AudioFlinger.cpp revision e34a44ff6cb12e65736c09e1baab1823de190e03 
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#include <media/MemoryLeakTrackUtil.h>
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 dumpMem = false;
475        bool unreachableMemory = false;
476        for (const auto &arg : args) {
477            if (arg == String16("-m")) {
478                dumpMem = true;
479            } else if (arg == String16("--unreachable")) {
480                unreachableMemory = true;
481            }
482        }
483
484        if (dumpMem) {
485            dprintf(fd, "\nDumping memory:\n");
486            std::string s = dumpMemoryAddresses(100 /* limit */);
487            write(fd, s.c_str(), s.size());
488        }
489        if (unreachableMemory) {
490            dprintf(fd, "\nDumping unreachable memory:\n");
491            // TODO - should limit be an argument parameter?
492            std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */);
493            write(fd, s.c_str(), s.size());
494        }
495    }
496    return NO_ERROR;
497}
498
499sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
500{
501    Mutex::Autolock _cl(mClientLock);
502    // If pid is already in the mClients wp<> map, then use that entry
503    // (for which promote() is always != 0), otherwise create a new entry and Client.
504    sp<Client> client = mClients.valueFor(pid).promote();
505    if (client == 0) {
506        client = new Client(this, pid);
507        mClients.add(pid, client);
508    }
509
510    return client;
511}
512
513sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
514{
515    // If there is no memory allocated for logs, return a dummy writer that does nothing
516    if (mLogMemoryDealer == 0) {
517        return new NBLog::Writer();
518    }
519    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
520    // Similarly if we can't contact the media.log service, also return a dummy writer
521    if (binder == 0) {
522        return new NBLog::Writer();
523    }
524    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
525    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
526    // If allocation fails, consult the vector of previously unregistered writers
527    // and garbage-collect one or more them until an allocation succeeds
528    if (shared == 0) {
529        Mutex::Autolock _l(mUnregisteredWritersLock);
530        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
531            {
532                // Pick the oldest stale writer to garbage-collect
533                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
534                mUnregisteredWriters.removeAt(0);
535                mediaLogService->unregisterWriter(iMemory);
536                // Now the media.log remote reference to IMemory is gone.  When our last local
537                // reference to IMemory also drops to zero at end of this block,
538                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
539            }
540            // Re-attempt the allocation
541            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
542            if (shared != 0) {
543                goto success;
544            }
545        }
546        // Even after garbage-collecting all old writers, there is still not enough memory,
547        // so return a dummy writer
548        return new NBLog::Writer();
549    }
550success:
551    mediaLogService->registerWriter(shared, size, name);
552    return new NBLog::Writer(size, shared);
553}
554
555void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
556{
557    if (writer == 0) {
558        return;
559    }
560    sp<IMemory> iMemory(writer->getIMemory());
561    if (iMemory == 0) {
562        return;
563    }
564    // Rather than removing the writer immediately, append it to a queue of old writers to
565    // be garbage-collected later.  This allows us to continue to view old logs for a while.
566    Mutex::Autolock _l(mUnregisteredWritersLock);
567    mUnregisteredWriters.push(writer);
568}
569
570// IAudioFlinger interface
571
572
573sp<IAudioTrack> AudioFlinger::createTrack(
574        audio_stream_type_t streamType,
575        uint32_t sampleRate,
576        audio_format_t format,
577        audio_channel_mask_t channelMask,
578        size_t *frameCount,
579        IAudioFlinger::track_flags_t *flags,
580        const sp<IMemory>& sharedBuffer,
581        audio_io_handle_t output,
582        pid_t pid,
583        pid_t tid,
584        audio_session_t *sessionId,
585        int clientUid,
586        status_t *status)
587{
588    sp<PlaybackThread::Track> track;
589    sp<TrackHandle> trackHandle;
590    sp<Client> client;
591    status_t lStatus;
592    audio_session_t lSessionId;
593
594    const uid_t callingUid = IPCThreadState::self()->getCallingUid();
595    if (pid == -1 || !isTrustedCallingUid(callingUid)) {
596        const pid_t callingPid = IPCThreadState::self()->getCallingPid();
597        ALOGW_IF(pid != -1 && pid != callingPid,
598                 "%s uid %d pid %d tried to pass itself off as pid %d",
599                 __func__, callingUid, callingPid, pid);
600        pid = callingPid;
601    }
602
603    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
604    // but if someone uses binder directly they could bypass that and cause us to crash
605    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
606        ALOGE("createTrack() invalid stream type %d", streamType);
607        lStatus = BAD_VALUE;
608        goto Exit;
609    }
610
611    // further sample rate checks are performed by createTrack_l() depending on the thread type
612    if (sampleRate == 0) {
613        ALOGE("createTrack() invalid sample rate %u", sampleRate);
614        lStatus = BAD_VALUE;
615        goto Exit;
616    }
617
618    // further channel mask checks are performed by createTrack_l() depending on the thread type
619    if (!audio_is_output_channel(channelMask)) {
620        ALOGE("createTrack() invalid channel mask %#x", channelMask);
621        lStatus = BAD_VALUE;
622        goto Exit;
623    }
624
625    // further format checks are performed by createTrack_l() depending on the thread type
626    if (!audio_is_valid_format(format)) {
627        ALOGE("createTrack() invalid format %#x", format);
628        lStatus = BAD_VALUE;
629        goto Exit;
630    }
631
632    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
633        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
634        lStatus = BAD_VALUE;
635        goto Exit;
636    }
637
638    {
639        Mutex::Autolock _l(mLock);
640        PlaybackThread *thread = checkPlaybackThread_l(output);
641        if (thread == NULL) {
642            ALOGE("no playback thread found for output handle %d", output);
643            lStatus = BAD_VALUE;
644            goto Exit;
645        }
646
647        client = registerPid(pid);
648
649        PlaybackThread *effectThread = NULL;
650        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
651            if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
652                ALOGE("createTrack() invalid session ID %d", *sessionId);
653                lStatus = BAD_VALUE;
654                goto Exit;
655            }
656            lSessionId = *sessionId;
657            // check if an effect chain with the same session ID is present on another
658            // output thread and move it here.
659            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
660                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
661                if (mPlaybackThreads.keyAt(i) != output) {
662                    uint32_t sessions = t->hasAudioSession(lSessionId);
663                    if (sessions & PlaybackThread::EFFECT_SESSION) {
664                        effectThread = t.get();
665                        break;
666                    }
667                }
668            }
669        } else {
670            // if no audio session id is provided, create one here
671            lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
672            if (sessionId != NULL) {
673                *sessionId = lSessionId;
674            }
675        }
676        ALOGV("createTrack() lSessionId: %d", lSessionId);
677
678        track = thread->createTrack_l(client, streamType, sampleRate, format,
679                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
680        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
681        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
682
683        // move effect chain to this output thread if an effect on same session was waiting
684        // for a track to be created
685        if (lStatus == NO_ERROR && effectThread != NULL) {
686            // no risk of deadlock because AudioFlinger::mLock is held
687            Mutex::Autolock _dl(thread->mLock);
688            Mutex::Autolock _sl(effectThread->mLock);
689            moveEffectChain_l(lSessionId, effectThread, thread, true);
690        }
691
692        // Look for sync events awaiting for a session to be used.
693        for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
694            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
695                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
696                    if (lStatus == NO_ERROR) {
697                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
698                    } else {
699                        mPendingSyncEvents[i]->cancel();
700                    }
701                    mPendingSyncEvents.removeAt(i);
702                    i--;
703                }
704            }
705        }
706
707        setAudioHwSyncForSession_l(thread, lSessionId);
708    }
709
710    if (lStatus != NO_ERROR) {
711        // remove local strong reference to Client before deleting the Track so that the
712        // Client destructor is called by the TrackBase destructor with mClientLock held
713        // Don't hold mClientLock when releasing the reference on the track as the
714        // destructor will acquire it.
715        {
716            Mutex::Autolock _cl(mClientLock);
717            client.clear();
718        }
719        track.clear();
720        goto Exit;
721    }
722
723    // return handle to client
724    trackHandle = new TrackHandle(track);
725
726Exit:
727    *status = lStatus;
728    return trackHandle;
729}
730
731uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const
732{
733    Mutex::Autolock _l(mLock);
734    ThreadBase *thread = checkThread_l(ioHandle);
735    if (thread == NULL) {
736        ALOGW("sampleRate() unknown thread %d", ioHandle);
737        return 0;
738    }
739    return thread->sampleRate();
740}
741
742audio_format_t AudioFlinger::format(audio_io_handle_t output) const
743{
744    Mutex::Autolock _l(mLock);
745    PlaybackThread *thread = checkPlaybackThread_l(output);
746    if (thread == NULL) {
747        ALOGW("format() unknown thread %d", output);
748        return AUDIO_FORMAT_INVALID;
749    }
750    return thread->format();
751}
752
753size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const
754{
755    Mutex::Autolock _l(mLock);
756    ThreadBase *thread = checkThread_l(ioHandle);
757    if (thread == NULL) {
758        ALOGW("frameCount() unknown thread %d", ioHandle);
759        return 0;
760    }
761    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
762    //       should examine all callers and fix them to handle smaller counts
763    return thread->frameCount();
764}
765
766size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const
767{
768    Mutex::Autolock _l(mLock);
769    ThreadBase *thread = checkThread_l(ioHandle);
770    if (thread == NULL) {
771        ALOGW("frameCountHAL() unknown thread %d", ioHandle);
772        return 0;
773    }
774    return thread->frameCountHAL();
775}
776
777uint32_t AudioFlinger::latency(audio_io_handle_t output) const
778{
779    Mutex::Autolock _l(mLock);
780    PlaybackThread *thread = checkPlaybackThread_l(output);
781    if (thread == NULL) {
782        ALOGW("latency(): no playback thread found for output handle %d", output);
783        return 0;
784    }
785    return thread->latency();
786}
787
788status_t AudioFlinger::setMasterVolume(float value)
789{
790    status_t ret = initCheck();
791    if (ret != NO_ERROR) {
792        return ret;
793    }
794
795    // check calling permissions
796    if (!settingsAllowed()) {
797        return PERMISSION_DENIED;
798    }
799
800    Mutex::Autolock _l(mLock);
801    mMasterVolume = value;
802
803    // Set master volume in the HALs which support it.
804    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
805        AutoMutex lock(mHardwareLock);
806        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
807
808        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
809        if (dev->canSetMasterVolume()) {
810            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
811        }
812        mHardwareStatus = AUDIO_HW_IDLE;
813    }
814
815    // Now set the master volume in each playback thread.  Playback threads
816    // assigned to HALs which do not have master volume support will apply
817    // master volume during the mix operation.  Threads with HALs which do
818    // support master volume will simply ignore the setting.
819    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
820        if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
821            continue;
822        }
823        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
824    }
825
826    return NO_ERROR;
827}
828
829status_t AudioFlinger::setMode(audio_mode_t mode)
830{
831    status_t ret = initCheck();
832    if (ret != NO_ERROR) {
833        return ret;
834    }
835
836    // check calling permissions
837    if (!settingsAllowed()) {
838        return PERMISSION_DENIED;
839    }
840    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
841        ALOGW("Illegal value: setMode(%d)", mode);
842        return BAD_VALUE;
843    }
844
845    { // scope for the lock
846        AutoMutex lock(mHardwareLock);
847        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
848        mHardwareStatus = AUDIO_HW_SET_MODE;
849        ret = dev->set_mode(dev, mode);
850        mHardwareStatus = AUDIO_HW_IDLE;
851    }
852
853    if (NO_ERROR == ret) {
854        Mutex::Autolock _l(mLock);
855        mMode = mode;
856        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
857            mPlaybackThreads.valueAt(i)->setMode(mode);
858    }
859
860    return ret;
861}
862
863status_t AudioFlinger::setMicMute(bool state)
864{
865    status_t ret = initCheck();
866    if (ret != NO_ERROR) {
867        return ret;
868    }
869
870    // check calling permissions
871    if (!settingsAllowed()) {
872        return PERMISSION_DENIED;
873    }
874
875    AutoMutex lock(mHardwareLock);
876    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
877    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
878        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
879        status_t result = dev->set_mic_mute(dev, state);
880        if (result != NO_ERROR) {
881            ret = result;
882        }
883    }
884    mHardwareStatus = AUDIO_HW_IDLE;
885    return ret;
886}
887
888bool AudioFlinger::getMicMute() const
889{
890    status_t ret = initCheck();
891    if (ret != NO_ERROR) {
892        return false;
893    }
894    bool mute = true;
895    bool state = AUDIO_MODE_INVALID;
896    AutoMutex lock(mHardwareLock);
897    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
898    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
899        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
900        status_t result = dev->get_mic_mute(dev, &state);
901        if (result == NO_ERROR) {
902            mute = mute && state;
903        }
904    }
905    mHardwareStatus = AUDIO_HW_IDLE;
906
907    return mute;
908}
909
910status_t AudioFlinger::setMasterMute(bool muted)
911{
912    status_t ret = initCheck();
913    if (ret != NO_ERROR) {
914        return ret;
915    }
916
917    // check calling permissions
918    if (!settingsAllowed()) {
919        return PERMISSION_DENIED;
920    }
921
922    Mutex::Autolock _l(mLock);
923    mMasterMute = muted;
924
925    // Set master mute in the HALs which support it.
926    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
927        AutoMutex lock(mHardwareLock);
928        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
929
930        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
931        if (dev->canSetMasterMute()) {
932            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
933        }
934        mHardwareStatus = AUDIO_HW_IDLE;
935    }
936
937    // Now set the master mute in each playback thread.  Playback threads
938    // assigned to HALs which do not have master mute support will apply master
939    // mute during the mix operation.  Threads with HALs which do support master
940    // mute will simply ignore the setting.
941    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
942        if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
943            continue;
944        }
945        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
946    }
947
948    return NO_ERROR;
949}
950
951float AudioFlinger::masterVolume() const
952{
953    Mutex::Autolock _l(mLock);
954    return masterVolume_l();
955}
956
957bool AudioFlinger::masterMute() const
958{
959    Mutex::Autolock _l(mLock);
960    return masterMute_l();
961}
962
963float AudioFlinger::masterVolume_l() const
964{
965    return mMasterVolume;
966}
967
968bool AudioFlinger::masterMute_l() const
969{
970    return mMasterMute;
971}
972
973status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const
974{
975    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
976        ALOGW("setStreamVolume() invalid stream %d", stream);
977        return BAD_VALUE;
978    }
979    pid_t caller = IPCThreadState::self()->getCallingPid();
980    if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) {
981        ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream);
982        return PERMISSION_DENIED;
983    }
984
985    return NO_ERROR;
986}
987
988status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
989        audio_io_handle_t output)
990{
991    // check calling permissions
992    if (!settingsAllowed()) {
993        return PERMISSION_DENIED;
994    }
995
996    status_t status = checkStreamType(stream);
997    if (status != NO_ERROR) {
998        return status;
999    }
1000    ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume");
1001
1002    AutoMutex lock(mLock);
1003    PlaybackThread *thread = NULL;
1004    if (output != AUDIO_IO_HANDLE_NONE) {
1005        thread = checkPlaybackThread_l(output);
1006        if (thread == NULL) {
1007            return BAD_VALUE;
1008        }
1009    }
1010
1011    mStreamTypes[stream].volume = value;
1012
1013    if (thread == NULL) {
1014        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1015            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
1016        }
1017    } else {
1018        thread->setStreamVolume(stream, value);
1019    }
1020
1021    return NO_ERROR;
1022}
1023
1024status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
1025{
1026    // check calling permissions
1027    if (!settingsAllowed()) {
1028        return PERMISSION_DENIED;
1029    }
1030
1031    status_t status = checkStreamType(stream);
1032    if (status != NO_ERROR) {
1033        return status;
1034    }
1035    ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH");
1036
1037    if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
1038        ALOGE("setStreamMute() invalid stream %d", stream);
1039        return BAD_VALUE;
1040    }
1041
1042    AutoMutex lock(mLock);
1043    mStreamTypes[stream].mute = muted;
1044    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
1045        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
1046
1047    return NO_ERROR;
1048}
1049
1050float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
1051{
1052    status_t status = checkStreamType(stream);
1053    if (status != NO_ERROR) {
1054        return 0.0f;
1055    }
1056
1057    AutoMutex lock(mLock);
1058    float volume;
1059    if (output != AUDIO_IO_HANDLE_NONE) {
1060        PlaybackThread *thread = checkPlaybackThread_l(output);
1061        if (thread == NULL) {
1062            return 0.0f;
1063        }
1064        volume = thread->streamVolume(stream);
1065    } else {
1066        volume = streamVolume_l(stream);
1067    }
1068
1069    return volume;
1070}
1071
1072bool AudioFlinger::streamMute(audio_stream_type_t stream) const
1073{
1074    status_t status = checkStreamType(stream);
1075    if (status != NO_ERROR) {
1076        return true;
1077    }
1078
1079    AutoMutex lock(mLock);
1080    return streamMute_l(stream);
1081}
1082
1083
1084void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs)
1085{
1086    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1087        mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1088    }
1089}
1090
1091status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
1092{
1093    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
1094            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
1095
1096    // check calling permissions
1097    if (!settingsAllowed()) {
1098        return PERMISSION_DENIED;
1099    }
1100
1101    // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
1102    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1103        Mutex::Autolock _l(mLock);
1104        status_t final_result = NO_ERROR;
1105        {
1106            AutoMutex lock(mHardwareLock);
1107            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
1108            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1109                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1110                status_t result = dev->set_parameters(dev, keyValuePairs.string());
1111                final_result = result ?: final_result;
1112            }
1113            mHardwareStatus = AUDIO_HW_IDLE;
1114        }
1115        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
1116        AudioParameter param = AudioParameter(keyValuePairs);
1117        String8 value;
1118        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
1119            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
1120            if (mBtNrecIsOff != btNrecIsOff) {
1121                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1122                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
1123                    audio_devices_t device = thread->inDevice();
1124                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
1125                    // collect all of the thread's session IDs
1126                    KeyedVector<audio_session_t, bool> ids = thread->sessionIds();
1127                    // suspend effects associated with those session IDs
1128                    for (size_t j = 0; j < ids.size(); ++j) {
1129                        audio_session_t sessionId = ids.keyAt(j);
1130                        thread->setEffectSuspended(FX_IID_AEC,
1131                                                   suspend,
1132                                                   sessionId);
1133                        thread->setEffectSuspended(FX_IID_NS,
1134                                                   suspend,
1135                                                   sessionId);
1136                    }
1137                }
1138                mBtNrecIsOff = btNrecIsOff;
1139            }
1140        }
1141        String8 screenState;
1142        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1143            bool isOff = screenState == "off";
1144            if (isOff != (AudioFlinger::mScreenState & 1)) {
1145                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1146            }
1147        }
1148        return final_result;
1149    }
1150
1151    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1152    // and the thread is exited once the lock is released
1153    sp<ThreadBase> thread;
1154    {
1155        Mutex::Autolock _l(mLock);
1156        thread = checkPlaybackThread_l(ioHandle);
1157        if (thread == 0) {
1158            thread = checkRecordThread_l(ioHandle);
1159        } else if (thread == primaryPlaybackThread_l()) {
1160            // indicate output device change to all input threads for pre processing
1161            AudioParameter param = AudioParameter(keyValuePairs);
1162            int value;
1163            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1164                    (value != 0)) {
1165                broacastParametersToRecordThreads_l(keyValuePairs);
1166            }
1167        }
1168    }
1169    if (thread != 0) {
1170        return thread->setParameters(keyValuePairs);
1171    }
1172    return BAD_VALUE;
1173}
1174
1175String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1176{
1177    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1178            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1179
1180    Mutex::Autolock _l(mLock);
1181
1182    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1183        String8 out_s8;
1184
1185        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1186            char *s;
1187            {
1188            AutoMutex lock(mHardwareLock);
1189            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1190            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1191            s = dev->get_parameters(dev, keys.string());
1192            mHardwareStatus = AUDIO_HW_IDLE;
1193            }
1194            out_s8 += String8(s ? s : "");
1195            free(s);
1196        }
1197        return out_s8;
1198    }
1199
1200    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1201    if (playbackThread != NULL) {
1202        return playbackThread->getParameters(keys);
1203    }
1204    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1205    if (recordThread != NULL) {
1206        return recordThread->getParameters(keys);
1207    }
1208    return String8("");
1209}
1210
1211size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1212        audio_channel_mask_t channelMask) const
1213{
1214    status_t ret = initCheck();
1215    if (ret != NO_ERROR) {
1216        return 0;
1217    }
1218    if ((sampleRate == 0) ||
1219            !audio_is_valid_format(format) || !audio_has_proportional_frames(format) ||
1220            !audio_is_input_channel(channelMask)) {
1221        return 0;
1222    }
1223
1224    AutoMutex lock(mHardwareLock);
1225    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1226    audio_config_t config, proposed;
1227    memset(&proposed, 0, sizeof(proposed));
1228    proposed.sample_rate = sampleRate;
1229    proposed.channel_mask = channelMask;
1230    proposed.format = format;
1231
1232    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1233    size_t frames;
1234    for (;;) {
1235        // Note: config is currently a const parameter for get_input_buffer_size()
1236        // but we use a copy from proposed in case config changes from the call.
1237        config = proposed;
1238        frames = dev->get_input_buffer_size(dev, &config);
1239        if (frames != 0) {
1240            break; // hal success, config is the result
1241        }
1242        // change one parameter of the configuration each iteration to a more "common" value
1243        // to see if the device will support it.
1244        if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
1245            proposed.format = AUDIO_FORMAT_PCM_16_BIT;
1246        } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
1247            proposed.sample_rate = 44100;           // legacy AudioRecord.java. TODO: Query hw?
1248        } else {
1249            ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
1250                    "format %#x, channelMask 0x%X",
1251                    sampleRate, format, channelMask);
1252            break; // retries failed, break out of loop with frames == 0.
1253        }
1254    }
1255    mHardwareStatus = AUDIO_HW_IDLE;
1256    if (frames > 0 && config.sample_rate != sampleRate) {
1257        frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
1258    }
1259    return frames; // may be converted to bytes at the Java level.
1260}
1261
1262uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1263{
1264    Mutex::Autolock _l(mLock);
1265
1266    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1267    if (recordThread != NULL) {
1268        return recordThread->getInputFramesLost();
1269    }
1270    return 0;
1271}
1272
1273status_t AudioFlinger::setVoiceVolume(float value)
1274{
1275    status_t ret = initCheck();
1276    if (ret != NO_ERROR) {
1277        return ret;
1278    }
1279
1280    // check calling permissions
1281    if (!settingsAllowed()) {
1282        return PERMISSION_DENIED;
1283    }
1284
1285    AutoMutex lock(mHardwareLock);
1286    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1287    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1288    ret = dev->set_voice_volume(dev, value);
1289    mHardwareStatus = AUDIO_HW_IDLE;
1290
1291    return ret;
1292}
1293
1294status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1295        audio_io_handle_t output) const
1296{
1297    Mutex::Autolock _l(mLock);
1298
1299    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1300    if (playbackThread != NULL) {
1301        return playbackThread->getRenderPosition(halFrames, dspFrames);
1302    }
1303
1304    return BAD_VALUE;
1305}
1306
1307void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1308{
1309    Mutex::Autolock _l(mLock);
1310    if (client == 0) {
1311        return;
1312    }
1313    pid_t pid = IPCThreadState::self()->getCallingPid();
1314    {
1315        Mutex::Autolock _cl(mClientLock);
1316        if (mNotificationClients.indexOfKey(pid) < 0) {
1317            sp<NotificationClient> notificationClient = new NotificationClient(this,
1318                                                                                client,
1319                                                                                pid);
1320            ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1321
1322            mNotificationClients.add(pid, notificationClient);
1323
1324            sp<IBinder> binder = IInterface::asBinder(client);
1325            binder->linkToDeath(notificationClient);
1326        }
1327    }
1328
1329    // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
1330    // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock.
1331    // the config change is always sent from playback or record threads to avoid deadlock
1332    // with AudioSystem::gLock
1333    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1334        mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid);
1335    }
1336
1337    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1338        mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid);
1339    }
1340}
1341
1342void AudioFlinger::removeNotificationClient(pid_t pid)
1343{
1344    Mutex::Autolock _l(mLock);
1345    {
1346        Mutex::Autolock _cl(mClientLock);
1347        mNotificationClients.removeItem(pid);
1348    }
1349
1350    ALOGV("%d died, releasing its sessions", pid);
1351    size_t num = mAudioSessionRefs.size();
1352    bool removed = false;
1353    for (size_t i = 0; i< num; ) {
1354        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1355        ALOGV(" pid %d @ %zu", ref->mPid, i);
1356        if (ref->mPid == pid) {
1357            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1358            mAudioSessionRefs.removeAt(i);
1359            delete ref;
1360            removed = true;
1361            num--;
1362        } else {
1363            i++;
1364        }
1365    }
1366    if (removed) {
1367        purgeStaleEffects_l();
1368    }
1369}
1370
1371void AudioFlinger::ioConfigChanged(audio_io_config_event event,
1372                                   const sp<AudioIoDescriptor>& ioDesc,
1373                                   pid_t pid)
1374{
1375    Mutex::Autolock _l(mClientLock);
1376    size_t size = mNotificationClients.size();
1377    for (size_t i = 0; i < size; i++) {
1378        if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) {
1379            mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc);
1380        }
1381    }
1382}
1383
1384// removeClient_l() must be called with AudioFlinger::mClientLock held
1385void AudioFlinger::removeClient_l(pid_t pid)
1386{
1387    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1388            IPCThreadState::self()->getCallingPid());
1389    mClients.removeItem(pid);
1390}
1391
1392// getEffectThread_l() must be called with AudioFlinger::mLock held
1393sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
1394        int EffectId)
1395{
1396    sp<PlaybackThread> thread;
1397
1398    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1399        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1400            ALOG_ASSERT(thread == 0);
1401            thread = mPlaybackThreads.valueAt(i);
1402        }
1403    }
1404
1405    return thread;
1406}
1407
1408
1409
1410// ----------------------------------------------------------------------------
1411
1412AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1413    :   RefBase(),
1414        mAudioFlinger(audioFlinger),
1415        mPid(pid)
1416{
1417    size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0);
1418    heapSize *= 1024;
1419    if (!heapSize) {
1420        heapSize = kClientSharedHeapSizeBytes;
1421        // Increase heap size on non low ram devices to limit risk of reconnection failure for
1422        // invalidated tracks
1423        if (!audioFlinger->isLowRamDevice()) {
1424            heapSize *= kClientSharedHeapSizeMultiplier;
1425        }
1426    }
1427    mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client");
1428}
1429
1430// Client destructor must be called with AudioFlinger::mClientLock held
1431AudioFlinger::Client::~Client()
1432{
1433    mAudioFlinger->removeClient_l(mPid);
1434}
1435
1436sp<MemoryDealer> AudioFlinger::Client::heap() const
1437{
1438    return mMemoryDealer;
1439}
1440
1441// ----------------------------------------------------------------------------
1442
1443AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1444                                                     const sp<IAudioFlingerClient>& client,
1445                                                     pid_t pid)
1446    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1447{
1448}
1449
1450AudioFlinger::NotificationClient::~NotificationClient()
1451{
1452}
1453
1454void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1455{
1456    sp<NotificationClient> keep(this);
1457    mAudioFlinger->removeNotificationClient(mPid);
1458}
1459
1460
1461// ----------------------------------------------------------------------------
1462
1463sp<IAudioRecord> AudioFlinger::openRecord(
1464        audio_io_handle_t input,
1465        uint32_t sampleRate,
1466        audio_format_t format,
1467        audio_channel_mask_t channelMask,
1468        const String16& opPackageName,
1469        size_t *frameCount,
1470        IAudioFlinger::track_flags_t *flags,
1471        pid_t pid,
1472        pid_t tid,
1473        int clientUid,
1474        audio_session_t *sessionId,
1475        size_t *notificationFrames,
1476        sp<IMemory>& cblk,
1477        sp<IMemory>& buffers,
1478        status_t *status)
1479{
1480    sp<RecordThread::RecordTrack> recordTrack;
1481    sp<RecordHandle> recordHandle;
1482    sp<Client> client;
1483    status_t lStatus;
1484    audio_session_t lSessionId;
1485
1486    cblk.clear();
1487    buffers.clear();
1488
1489    bool updatePid = (pid == -1);
1490    const uid_t callingUid = IPCThreadState::self()->getCallingUid();
1491    if (!isTrustedCallingUid(callingUid)) {
1492        ALOGW_IF((uid_t)clientUid != callingUid,
1493                "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid);
1494        clientUid = callingUid;
1495        updatePid = true;
1496    }
1497
1498    if (updatePid) {
1499        const pid_t callingPid = IPCThreadState::self()->getCallingPid();
1500        ALOGW_IF(pid != -1 && pid != callingPid,
1501                 "%s uid %d pid %d tried to pass itself off as pid %d",
1502                 __func__, callingUid, callingPid, pid);
1503        pid = callingPid;
1504    }
1505
1506    // check calling permissions
1507    if (!recordingAllowed(opPackageName, tid, clientUid)) {
1508        ALOGE("openRecord() permission denied: recording not allowed");
1509        lStatus = PERMISSION_DENIED;
1510        goto Exit;
1511    }
1512
1513    // further sample rate checks are performed by createRecordTrack_l()
1514    if (sampleRate == 0) {
1515        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1516        lStatus = BAD_VALUE;
1517        goto Exit;
1518    }
1519
1520    // we don't yet support anything other than linear PCM
1521    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
1522        ALOGE("openRecord() invalid format %#x", format);
1523        lStatus = BAD_VALUE;
1524        goto Exit;
1525    }
1526
1527    // further channel mask checks are performed by createRecordTrack_l()
1528    if (!audio_is_input_channel(channelMask)) {
1529        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1530        lStatus = BAD_VALUE;
1531        goto Exit;
1532    }
1533
1534    {
1535        Mutex::Autolock _l(mLock);
1536        RecordThread *thread = checkRecordThread_l(input);
1537        if (thread == NULL) {
1538            ALOGE("openRecord() checkRecordThread_l failed");
1539            lStatus = BAD_VALUE;
1540            goto Exit;
1541        }
1542
1543        client = registerPid(pid);
1544
1545        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1546            if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
1547                lStatus = BAD_VALUE;
1548                goto Exit;
1549            }
1550            lSessionId = *sessionId;
1551        } else {
1552            // if no audio session id is provided, create one here
1553            lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
1554            if (sessionId != NULL) {
1555                *sessionId = lSessionId;
1556            }
1557        }
1558        ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input);
1559
1560        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1561                                                  frameCount, lSessionId, notificationFrames,
1562                                                  clientUid, flags, tid, &lStatus);
1563        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1564
1565        if (lStatus == NO_ERROR) {
1566            // Check if one effect chain was awaiting for an AudioRecord to be created on this
1567            // session and move it to this thread.
1568            sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId);
1569            if (chain != 0) {
1570                Mutex::Autolock _l(thread->mLock);
1571                thread->addEffectChain_l(chain);
1572            }
1573        }
1574    }
1575
1576    if (lStatus != NO_ERROR) {
1577        // remove local strong reference to Client before deleting the RecordTrack so that the
1578        // Client destructor is called by the TrackBase destructor with mClientLock held
1579        // Don't hold mClientLock when releasing the reference on the track as the
1580        // destructor will acquire it.
1581        {
1582            Mutex::Autolock _cl(mClientLock);
1583            client.clear();
1584        }
1585        recordTrack.clear();
1586        goto Exit;
1587    }
1588
1589    cblk = recordTrack->getCblk();
1590    buffers = recordTrack->getBuffers();
1591
1592    // return handle to client
1593    recordHandle = new RecordHandle(recordTrack);
1594
1595Exit:
1596    *status = lStatus;
1597    return recordHandle;
1598}
1599
1600
1601
1602// ----------------------------------------------------------------------------
1603
1604audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1605{
1606    if (name == NULL) {
1607        return AUDIO_MODULE_HANDLE_NONE;
1608    }
1609    if (!settingsAllowed()) {
1610        return AUDIO_MODULE_HANDLE_NONE;
1611    }
1612    Mutex::Autolock _l(mLock);
1613    return loadHwModule_l(name);
1614}
1615
1616// loadHwModule_l() must be called with AudioFlinger::mLock held
1617audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1618{
1619    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1620        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1621            ALOGW("loadHwModule() module %s already loaded", name);
1622            return mAudioHwDevs.keyAt(i);
1623        }
1624    }
1625
1626    audio_hw_device_t *dev;
1627
1628    int rc = load_audio_interface(name, &dev);
1629    if (rc) {
1630        ALOGE("loadHwModule() error %d loading module %s", rc, name);
1631        return AUDIO_MODULE_HANDLE_NONE;
1632    }
1633
1634    mHardwareStatus = AUDIO_HW_INIT;
1635    rc = dev->init_check(dev);
1636    mHardwareStatus = AUDIO_HW_IDLE;
1637    if (rc) {
1638        ALOGE("loadHwModule() init check error %d for module %s", rc, name);
1639        return AUDIO_MODULE_HANDLE_NONE;
1640    }
1641
1642    // Check and cache this HAL's level of support for master mute and master
1643    // volume.  If this is the first HAL opened, and it supports the get
1644    // methods, use the initial values provided by the HAL as the current
1645    // master mute and volume settings.
1646
1647    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1648    {  // scope for auto-lock pattern
1649        AutoMutex lock(mHardwareLock);
1650
1651        if (0 == mAudioHwDevs.size()) {
1652            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1653            if (NULL != dev->get_master_volume) {
1654                float mv;
1655                if (OK == dev->get_master_volume(dev, &mv)) {
1656                    mMasterVolume = mv;
1657                }
1658            }
1659
1660            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1661            if (NULL != dev->get_master_mute) {
1662                bool mm;
1663                if (OK == dev->get_master_mute(dev, &mm)) {
1664                    mMasterMute = mm;
1665                }
1666            }
1667        }
1668
1669        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1670        if ((NULL != dev->set_master_volume) &&
1671            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1672            flags = static_cast<AudioHwDevice::Flags>(flags |
1673                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1674        }
1675
1676        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1677        if ((NULL != dev->set_master_mute) &&
1678            (OK == dev->set_master_mute(dev, mMasterMute))) {
1679            flags = static_cast<AudioHwDevice::Flags>(flags |
1680                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1681        }
1682
1683        mHardwareStatus = AUDIO_HW_IDLE;
1684    }
1685
1686    audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE);
1687    mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags));
1688
1689    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1690          name, dev->common.module->name, dev->common.module->id, handle);
1691
1692    return handle;
1693
1694}
1695
1696// ----------------------------------------------------------------------------
1697
1698uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1699{
1700    Mutex::Autolock _l(mLock);
1701    PlaybackThread *thread = primaryPlaybackThread_l();
1702    return thread != NULL ? thread->sampleRate() : 0;
1703}
1704
1705size_t AudioFlinger::getPrimaryOutputFrameCount()
1706{
1707    Mutex::Autolock _l(mLock);
1708    PlaybackThread *thread = primaryPlaybackThread_l();
1709    return thread != NULL ? thread->frameCountHAL() : 0;
1710}
1711
1712// ----------------------------------------------------------------------------
1713
1714status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1715{
1716    uid_t uid = IPCThreadState::self()->getCallingUid();
1717    if (uid != AID_SYSTEM) {
1718        return PERMISSION_DENIED;
1719    }
1720    Mutex::Autolock _l(mLock);
1721    if (mIsDeviceTypeKnown) {
1722        return INVALID_OPERATION;
1723    }
1724    mIsLowRamDevice = isLowRamDevice;
1725    mIsDeviceTypeKnown = true;
1726    return NO_ERROR;
1727}
1728
1729audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId)
1730{
1731    Mutex::Autolock _l(mLock);
1732
1733    ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
1734    if (index >= 0) {
1735        ALOGV("getAudioHwSyncForSession found ID %d for session %d",
1736              mHwAvSyncIds.valueAt(index), sessionId);
1737        return mHwAvSyncIds.valueAt(index);
1738    }
1739
1740    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1741    if (dev == NULL) {
1742        return AUDIO_HW_SYNC_INVALID;
1743    }
1744    char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC);
1745    AudioParameter param = AudioParameter(String8(reply));
1746    free(reply);
1747
1748    int value;
1749    if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) {
1750        ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId);
1751        return AUDIO_HW_SYNC_INVALID;
1752    }
1753
1754    // allow only one session for a given HW A/V sync ID.
1755    for (size_t i = 0; i < mHwAvSyncIds.size(); i++) {
1756        if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) {
1757            ALOGV("getAudioHwSyncForSession removing ID %d for session %d",
1758                  value, mHwAvSyncIds.keyAt(i));
1759            mHwAvSyncIds.removeItemsAt(i);
1760            break;
1761        }
1762    }
1763
1764    mHwAvSyncIds.add(sessionId, value);
1765
1766    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1767        sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i);
1768        uint32_t sessions = thread->hasAudioSession(sessionId);
1769        if (sessions & PlaybackThread::TRACK_SESSION) {
1770            AudioParameter param = AudioParameter();
1771            param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value);
1772            thread->setParameters(param.toString());
1773            break;
1774        }
1775    }
1776
1777    ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId);
1778    return (audio_hw_sync_t)value;
1779}
1780
1781status_t AudioFlinger::systemReady()
1782{
1783    Mutex::Autolock _l(mLock);
1784    ALOGI("%s", __FUNCTION__);
1785    if (mSystemReady) {
1786        ALOGW("%s called twice", __FUNCTION__);
1787        return NO_ERROR;
1788    }
1789    mSystemReady = true;
1790    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1791        ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get();
1792        thread->systemReady();
1793    }
1794    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1795        ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
1796        thread->systemReady();
1797    }
1798    return NO_ERROR;
1799}
1800
1801// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held
1802void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId)
1803{
1804    ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
1805    if (index >= 0) {
1806        audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index);
1807        ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId);
1808        AudioParameter param = AudioParameter();
1809        param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId);
1810        thread->setParameters(param.toString());
1811    }
1812}
1813
1814
1815// ----------------------------------------------------------------------------
1816
1817
1818sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module,
1819                                                            audio_io_handle_t *output,
1820                                                            audio_config_t *config,
1821                                                            audio_devices_t devices,
1822                                                            const String8& address,
1823                                                            audio_output_flags_t flags)
1824{
1825    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices);
1826    if (outHwDev == NULL) {
1827        return 0;
1828    }
1829
1830    if (*output == AUDIO_IO_HANDLE_NONE) {
1831        *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
1832    } else {
1833        // Audio Policy does not currently request a specific output handle.
1834        // If this is ever needed, see openInput_l() for example code.
1835        ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output);
1836        return 0;
1837    }
1838
1839    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1840
1841    // FOR TESTING ONLY:
1842    // This if statement allows overriding the audio policy settings
1843    // and forcing a specific format or channel mask to the HAL/Sink device for testing.
1844    if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
1845        // Check only for Normal Mixing mode
1846        if (kEnableExtendedPrecision) {
1847            // Specify format (uncomment one below to choose)
1848            //config->format = AUDIO_FORMAT_PCM_FLOAT;
1849            //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
1850            //config->format = AUDIO_FORMAT_PCM_32_BIT;
1851            //config->format = AUDIO_FORMAT_PCM_8_24_BIT;
1852            // ALOGV("openOutput_l() upgrading format to %#08x", config->format);
1853        }
1854        if (kEnableExtendedChannels) {
1855            // Specify channel mask (uncomment one below to choose)
1856            //config->channel_mask = audio_channel_out_mask_from_count(4);  // for USB 4ch
1857            //config->channel_mask = audio_channel_mask_from_representation_and_bits(
1858            //        AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1);  // another 4ch example
1859        }
1860    }
1861
1862    AudioStreamOut *outputStream = NULL;
1863    status_t status = outHwDev->openOutputStream(
1864            &outputStream,
1865            *output,
1866            devices,
1867            flags,
1868            config,
1869            address.string());
1870
1871    mHardwareStatus = AUDIO_HW_IDLE;
1872
1873    if (status == NO_ERROR) {
1874
1875        PlaybackThread *thread;
1876        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1877            thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady);
1878            ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread);
1879        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
1880                || !isValidPcmSinkFormat(config->format)
1881                || !isValidPcmSinkChannelMask(config->channel_mask)) {
1882            thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
1883            ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread);
1884        } else {
1885            thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
1886            ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread);
1887        }
1888        mPlaybackThreads.add(*output, thread);
1889        return thread;
1890    }
1891
1892    return 0;
1893}
1894
1895status_t AudioFlinger::openOutput(audio_module_handle_t module,
1896                                  audio_io_handle_t *output,
1897                                  audio_config_t *config,
1898                                  audio_devices_t *devices,
1899                                  const String8& address,
1900                                  uint32_t *latencyMs,
1901                                  audio_output_flags_t flags)
1902{
1903    ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1904              module,
1905              (devices != NULL) ? *devices : 0,
1906              config->sample_rate,
1907              config->format,
1908              config->channel_mask,
1909              flags);
1910
1911    if (*devices == AUDIO_DEVICE_NONE) {
1912        return BAD_VALUE;
1913    }
1914
1915    Mutex::Autolock _l(mLock);
1916
1917    sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags);
1918    if (thread != 0) {
1919        *latencyMs = thread->latency();
1920
1921        // notify client processes of the new output creation
1922        thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
1923
1924        // the first primary output opened designates the primary hw device
1925        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1926            ALOGI("Using module %d has the primary audio interface", module);
1927            mPrimaryHardwareDev = thread->getOutput()->audioHwDev;
1928
1929            AutoMutex lock(mHardwareLock);
1930            mHardwareStatus = AUDIO_HW_SET_MODE;
1931            mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode);
1932            mHardwareStatus = AUDIO_HW_IDLE;
1933        }
1934        return NO_ERROR;
1935    }
1936
1937    return NO_INIT;
1938}
1939
1940audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1941        audio_io_handle_t output2)
1942{
1943    Mutex::Autolock _l(mLock);
1944    MixerThread *thread1 = checkMixerThread_l(output1);
1945    MixerThread *thread2 = checkMixerThread_l(output2);
1946
1947    if (thread1 == NULL || thread2 == NULL) {
1948        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1949                output2);
1950        return AUDIO_IO_HANDLE_NONE;
1951    }
1952
1953    audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
1954    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady);
1955    thread->addOutputTrack(thread2);
1956    mPlaybackThreads.add(id, thread);
1957    // notify client processes of the new output creation
1958    thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
1959    return id;
1960}
1961
1962status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1963{
1964    return closeOutput_nonvirtual(output);
1965}
1966
1967status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1968{
1969    // keep strong reference on the playback thread so that
1970    // it is not destroyed while exit() is executed
1971    sp<PlaybackThread> thread;
1972    {
1973        Mutex::Autolock _l(mLock);
1974        thread = checkPlaybackThread_l(output);
1975        if (thread == NULL) {
1976            return BAD_VALUE;
1977        }
1978
1979        ALOGV("closeOutput() %d", output);
1980
1981        if (thread->type() == ThreadBase::MIXER) {
1982            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1983                if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
1984                    DuplicatingThread *dupThread =
1985                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1986                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1987                }
1988            }
1989        }
1990
1991
1992        mPlaybackThreads.removeItem(output);
1993        // save all effects to the default thread
1994        if (mPlaybackThreads.size()) {
1995            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1996            if (dstThread != NULL) {
1997                // audioflinger lock is held here so the acquisition order of thread locks does not
1998                // matter
1999                Mutex::Autolock _dl(dstThread->mLock);
2000                Mutex::Autolock _sl(thread->mLock);
2001                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
2002                for (size_t i = 0; i < effectChains.size(); i ++) {
2003                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
2004                }
2005            }
2006        }
2007        const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
2008        ioDesc->mIoHandle = output;
2009        ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc);
2010    }
2011    thread->exit();
2012    // The thread entity (active unit of execution) is no longer running here,
2013    // but the ThreadBase container still exists.
2014
2015    if (!thread->isDuplicating()) {
2016        closeOutputFinish(thread);
2017    }
2018
2019    return NO_ERROR;
2020}
2021
2022void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread)
2023{
2024    AudioStreamOut *out = thread->clearOutput();
2025    ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
2026    // from now on thread->mOutput is NULL
2027    out->hwDev()->close_output_stream(out->hwDev(), out->stream);
2028    delete out;
2029}
2030
2031void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread)
2032{
2033    mPlaybackThreads.removeItem(thread->mId);
2034    thread->exit();
2035    closeOutputFinish(thread);
2036}
2037
2038status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
2039{
2040    Mutex::Autolock _l(mLock);
2041    PlaybackThread *thread = checkPlaybackThread_l(output);
2042
2043    if (thread == NULL) {
2044        return BAD_VALUE;
2045    }
2046
2047    ALOGV("suspendOutput() %d", output);
2048    thread->suspend();
2049
2050    return NO_ERROR;
2051}
2052
2053status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
2054{
2055    Mutex::Autolock _l(mLock);
2056    PlaybackThread *thread = checkPlaybackThread_l(output);
2057
2058    if (thread == NULL) {
2059        return BAD_VALUE;
2060    }
2061
2062    ALOGV("restoreOutput() %d", output);
2063
2064    thread->restore();
2065
2066    return NO_ERROR;
2067}
2068
2069status_t AudioFlinger::openInput(audio_module_handle_t module,
2070                                          audio_io_handle_t *input,
2071                                          audio_config_t *config,
2072                                          audio_devices_t *devices,
2073                                          const String8& address,
2074                                          audio_source_t source,
2075                                          audio_input_flags_t flags)
2076{
2077    Mutex::Autolock _l(mLock);
2078
2079    if (*devices == AUDIO_DEVICE_NONE) {
2080        return BAD_VALUE;
2081    }
2082
2083    sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags);
2084
2085    if (thread != 0) {
2086        // notify client processes of the new input creation
2087        thread->ioConfigChanged(AUDIO_INPUT_OPENED);
2088        return NO_ERROR;
2089    }
2090    return NO_INIT;
2091}
2092
2093sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module,
2094                                                         audio_io_handle_t *input,
2095                                                         audio_config_t *config,
2096                                                         audio_devices_t devices,
2097                                                         const String8& address,
2098                                                         audio_source_t source,
2099                                                         audio_input_flags_t flags)
2100{
2101    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
2102    if (inHwDev == NULL) {
2103        *input = AUDIO_IO_HANDLE_NONE;
2104        return 0;
2105    }
2106
2107    // Audio Policy can request a specific handle for hardware hotword.
2108    // The goal here is not to re-open an already opened input.
2109    // It is to use a pre-assigned I/O handle.
2110    if (*input == AUDIO_IO_HANDLE_NONE) {
2111        *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT);
2112    } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) {
2113        ALOGE("openInput_l() requested input handle %d is invalid", *input);
2114        return 0;
2115    } else if (mRecordThreads.indexOfKey(*input) >= 0) {
2116        // This should not happen in a transient state with current design.
2117        ALOGE("openInput_l() requested input handle %d is already assigned", *input);
2118        return 0;
2119    }
2120
2121    audio_config_t halconfig = *config;
2122    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
2123    audio_stream_in_t *inStream = NULL;
2124    status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
2125                                        &inStream, flags, address.string(), source);
2126    ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d"
2127           ", Format %#x, Channels %x, flags %#x, status %d addr %s",
2128            inStream,
2129            halconfig.sample_rate,
2130            halconfig.format,
2131            halconfig.channel_mask,
2132            flags,
2133            status, address.string());
2134
2135    // If the input could not be opened with the requested parameters and we can handle the
2136    // conversion internally, try to open again with the proposed parameters.
2137    if (status == BAD_VALUE &&
2138        audio_is_linear_pcm(config->format) &&
2139        audio_is_linear_pcm(halconfig.format) &&
2140        (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
2141        (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) &&
2142        (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) {
2143        // FIXME describe the change proposed by HAL (save old values so we can log them here)
2144        ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
2145        inStream = NULL;
2146        status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
2147                                            &inStream, flags, address.string(), source);
2148        // FIXME log this new status; HAL should not propose any further changes
2149    }
2150
2151    if (status == NO_ERROR && inStream != NULL) {
2152
2153#ifdef TEE_SINK
2154        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
2155        // or (re-)create if current Pipe is idle and does not match the new format
2156        sp<NBAIO_Sink> teeSink;
2157        enum {
2158            TEE_SINK_NO,    // don't copy input
2159            TEE_SINK_NEW,   // copy input using a new pipe
2160            TEE_SINK_OLD,   // copy input using an existing pipe
2161        } kind;
2162        NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate,
2163                audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format);
2164        if (!mTeeSinkInputEnabled) {
2165            kind = TEE_SINK_NO;
2166        } else if (!Format_isValid(format)) {
2167            kind = TEE_SINK_NO;
2168        } else if (mRecordTeeSink == 0) {
2169            kind = TEE_SINK_NEW;
2170        } else if (mRecordTeeSink->getStrongCount() != 1) {
2171            kind = TEE_SINK_NO;
2172        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
2173            kind = TEE_SINK_OLD;
2174        } else {
2175            kind = TEE_SINK_NEW;
2176        }
2177        switch (kind) {
2178        case TEE_SINK_NEW: {
2179            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
2180            size_t numCounterOffers = 0;
2181            const NBAIO_Format offers[1] = {format};
2182            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
2183            ALOG_ASSERT(index == 0);
2184            PipeReader *pipeReader = new PipeReader(*pipe);
2185            numCounterOffers = 0;
2186            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
2187            ALOG_ASSERT(index == 0);
2188            mRecordTeeSink = pipe;
2189            mRecordTeeSource = pipeReader;
2190            teeSink = pipe;
2191            }
2192            break;
2193        case TEE_SINK_OLD:
2194            teeSink = mRecordTeeSink;
2195            break;
2196        case TEE_SINK_NO:
2197        default:
2198            break;
2199        }
2200#endif
2201
2202        AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream);
2203
2204        // Start record thread
2205        // RecordThread requires both input and output device indication to forward to audio
2206        // pre processing modules
2207        sp<RecordThread> thread = new RecordThread(this,
2208                                  inputStream,
2209                                  *input,
2210                                  primaryOutputDevice_l(),
2211                                  devices,
2212                                  mSystemReady
2213#ifdef TEE_SINK
2214                                  , teeSink
2215#endif
2216                                  );
2217        mRecordThreads.add(*input, thread);
2218        ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
2219        return thread;
2220    }
2221
2222    *input = AUDIO_IO_HANDLE_NONE;
2223    return 0;
2224}
2225
2226status_t AudioFlinger::closeInput(audio_io_handle_t input)
2227{
2228    return closeInput_nonvirtual(input);
2229}
2230
2231status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
2232{
2233    // keep strong reference on the record thread so that
2234    // it is not destroyed while exit() is executed
2235    sp<RecordThread> thread;
2236    {
2237        Mutex::Autolock _l(mLock);
2238        thread = checkRecordThread_l(input);
2239        if (thread == 0) {
2240            return BAD_VALUE;
2241        }
2242
2243        ALOGV("closeInput() %d", input);
2244
2245        // If we still have effect chains, it means that a client still holds a handle
2246        // on at least one effect. We must either move the chain to an existing thread with the
2247        // same session ID or put it aside in case a new record thread is opened for a
2248        // new capture on the same session
2249        sp<EffectChain> chain;
2250        {
2251            Mutex::Autolock _sl(thread->mLock);
2252            Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
2253            // Note: maximum one chain per record thread
2254            if (effectChains.size() != 0) {
2255                chain = effectChains[0];
2256            }
2257        }
2258        if (chain != 0) {
2259            // first check if a record thread is already opened with a client on the same session.
2260            // This should only happen in case of overlap between one thread tear down and the
2261            // creation of its replacement
2262            size_t i;
2263            for (i = 0; i < mRecordThreads.size(); i++) {
2264                sp<RecordThread> t = mRecordThreads.valueAt(i);
2265                if (t == thread) {
2266                    continue;
2267                }
2268                if (t->hasAudioSession(chain->sessionId()) != 0) {
2269                    Mutex::Autolock _l(t->mLock);
2270                    ALOGV("closeInput() found thread %d for effect session %d",
2271                          t->id(), chain->sessionId());
2272                    t->addEffectChain_l(chain);
2273                    break;
2274                }
2275            }
2276            // put the chain aside if we could not find a record thread with the same session id.
2277            if (i == mRecordThreads.size()) {
2278                putOrphanEffectChain_l(chain);
2279            }
2280        }
2281        const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
2282        ioDesc->mIoHandle = input;
2283        ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc);
2284        mRecordThreads.removeItem(input);
2285    }
2286    // FIXME: calling thread->exit() without mLock held should not be needed anymore now that
2287    // we have a different lock for notification client
2288    closeInputFinish(thread);
2289    return NO_ERROR;
2290}
2291
2292void AudioFlinger::closeInputFinish(sp<RecordThread> thread)
2293{
2294    thread->exit();
2295    AudioStreamIn *in = thread->clearInput();
2296    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
2297    // from now on thread->mInput is NULL
2298    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
2299    delete in;
2300}
2301
2302void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread)
2303{
2304    mRecordThreads.removeItem(thread->mId);
2305    closeInputFinish(thread);
2306}
2307
2308status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
2309{
2310    Mutex::Autolock _l(mLock);
2311    ALOGV("invalidateStream() stream %d", stream);
2312
2313    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2314        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2315        thread->invalidateTracks(stream);
2316    }
2317
2318    return NO_ERROR;
2319}
2320
2321
2322audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use)
2323{
2324    // This is a binder API, so a malicious client could pass in a bad parameter.
2325    // Check for that before calling the internal API nextUniqueId().
2326    if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) {
2327        ALOGE("newAudioUniqueId invalid use %d", use);
2328        return AUDIO_UNIQUE_ID_ALLOCATE;
2329    }
2330    return nextUniqueId(use);
2331}
2332
2333void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid)
2334{
2335    Mutex::Autolock _l(mLock);
2336    pid_t caller = IPCThreadState::self()->getCallingPid();
2337    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
2338    if (pid != -1 && (caller == getpid_cached)) {
2339        caller = pid;
2340    }
2341
2342    {
2343        Mutex::Autolock _cl(mClientLock);
2344        // Ignore requests received from processes not known as notification client. The request
2345        // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
2346        // called from a different pid leaving a stale session reference.  Also we don't know how
2347        // to clear this reference if the client process dies.
2348        if (mNotificationClients.indexOfKey(caller) < 0) {
2349            ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
2350            return;
2351        }
2352    }
2353
2354    size_t num = mAudioSessionRefs.size();
2355    for (size_t i = 0; i< num; i++) {
2356        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
2357        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2358            ref->mCnt++;
2359            ALOGV(" incremented refcount to %d", ref->mCnt);
2360            return;
2361        }
2362    }
2363    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
2364    ALOGV(" added new entry for %d", audioSession);
2365}
2366
2367void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid)
2368{
2369    Mutex::Autolock _l(mLock);
2370    pid_t caller = IPCThreadState::self()->getCallingPid();
2371    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
2372    if (pid != -1 && (caller == getpid_cached)) {
2373        caller = pid;
2374    }
2375    size_t num = mAudioSessionRefs.size();
2376    for (size_t i = 0; i< num; i++) {
2377        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2378        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2379            ref->mCnt--;
2380            ALOGV(" decremented refcount to %d", ref->mCnt);
2381            if (ref->mCnt == 0) {
2382                mAudioSessionRefs.removeAt(i);
2383                delete ref;
2384                purgeStaleEffects_l();
2385            }
2386            return;
2387        }
2388    }
2389    // If the caller is mediaserver it is likely that the session being released was acquired
2390    // on behalf of a process not in notification clients and we ignore the warning.
2391    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2392}
2393
2394void AudioFlinger::purgeStaleEffects_l() {
2395
2396    ALOGV("purging stale effects");
2397
2398    Vector< sp<EffectChain> > chains;
2399
2400    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2401        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2402        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2403            sp<EffectChain> ec = t->mEffectChains[j];
2404            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2405                chains.push(ec);
2406            }
2407        }
2408    }
2409    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2410        sp<RecordThread> t = mRecordThreads.valueAt(i);
2411        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2412            sp<EffectChain> ec = t->mEffectChains[j];
2413            chains.push(ec);
2414        }
2415    }
2416
2417    for (size_t i = 0; i < chains.size(); i++) {
2418        sp<EffectChain> ec = chains[i];
2419        int sessionid = ec->sessionId();
2420        sp<ThreadBase> t = ec->mThread.promote();
2421        if (t == 0) {
2422            continue;
2423        }
2424        size_t numsessionrefs = mAudioSessionRefs.size();
2425        bool found = false;
2426        for (size_t k = 0; k < numsessionrefs; k++) {
2427            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2428            if (ref->mSessionid == sessionid) {
2429                ALOGV(" session %d still exists for %d with %d refs",
2430                    sessionid, ref->mPid, ref->mCnt);
2431                found = true;
2432                break;
2433            }
2434        }
2435        if (!found) {
2436            Mutex::Autolock _l(t->mLock);
2437            // remove all effects from the chain
2438            while (ec->mEffects.size()) {
2439                sp<EffectModule> effect = ec->mEffects[0];
2440                effect->unPin();
2441                t->removeEffect_l(effect);
2442                if (effect->purgeHandles()) {
2443                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2444                }
2445                AudioSystem::unregisterEffect(effect->id());
2446            }
2447        }
2448    }
2449    return;
2450}
2451
2452// checkThread_l() must be called with AudioFlinger::mLock held
2453AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const
2454{
2455    ThreadBase *thread = NULL;
2456    switch (audio_unique_id_get_use(ioHandle)) {
2457    case AUDIO_UNIQUE_ID_USE_OUTPUT:
2458        thread = checkPlaybackThread_l(ioHandle);
2459        break;
2460    case AUDIO_UNIQUE_ID_USE_INPUT:
2461        thread = checkRecordThread_l(ioHandle);
2462        break;
2463    default:
2464        break;
2465    }
2466    return thread;
2467}
2468
2469// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2470AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2471{
2472    return mPlaybackThreads.valueFor(output).get();
2473}
2474
2475// checkMixerThread_l() must be called with AudioFlinger::mLock held
2476AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2477{
2478    PlaybackThread *thread = checkPlaybackThread_l(output);
2479    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2480}
2481
2482// checkRecordThread_l() must be called with AudioFlinger::mLock held
2483AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2484{
2485    return mRecordThreads.valueFor(input).get();
2486}
2487
2488audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use)
2489{
2490    // This is the internal API, so it is OK to assert on bad parameter.
2491    LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX);
2492    const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1;
2493    for (int retry = 0; retry < maxRetries; retry++) {
2494        // The cast allows wraparound from max positive to min negative instead of abort
2495        uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use],
2496                (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel);
2497        ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED);
2498        // allow wrap by skipping 0 and -1 for session ids
2499        if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) {
2500            ALOGW_IF(retry != 0, "unique ID overflow for use %d", use);
2501            return (audio_unique_id_t) (base | use);
2502        }
2503    }
2504    // We have no way of recovering from wraparound
2505    LOG_ALWAYS_FATAL("unique ID overflow for use %d", use);
2506    // TODO Use a floor after wraparound.  This may need a mutex.
2507}
2508
2509AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2510{
2511    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2512        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2513        if(thread->isDuplicating()) {
2514            continue;
2515        }
2516        AudioStreamOut *output = thread->getOutput();
2517        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2518            return thread;
2519        }
2520    }
2521    return NULL;
2522}
2523
2524audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2525{
2526    PlaybackThread *thread = primaryPlaybackThread_l();
2527
2528    if (thread == NULL) {
2529        return 0;
2530    }
2531
2532    return thread->outDevice();
2533}
2534
2535sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2536                                    audio_session_t triggerSession,
2537                                    audio_session_t listenerSession,
2538                                    sync_event_callback_t callBack,
2539                                    wp<RefBase> cookie)
2540{
2541    Mutex::Autolock _l(mLock);
2542
2543    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2544    status_t playStatus = NAME_NOT_FOUND;
2545    status_t recStatus = NAME_NOT_FOUND;
2546    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2547        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2548        if (playStatus == NO_ERROR) {
2549            return event;
2550        }
2551    }
2552    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2553        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2554        if (recStatus == NO_ERROR) {
2555            return event;
2556        }
2557    }
2558    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2559        mPendingSyncEvents.add(event);
2560    } else {
2561        ALOGV("createSyncEvent() invalid event %d", event->type());
2562        event.clear();
2563    }
2564    return event;
2565}
2566
2567// ----------------------------------------------------------------------------
2568//  Effect management
2569// ----------------------------------------------------------------------------
2570
2571
2572status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2573{
2574    Mutex::Autolock _l(mLock);
2575    return EffectQueryNumberEffects(numEffects);
2576}
2577
2578status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2579{
2580    Mutex::Autolock _l(mLock);
2581    return EffectQueryEffect(index, descriptor);
2582}
2583
2584status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2585        effect_descriptor_t *descriptor) const
2586{
2587    Mutex::Autolock _l(mLock);
2588    return EffectGetDescriptor(pUuid, descriptor);
2589}
2590
2591
2592sp<IEffect> AudioFlinger::createEffect(
2593        effect_descriptor_t *pDesc,
2594        const sp<IEffectClient>& effectClient,
2595        int32_t priority,
2596        audio_io_handle_t io,
2597        audio_session_t sessionId,
2598        const String16& opPackageName,
2599        status_t *status,
2600        int *id,
2601        int *enabled)
2602{
2603    status_t lStatus = NO_ERROR;
2604    sp<EffectHandle> handle;
2605    effect_descriptor_t desc;
2606
2607    pid_t pid = IPCThreadState::self()->getCallingPid();
2608    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2609            pid, effectClient.get(), priority, sessionId, io);
2610
2611    if (pDesc == NULL) {
2612        lStatus = BAD_VALUE;
2613        goto Exit;
2614    }
2615
2616    // check audio settings permission for global effects
2617    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2618        lStatus = PERMISSION_DENIED;
2619        goto Exit;
2620    }
2621
2622    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2623    // that can only be created by audio policy manager (running in same process)
2624    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2625        lStatus = PERMISSION_DENIED;
2626        goto Exit;
2627    }
2628
2629    {
2630        if (!EffectIsNullUuid(&pDesc->uuid)) {
2631            // if uuid is specified, request effect descriptor
2632            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2633            if (lStatus < 0) {
2634                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2635                goto Exit;
2636            }
2637        } else {
2638            // if uuid is not specified, look for an available implementation
2639            // of the required type in effect factory
2640            if (EffectIsNullUuid(&pDesc->type)) {
2641                ALOGW("createEffect() no effect type");
2642                lStatus = BAD_VALUE;
2643                goto Exit;
2644            }
2645            uint32_t numEffects = 0;
2646            effect_descriptor_t d;
2647            d.flags = 0; // prevent compiler warning
2648            bool found = false;
2649
2650            lStatus = EffectQueryNumberEffects(&numEffects);
2651            if (lStatus < 0) {
2652                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2653                goto Exit;
2654            }
2655            for (uint32_t i = 0; i < numEffects; i++) {
2656                lStatus = EffectQueryEffect(i, &desc);
2657                if (lStatus < 0) {
2658                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2659                    continue;
2660                }
2661                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2662                    // If matching type found save effect descriptor. If the session is
2663                    // 0 and the effect is not auxiliary, continue enumeration in case
2664                    // an auxiliary version of this effect type is available
2665                    found = true;
2666                    d = desc;
2667                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2668                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2669                        break;
2670                    }
2671                }
2672            }
2673            if (!found) {
2674                lStatus = BAD_VALUE;
2675                ALOGW("createEffect() effect not found");
2676                goto Exit;
2677            }
2678            // For same effect type, chose auxiliary version over insert version if
2679            // connect to output mix (Compliance to OpenSL ES)
2680            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2681                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2682                desc = d;
2683            }
2684        }
2685
2686        // Do not allow auxiliary effects on a session different from 0 (output mix)
2687        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2688             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2689            lStatus = INVALID_OPERATION;
2690            goto Exit;
2691        }
2692
2693        // check recording permission for visualizer
2694        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2695            !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) {
2696            lStatus = PERMISSION_DENIED;
2697            goto Exit;
2698        }
2699
2700        // return effect descriptor
2701        *pDesc = desc;
2702        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2703            // if the output returned by getOutputForEffect() is removed before we lock the
2704            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2705            // and we will exit safely
2706            io = AudioSystem::getOutputForEffect(&desc);
2707            ALOGV("createEffect got output %d", io);
2708        }
2709
2710        Mutex::Autolock _l(mLock);
2711
2712        // If output is not specified try to find a matching audio session ID in one of the
2713        // output threads.
2714        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2715        // because of code checking output when entering the function.
2716        // Note: io is never 0 when creating an effect on an input
2717        if (io == AUDIO_IO_HANDLE_NONE) {
2718            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2719                // output must be specified by AudioPolicyManager when using session
2720                // AUDIO_SESSION_OUTPUT_STAGE
2721                lStatus = BAD_VALUE;
2722                goto Exit;
2723            }
2724            // look for the thread where the specified audio session is present
2725            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2726                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2727                    io = mPlaybackThreads.keyAt(i);
2728                    break;
2729                }
2730            }
2731            if (io == 0) {
2732                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2733                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2734                        io = mRecordThreads.keyAt(i);
2735                        break;
2736                    }
2737                }
2738            }
2739            // If no output thread contains the requested session ID, default to
2740            // first output. The effect chain will be moved to the correct output
2741            // thread when a track with the same session ID is created
2742            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2743                io = mPlaybackThreads.keyAt(0);
2744            }
2745            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2746        }
2747        ThreadBase *thread = checkRecordThread_l(io);
2748        if (thread == NULL) {
2749            thread = checkPlaybackThread_l(io);
2750            if (thread == NULL) {
2751                ALOGE("createEffect() unknown output thread");
2752                lStatus = BAD_VALUE;
2753                goto Exit;
2754            }
2755        } else {
2756            // Check if one effect chain was awaiting for an effect to be created on this
2757            // session and used it instead of creating a new one.
2758            sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
2759            if (chain != 0) {
2760                Mutex::Autolock _l(thread->mLock);
2761                thread->addEffectChain_l(chain);
2762            }
2763        }
2764
2765        sp<Client> client = registerPid(pid);
2766
2767        // create effect on selected output thread
2768        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2769                &desc, enabled, &lStatus);
2770        if (handle != 0 && id != NULL) {
2771            *id = handle->id();
2772        }
2773        if (handle == 0) {
2774            // remove local strong reference to Client with mClientLock held
2775            Mutex::Autolock _cl(mClientLock);
2776            client.clear();
2777        }
2778    }
2779
2780Exit:
2781    *status = lStatus;
2782    return handle;
2783}
2784
2785status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
2786        audio_io_handle_t dstOutput)
2787{
2788    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2789            sessionId, srcOutput, dstOutput);
2790    Mutex::Autolock _l(mLock);
2791    if (srcOutput == dstOutput) {
2792        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2793        return NO_ERROR;
2794    }
2795    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2796    if (srcThread == NULL) {
2797        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2798        return BAD_VALUE;
2799    }
2800    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2801    if (dstThread == NULL) {
2802        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2803        return BAD_VALUE;
2804    }
2805
2806    Mutex::Autolock _dl(dstThread->mLock);
2807    Mutex::Autolock _sl(srcThread->mLock);
2808    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2809}
2810
2811// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2812status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId,
2813                                   AudioFlinger::PlaybackThread *srcThread,
2814                                   AudioFlinger::PlaybackThread *dstThread,
2815                                   bool reRegister)
2816{
2817    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2818            sessionId, srcThread, dstThread);
2819
2820    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2821    if (chain == 0) {
2822        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2823                sessionId, srcThread);
2824        return INVALID_OPERATION;
2825    }
2826
2827    // Check whether the destination thread has a channel count of FCC_2, which is
2828    // currently required for (most) effects. Prevent moving the effect chain here rather
2829    // than disabling the addEffect_l() call in dstThread below.
2830    if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) &&
2831            dstThread->mChannelCount != FCC_2) {
2832        ALOGW("moveEffectChain_l() effect chain failed because"
2833                " destination thread %p channel count(%u) != %u",
2834                dstThread, dstThread->mChannelCount, FCC_2);
2835        return INVALID_OPERATION;
2836    }
2837
2838    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2839    // so that a new chain is created with correct parameters when first effect is added. This is
2840    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2841    // removed.
2842    srcThread->removeEffectChain_l(chain);
2843
2844    // transfer all effects one by one so that new effect chain is created on new thread with
2845    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2846    sp<EffectChain> dstChain;
2847    uint32_t strategy = 0; // prevent compiler warning
2848    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2849    Vector< sp<EffectModule> > removed;
2850    status_t status = NO_ERROR;
2851    while (effect != 0) {
2852        srcThread->removeEffect_l(effect);
2853        removed.add(effect);
2854        status = dstThread->addEffect_l(effect);
2855        if (status != NO_ERROR) {
2856            break;
2857        }
2858        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2859        if (effect->state() == EffectModule::ACTIVE ||
2860                effect->state() == EffectModule::STOPPING) {
2861            effect->start();
2862        }
2863        // if the move request is not received from audio policy manager, the effect must be
2864        // re-registered with the new strategy and output
2865        if (dstChain == 0) {
2866            dstChain = effect->chain().promote();
2867            if (dstChain == 0) {
2868                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2869                status = NO_INIT;
2870                break;
2871            }
2872            strategy = dstChain->strategy();
2873        }
2874        if (reRegister) {
2875            AudioSystem::unregisterEffect(effect->id());
2876            AudioSystem::registerEffect(&effect->desc(),
2877                                        dstThread->id(),
2878                                        strategy,
2879                                        sessionId,
2880                                        effect->id());
2881            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2882        }
2883        effect = chain->getEffectFromId_l(0);
2884    }
2885
2886    if (status != NO_ERROR) {
2887        for (size_t i = 0; i < removed.size(); i++) {
2888            srcThread->addEffect_l(removed[i]);
2889            if (dstChain != 0 && reRegister) {
2890                AudioSystem::unregisterEffect(removed[i]->id());
2891                AudioSystem::registerEffect(&removed[i]->desc(),
2892                                            srcThread->id(),
2893                                            strategy,
2894                                            sessionId,
2895                                            removed[i]->id());
2896                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2897            }
2898        }
2899    }
2900
2901    return status;
2902}
2903
2904bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2905{
2906    if (mGlobalEffectEnableTime != 0 &&
2907            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2908        return true;
2909    }
2910
2911    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2912        sp<EffectChain> ec =
2913                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2914        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2915            return true;
2916        }
2917    }
2918    return false;
2919}
2920
2921void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2922{
2923    Mutex::Autolock _l(mLock);
2924
2925    mGlobalEffectEnableTime = systemTime();
2926
2927    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2928        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2929        if (t->mType == ThreadBase::OFFLOAD) {
2930            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2931        }
2932    }
2933
2934}
2935
2936status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain)
2937{
2938    audio_session_t session = chain->sessionId();
2939    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2940    ALOGV("putOrphanEffectChain_l session %d index %zd", session, index);
2941    if (index >= 0) {
2942        ALOGW("putOrphanEffectChain_l chain for session %d already present", session);
2943        return ALREADY_EXISTS;
2944    }
2945    mOrphanEffectChains.add(session, chain);
2946    return NO_ERROR;
2947}
2948
2949sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session)
2950{
2951    sp<EffectChain> chain;
2952    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2953    ALOGV("getOrphanEffectChain_l session %d index %zd", session, index);
2954    if (index >= 0) {
2955        chain = mOrphanEffectChains.valueAt(index);
2956        mOrphanEffectChains.removeItemsAt(index);
2957    }
2958    return chain;
2959}
2960
2961bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect)
2962{
2963    Mutex::Autolock _l(mLock);
2964    audio_session_t session = effect->sessionId();
2965    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2966    ALOGV("updateOrphanEffectChains session %d index %zd", session, index);
2967    if (index >= 0) {
2968        sp<EffectChain> chain = mOrphanEffectChains.valueAt(index);
2969        if (chain->removeEffect_l(effect) == 0) {
2970            ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index);
2971            mOrphanEffectChains.removeItemsAt(index);
2972        }
2973        return true;
2974    }
2975    return false;
2976}
2977
2978
2979struct Entry {
2980#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21
2981    char mFileName[TEE_MAX_FILENAME];
2982};
2983
2984int comparEntry(const void *p1, const void *p2)
2985{
2986    return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName);
2987}
2988
2989#ifdef TEE_SINK
2990void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2991{
2992    NBAIO_Source *teeSource = source.get();
2993    if (teeSource != NULL) {
2994        // .wav rotation
2995        // There is a benign race condition if 2 threads call this simultaneously.
2996        // They would both traverse the directory, but the result would simply be
2997        // failures at unlink() which are ignored.  It's also unlikely since
2998        // normally dumpsys is only done by bugreport or from the command line.
2999        char teePath[32+256];
3000        strcpy(teePath, "/data/misc/audioserver");
3001        size_t teePathLen = strlen(teePath);
3002        DIR *dir = opendir(teePath);
3003        teePath[teePathLen++] = '/';
3004        if (dir != NULL) {
3005#define TEE_MAX_SORT 20 // number of entries to sort
3006#define TEE_MAX_KEEP 10 // number of entries to keep
3007            struct Entry entries[TEE_MAX_SORT];
3008            size_t entryCount = 0;
3009            while (entryCount < TEE_MAX_SORT) {
3010                struct dirent de;
3011                struct dirent *result = NULL;
3012                int rc = readdir_r(dir, &de, &result);
3013                if (rc != 0) {
3014                    ALOGW("readdir_r failed %d", rc);
3015                    break;
3016                }
3017                if (result == NULL) {
3018                    break;
3019                }
3020                if (result != &de) {
3021                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
3022                    break;
3023                }
3024                // ignore non .wav file entries
3025                size_t nameLen = strlen(de.d_name);
3026                if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME ||
3027                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
3028                    continue;
3029                }
3030                strcpy(entries[entryCount++].mFileName, de.d_name);
3031            }
3032            (void) closedir(dir);
3033            if (entryCount > TEE_MAX_KEEP) {
3034                qsort(entries, entryCount, sizeof(Entry), comparEntry);
3035                for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) {
3036                    strcpy(&teePath[teePathLen], entries[i].mFileName);
3037                    (void) unlink(teePath);
3038                }
3039            }
3040        } else {
3041            if (fd >= 0) {
3042                dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath,
3043                        strerror(errno));
3044            }
3045        }
3046        char teeTime[16];
3047        struct timeval tv;
3048        gettimeofday(&tv, NULL);
3049        struct tm tm;
3050        localtime_r(&tv.tv_sec, &tm);
3051        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
3052        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
3053        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
3054        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
3055        if (teeFd >= 0) {
3056            // FIXME use libsndfile
3057            char wavHeader[44];
3058            memcpy(wavHeader,
3059                "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",
3060                sizeof(wavHeader));
3061            NBAIO_Format format = teeSource->format();
3062            unsigned channelCount = Format_channelCount(format);
3063            uint32_t sampleRate = Format_sampleRate(format);
3064            size_t frameSize = Format_frameSize(format);
3065            wavHeader[22] = channelCount;       // number of channels
3066            wavHeader[24] = sampleRate;         // sample rate
3067            wavHeader[25] = sampleRate >> 8;
3068            wavHeader[32] = frameSize;          // block alignment
3069            wavHeader[33] = frameSize >> 8;
3070            write(teeFd, wavHeader, sizeof(wavHeader));
3071            size_t total = 0;
3072            bool firstRead = true;
3073#define TEE_SINK_READ 1024                      // frames per I/O operation
3074            void *buffer = malloc(TEE_SINK_READ * frameSize);
3075            for (;;) {
3076                size_t count = TEE_SINK_READ;
3077                ssize_t actual = teeSource->read(buffer, count);
3078                bool wasFirstRead = firstRead;
3079                firstRead = false;
3080                if (actual <= 0) {
3081                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
3082                        continue;
3083                    }
3084                    break;
3085                }
3086                ALOG_ASSERT(actual <= (ssize_t)count);
3087                write(teeFd, buffer, actual * frameSize);
3088                total += actual;
3089            }
3090            free(buffer);
3091            lseek(teeFd, (off_t) 4, SEEK_SET);
3092            uint32_t temp = 44 + total * frameSize - 8;
3093            // FIXME not big-endian safe
3094            write(teeFd, &temp, sizeof(temp));
3095            lseek(teeFd, (off_t) 40, SEEK_SET);
3096            temp =  total * frameSize;
3097            // FIXME not big-endian safe
3098            write(teeFd, &temp, sizeof(temp));
3099            close(teeFd);
3100            if (fd >= 0) {
3101                dprintf(fd, "tee copied to %s\n", teePath);
3102            }
3103        } else {
3104            if (fd >= 0) {
3105                dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
3106            }
3107        }
3108    }
3109}
3110#endif
3111
3112// ----------------------------------------------------------------------------
3113
3114status_t AudioFlinger::onTransact(
3115        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
3116{
3117    return BnAudioFlinger::onTransact(code, data, reply, flags);
3118}
3119
3120} // namespace android
3121