AudioFlinger.cpp revision 4a3d5c23f79189eb7ab9f31c440c7da5b15947a2 
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 "EffectsFactoryHalInterface.h"
48#include "ServiceUtilities.h"
49
50#include <media/AudioResamplerPublic.h>
51
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";
88static const char kNoEffectsFactory[] = "Effects Factory is absent\n";
89
90
91nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
92
93uint32_t AudioFlinger::mScreenState;
94
95#ifdef TEE_SINK
96bool AudioFlinger::mTeeSinkInputEnabled = false;
97bool AudioFlinger::mTeeSinkOutputEnabled = false;
98bool AudioFlinger::mTeeSinkTrackEnabled = false;
99
100size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
101size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
102size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
103#endif
104
105// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
106// we define a minimum time during which a global effect is considered enabled.
107static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
108
109// ----------------------------------------------------------------------------
110
111const char *formatToString(audio_format_t format) {
112    switch (audio_get_main_format(format)) {
113    case AUDIO_FORMAT_PCM:
114        switch (format) {
115        case AUDIO_FORMAT_PCM_16_BIT: return "pcm16";
116        case AUDIO_FORMAT_PCM_8_BIT: return "pcm8";
117        case AUDIO_FORMAT_PCM_32_BIT: return "pcm32";
118        case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24";
119        case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat";
120        case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24";
121        default:
122            break;
123        }
124        break;
125    case AUDIO_FORMAT_MP3: return "mp3";
126    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
127    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
128    case AUDIO_FORMAT_AAC: return "aac";
129    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
130    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
131    case AUDIO_FORMAT_VORBIS: return "vorbis";
132    case AUDIO_FORMAT_OPUS: return "opus";
133    case AUDIO_FORMAT_AC3: return "ac-3";
134    case AUDIO_FORMAT_E_AC3: return "e-ac-3";
135    case AUDIO_FORMAT_IEC61937: return "iec61937";
136    case AUDIO_FORMAT_DTS: return "dts";
137    case AUDIO_FORMAT_DTS_HD: return "dts-hd";
138    case AUDIO_FORMAT_DOLBY_TRUEHD: return "dolby-truehd";
139    default:
140        break;
141    }
142    return "unknown";
143}
144
145static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
146{
147    const hw_module_t *mod;
148    int rc;
149
150    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
151    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
152                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
153    if (rc) {
154        goto out;
155    }
156    rc = audio_hw_device_open(mod, dev);
157    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
158                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
159    if (rc) {
160        goto out;
161    }
162    if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
163        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
164        rc = BAD_VALUE;
165        goto out;
166    }
167    return 0;
168
169out:
170    *dev = NULL;
171    return rc;
172}
173
174// ----------------------------------------------------------------------------
175
176AudioFlinger::AudioFlinger()
177    : BnAudioFlinger(),
178      mPrimaryHardwareDev(NULL),
179      mAudioHwDevs(NULL),
180      mHardwareStatus(AUDIO_HW_IDLE),
181      mMasterVolume(1.0f),
182      mMasterMute(false),
183      // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX),
184      mMode(AUDIO_MODE_INVALID),
185      mBtNrecIsOff(false),
186      mIsLowRamDevice(true),
187      mIsDeviceTypeKnown(false),
188      mGlobalEffectEnableTime(0),
189      mSystemReady(false)
190{
191    // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum
192    for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) {
193        // zero ID has a special meaning, so unavailable
194        mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX;
195    }
196
197    getpid_cached = getpid();
198    const bool doLog = property_get_bool("ro.test_harness", false);
199    if (doLog) {
200        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
201                MemoryHeapBase::READ_ONLY);
202    }
203
204    // reset battery stats.
205    // if the audio service has crashed, battery stats could be left
206    // in bad state, reset the state upon service start.
207    BatteryNotifier::getInstance().noteResetAudio();
208
209    mEffectsFactoryHal = EffectsFactoryHalInterface::create();
210
211#ifdef TEE_SINK
212    char value[PROPERTY_VALUE_MAX];
213    (void) property_get("ro.debuggable", value, "0");
214    int debuggable = atoi(value);
215    int teeEnabled = 0;
216    if (debuggable) {
217        (void) property_get("af.tee", value, "0");
218        teeEnabled = atoi(value);
219    }
220    // FIXME symbolic constants here
221    if (teeEnabled & 1) {
222        mTeeSinkInputEnabled = true;
223    }
224    if (teeEnabled & 2) {
225        mTeeSinkOutputEnabled = true;
226    }
227    if (teeEnabled & 4) {
228        mTeeSinkTrackEnabled = true;
229    }
230#endif
231}
232
233void AudioFlinger::onFirstRef()
234{
235    Mutex::Autolock _l(mLock);
236
237    /* TODO: move all this work into an Init() function */
238    char val_str[PROPERTY_VALUE_MAX] = { 0 };
239    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
240        uint32_t int_val;
241        if (1 == sscanf(val_str, "%u", &int_val)) {
242            mStandbyTimeInNsecs = milliseconds(int_val);
243            ALOGI("Using %u mSec as standby time.", int_val);
244        } else {
245            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
246            ALOGI("Using default %u mSec as standby time.",
247                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
248        }
249    }
250
251    mPatchPanel = new PatchPanel(this);
252    // FIXME: bug 30737845: trigger audioserver restart if main audioflinger lock
253    // is held continuously for more than 3 seconds
254    mLockWatch = new LockWatch(mLock, String8("AudioFlinger"));
255    mMode = AUDIO_MODE_NORMAL;
256}
257
258AudioFlinger::~AudioFlinger()
259{
260    while (!mRecordThreads.isEmpty()) {
261        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
262        closeInput_nonvirtual(mRecordThreads.keyAt(0));
263    }
264    while (!mPlaybackThreads.isEmpty()) {
265        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
266        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
267    }
268
269    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
270        // no mHardwareLock needed, as there are no other references to this
271        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
272        delete mAudioHwDevs.valueAt(i);
273    }
274
275    // Tell media.log service about any old writers that still need to be unregistered
276    if (mLogMemoryDealer != 0) {
277        sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
278        if (binder != 0) {
279            sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
280            for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
281                sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
282                mUnregisteredWriters.pop();
283                mediaLogService->unregisterWriter(iMemory);
284            }
285        }
286    }
287    mLockWatch->requestExitAndWait();
288}
289
290static const char * const audio_interfaces[] = {
291    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
292    AUDIO_HARDWARE_MODULE_ID_A2DP,
293    AUDIO_HARDWARE_MODULE_ID_USB,
294};
295#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
296
297AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
298        audio_module_handle_t module,
299        audio_devices_t devices)
300{
301    // if module is 0, the request comes from an old policy manager and we should load
302    // well known modules
303    if (module == 0) {
304        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
305        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
306            loadHwModule_l(audio_interfaces[i]);
307        }
308        // then try to find a module supporting the requested device.
309        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
310            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
311            audio_hw_device_t *dev = audioHwDevice->hwDevice();
312            if ((dev->get_supported_devices != NULL) &&
313                    (dev->get_supported_devices(dev) & devices) == devices)
314                return audioHwDevice;
315        }
316    } else {
317        // check a match for the requested module handle
318        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
319        if (audioHwDevice != NULL) {
320            return audioHwDevice;
321        }
322    }
323
324    return NULL;
325}
326
327void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
328{
329    const size_t SIZE = 256;
330    char buffer[SIZE];
331    String8 result;
332
333    result.append("Clients:\n");
334    for (size_t i = 0; i < mClients.size(); ++i) {
335        sp<Client> client = mClients.valueAt(i).promote();
336        if (client != 0) {
337            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
338            result.append(buffer);
339        }
340    }
341
342    result.append("Notification Clients:\n");
343    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
344        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
345        result.append(buffer);
346    }
347
348    result.append("Global session refs:\n");
349    result.append("  session   pid count\n");
350    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
351        AudioSessionRef *r = mAudioSessionRefs[i];
352        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
353        result.append(buffer);
354    }
355    write(fd, result.string(), result.size());
356}
357
358
359void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
360{
361    const size_t SIZE = 256;
362    char buffer[SIZE];
363    String8 result;
364    hardware_call_state hardwareStatus = mHardwareStatus;
365
366    snprintf(buffer, SIZE, "Hardware status: %d\n"
367                           "Standby Time mSec: %u\n",
368                            hardwareStatus,
369                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
370    result.append(buffer);
371    write(fd, result.string(), result.size());
372}
373
374void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
375{
376    const size_t SIZE = 256;
377    char buffer[SIZE];
378    String8 result;
379    snprintf(buffer, SIZE, "Permission Denial: "
380            "can't dump AudioFlinger from pid=%d, uid=%d\n",
381            IPCThreadState::self()->getCallingPid(),
382            IPCThreadState::self()->getCallingUid());
383    result.append(buffer);
384    write(fd, result.string(), result.size());
385}
386
387bool AudioFlinger::dumpTryLock(Mutex& mutex)
388{
389    bool locked = false;
390    for (int i = 0; i < kDumpLockRetries; ++i) {
391        if (mutex.tryLock() == NO_ERROR) {
392            locked = true;
393            break;
394        }
395        usleep(kDumpLockSleepUs);
396    }
397    return locked;
398}
399
400status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
401{
402    if (!dumpAllowed()) {
403        dumpPermissionDenial(fd, args);
404    } else {
405        // get state of hardware lock
406        bool hardwareLocked = dumpTryLock(mHardwareLock);
407        if (!hardwareLocked) {
408            String8 result(kHardwareLockedString);
409            write(fd, result.string(), result.size());
410        } else {
411            mHardwareLock.unlock();
412        }
413
414        bool locked = dumpTryLock(mLock);
415
416        // failed to lock - AudioFlinger is probably deadlocked
417        if (!locked) {
418            String8 result(kDeadlockedString);
419            write(fd, result.string(), result.size());
420        }
421
422        bool clientLocked = dumpTryLock(mClientLock);
423        if (!clientLocked) {
424            String8 result(kClientLockedString);
425            write(fd, result.string(), result.size());
426        }
427
428        if (mEffectsFactoryHal.get() != NULL) {
429            mEffectsFactoryHal->dumpEffects(fd);
430        } else {
431            String8 result(kNoEffectsFactory);
432            write(fd, result.string(), result.size());
433        }
434
435        dumpClients(fd, args);
436        if (clientLocked) {
437            mClientLock.unlock();
438        }
439
440        dumpInternals(fd, args);
441
442        // dump playback threads
443        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
444            mPlaybackThreads.valueAt(i)->dump(fd, args);
445        }
446
447        // dump record threads
448        for (size_t i = 0; i < mRecordThreads.size(); i++) {
449            mRecordThreads.valueAt(i)->dump(fd, args);
450        }
451
452        // dump orphan effect chains
453        if (mOrphanEffectChains.size() != 0) {
454            write(fd, "  Orphan Effect Chains\n", strlen("  Orphan Effect Chains\n"));
455            for (size_t i = 0; i < mOrphanEffectChains.size(); i++) {
456                mOrphanEffectChains.valueAt(i)->dump(fd, args);
457            }
458        }
459        // dump all hardware devs
460        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
461            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
462            dev->dump(dev, fd);
463        }
464
465#ifdef TEE_SINK
466        // dump the serially shared record tee sink
467        if (mRecordTeeSource != 0) {
468            dumpTee(fd, mRecordTeeSource);
469        }
470#endif
471
472        if (locked) {
473            mLock.unlock();
474        }
475
476        // append a copy of media.log here by forwarding fd to it, but don't attempt
477        // to lookup the service if it's not running, as it will block for a second
478        if (mLogMemoryDealer != 0) {
479            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
480            if (binder != 0) {
481                dprintf(fd, "\nmedia.log:\n");
482                Vector<String16> args;
483                binder->dump(fd, args);
484            }
485        }
486
487        // check for optional arguments
488        bool dumpMem = false;
489        bool unreachableMemory = false;
490        for (const auto &arg : args) {
491            if (arg == String16("-m")) {
492                dumpMem = true;
493            } else if (arg == String16("--unreachable")) {
494                unreachableMemory = true;
495            }
496        }
497
498        if (dumpMem) {
499            dprintf(fd, "\nDumping memory:\n");
500            std::string s = dumpMemoryAddresses(100 /* limit */);
501            write(fd, s.c_str(), s.size());
502        }
503        if (unreachableMemory) {
504            dprintf(fd, "\nDumping unreachable memory:\n");
505            // TODO - should limit be an argument parameter?
506            std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */);
507            write(fd, s.c_str(), s.size());
508        }
509    }
510    return NO_ERROR;
511}
512
513sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
514{
515    Mutex::Autolock _cl(mClientLock);
516    // If pid is already in the mClients wp<> map, then use that entry
517    // (for which promote() is always != 0), otherwise create a new entry and Client.
518    sp<Client> client = mClients.valueFor(pid).promote();
519    if (client == 0) {
520        client = new Client(this, pid);
521        mClients.add(pid, client);
522    }
523
524    return client;
525}
526
527sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
528{
529    // If there is no memory allocated for logs, return a dummy writer that does nothing
530    if (mLogMemoryDealer == 0) {
531        return new NBLog::Writer();
532    }
533    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
534    // Similarly if we can't contact the media.log service, also return a dummy writer
535    if (binder == 0) {
536        return new NBLog::Writer();
537    }
538    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
539    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
540    // If allocation fails, consult the vector of previously unregistered writers
541    // and garbage-collect one or more them until an allocation succeeds
542    if (shared == 0) {
543        Mutex::Autolock _l(mUnregisteredWritersLock);
544        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
545            {
546                // Pick the oldest stale writer to garbage-collect
547                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
548                mUnregisteredWriters.removeAt(0);
549                mediaLogService->unregisterWriter(iMemory);
550                // Now the media.log remote reference to IMemory is gone.  When our last local
551                // reference to IMemory also drops to zero at end of this block,
552                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
553            }
554            // Re-attempt the allocation
555            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
556            if (shared != 0) {
557                goto success;
558            }
559        }
560        // Even after garbage-collecting all old writers, there is still not enough memory,
561        // so return a dummy writer
562        return new NBLog::Writer();
563    }
564success:
565    mediaLogService->registerWriter(shared, size, name);
566    return new NBLog::Writer(size, shared);
567}
568
569void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
570{
571    if (writer == 0) {
572        return;
573    }
574    sp<IMemory> iMemory(writer->getIMemory());
575    if (iMemory == 0) {
576        return;
577    }
578    // Rather than removing the writer immediately, append it to a queue of old writers to
579    // be garbage-collected later.  This allows us to continue to view old logs for a while.
580    Mutex::Autolock _l(mUnregisteredWritersLock);
581    mUnregisteredWriters.push(writer);
582}
583
584// IAudioFlinger interface
585
586
587sp<IAudioTrack> AudioFlinger::createTrack(
588        audio_stream_type_t streamType,
589        uint32_t sampleRate,
590        audio_format_t format,
591        audio_channel_mask_t channelMask,
592        size_t *frameCount,
593        audio_output_flags_t *flags,
594        const sp<IMemory>& sharedBuffer,
595        audio_io_handle_t output,
596        pid_t pid,
597        pid_t tid,
598        audio_session_t *sessionId,
599        int clientUid,
600        status_t *status)
601{
602    sp<PlaybackThread::Track> track;
603    sp<TrackHandle> trackHandle;
604    sp<Client> client;
605    status_t lStatus;
606    audio_session_t lSessionId;
607
608    const uid_t callingUid = IPCThreadState::self()->getCallingUid();
609    if (pid == -1 || !isTrustedCallingUid(callingUid)) {
610        const pid_t callingPid = IPCThreadState::self()->getCallingPid();
611        ALOGW_IF(pid != -1 && pid != callingPid,
612                 "%s uid %d pid %d tried to pass itself off as pid %d",
613                 __func__, callingUid, callingPid, pid);
614        pid = callingPid;
615    }
616
617    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
618    // but if someone uses binder directly they could bypass that and cause us to crash
619    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
620        ALOGE("createTrack() invalid stream type %d", streamType);
621        lStatus = BAD_VALUE;
622        goto Exit;
623    }
624
625    // further sample rate checks are performed by createTrack_l() depending on the thread type
626    if (sampleRate == 0) {
627        ALOGE("createTrack() invalid sample rate %u", sampleRate);
628        lStatus = BAD_VALUE;
629        goto Exit;
630    }
631
632    // further channel mask checks are performed by createTrack_l() depending on the thread type
633    if (!audio_is_output_channel(channelMask)) {
634        ALOGE("createTrack() invalid channel mask %#x", channelMask);
635        lStatus = BAD_VALUE;
636        goto Exit;
637    }
638
639    // further format checks are performed by createTrack_l() depending on the thread type
640    if (!audio_is_valid_format(format)) {
641        ALOGE("createTrack() invalid format %#x", format);
642        lStatus = BAD_VALUE;
643        goto Exit;
644    }
645
646    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
647        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
648        lStatus = BAD_VALUE;
649        goto Exit;
650    }
651
652    {
653        Mutex::Autolock _l(mLock);
654        PlaybackThread *thread = checkPlaybackThread_l(output);
655        if (thread == NULL) {
656            ALOGE("no playback thread found for output handle %d", output);
657            lStatus = BAD_VALUE;
658            goto Exit;
659        }
660
661        client = registerPid(pid);
662
663        PlaybackThread *effectThread = NULL;
664        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
665            if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
666                ALOGE("createTrack() invalid session ID %d", *sessionId);
667                lStatus = BAD_VALUE;
668                goto Exit;
669            }
670            lSessionId = *sessionId;
671            // check if an effect chain with the same session ID is present on another
672            // output thread and move it here.
673            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
674                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
675                if (mPlaybackThreads.keyAt(i) != output) {
676                    uint32_t sessions = t->hasAudioSession(lSessionId);
677                    if (sessions & ThreadBase::EFFECT_SESSION) {
678                        effectThread = t.get();
679                        break;
680                    }
681                }
682            }
683        } else {
684            // if no audio session id is provided, create one here
685            lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
686            if (sessionId != NULL) {
687                *sessionId = lSessionId;
688            }
689        }
690        ALOGV("createTrack() lSessionId: %d", lSessionId);
691
692        track = thread->createTrack_l(client, streamType, sampleRate, format,
693                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
694        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
695        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
696
697        // move effect chain to this output thread if an effect on same session was waiting
698        // for a track to be created
699        if (lStatus == NO_ERROR && effectThread != NULL) {
700            // no risk of deadlock because AudioFlinger::mLock is held
701            Mutex::Autolock _dl(thread->mLock);
702            Mutex::Autolock _sl(effectThread->mLock);
703            moveEffectChain_l(lSessionId, effectThread, thread, true);
704        }
705
706        // Look for sync events awaiting for a session to be used.
707        for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
708            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
709                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
710                    if (lStatus == NO_ERROR) {
711                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
712                    } else {
713                        mPendingSyncEvents[i]->cancel();
714                    }
715                    mPendingSyncEvents.removeAt(i);
716                    i--;
717                }
718            }
719        }
720
721        setAudioHwSyncForSession_l(thread, lSessionId);
722    }
723
724    if (lStatus != NO_ERROR) {
725        // remove local strong reference to Client before deleting the Track so that the
726        // Client destructor is called by the TrackBase destructor with mClientLock held
727        // Don't hold mClientLock when releasing the reference on the track as the
728        // destructor will acquire it.
729        {
730            Mutex::Autolock _cl(mClientLock);
731            client.clear();
732        }
733        track.clear();
734        goto Exit;
735    }
736
737    // return handle to client
738    trackHandle = new TrackHandle(track);
739
740Exit:
741    *status = lStatus;
742    return trackHandle;
743}
744
745uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const
746{
747    Mutex::Autolock _l(mLock);
748    ThreadBase *thread = checkThread_l(ioHandle);
749    if (thread == NULL) {
750        ALOGW("sampleRate() unknown thread %d", ioHandle);
751        return 0;
752    }
753    return thread->sampleRate();
754}
755
756audio_format_t AudioFlinger::format(audio_io_handle_t output) const
757{
758    Mutex::Autolock _l(mLock);
759    PlaybackThread *thread = checkPlaybackThread_l(output);
760    if (thread == NULL) {
761        ALOGW("format() unknown thread %d", output);
762        return AUDIO_FORMAT_INVALID;
763    }
764    return thread->format();
765}
766
767size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const
768{
769    Mutex::Autolock _l(mLock);
770    ThreadBase *thread = checkThread_l(ioHandle);
771    if (thread == NULL) {
772        ALOGW("frameCount() unknown thread %d", ioHandle);
773        return 0;
774    }
775    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
776    //       should examine all callers and fix them to handle smaller counts
777    return thread->frameCount();
778}
779
780size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const
781{
782    Mutex::Autolock _l(mLock);
783    ThreadBase *thread = checkThread_l(ioHandle);
784    if (thread == NULL) {
785        ALOGW("frameCountHAL() unknown thread %d", ioHandle);
786        return 0;
787    }
788    return thread->frameCountHAL();
789}
790
791uint32_t AudioFlinger::latency(audio_io_handle_t output) const
792{
793    Mutex::Autolock _l(mLock);
794    PlaybackThread *thread = checkPlaybackThread_l(output);
795    if (thread == NULL) {
796        ALOGW("latency(): no playback thread found for output handle %d", output);
797        return 0;
798    }
799    return thread->latency();
800}
801
802status_t AudioFlinger::setMasterVolume(float value)
803{
804    status_t ret = initCheck();
805    if (ret != NO_ERROR) {
806        return ret;
807    }
808
809    // check calling permissions
810    if (!settingsAllowed()) {
811        return PERMISSION_DENIED;
812    }
813
814    Mutex::Autolock _l(mLock);
815    mMasterVolume = value;
816
817    // Set master volume in the HALs which support it.
818    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
819        AutoMutex lock(mHardwareLock);
820        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
821
822        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
823        if (dev->canSetMasterVolume()) {
824            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
825        }
826        mHardwareStatus = AUDIO_HW_IDLE;
827    }
828
829    // Now set the master volume in each playback thread.  Playback threads
830    // assigned to HALs which do not have master volume support will apply
831    // master volume during the mix operation.  Threads with HALs which do
832    // support master volume will simply ignore the setting.
833    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
834        if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
835            continue;
836        }
837        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
838    }
839
840    return NO_ERROR;
841}
842
843status_t AudioFlinger::setMode(audio_mode_t mode)
844{
845    status_t ret = initCheck();
846    if (ret != NO_ERROR) {
847        return ret;
848    }
849
850    // check calling permissions
851    if (!settingsAllowed()) {
852        return PERMISSION_DENIED;
853    }
854    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
855        ALOGW("Illegal value: setMode(%d)", mode);
856        return BAD_VALUE;
857    }
858
859    { // scope for the lock
860        AutoMutex lock(mHardwareLock);
861        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
862        mHardwareStatus = AUDIO_HW_SET_MODE;
863        ret = dev->set_mode(dev, mode);
864        mHardwareStatus = AUDIO_HW_IDLE;
865    }
866
867    if (NO_ERROR == ret) {
868        Mutex::Autolock _l(mLock);
869        mMode = mode;
870        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
871            mPlaybackThreads.valueAt(i)->setMode(mode);
872    }
873
874    return ret;
875}
876
877status_t AudioFlinger::setMicMute(bool state)
878{
879    status_t ret = initCheck();
880    if (ret != NO_ERROR) {
881        return ret;
882    }
883
884    // check calling permissions
885    if (!settingsAllowed()) {
886        return PERMISSION_DENIED;
887    }
888
889    AutoMutex lock(mHardwareLock);
890    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
891    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
892        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
893        status_t result = dev->set_mic_mute(dev, state);
894        if (result != NO_ERROR) {
895            ret = result;
896        }
897    }
898    mHardwareStatus = AUDIO_HW_IDLE;
899    return ret;
900}
901
902bool AudioFlinger::getMicMute() const
903{
904    status_t ret = initCheck();
905    if (ret != NO_ERROR) {
906        return false;
907    }
908    bool mute = true;
909    bool state = AUDIO_MODE_INVALID;
910    AutoMutex lock(mHardwareLock);
911    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
912    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
913        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
914        status_t result = dev->get_mic_mute(dev, &state);
915        if (result == NO_ERROR) {
916            mute = mute && state;
917        }
918    }
919    mHardwareStatus = AUDIO_HW_IDLE;
920
921    return mute;
922}
923
924status_t AudioFlinger::setMasterMute(bool muted)
925{
926    status_t ret = initCheck();
927    if (ret != NO_ERROR) {
928        return ret;
929    }
930
931    // check calling permissions
932    if (!settingsAllowed()) {
933        return PERMISSION_DENIED;
934    }
935
936    Mutex::Autolock _l(mLock);
937    mMasterMute = muted;
938
939    // Set master mute in the HALs which support it.
940    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
941        AutoMutex lock(mHardwareLock);
942        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
943
944        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
945        if (dev->canSetMasterMute()) {
946            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
947        }
948        mHardwareStatus = AUDIO_HW_IDLE;
949    }
950
951    // Now set the master mute in each playback thread.  Playback threads
952    // assigned to HALs which do not have master mute support will apply master
953    // mute during the mix operation.  Threads with HALs which do support master
954    // mute will simply ignore the setting.
955    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
956        if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
957            continue;
958        }
959        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
960    }
961
962    return NO_ERROR;
963}
964
965float AudioFlinger::masterVolume() const
966{
967    Mutex::Autolock _l(mLock);
968    return masterVolume_l();
969}
970
971bool AudioFlinger::masterMute() const
972{
973    Mutex::Autolock _l(mLock);
974    return masterMute_l();
975}
976
977float AudioFlinger::masterVolume_l() const
978{
979    return mMasterVolume;
980}
981
982bool AudioFlinger::masterMute_l() const
983{
984    return mMasterMute;
985}
986
987status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const
988{
989    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
990        ALOGW("setStreamVolume() invalid stream %d", stream);
991        return BAD_VALUE;
992    }
993    pid_t caller = IPCThreadState::self()->getCallingPid();
994    if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) {
995        ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream);
996        return PERMISSION_DENIED;
997    }
998
999    return NO_ERROR;
1000}
1001
1002status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
1003        audio_io_handle_t output)
1004{
1005    // check calling permissions
1006    if (!settingsAllowed()) {
1007        return PERMISSION_DENIED;
1008    }
1009
1010    status_t status = checkStreamType(stream);
1011    if (status != NO_ERROR) {
1012        return status;
1013    }
1014    ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume");
1015
1016    AutoMutex lock(mLock);
1017    PlaybackThread *thread = NULL;
1018    if (output != AUDIO_IO_HANDLE_NONE) {
1019        thread = checkPlaybackThread_l(output);
1020        if (thread == NULL) {
1021            return BAD_VALUE;
1022        }
1023    }
1024
1025    mStreamTypes[stream].volume = value;
1026
1027    if (thread == NULL) {
1028        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1029            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
1030        }
1031    } else {
1032        thread->setStreamVolume(stream, value);
1033    }
1034
1035    return NO_ERROR;
1036}
1037
1038status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
1039{
1040    // check calling permissions
1041    if (!settingsAllowed()) {
1042        return PERMISSION_DENIED;
1043    }
1044
1045    status_t status = checkStreamType(stream);
1046    if (status != NO_ERROR) {
1047        return status;
1048    }
1049    ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH");
1050
1051    if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
1052        ALOGE("setStreamMute() invalid stream %d", stream);
1053        return BAD_VALUE;
1054    }
1055
1056    AutoMutex lock(mLock);
1057    mStreamTypes[stream].mute = muted;
1058    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
1059        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
1060
1061    return NO_ERROR;
1062}
1063
1064float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
1065{
1066    status_t status = checkStreamType(stream);
1067    if (status != NO_ERROR) {
1068        return 0.0f;
1069    }
1070
1071    AutoMutex lock(mLock);
1072    float volume;
1073    if (output != AUDIO_IO_HANDLE_NONE) {
1074        PlaybackThread *thread = checkPlaybackThread_l(output);
1075        if (thread == NULL) {
1076            return 0.0f;
1077        }
1078        volume = thread->streamVolume(stream);
1079    } else {
1080        volume = streamVolume_l(stream);
1081    }
1082
1083    return volume;
1084}
1085
1086bool AudioFlinger::streamMute(audio_stream_type_t stream) const
1087{
1088    status_t status = checkStreamType(stream);
1089    if (status != NO_ERROR) {
1090        return true;
1091    }
1092
1093    AutoMutex lock(mLock);
1094    return streamMute_l(stream);
1095}
1096
1097
1098void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs)
1099{
1100    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1101        mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1102    }
1103}
1104
1105status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
1106{
1107    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
1108            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
1109
1110    // check calling permissions
1111    if (!settingsAllowed()) {
1112        return PERMISSION_DENIED;
1113    }
1114
1115    // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
1116    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1117        Mutex::Autolock _l(mLock);
1118        status_t final_result = NO_ERROR;
1119        {
1120            AutoMutex lock(mHardwareLock);
1121            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
1122            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1123                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1124                status_t result = dev->set_parameters(dev, keyValuePairs.string());
1125                final_result = result ?: final_result;
1126            }
1127            mHardwareStatus = AUDIO_HW_IDLE;
1128        }
1129        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
1130        AudioParameter param = AudioParameter(keyValuePairs);
1131        String8 value;
1132        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
1133            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
1134            if (mBtNrecIsOff != btNrecIsOff) {
1135                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1136                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
1137                    audio_devices_t device = thread->inDevice();
1138                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
1139                    // collect all of the thread's session IDs
1140                    KeyedVector<audio_session_t, bool> ids = thread->sessionIds();
1141                    // suspend effects associated with those session IDs
1142                    for (size_t j = 0; j < ids.size(); ++j) {
1143                        audio_session_t sessionId = ids.keyAt(j);
1144                        thread->setEffectSuspended(FX_IID_AEC,
1145                                                   suspend,
1146                                                   sessionId);
1147                        thread->setEffectSuspended(FX_IID_NS,
1148                                                   suspend,
1149                                                   sessionId);
1150                    }
1151                }
1152                mBtNrecIsOff = btNrecIsOff;
1153            }
1154        }
1155        String8 screenState;
1156        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1157            bool isOff = screenState == "off";
1158            if (isOff != (AudioFlinger::mScreenState & 1)) {
1159                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1160            }
1161        }
1162        return final_result;
1163    }
1164
1165    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1166    // and the thread is exited once the lock is released
1167    sp<ThreadBase> thread;
1168    {
1169        Mutex::Autolock _l(mLock);
1170        thread = checkPlaybackThread_l(ioHandle);
1171        if (thread == 0) {
1172            thread = checkRecordThread_l(ioHandle);
1173        } else if (thread == primaryPlaybackThread_l()) {
1174            // indicate output device change to all input threads for pre processing
1175            AudioParameter param = AudioParameter(keyValuePairs);
1176            int value;
1177            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1178                    (value != 0)) {
1179                broacastParametersToRecordThreads_l(keyValuePairs);
1180            }
1181        }
1182    }
1183    if (thread != 0) {
1184        return thread->setParameters(keyValuePairs);
1185    }
1186    return BAD_VALUE;
1187}
1188
1189String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1190{
1191    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1192            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1193
1194    Mutex::Autolock _l(mLock);
1195
1196    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1197        String8 out_s8;
1198
1199        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1200            char *s;
1201            {
1202            AutoMutex lock(mHardwareLock);
1203            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1204            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1205            s = dev->get_parameters(dev, keys.string());
1206            mHardwareStatus = AUDIO_HW_IDLE;
1207            }
1208            out_s8 += String8(s ? s : "");
1209            free(s);
1210        }
1211        return out_s8;
1212    }
1213
1214    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1215    if (playbackThread != NULL) {
1216        return playbackThread->getParameters(keys);
1217    }
1218    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1219    if (recordThread != NULL) {
1220        return recordThread->getParameters(keys);
1221    }
1222    return String8("");
1223}
1224
1225size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1226        audio_channel_mask_t channelMask) const
1227{
1228    status_t ret = initCheck();
1229    if (ret != NO_ERROR) {
1230        return 0;
1231    }
1232    if ((sampleRate == 0) ||
1233            !audio_is_valid_format(format) || !audio_has_proportional_frames(format) ||
1234            !audio_is_input_channel(channelMask)) {
1235        return 0;
1236    }
1237
1238    AutoMutex lock(mHardwareLock);
1239    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1240    audio_config_t config, proposed;
1241    memset(&proposed, 0, sizeof(proposed));
1242    proposed.sample_rate = sampleRate;
1243    proposed.channel_mask = channelMask;
1244    proposed.format = format;
1245
1246    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1247    size_t frames;
1248    for (;;) {
1249        // Note: config is currently a const parameter for get_input_buffer_size()
1250        // but we use a copy from proposed in case config changes from the call.
1251        config = proposed;
1252        frames = dev->get_input_buffer_size(dev, &config);
1253        if (frames != 0) {
1254            break; // hal success, config is the result
1255        }
1256        // change one parameter of the configuration each iteration to a more "common" value
1257        // to see if the device will support it.
1258        if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
1259            proposed.format = AUDIO_FORMAT_PCM_16_BIT;
1260        } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
1261            proposed.sample_rate = 44100;           // legacy AudioRecord.java. TODO: Query hw?
1262        } else {
1263            ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
1264                    "format %#x, channelMask 0x%X",
1265                    sampleRate, format, channelMask);
1266            break; // retries failed, break out of loop with frames == 0.
1267        }
1268    }
1269    mHardwareStatus = AUDIO_HW_IDLE;
1270    if (frames > 0 && config.sample_rate != sampleRate) {
1271        frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
1272    }
1273    return frames; // may be converted to bytes at the Java level.
1274}
1275
1276uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1277{
1278    Mutex::Autolock _l(mLock);
1279
1280    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1281    if (recordThread != NULL) {
1282        return recordThread->getInputFramesLost();
1283    }
1284    return 0;
1285}
1286
1287status_t AudioFlinger::setVoiceVolume(float value)
1288{
1289    status_t ret = initCheck();
1290    if (ret != NO_ERROR) {
1291        return ret;
1292    }
1293
1294    // check calling permissions
1295    if (!settingsAllowed()) {
1296        return PERMISSION_DENIED;
1297    }
1298
1299    AutoMutex lock(mHardwareLock);
1300    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1301    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1302    ret = dev->set_voice_volume(dev, value);
1303    mHardwareStatus = AUDIO_HW_IDLE;
1304
1305    return ret;
1306}
1307
1308status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1309        audio_io_handle_t output) const
1310{
1311    Mutex::Autolock _l(mLock);
1312
1313    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1314    if (playbackThread != NULL) {
1315        return playbackThread->getRenderPosition(halFrames, dspFrames);
1316    }
1317
1318    return BAD_VALUE;
1319}
1320
1321void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1322{
1323    Mutex::Autolock _l(mLock);
1324    if (client == 0) {
1325        return;
1326    }
1327    pid_t pid = IPCThreadState::self()->getCallingPid();
1328    {
1329        Mutex::Autolock _cl(mClientLock);
1330        if (mNotificationClients.indexOfKey(pid) < 0) {
1331            sp<NotificationClient> notificationClient = new NotificationClient(this,
1332                                                                                client,
1333                                                                                pid);
1334            ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1335
1336            mNotificationClients.add(pid, notificationClient);
1337
1338            sp<IBinder> binder = IInterface::asBinder(client);
1339            binder->linkToDeath(notificationClient);
1340        }
1341    }
1342
1343    // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
1344    // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock.
1345    // the config change is always sent from playback or record threads to avoid deadlock
1346    // with AudioSystem::gLock
1347    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1348        mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid);
1349    }
1350
1351    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1352        mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid);
1353    }
1354}
1355
1356void AudioFlinger::removeNotificationClient(pid_t pid)
1357{
1358    Mutex::Autolock _l(mLock);
1359    {
1360        Mutex::Autolock _cl(mClientLock);
1361        mNotificationClients.removeItem(pid);
1362    }
1363
1364    ALOGV("%d died, releasing its sessions", pid);
1365    size_t num = mAudioSessionRefs.size();
1366    bool removed = false;
1367    for (size_t i = 0; i< num; ) {
1368        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1369        ALOGV(" pid %d @ %zu", ref->mPid, i);
1370        if (ref->mPid == pid) {
1371            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1372            mAudioSessionRefs.removeAt(i);
1373            delete ref;
1374            removed = true;
1375            num--;
1376        } else {
1377            i++;
1378        }
1379    }
1380    if (removed) {
1381        purgeStaleEffects_l();
1382    }
1383}
1384
1385void AudioFlinger::ioConfigChanged(audio_io_config_event event,
1386                                   const sp<AudioIoDescriptor>& ioDesc,
1387                                   pid_t pid)
1388{
1389    Mutex::Autolock _l(mClientLock);
1390    size_t size = mNotificationClients.size();
1391    for (size_t i = 0; i < size; i++) {
1392        if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) {
1393            mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc);
1394        }
1395    }
1396}
1397
1398// removeClient_l() must be called with AudioFlinger::mClientLock held
1399void AudioFlinger::removeClient_l(pid_t pid)
1400{
1401    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1402            IPCThreadState::self()->getCallingPid());
1403    mClients.removeItem(pid);
1404}
1405
1406// getEffectThread_l() must be called with AudioFlinger::mLock held
1407sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
1408        int EffectId)
1409{
1410    sp<PlaybackThread> thread;
1411
1412    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1413        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1414            ALOG_ASSERT(thread == 0);
1415            thread = mPlaybackThreads.valueAt(i);
1416        }
1417    }
1418
1419    return thread;
1420}
1421
1422
1423
1424// ----------------------------------------------------------------------------
1425
1426AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1427    :   RefBase(),
1428        mAudioFlinger(audioFlinger),
1429        mPid(pid)
1430{
1431    size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0);
1432    heapSize *= 1024;
1433    if (!heapSize) {
1434        heapSize = kClientSharedHeapSizeBytes;
1435        // Increase heap size on non low ram devices to limit risk of reconnection failure for
1436        // invalidated tracks
1437        if (!audioFlinger->isLowRamDevice()) {
1438            heapSize *= kClientSharedHeapSizeMultiplier;
1439        }
1440    }
1441    mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client");
1442}
1443
1444// Client destructor must be called with AudioFlinger::mClientLock held
1445AudioFlinger::Client::~Client()
1446{
1447    mAudioFlinger->removeClient_l(mPid);
1448}
1449
1450sp<MemoryDealer> AudioFlinger::Client::heap() const
1451{
1452    return mMemoryDealer;
1453}
1454
1455// ----------------------------------------------------------------------------
1456
1457AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1458                                                     const sp<IAudioFlingerClient>& client,
1459                                                     pid_t pid)
1460    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1461{
1462}
1463
1464AudioFlinger::NotificationClient::~NotificationClient()
1465{
1466}
1467
1468void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1469{
1470    sp<NotificationClient> keep(this);
1471    mAudioFlinger->removeNotificationClient(mPid);
1472}
1473
1474
1475// ----------------------------------------------------------------------------
1476
1477sp<IAudioRecord> AudioFlinger::openRecord(
1478        audio_io_handle_t input,
1479        uint32_t sampleRate,
1480        audio_format_t format,
1481        audio_channel_mask_t channelMask,
1482        const String16& opPackageName,
1483        size_t *frameCount,
1484        audio_input_flags_t *flags,
1485        pid_t pid,
1486        pid_t tid,
1487        int clientUid,
1488        audio_session_t *sessionId,
1489        size_t *notificationFrames,
1490        sp<IMemory>& cblk,
1491        sp<IMemory>& buffers,
1492        status_t *status)
1493{
1494    sp<RecordThread::RecordTrack> recordTrack;
1495    sp<RecordHandle> recordHandle;
1496    sp<Client> client;
1497    status_t lStatus;
1498    audio_session_t lSessionId;
1499
1500    cblk.clear();
1501    buffers.clear();
1502
1503    bool updatePid = (pid == -1);
1504    const uid_t callingUid = IPCThreadState::self()->getCallingUid();
1505    if (!isTrustedCallingUid(callingUid)) {
1506        ALOGW_IF((uid_t)clientUid != callingUid,
1507                "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid);
1508        clientUid = callingUid;
1509        updatePid = true;
1510    }
1511
1512    if (updatePid) {
1513        const pid_t callingPid = IPCThreadState::self()->getCallingPid();
1514        ALOGW_IF(pid != -1 && pid != callingPid,
1515                 "%s uid %d pid %d tried to pass itself off as pid %d",
1516                 __func__, callingUid, callingPid, pid);
1517        pid = callingPid;
1518    }
1519
1520    // check calling permissions
1521    if (!recordingAllowed(opPackageName, tid, clientUid)) {
1522        ALOGE("openRecord() permission denied: recording not allowed");
1523        lStatus = PERMISSION_DENIED;
1524        goto Exit;
1525    }
1526
1527    // further sample rate checks are performed by createRecordTrack_l()
1528    if (sampleRate == 0) {
1529        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1530        lStatus = BAD_VALUE;
1531        goto Exit;
1532    }
1533
1534    // we don't yet support anything other than linear PCM
1535    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
1536        ALOGE("openRecord() invalid format %#x", format);
1537        lStatus = BAD_VALUE;
1538        goto Exit;
1539    }
1540
1541    // further channel mask checks are performed by createRecordTrack_l()
1542    if (!audio_is_input_channel(channelMask)) {
1543        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1544        lStatus = BAD_VALUE;
1545        goto Exit;
1546    }
1547
1548    {
1549        Mutex::Autolock _l(mLock);
1550        RecordThread *thread = checkRecordThread_l(input);
1551        if (thread == NULL) {
1552            ALOGE("openRecord() checkRecordThread_l failed");
1553            lStatus = BAD_VALUE;
1554            goto Exit;
1555        }
1556
1557        client = registerPid(pid);
1558
1559        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1560            if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
1561                lStatus = BAD_VALUE;
1562                goto Exit;
1563            }
1564            lSessionId = *sessionId;
1565        } else {
1566            // if no audio session id is provided, create one here
1567            lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
1568            if (sessionId != NULL) {
1569                *sessionId = lSessionId;
1570            }
1571        }
1572        ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input);
1573
1574        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1575                                                  frameCount, lSessionId, notificationFrames,
1576                                                  clientUid, flags, tid, &lStatus);
1577        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1578
1579        if (lStatus == NO_ERROR) {
1580            // Check if one effect chain was awaiting for an AudioRecord to be created on this
1581            // session and move it to this thread.
1582            sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId);
1583            if (chain != 0) {
1584                Mutex::Autolock _l(thread->mLock);
1585                thread->addEffectChain_l(chain);
1586            }
1587        }
1588    }
1589
1590    if (lStatus != NO_ERROR) {
1591        // remove local strong reference to Client before deleting the RecordTrack so that the
1592        // Client destructor is called by the TrackBase destructor with mClientLock held
1593        // Don't hold mClientLock when releasing the reference on the track as the
1594        // destructor will acquire it.
1595        {
1596            Mutex::Autolock _cl(mClientLock);
1597            client.clear();
1598        }
1599        recordTrack.clear();
1600        goto Exit;
1601    }
1602
1603    cblk = recordTrack->getCblk();
1604    buffers = recordTrack->getBuffers();
1605
1606    // return handle to client
1607    recordHandle = new RecordHandle(recordTrack);
1608
1609Exit:
1610    *status = lStatus;
1611    return recordHandle;
1612}
1613
1614
1615
1616// ----------------------------------------------------------------------------
1617
1618audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1619{
1620    if (name == NULL) {
1621        return AUDIO_MODULE_HANDLE_NONE;
1622    }
1623    if (!settingsAllowed()) {
1624        return AUDIO_MODULE_HANDLE_NONE;
1625    }
1626    Mutex::Autolock _l(mLock);
1627    return loadHwModule_l(name);
1628}
1629
1630// loadHwModule_l() must be called with AudioFlinger::mLock held
1631audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1632{
1633    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1634        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1635            ALOGW("loadHwModule() module %s already loaded", name);
1636            return mAudioHwDevs.keyAt(i);
1637        }
1638    }
1639
1640    audio_hw_device_t *dev;
1641
1642    int rc = load_audio_interface(name, &dev);
1643    if (rc) {
1644        ALOGE("loadHwModule() error %d loading module %s", rc, name);
1645        return AUDIO_MODULE_HANDLE_NONE;
1646    }
1647
1648    mHardwareStatus = AUDIO_HW_INIT;
1649    rc = dev->init_check(dev);
1650    mHardwareStatus = AUDIO_HW_IDLE;
1651    if (rc) {
1652        ALOGE("loadHwModule() init check error %d for module %s", rc, name);
1653        return AUDIO_MODULE_HANDLE_NONE;
1654    }
1655
1656    // Check and cache this HAL's level of support for master mute and master
1657    // volume.  If this is the first HAL opened, and it supports the get
1658    // methods, use the initial values provided by the HAL as the current
1659    // master mute and volume settings.
1660
1661    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1662    {  // scope for auto-lock pattern
1663        AutoMutex lock(mHardwareLock);
1664
1665        if (0 == mAudioHwDevs.size()) {
1666            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1667            if (NULL != dev->get_master_volume) {
1668                float mv;
1669                if (OK == dev->get_master_volume(dev, &mv)) {
1670                    mMasterVolume = mv;
1671                }
1672            }
1673
1674            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1675            if (NULL != dev->get_master_mute) {
1676                bool mm;
1677                if (OK == dev->get_master_mute(dev, &mm)) {
1678                    mMasterMute = mm;
1679                }
1680            }
1681        }
1682
1683        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1684        if ((NULL != dev->set_master_volume) &&
1685            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1686            flags = static_cast<AudioHwDevice::Flags>(flags |
1687                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1688        }
1689
1690        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1691        if ((NULL != dev->set_master_mute) &&
1692            (OK == dev->set_master_mute(dev, mMasterMute))) {
1693            flags = static_cast<AudioHwDevice::Flags>(flags |
1694                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1695        }
1696
1697        mHardwareStatus = AUDIO_HW_IDLE;
1698    }
1699
1700    audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE);
1701    mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags));
1702
1703    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1704          name, dev->common.module->name, dev->common.module->id, handle);
1705
1706    return handle;
1707
1708}
1709
1710// ----------------------------------------------------------------------------
1711
1712uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1713{
1714    Mutex::Autolock _l(mLock);
1715    PlaybackThread *thread = fastPlaybackThread_l();
1716    return thread != NULL ? thread->sampleRate() : 0;
1717}
1718
1719size_t AudioFlinger::getPrimaryOutputFrameCount()
1720{
1721    Mutex::Autolock _l(mLock);
1722    PlaybackThread *thread = fastPlaybackThread_l();
1723    return thread != NULL ? thread->frameCountHAL() : 0;
1724}
1725
1726// ----------------------------------------------------------------------------
1727
1728status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1729{
1730    uid_t uid = IPCThreadState::self()->getCallingUid();
1731    if (uid != AID_SYSTEM) {
1732        return PERMISSION_DENIED;
1733    }
1734    Mutex::Autolock _l(mLock);
1735    if (mIsDeviceTypeKnown) {
1736        return INVALID_OPERATION;
1737    }
1738    mIsLowRamDevice = isLowRamDevice;
1739    mIsDeviceTypeKnown = true;
1740    return NO_ERROR;
1741}
1742
1743audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId)
1744{
1745    Mutex::Autolock _l(mLock);
1746
1747    ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
1748    if (index >= 0) {
1749        ALOGV("getAudioHwSyncForSession found ID %d for session %d",
1750              mHwAvSyncIds.valueAt(index), sessionId);
1751        return mHwAvSyncIds.valueAt(index);
1752    }
1753
1754    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1755    if (dev == NULL) {
1756        return AUDIO_HW_SYNC_INVALID;
1757    }
1758    char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC);
1759    AudioParameter param = AudioParameter(String8(reply));
1760    free(reply);
1761
1762    int value;
1763    if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) {
1764        ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId);
1765        return AUDIO_HW_SYNC_INVALID;
1766    }
1767
1768    // allow only one session for a given HW A/V sync ID.
1769    for (size_t i = 0; i < mHwAvSyncIds.size(); i++) {
1770        if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) {
1771            ALOGV("getAudioHwSyncForSession removing ID %d for session %d",
1772                  value, mHwAvSyncIds.keyAt(i));
1773            mHwAvSyncIds.removeItemsAt(i);
1774            break;
1775        }
1776    }
1777
1778    mHwAvSyncIds.add(sessionId, value);
1779
1780    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1781        sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i);
1782        uint32_t sessions = thread->hasAudioSession(sessionId);
1783        if (sessions & ThreadBase::TRACK_SESSION) {
1784            AudioParameter param = AudioParameter();
1785            param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value);
1786            thread->setParameters(param.toString());
1787            break;
1788        }
1789    }
1790
1791    ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId);
1792    return (audio_hw_sync_t)value;
1793}
1794
1795status_t AudioFlinger::systemReady()
1796{
1797    Mutex::Autolock _l(mLock);
1798    ALOGI("%s", __FUNCTION__);
1799    if (mSystemReady) {
1800        ALOGW("%s called twice", __FUNCTION__);
1801        return NO_ERROR;
1802    }
1803    mSystemReady = true;
1804    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1805        ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get();
1806        thread->systemReady();
1807    }
1808    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1809        ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
1810        thread->systemReady();
1811    }
1812    return NO_ERROR;
1813}
1814
1815// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held
1816void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId)
1817{
1818    ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
1819    if (index >= 0) {
1820        audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index);
1821        ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId);
1822        AudioParameter param = AudioParameter();
1823        param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId);
1824        thread->setParameters(param.toString());
1825    }
1826}
1827
1828
1829// ----------------------------------------------------------------------------
1830
1831
1832sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module,
1833                                                            audio_io_handle_t *output,
1834                                                            audio_config_t *config,
1835                                                            audio_devices_t devices,
1836                                                            const String8& address,
1837                                                            audio_output_flags_t flags)
1838{
1839    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices);
1840    if (outHwDev == NULL) {
1841        return 0;
1842    }
1843
1844    if (*output == AUDIO_IO_HANDLE_NONE) {
1845        *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
1846    } else {
1847        // Audio Policy does not currently request a specific output handle.
1848        // If this is ever needed, see openInput_l() for example code.
1849        ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output);
1850        return 0;
1851    }
1852
1853    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1854
1855    // FOR TESTING ONLY:
1856    // This if statement allows overriding the audio policy settings
1857    // and forcing a specific format or channel mask to the HAL/Sink device for testing.
1858    if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
1859        // Check only for Normal Mixing mode
1860        if (kEnableExtendedPrecision) {
1861            // Specify format (uncomment one below to choose)
1862            //config->format = AUDIO_FORMAT_PCM_FLOAT;
1863            //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
1864            //config->format = AUDIO_FORMAT_PCM_32_BIT;
1865            //config->format = AUDIO_FORMAT_PCM_8_24_BIT;
1866            // ALOGV("openOutput_l() upgrading format to %#08x", config->format);
1867        }
1868        if (kEnableExtendedChannels) {
1869            // Specify channel mask (uncomment one below to choose)
1870            //config->channel_mask = audio_channel_out_mask_from_count(4);  // for USB 4ch
1871            //config->channel_mask = audio_channel_mask_from_representation_and_bits(
1872            //        AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1);  // another 4ch example
1873        }
1874    }
1875
1876    AudioStreamOut *outputStream = NULL;
1877    status_t status = outHwDev->openOutputStream(
1878            &outputStream,
1879            *output,
1880            devices,
1881            flags,
1882            config,
1883            address.string());
1884
1885    mHardwareStatus = AUDIO_HW_IDLE;
1886
1887    if (status == NO_ERROR) {
1888
1889        PlaybackThread *thread;
1890        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1891            thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady);
1892            ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread);
1893        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
1894                || !isValidPcmSinkFormat(config->format)
1895                || !isValidPcmSinkChannelMask(config->channel_mask)) {
1896            thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
1897            ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread);
1898        } else {
1899            thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
1900            ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread);
1901        }
1902        mPlaybackThreads.add(*output, thread);
1903        return thread;
1904    }
1905
1906    return 0;
1907}
1908
1909status_t AudioFlinger::openOutput(audio_module_handle_t module,
1910                                  audio_io_handle_t *output,
1911                                  audio_config_t *config,
1912                                  audio_devices_t *devices,
1913                                  const String8& address,
1914                                  uint32_t *latencyMs,
1915                                  audio_output_flags_t flags)
1916{
1917    ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1918              module,
1919              (devices != NULL) ? *devices : 0,
1920              config->sample_rate,
1921              config->format,
1922              config->channel_mask,
1923              flags);
1924
1925    if (*devices == AUDIO_DEVICE_NONE) {
1926        return BAD_VALUE;
1927    }
1928
1929    Mutex::Autolock _l(mLock);
1930
1931    sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags);
1932    if (thread != 0) {
1933        *latencyMs = thread->latency();
1934
1935        // notify client processes of the new output creation
1936        thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
1937
1938        // the first primary output opened designates the primary hw device
1939        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1940            ALOGI("Using module %d has the primary audio interface", module);
1941            mPrimaryHardwareDev = thread->getOutput()->audioHwDev;
1942
1943            AutoMutex lock(mHardwareLock);
1944            mHardwareStatus = AUDIO_HW_SET_MODE;
1945            mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode);
1946            mHardwareStatus = AUDIO_HW_IDLE;
1947        }
1948        return NO_ERROR;
1949    }
1950
1951    return NO_INIT;
1952}
1953
1954audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1955        audio_io_handle_t output2)
1956{
1957    Mutex::Autolock _l(mLock);
1958    MixerThread *thread1 = checkMixerThread_l(output1);
1959    MixerThread *thread2 = checkMixerThread_l(output2);
1960
1961    if (thread1 == NULL || thread2 == NULL) {
1962        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1963                output2);
1964        return AUDIO_IO_HANDLE_NONE;
1965    }
1966
1967    audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
1968    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady);
1969    thread->addOutputTrack(thread2);
1970    mPlaybackThreads.add(id, thread);
1971    // notify client processes of the new output creation
1972    thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
1973    return id;
1974}
1975
1976status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1977{
1978    return closeOutput_nonvirtual(output);
1979}
1980
1981status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1982{
1983    // keep strong reference on the playback thread so that
1984    // it is not destroyed while exit() is executed
1985    sp<PlaybackThread> thread;
1986    {
1987        Mutex::Autolock _l(mLock);
1988        thread = checkPlaybackThread_l(output);
1989        if (thread == NULL) {
1990            return BAD_VALUE;
1991        }
1992
1993        ALOGV("closeOutput() %d", output);
1994
1995        if (thread->type() == ThreadBase::MIXER) {
1996            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1997                if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
1998                    DuplicatingThread *dupThread =
1999                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
2000                    dupThread->removeOutputTrack((MixerThread *)thread.get());
2001                }
2002            }
2003        }
2004
2005
2006        mPlaybackThreads.removeItem(output);
2007        // save all effects to the default thread
2008        if (mPlaybackThreads.size()) {
2009            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
2010            if (dstThread != NULL) {
2011                // audioflinger lock is held here so the acquisition order of thread locks does not
2012                // matter
2013                Mutex::Autolock _dl(dstThread->mLock);
2014                Mutex::Autolock _sl(thread->mLock);
2015                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
2016                for (size_t i = 0; i < effectChains.size(); i ++) {
2017                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
2018                }
2019            }
2020        }
2021        const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
2022        ioDesc->mIoHandle = output;
2023        ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc);
2024    }
2025    thread->exit();
2026    // The thread entity (active unit of execution) is no longer running here,
2027    // but the ThreadBase container still exists.
2028
2029    if (!thread->isDuplicating()) {
2030        closeOutputFinish(thread);
2031    }
2032
2033    return NO_ERROR;
2034}
2035
2036void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread)
2037{
2038    AudioStreamOut *out = thread->clearOutput();
2039    ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
2040    // from now on thread->mOutput is NULL
2041    out->hwDev()->close_output_stream(out->hwDev(), out->stream);
2042    delete out;
2043}
2044
2045void AudioFlinger::closeOutputInternal_l(const sp<PlaybackThread>& thread)
2046{
2047    mPlaybackThreads.removeItem(thread->mId);
2048    thread->exit();
2049    closeOutputFinish(thread);
2050}
2051
2052status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
2053{
2054    Mutex::Autolock _l(mLock);
2055    PlaybackThread *thread = checkPlaybackThread_l(output);
2056
2057    if (thread == NULL) {
2058        return BAD_VALUE;
2059    }
2060
2061    ALOGV("suspendOutput() %d", output);
2062    thread->suspend();
2063
2064    return NO_ERROR;
2065}
2066
2067status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
2068{
2069    Mutex::Autolock _l(mLock);
2070    PlaybackThread *thread = checkPlaybackThread_l(output);
2071
2072    if (thread == NULL) {
2073        return BAD_VALUE;
2074    }
2075
2076    ALOGV("restoreOutput() %d", output);
2077
2078    thread->restore();
2079
2080    return NO_ERROR;
2081}
2082
2083status_t AudioFlinger::openInput(audio_module_handle_t module,
2084                                          audio_io_handle_t *input,
2085                                          audio_config_t *config,
2086                                          audio_devices_t *devices,
2087                                          const String8& address,
2088                                          audio_source_t source,
2089                                          audio_input_flags_t flags)
2090{
2091    Mutex::Autolock _l(mLock);
2092
2093    if (*devices == AUDIO_DEVICE_NONE) {
2094        return BAD_VALUE;
2095    }
2096
2097    sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags);
2098
2099    if (thread != 0) {
2100        // notify client processes of the new input creation
2101        thread->ioConfigChanged(AUDIO_INPUT_OPENED);
2102        return NO_ERROR;
2103    }
2104    return NO_INIT;
2105}
2106
2107sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module,
2108                                                         audio_io_handle_t *input,
2109                                                         audio_config_t *config,
2110                                                         audio_devices_t devices,
2111                                                         const String8& address,
2112                                                         audio_source_t source,
2113                                                         audio_input_flags_t flags)
2114{
2115    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
2116    if (inHwDev == NULL) {
2117        *input = AUDIO_IO_HANDLE_NONE;
2118        return 0;
2119    }
2120
2121    // Audio Policy can request a specific handle for hardware hotword.
2122    // The goal here is not to re-open an already opened input.
2123    // It is to use a pre-assigned I/O handle.
2124    if (*input == AUDIO_IO_HANDLE_NONE) {
2125        *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT);
2126    } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) {
2127        ALOGE("openInput_l() requested input handle %d is invalid", *input);
2128        return 0;
2129    } else if (mRecordThreads.indexOfKey(*input) >= 0) {
2130        // This should not happen in a transient state with current design.
2131        ALOGE("openInput_l() requested input handle %d is already assigned", *input);
2132        return 0;
2133    }
2134
2135    audio_config_t halconfig = *config;
2136    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
2137    audio_stream_in_t *inStream = NULL;
2138    status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
2139                                        &inStream, flags, address.string(), source);
2140    ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d"
2141           ", Format %#x, Channels %x, flags %#x, status %d addr %s",
2142            inStream,
2143            halconfig.sample_rate,
2144            halconfig.format,
2145            halconfig.channel_mask,
2146            flags,
2147            status, address.string());
2148
2149    // If the input could not be opened with the requested parameters and we can handle the
2150    // conversion internally, try to open again with the proposed parameters.
2151    if (status == BAD_VALUE &&
2152        audio_is_linear_pcm(config->format) &&
2153        audio_is_linear_pcm(halconfig.format) &&
2154        (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
2155        (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) &&
2156        (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) {
2157        // FIXME describe the change proposed by HAL (save old values so we can log them here)
2158        ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
2159        inStream = NULL;
2160        status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
2161                                            &inStream, flags, address.string(), source);
2162        // FIXME log this new status; HAL should not propose any further changes
2163    }
2164
2165    if (status == NO_ERROR && inStream != NULL) {
2166
2167#ifdef TEE_SINK
2168        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
2169        // or (re-)create if current Pipe is idle and does not match the new format
2170        sp<NBAIO_Sink> teeSink;
2171        enum {
2172            TEE_SINK_NO,    // don't copy input
2173            TEE_SINK_NEW,   // copy input using a new pipe
2174            TEE_SINK_OLD,   // copy input using an existing pipe
2175        } kind;
2176        NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate,
2177                audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format);
2178        if (!mTeeSinkInputEnabled) {
2179            kind = TEE_SINK_NO;
2180        } else if (!Format_isValid(format)) {
2181            kind = TEE_SINK_NO;
2182        } else if (mRecordTeeSink == 0) {
2183            kind = TEE_SINK_NEW;
2184        } else if (mRecordTeeSink->getStrongCount() != 1) {
2185            kind = TEE_SINK_NO;
2186        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
2187            kind = TEE_SINK_OLD;
2188        } else {
2189            kind = TEE_SINK_NEW;
2190        }
2191        switch (kind) {
2192        case TEE_SINK_NEW: {
2193            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
2194            size_t numCounterOffers = 0;
2195            const NBAIO_Format offers[1] = {format};
2196            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
2197            ALOG_ASSERT(index == 0);
2198            PipeReader *pipeReader = new PipeReader(*pipe);
2199            numCounterOffers = 0;
2200            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
2201            ALOG_ASSERT(index == 0);
2202            mRecordTeeSink = pipe;
2203            mRecordTeeSource = pipeReader;
2204            teeSink = pipe;
2205            }
2206            break;
2207        case TEE_SINK_OLD:
2208            teeSink = mRecordTeeSink;
2209            break;
2210        case TEE_SINK_NO:
2211        default:
2212            break;
2213        }
2214#endif
2215
2216        AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags);
2217
2218        // Start record thread
2219        // RecordThread requires both input and output device indication to forward to audio
2220        // pre processing modules
2221        sp<RecordThread> thread = new RecordThread(this,
2222                                  inputStream,
2223                                  *input,
2224                                  primaryOutputDevice_l(),
2225                                  devices,
2226                                  mSystemReady
2227#ifdef TEE_SINK
2228                                  , teeSink
2229#endif
2230                                  );
2231        mRecordThreads.add(*input, thread);
2232        ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
2233        return thread;
2234    }
2235
2236    *input = AUDIO_IO_HANDLE_NONE;
2237    return 0;
2238}
2239
2240status_t AudioFlinger::closeInput(audio_io_handle_t input)
2241{
2242    return closeInput_nonvirtual(input);
2243}
2244
2245status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
2246{
2247    // keep strong reference on the record thread so that
2248    // it is not destroyed while exit() is executed
2249    sp<RecordThread> thread;
2250    {
2251        Mutex::Autolock _l(mLock);
2252        thread = checkRecordThread_l(input);
2253        if (thread == 0) {
2254            return BAD_VALUE;
2255        }
2256
2257        ALOGV("closeInput() %d", input);
2258
2259        // If we still have effect chains, it means that a client still holds a handle
2260        // on at least one effect. We must either move the chain to an existing thread with the
2261        // same session ID or put it aside in case a new record thread is opened for a
2262        // new capture on the same session
2263        sp<EffectChain> chain;
2264        {
2265            Mutex::Autolock _sl(thread->mLock);
2266            Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
2267            // Note: maximum one chain per record thread
2268            if (effectChains.size() != 0) {
2269                chain = effectChains[0];
2270            }
2271        }
2272        if (chain != 0) {
2273            // first check if a record thread is already opened with a client on the same session.
2274            // This should only happen in case of overlap between one thread tear down and the
2275            // creation of its replacement
2276            size_t i;
2277            for (i = 0; i < mRecordThreads.size(); i++) {
2278                sp<RecordThread> t = mRecordThreads.valueAt(i);
2279                if (t == thread) {
2280                    continue;
2281                }
2282                if (t->hasAudioSession(chain->sessionId()) != 0) {
2283                    Mutex::Autolock _l(t->mLock);
2284                    ALOGV("closeInput() found thread %d for effect session %d",
2285                          t->id(), chain->sessionId());
2286                    t->addEffectChain_l(chain);
2287                    break;
2288                }
2289            }
2290            // put the chain aside if we could not find a record thread with the same session id.
2291            if (i == mRecordThreads.size()) {
2292                putOrphanEffectChain_l(chain);
2293            }
2294        }
2295        const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
2296        ioDesc->mIoHandle = input;
2297        ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc);
2298        mRecordThreads.removeItem(input);
2299    }
2300    // FIXME: calling thread->exit() without mLock held should not be needed anymore now that
2301    // we have a different lock for notification client
2302    closeInputFinish(thread);
2303    return NO_ERROR;
2304}
2305
2306void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread)
2307{
2308    thread->exit();
2309    AudioStreamIn *in = thread->clearInput();
2310    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
2311    // from now on thread->mInput is NULL
2312    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
2313    delete in;
2314}
2315
2316void AudioFlinger::closeInputInternal_l(const sp<RecordThread>& thread)
2317{
2318    mRecordThreads.removeItem(thread->mId);
2319    closeInputFinish(thread);
2320}
2321
2322status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
2323{
2324    Mutex::Autolock _l(mLock);
2325    ALOGV("invalidateStream() stream %d", stream);
2326
2327    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2328        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2329        thread->invalidateTracks(stream);
2330    }
2331
2332    return NO_ERROR;
2333}
2334
2335
2336audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use)
2337{
2338    // This is a binder API, so a malicious client could pass in a bad parameter.
2339    // Check for that before calling the internal API nextUniqueId().
2340    if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) {
2341        ALOGE("newAudioUniqueId invalid use %d", use);
2342        return AUDIO_UNIQUE_ID_ALLOCATE;
2343    }
2344    return nextUniqueId(use);
2345}
2346
2347void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid)
2348{
2349    Mutex::Autolock _l(mLock);
2350    pid_t caller = IPCThreadState::self()->getCallingPid();
2351    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
2352    if (pid != -1 && (caller == getpid_cached)) {
2353        caller = pid;
2354    }
2355
2356    {
2357        Mutex::Autolock _cl(mClientLock);
2358        // Ignore requests received from processes not known as notification client. The request
2359        // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
2360        // called from a different pid leaving a stale session reference.  Also we don't know how
2361        // to clear this reference if the client process dies.
2362        if (mNotificationClients.indexOfKey(caller) < 0) {
2363            ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
2364            return;
2365        }
2366    }
2367
2368    size_t num = mAudioSessionRefs.size();
2369    for (size_t i = 0; i< num; i++) {
2370        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
2371        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2372            ref->mCnt++;
2373            ALOGV(" incremented refcount to %d", ref->mCnt);
2374            return;
2375        }
2376    }
2377    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
2378    ALOGV(" added new entry for %d", audioSession);
2379}
2380
2381void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid)
2382{
2383    Mutex::Autolock _l(mLock);
2384    pid_t caller = IPCThreadState::self()->getCallingPid();
2385    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
2386    if (pid != -1 && (caller == getpid_cached)) {
2387        caller = pid;
2388    }
2389    size_t num = mAudioSessionRefs.size();
2390    for (size_t i = 0; i< num; i++) {
2391        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2392        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2393            ref->mCnt--;
2394            ALOGV(" decremented refcount to %d", ref->mCnt);
2395            if (ref->mCnt == 0) {
2396                mAudioSessionRefs.removeAt(i);
2397                delete ref;
2398                purgeStaleEffects_l();
2399            }
2400            return;
2401        }
2402    }
2403    // If the caller is mediaserver it is likely that the session being released was acquired
2404    // on behalf of a process not in notification clients and we ignore the warning.
2405    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2406}
2407
2408void AudioFlinger::purgeStaleEffects_l() {
2409
2410    ALOGV("purging stale effects");
2411
2412    Vector< sp<EffectChain> > chains;
2413
2414    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2415        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2416        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2417            sp<EffectChain> ec = t->mEffectChains[j];
2418            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2419                chains.push(ec);
2420            }
2421        }
2422    }
2423    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2424        sp<RecordThread> t = mRecordThreads.valueAt(i);
2425        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2426            sp<EffectChain> ec = t->mEffectChains[j];
2427            chains.push(ec);
2428        }
2429    }
2430
2431    for (size_t i = 0; i < chains.size(); i++) {
2432        sp<EffectChain> ec = chains[i];
2433        int sessionid = ec->sessionId();
2434        sp<ThreadBase> t = ec->mThread.promote();
2435        if (t == 0) {
2436            continue;
2437        }
2438        size_t numsessionrefs = mAudioSessionRefs.size();
2439        bool found = false;
2440        for (size_t k = 0; k < numsessionrefs; k++) {
2441            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2442            if (ref->mSessionid == sessionid) {
2443                ALOGV(" session %d still exists for %d with %d refs",
2444                    sessionid, ref->mPid, ref->mCnt);
2445                found = true;
2446                break;
2447            }
2448        }
2449        if (!found) {
2450            Mutex::Autolock _l(t->mLock);
2451            // remove all effects from the chain
2452            while (ec->mEffects.size()) {
2453                sp<EffectModule> effect = ec->mEffects[0];
2454                effect->unPin();
2455                t->removeEffect_l(effect);
2456                if (effect->purgeHandles()) {
2457                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2458                }
2459                AudioSystem::unregisterEffect(effect->id());
2460            }
2461        }
2462    }
2463    return;
2464}
2465
2466// checkThread_l() must be called with AudioFlinger::mLock held
2467AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const
2468{
2469    ThreadBase *thread = NULL;
2470    switch (audio_unique_id_get_use(ioHandle)) {
2471    case AUDIO_UNIQUE_ID_USE_OUTPUT:
2472        thread = checkPlaybackThread_l(ioHandle);
2473        break;
2474    case AUDIO_UNIQUE_ID_USE_INPUT:
2475        thread = checkRecordThread_l(ioHandle);
2476        break;
2477    default:
2478        break;
2479    }
2480    return thread;
2481}
2482
2483// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2484AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2485{
2486    return mPlaybackThreads.valueFor(output).get();
2487}
2488
2489// checkMixerThread_l() must be called with AudioFlinger::mLock held
2490AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2491{
2492    PlaybackThread *thread = checkPlaybackThread_l(output);
2493    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2494}
2495
2496// checkRecordThread_l() must be called with AudioFlinger::mLock held
2497AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2498{
2499    return mRecordThreads.valueFor(input).get();
2500}
2501
2502audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use)
2503{
2504    // This is the internal API, so it is OK to assert on bad parameter.
2505    LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX);
2506    const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1;
2507    for (int retry = 0; retry < maxRetries; retry++) {
2508        // The cast allows wraparound from max positive to min negative instead of abort
2509        uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use],
2510                (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel);
2511        ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED);
2512        // allow wrap by skipping 0 and -1 for session ids
2513        if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) {
2514            ALOGW_IF(retry != 0, "unique ID overflow for use %d", use);
2515            return (audio_unique_id_t) (base | use);
2516        }
2517    }
2518    // We have no way of recovering from wraparound
2519    LOG_ALWAYS_FATAL("unique ID overflow for use %d", use);
2520    // TODO Use a floor after wraparound.  This may need a mutex.
2521}
2522
2523AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2524{
2525    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2526        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2527        if(thread->isDuplicating()) {
2528            continue;
2529        }
2530        AudioStreamOut *output = thread->getOutput();
2531        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2532            return thread;
2533        }
2534    }
2535    return NULL;
2536}
2537
2538audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2539{
2540    PlaybackThread *thread = primaryPlaybackThread_l();
2541
2542    if (thread == NULL) {
2543        return 0;
2544    }
2545
2546    return thread->outDevice();
2547}
2548
2549AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const
2550{
2551    size_t minFrameCount = 0;
2552    PlaybackThread *minThread = NULL;
2553    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2554        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2555        if (!thread->isDuplicating()) {
2556            size_t frameCount = thread->frameCountHAL();
2557            if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount ||
2558                    (frameCount == minFrameCount && thread->hasFastMixer() &&
2559                    /*minThread != NULL &&*/ !minThread->hasFastMixer()))) {
2560                minFrameCount = frameCount;
2561                minThread = thread;
2562            }
2563        }
2564    }
2565    return minThread;
2566}
2567
2568sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2569                                    audio_session_t triggerSession,
2570                                    audio_session_t listenerSession,
2571                                    sync_event_callback_t callBack,
2572                                    const wp<RefBase>& cookie)
2573{
2574    Mutex::Autolock _l(mLock);
2575
2576    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2577    status_t playStatus = NAME_NOT_FOUND;
2578    status_t recStatus = NAME_NOT_FOUND;
2579    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2580        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2581        if (playStatus == NO_ERROR) {
2582            return event;
2583        }
2584    }
2585    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2586        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2587        if (recStatus == NO_ERROR) {
2588            return event;
2589        }
2590    }
2591    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2592        mPendingSyncEvents.add(event);
2593    } else {
2594        ALOGV("createSyncEvent() invalid event %d", event->type());
2595        event.clear();
2596    }
2597    return event;
2598}
2599
2600// ----------------------------------------------------------------------------
2601//  Effect management
2602// ----------------------------------------------------------------------------
2603
2604sp<EffectsFactoryHalInterface> AudioFlinger::getEffectsFactory() {
2605    return mEffectsFactoryHal;
2606}
2607
2608status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2609{
2610    Mutex::Autolock _l(mLock);
2611    if (mEffectsFactoryHal.get()) {
2612        return mEffectsFactoryHal->queryNumberEffects(numEffects);
2613    } else {
2614        return -ENODEV;
2615    }
2616}
2617
2618status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2619{
2620    Mutex::Autolock _l(mLock);
2621    if (mEffectsFactoryHal.get()) {
2622        return mEffectsFactoryHal->getDescriptor(index, descriptor);
2623    } else {
2624        return -ENODEV;
2625    }
2626}
2627
2628status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2629        effect_descriptor_t *descriptor) const
2630{
2631    Mutex::Autolock _l(mLock);
2632    if (mEffectsFactoryHal.get()) {
2633        return mEffectsFactoryHal->getDescriptor(pUuid, descriptor);
2634    } else {
2635        return -ENODEV;
2636    }
2637}
2638
2639
2640sp<IEffect> AudioFlinger::createEffect(
2641        effect_descriptor_t *pDesc,
2642        const sp<IEffectClient>& effectClient,
2643        int32_t priority,
2644        audio_io_handle_t io,
2645        audio_session_t sessionId,
2646        const String16& opPackageName,
2647        status_t *status,
2648        int *id,
2649        int *enabled)
2650{
2651    status_t lStatus = NO_ERROR;
2652    sp<EffectHandle> handle;
2653    effect_descriptor_t desc;
2654
2655    pid_t pid = IPCThreadState::self()->getCallingPid();
2656    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d, factory %p",
2657            pid, effectClient.get(), priority, sessionId, io, mEffectsFactoryHal.get());
2658
2659    if (pDesc == NULL) {
2660        lStatus = BAD_VALUE;
2661        goto Exit;
2662    }
2663
2664    // check audio settings permission for global effects
2665    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2666        lStatus = PERMISSION_DENIED;
2667        goto Exit;
2668    }
2669
2670    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2671    // that can only be created by audio policy manager (running in same process)
2672    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2673        lStatus = PERMISSION_DENIED;
2674        goto Exit;
2675    }
2676
2677    if (mEffectsFactoryHal.get() == NULL) {
2678        lStatus = NO_INIT;
2679        goto Exit;
2680    }
2681
2682    {
2683        if (!EffectsFactoryHalInterface::isNullUuid(&pDesc->uuid)) {
2684            // if uuid is specified, request effect descriptor
2685            lStatus = mEffectsFactoryHal->getDescriptor(&pDesc->uuid, &desc);
2686            if (lStatus < 0) {
2687                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2688                goto Exit;
2689            }
2690        } else {
2691            // if uuid is not specified, look for an available implementation
2692            // of the required type in effect factory
2693            if (EffectsFactoryHalInterface::isNullUuid(&pDesc->type)) {
2694                ALOGW("createEffect() no effect type");
2695                lStatus = BAD_VALUE;
2696                goto Exit;
2697            }
2698            uint32_t numEffects = 0;
2699            effect_descriptor_t d;
2700            d.flags = 0; // prevent compiler warning
2701            bool found = false;
2702
2703            lStatus = mEffectsFactoryHal->queryNumberEffects(&numEffects);
2704            if (lStatus < 0) {
2705                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2706                goto Exit;
2707            }
2708            for (uint32_t i = 0; i < numEffects; i++) {
2709                lStatus = mEffectsFactoryHal->getDescriptor(i, &desc);
2710                if (lStatus < 0) {
2711                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2712                    continue;
2713                }
2714                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2715                    // If matching type found save effect descriptor. If the session is
2716                    // 0 and the effect is not auxiliary, continue enumeration in case
2717                    // an auxiliary version of this effect type is available
2718                    found = true;
2719                    d = desc;
2720                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2721                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2722                        break;
2723                    }
2724                }
2725            }
2726            if (!found) {
2727                lStatus = BAD_VALUE;
2728                ALOGW("createEffect() effect not found");
2729                goto Exit;
2730            }
2731            // For same effect type, chose auxiliary version over insert version if
2732            // connect to output mix (Compliance to OpenSL ES)
2733            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2734                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2735                desc = d;
2736            }
2737        }
2738
2739        // Do not allow auxiliary effects on a session different from 0 (output mix)
2740        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2741             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2742            lStatus = INVALID_OPERATION;
2743            goto Exit;
2744        }
2745
2746        // check recording permission for visualizer
2747        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2748            !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) {
2749            lStatus = PERMISSION_DENIED;
2750            goto Exit;
2751        }
2752
2753        // return effect descriptor
2754        *pDesc = desc;
2755        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2756            // if the output returned by getOutputForEffect() is removed before we lock the
2757            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2758            // and we will exit safely
2759            io = AudioSystem::getOutputForEffect(&desc);
2760            ALOGV("createEffect got output %d", io);
2761        }
2762
2763        Mutex::Autolock _l(mLock);
2764
2765        // If output is not specified try to find a matching audio session ID in one of the
2766        // output threads.
2767        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2768        // because of code checking output when entering the function.
2769        // Note: io is never 0 when creating an effect on an input
2770        if (io == AUDIO_IO_HANDLE_NONE) {
2771            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2772                // output must be specified by AudioPolicyManager when using session
2773                // AUDIO_SESSION_OUTPUT_STAGE
2774                lStatus = BAD_VALUE;
2775                goto Exit;
2776            }
2777            // look for the thread where the specified audio session is present
2778            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2779                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2780                    io = mPlaybackThreads.keyAt(i);
2781                    break;
2782                }
2783            }
2784            if (io == 0) {
2785                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2786                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2787                        io = mRecordThreads.keyAt(i);
2788                        break;
2789                    }
2790                }
2791            }
2792            // If no output thread contains the requested session ID, default to
2793            // first output. The effect chain will be moved to the correct output
2794            // thread when a track with the same session ID is created
2795            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2796                io = mPlaybackThreads.keyAt(0);
2797            }
2798            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2799        }
2800        ThreadBase *thread = checkRecordThread_l(io);
2801        if (thread == NULL) {
2802            thread = checkPlaybackThread_l(io);
2803            if (thread == NULL) {
2804                ALOGE("createEffect() unknown output thread");
2805                lStatus = BAD_VALUE;
2806                goto Exit;
2807            }
2808        } else {
2809            // Check if one effect chain was awaiting for an effect to be created on this
2810            // session and used it instead of creating a new one.
2811            sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
2812            if (chain != 0) {
2813                Mutex::Autolock _l(thread->mLock);
2814                thread->addEffectChain_l(chain);
2815            }
2816        }
2817
2818        sp<Client> client = registerPid(pid);
2819
2820        // create effect on selected output thread
2821        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2822                &desc, enabled, &lStatus);
2823        if (handle != 0 && id != NULL) {
2824            *id = handle->id();
2825        }
2826        if (handle == 0) {
2827            // remove local strong reference to Client with mClientLock held
2828            Mutex::Autolock _cl(mClientLock);
2829            client.clear();
2830        }
2831    }
2832
2833Exit:
2834    *status = lStatus;
2835    return handle;
2836}
2837
2838status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
2839        audio_io_handle_t dstOutput)
2840{
2841    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2842            sessionId, srcOutput, dstOutput);
2843    Mutex::Autolock _l(mLock);
2844    if (srcOutput == dstOutput) {
2845        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2846        return NO_ERROR;
2847    }
2848    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2849    if (srcThread == NULL) {
2850        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2851        return BAD_VALUE;
2852    }
2853    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2854    if (dstThread == NULL) {
2855        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2856        return BAD_VALUE;
2857    }
2858
2859    Mutex::Autolock _dl(dstThread->mLock);
2860    Mutex::Autolock _sl(srcThread->mLock);
2861    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2862}
2863
2864// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2865status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId,
2866                                   AudioFlinger::PlaybackThread *srcThread,
2867                                   AudioFlinger::PlaybackThread *dstThread,
2868                                   bool reRegister)
2869{
2870    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2871            sessionId, srcThread, dstThread);
2872
2873    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2874    if (chain == 0) {
2875        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2876                sessionId, srcThread);
2877        return INVALID_OPERATION;
2878    }
2879
2880    // Check whether the destination thread and all effects in the chain are compatible
2881    if (!chain->isCompatibleWithThread_l(dstThread)) {
2882        ALOGW("moveEffectChain_l() effect chain failed because"
2883                " destination thread %p is not compatible with effects in the chain",
2884                dstThread);
2885        return INVALID_OPERATION;
2886    }
2887
2888    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2889    // so that a new chain is created with correct parameters when first effect is added. This is
2890    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2891    // removed.
2892    srcThread->removeEffectChain_l(chain);
2893
2894    // transfer all effects one by one so that new effect chain is created on new thread with
2895    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2896    sp<EffectChain> dstChain;
2897    uint32_t strategy = 0; // prevent compiler warning
2898    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2899    Vector< sp<EffectModule> > removed;
2900    status_t status = NO_ERROR;
2901    while (effect != 0) {
2902        srcThread->removeEffect_l(effect);
2903        removed.add(effect);
2904        status = dstThread->addEffect_l(effect);
2905        if (status != NO_ERROR) {
2906            break;
2907        }
2908        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2909        if (effect->state() == EffectModule::ACTIVE ||
2910                effect->state() == EffectModule::STOPPING) {
2911            effect->start();
2912        }
2913        // if the move request is not received from audio policy manager, the effect must be
2914        // re-registered with the new strategy and output
2915        if (dstChain == 0) {
2916            dstChain = effect->chain().promote();
2917            if (dstChain == 0) {
2918                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2919                status = NO_INIT;
2920                break;
2921            }
2922            strategy = dstChain->strategy();
2923        }
2924        if (reRegister) {
2925            AudioSystem::unregisterEffect(effect->id());
2926            AudioSystem::registerEffect(&effect->desc(),
2927                                        dstThread->id(),
2928                                        strategy,
2929                                        sessionId,
2930                                        effect->id());
2931            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2932        }
2933        effect = chain->getEffectFromId_l(0);
2934    }
2935
2936    if (status != NO_ERROR) {
2937        for (size_t i = 0; i < removed.size(); i++) {
2938            srcThread->addEffect_l(removed[i]);
2939            if (dstChain != 0 && reRegister) {
2940                AudioSystem::unregisterEffect(removed[i]->id());
2941                AudioSystem::registerEffect(&removed[i]->desc(),
2942                                            srcThread->id(),
2943                                            strategy,
2944                                            sessionId,
2945                                            removed[i]->id());
2946                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2947            }
2948        }
2949    }
2950
2951    return status;
2952}
2953
2954bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2955{
2956    if (mGlobalEffectEnableTime != 0 &&
2957            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2958        return true;
2959    }
2960
2961    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2962        sp<EffectChain> ec =
2963                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2964        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2965            return true;
2966        }
2967    }
2968    return false;
2969}
2970
2971void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2972{
2973    Mutex::Autolock _l(mLock);
2974
2975    mGlobalEffectEnableTime = systemTime();
2976
2977    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2978        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2979        if (t->mType == ThreadBase::OFFLOAD) {
2980            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2981        }
2982    }
2983
2984}
2985
2986status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain)
2987{
2988    audio_session_t session = chain->sessionId();
2989    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2990    ALOGV("putOrphanEffectChain_l session %d index %zd", session, index);
2991    if (index >= 0) {
2992        ALOGW("putOrphanEffectChain_l chain for session %d already present", session);
2993        return ALREADY_EXISTS;
2994    }
2995    mOrphanEffectChains.add(session, chain);
2996    return NO_ERROR;
2997}
2998
2999sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session)
3000{
3001    sp<EffectChain> chain;
3002    ssize_t index = mOrphanEffectChains.indexOfKey(session);
3003    ALOGV("getOrphanEffectChain_l session %d index %zd", session, index);
3004    if (index >= 0) {
3005        chain = mOrphanEffectChains.valueAt(index);
3006        mOrphanEffectChains.removeItemsAt(index);
3007    }
3008    return chain;
3009}
3010
3011bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect)
3012{
3013    Mutex::Autolock _l(mLock);
3014    audio_session_t session = effect->sessionId();
3015    ssize_t index = mOrphanEffectChains.indexOfKey(session);
3016    ALOGV("updateOrphanEffectChains session %d index %zd", session, index);
3017    if (index >= 0) {
3018        sp<EffectChain> chain = mOrphanEffectChains.valueAt(index);
3019        if (chain->removeEffect_l(effect) == 0) {
3020            ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index);
3021            mOrphanEffectChains.removeItemsAt(index);
3022        }
3023        return true;
3024    }
3025    return false;
3026}
3027
3028
3029struct Entry {
3030#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21
3031    char mFileName[TEE_MAX_FILENAME];
3032};
3033
3034int comparEntry(const void *p1, const void *p2)
3035{
3036    return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName);
3037}
3038
3039#ifdef TEE_SINK
3040void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
3041{
3042    NBAIO_Source *teeSource = source.get();
3043    if (teeSource != NULL) {
3044        // .wav rotation
3045        // There is a benign race condition if 2 threads call this simultaneously.
3046        // They would both traverse the directory, but the result would simply be
3047        // failures at unlink() which are ignored.  It's also unlikely since
3048        // normally dumpsys is only done by bugreport or from the command line.
3049        char teePath[32+256];
3050        strcpy(teePath, "/data/misc/audioserver");
3051        size_t teePathLen = strlen(teePath);
3052        DIR *dir = opendir(teePath);
3053        teePath[teePathLen++] = '/';
3054        if (dir != NULL) {
3055#define TEE_MAX_SORT 20 // number of entries to sort
3056#define TEE_MAX_KEEP 10 // number of entries to keep
3057            struct Entry entries[TEE_MAX_SORT];
3058            size_t entryCount = 0;
3059            while (entryCount < TEE_MAX_SORT) {
3060                struct dirent de;
3061                struct dirent *result = NULL;
3062                int rc = readdir_r(dir, &de, &result);
3063                if (rc != 0) {
3064                    ALOGW("readdir_r failed %d", rc);
3065                    break;
3066                }
3067                if (result == NULL) {
3068                    break;
3069                }
3070                if (result != &de) {
3071                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
3072                    break;
3073                }
3074                // ignore non .wav file entries
3075                size_t nameLen = strlen(de.d_name);
3076                if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME ||
3077                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
3078                    continue;
3079                }
3080                strcpy(entries[entryCount++].mFileName, de.d_name);
3081            }
3082            (void) closedir(dir);
3083            if (entryCount > TEE_MAX_KEEP) {
3084                qsort(entries, entryCount, sizeof(Entry), comparEntry);
3085                for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) {
3086                    strcpy(&teePath[teePathLen], entries[i].mFileName);
3087                    (void) unlink(teePath);
3088                }
3089            }
3090        } else {
3091            if (fd >= 0) {
3092                dprintf(fd, "unable to rotate tees in %.*s: %s\n", (int) teePathLen, teePath,
3093                        strerror(errno));
3094            }
3095        }
3096        char teeTime[16];
3097        struct timeval tv;
3098        gettimeofday(&tv, NULL);
3099        struct tm tm;
3100        localtime_r(&tv.tv_sec, &tm);
3101        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
3102        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
3103        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
3104        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
3105        if (teeFd >= 0) {
3106            // FIXME use libsndfile
3107            char wavHeader[44];
3108            memcpy(wavHeader,
3109                "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",
3110                sizeof(wavHeader));
3111            NBAIO_Format format = teeSource->format();
3112            unsigned channelCount = Format_channelCount(format);
3113            uint32_t sampleRate = Format_sampleRate(format);
3114            size_t frameSize = Format_frameSize(format);
3115            wavHeader[22] = channelCount;       // number of channels
3116            wavHeader[24] = sampleRate;         // sample rate
3117            wavHeader[25] = sampleRate >> 8;
3118            wavHeader[32] = frameSize;          // block alignment
3119            wavHeader[33] = frameSize >> 8;
3120            write(teeFd, wavHeader, sizeof(wavHeader));
3121            size_t total = 0;
3122            bool firstRead = true;
3123#define TEE_SINK_READ 1024                      // frames per I/O operation
3124            void *buffer = malloc(TEE_SINK_READ * frameSize);
3125            for (;;) {
3126                size_t count = TEE_SINK_READ;
3127                ssize_t actual = teeSource->read(buffer, count);
3128                bool wasFirstRead = firstRead;
3129                firstRead = false;
3130                if (actual <= 0) {
3131                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
3132                        continue;
3133                    }
3134                    break;
3135                }
3136                ALOG_ASSERT(actual <= (ssize_t)count);
3137                write(teeFd, buffer, actual * frameSize);
3138                total += actual;
3139            }
3140            free(buffer);
3141            lseek(teeFd, (off_t) 4, SEEK_SET);
3142            uint32_t temp = 44 + total * frameSize - 8;
3143            // FIXME not big-endian safe
3144            write(teeFd, &temp, sizeof(temp));
3145            lseek(teeFd, (off_t) 40, SEEK_SET);
3146            temp =  total * frameSize;
3147            // FIXME not big-endian safe
3148            write(teeFd, &temp, sizeof(temp));
3149            close(teeFd);
3150            if (fd >= 0) {
3151                dprintf(fd, "tee copied to %s\n", teePath);
3152            }
3153        } else {
3154            if (fd >= 0) {
3155                dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
3156            }
3157        }
3158    }
3159}
3160#endif
3161
3162// ----------------------------------------------------------------------------
3163
3164status_t AudioFlinger::onTransact(
3165        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
3166{
3167    return BnAudioFlinger::onTransact(code, data, reply, flags);
3168}
3169
3170} // namespace android
3171