to keep
2981            struct Entry entries[TEE_MAX_SORT];
2982            size_t entryCount = 0;
2983            while (entryCount < TEE_MAX_SORT) {
2984                struct dirent de;
2985                struct dirent *result = NULL;
2986                int rc = readdir_r(dir, &de, &result);
2987                if (rc != 0) {
2988                    ALOGW("readdir_r failed %d", rc);
2989                    break;
2990                }
2991                if (result == NULL) {
2992                    break;
2993                }
2994                if (result != &de) {
2995                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2996                    break;
2997                }
2998                // ignore non .wav file entries
2999                size_t nameLen = strlen(de.d_name);
3000                if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME ||
3001                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
3002                    continue;
3003                }
3004                strcpy(entries[entryCount++].mFileName, de.d_name);
3005            }
3006            (void) closedir(dir);
3007            if (entryCount > TEE_MAX_KEEP) {
3008                qsort(entries, entryCount, sizeof(Entry), comparEntry);
3009                for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) {
3010                    strcpy(&teePath[teePathLen], entries[i].mFileName);
3011                    (void) unlink(teePath);
3012                }
3013            }
3014        } else {
3015            if (fd >= 0) {
3016                dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath,
3017                        strerror(errno));
3018            }
3019        }
3020        char teeTime[16];
3021        struct timeval tv;
3022        gettimeofday(&tv, NULL);
3023        struct tm tm;
3024        localtime_r(&tv.tv_sec, &tm);
3025        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
3026        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
3027        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
3028        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
3029        if (teeFd >= 0) {
3030            // FIXME use libsndfile
3031            char wavHeader[44];
3032            memcpy(wavHeader,
3033                "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",
3034                sizeof(wavHeader));
3035            NBAIO_Format format = teeSource->format();
3036            unsigned channelCount = Format_channelCount(format);
3037            uint32_t sampleRate = Format_sampleRate(format);
3038            size_t frameSize = Format_frameSize(format);
3039            wavHeader[22] = channelCount;       // number of channels
3040            wavHeader[24] = sampleRate;         // sample rate
3041            wavHeader[25] = sampleRate >> 8;
3042            wavHeader[32] = frameSize;          // block alignment
3043            wavHeader[33] = frameSize >> 8;
3044            write(teeFd, wavHeader, sizeof(wavHeader));
3045            size_t total = 0;
3046            bool firstRead = true;
3047#define TEE_SINK_READ 1024                      // frames per I/O operation
3048            void *buffer = malloc(TEE_SINK_READ * frameSize);
3049            for (;;) {
3050                size_t count = TEE_SINK_READ;
3051                ssize_t actual = teeSource->read(buffer, count);
3052                bool wasFirstRead = firstRead;
3053                firstRead = false;
3054                if (actual <= 0) {
3055                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
3056                        continue;
3057                    }
3058                    break;
3059                }
3060                ALOG_ASSERT(actual <= (ssize_t)count);
3061                write(teeFd, buffer, actual * frameSize);
3062                total += actual;
3063            }
3064            free(buffer);
3065            lseek(teeFd, (off_t) 4, SEEK_SET);
3066            uint32_t temp = 44 + total * frameSize - 8;
3067            // FIXME not big-endian safe
3068            write(teeFd, &temp, sizeof(temp));
3069            lseek(teeFd, (off_t) 40, SEEK_SET);
3070            temp =  total * frameSize;
3071            // FIXME not big-endian safe
3072            write(teeFd, &temp, sizeof(temp));
3073            close(teeFd);
3074            if (fd >= 0) {
3075                dprintf(fd, "tee copied to %s\n", teePath);
3076            }
3077        } else {
3078            if (fd >= 0) {
3079                dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
3080            }
3081        }
3082    }
3083}
3084#endif
3085
3086// ----------------------------------------------------------------------------
3087
3088status_t AudioFlinger::onTransact(
3089        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
3090{
3091    return BnAudioFlinger::onTransact(code, data, reply, flags);
3092}
3093
3094} // namespace android
3095