AudioFlinger.cpp revision 26d5ff926fa3323b39ae4408bcd29826a9523c9b 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <dirent.h>
24#include <math.h>
25#include <signal.h>
26#include <sys/time.h>
27#include <sys/resource.h>
28
29#include <binder/IPCThreadState.h>
30#include <binder/IServiceManager.h>
31#include <utils/Log.h>
32#include <utils/Trace.h>
33#include <binder/Parcel.h>
34#include <utils/String16.h>
35#include <utils/threads.h>
36#include <utils/Atomic.h>
37
38#include <cutils/bitops.h>
39#include <cutils/properties.h>
40
41#include <system/audio.h>
42#include <hardware/audio.h>
43
44#include "AudioMixer.h"
45#include "AudioFlinger.h"
46#include "ServiceUtilities.h"
47
48#include <media/EffectsFactoryApi.h>
49#include <audio_effects/effect_visualizer.h>
50#include <audio_effects/effect_ns.h>
51#include <audio_effects/effect_aec.h>
52
53#include <audio_utils/primitives.h>
54
55#include <powermanager/PowerManager.h>
56
57#include <common_time/cc_helper.h>
58
59#include <media/IMediaLogService.h>
60
61#include <media/nbaio/Pipe.h>
62#include <media/nbaio/PipeReader.h>
63#include <media/AudioParameter.h>
64#include <private/android_filesystem_config.h>
65
66// ----------------------------------------------------------------------------
67
68// Note: the following macro is used for extremely verbose logging message.  In
69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
70// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
71// are so verbose that we want to suppress them even when we have ALOG_ASSERT
72// turned on.  Do not uncomment the #def below unless you really know what you
73// are doing and want to see all of the extremely verbose messages.
74//#define VERY_VERY_VERBOSE_LOGGING
75#ifdef VERY_VERY_VERBOSE_LOGGING
76#define ALOGVV ALOGV
77#else
78#define ALOGVV(a...) do { } while(0)
79#endif
80
81namespace android {
82
83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
84static const char kHardwareLockedString[] = "Hardware lock is taken\n";
85
86
87nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
88
89uint32_t AudioFlinger::mScreenState;
90
91#ifdef TEE_SINK
92bool AudioFlinger::mTeeSinkInputEnabled = false;
93bool AudioFlinger::mTeeSinkOutputEnabled = false;
94bool AudioFlinger::mTeeSinkTrackEnabled = false;
95
96size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
97size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
98size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
99#endif
100
101// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
102// we define a minimum time during which a global effect is considered enabled.
103static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
104
105// ----------------------------------------------------------------------------
106
107const char *formatToString(audio_format_t format) {
108    switch(format) {
109    case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8";
110    case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16";
111    case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32";
112    case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24";
113    case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24";
114    case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat";
115    case AUDIO_FORMAT_MP3: return "mp3";
116    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
117    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
118    case AUDIO_FORMAT_AAC: return "aac";
119    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
120    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
121    case AUDIO_FORMAT_VORBIS: return "vorbis";
122    default:
123        break;
124    }
125    return "unknown";
126}
127
128static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
129{
130    const hw_module_t *mod;
131    int rc;
132
133    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
134    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
135                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
136    if (rc) {
137        goto out;
138    }
139    rc = audio_hw_device_open(mod, dev);
140    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
141                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
142    if (rc) {
143        goto out;
144    }
145    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
146        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
147        rc = BAD_VALUE;
148        goto out;
149    }
150    return 0;
151
152out:
153    *dev = NULL;
154    return rc;
155}
156
157// ----------------------------------------------------------------------------
158
159AudioFlinger::AudioFlinger()
160    : BnAudioFlinger(),
161      mPrimaryHardwareDev(NULL),
162      mAudioHwDevs(NULL),
163      mHardwareStatus(AUDIO_HW_IDLE),
164      mMasterVolume(1.0f),
165      mMasterMute(false),
166      mNextUniqueId(1),
167      mMode(AUDIO_MODE_INVALID),
168      mBtNrecIsOff(false),
169      mIsLowRamDevice(true),
170      mIsDeviceTypeKnown(false),
171      mGlobalEffectEnableTime(0)
172{
173    getpid_cached = getpid();
174    char value[PROPERTY_VALUE_MAX];
175    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
176    if (doLog) {
177        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY);
178    }
179#ifdef TEE_SINK
180    (void) property_get("ro.debuggable", value, "0");
181    int debuggable = atoi(value);
182    int teeEnabled = 0;
183    if (debuggable) {
184        (void) property_get("af.tee", value, "0");
185        teeEnabled = atoi(value);
186    }
187    // FIXME symbolic constants here
188    if (teeEnabled & 1) {
189        mTeeSinkInputEnabled = true;
190    }
191    if (teeEnabled & 2) {
192        mTeeSinkOutputEnabled = true;
193    }
194    if (teeEnabled & 4) {
195        mTeeSinkTrackEnabled = true;
196    }
197#endif
198}
199
200void AudioFlinger::onFirstRef()
201{
202    int rc = 0;
203
204    Mutex::Autolock _l(mLock);
205
206    /* TODO: move all this work into an Init() function */
207    char val_str[PROPERTY_VALUE_MAX] = { 0 };
208    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
209        uint32_t int_val;
210        if (1 == sscanf(val_str, "%u", &int_val)) {
211            mStandbyTimeInNsecs = milliseconds(int_val);
212            ALOGI("Using %u mSec as standby time.", int_val);
213        } else {
214            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
215            ALOGI("Using default %u mSec as standby time.",
216                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
217        }
218    }
219
220    mMode = AUDIO_MODE_NORMAL;
221}
222
223AudioFlinger::~AudioFlinger()
224{
225    while (!mRecordThreads.isEmpty()) {
226        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
227        closeInput_nonvirtual(mRecordThreads.keyAt(0));
228    }
229    while (!mPlaybackThreads.isEmpty()) {
230        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
231        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
232    }
233
234    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
235        // no mHardwareLock needed, as there are no other references to this
236        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
237        delete mAudioHwDevs.valueAt(i);
238    }
239
240    // Tell media.log service about any old writers that still need to be unregistered
241    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
242    if (binder != 0) {
243        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
244        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
245            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
246            mUnregisteredWriters.pop();
247            mediaLogService->unregisterWriter(iMemory);
248        }
249    }
250
251}
252
253static const char * const audio_interfaces[] = {
254    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
255    AUDIO_HARDWARE_MODULE_ID_A2DP,
256    AUDIO_HARDWARE_MODULE_ID_USB,
257};
258#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
259
260AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
261        audio_module_handle_t module,
262        audio_devices_t devices)
263{
264    // if module is 0, the request comes from an old policy manager and we should load
265    // well known modules
266    if (module == 0) {
267        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
268        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
269            loadHwModule_l(audio_interfaces[i]);
270        }
271        // then try to find a module supporting the requested device.
272        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
273            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
274            audio_hw_device_t *dev = audioHwDevice->hwDevice();
275            if ((dev->get_supported_devices != NULL) &&
276                    (dev->get_supported_devices(dev) & devices) == devices)
277                return audioHwDevice;
278        }
279    } else {
280        // check a match for the requested module handle
281        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
282        if (audioHwDevice != NULL) {
283            return audioHwDevice;
284        }
285    }
286
287    return NULL;
288}
289
290void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
291{
292    const size_t SIZE = 256;
293    char buffer[SIZE];
294    String8 result;
295
296    result.append("Clients:\n");
297    for (size_t i = 0; i < mClients.size(); ++i) {
298        sp<Client> client = mClients.valueAt(i).promote();
299        if (client != 0) {
300            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
301            result.append(buffer);
302        }
303    }
304
305    result.append("Notification Clients:\n");
306    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
307        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
308        result.append(buffer);
309    }
310
311    result.append("Global session refs:\n");
312    result.append("  session   pid count\n");
313    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
314        AudioSessionRef *r = mAudioSessionRefs[i];
315        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
316        result.append(buffer);
317    }
318    write(fd, result.string(), result.size());
319}
320
321
322void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
323{
324    const size_t SIZE = 256;
325    char buffer[SIZE];
326    String8 result;
327    hardware_call_state hardwareStatus = mHardwareStatus;
328
329    snprintf(buffer, SIZE, "Hardware status: %d\n"
330                           "Standby Time mSec: %u\n",
331                            hardwareStatus,
332                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
333    result.append(buffer);
334    write(fd, result.string(), result.size());
335}
336
337void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
338{
339    const size_t SIZE = 256;
340    char buffer[SIZE];
341    String8 result;
342    snprintf(buffer, SIZE, "Permission Denial: "
343            "can't dump AudioFlinger from pid=%d, uid=%d\n",
344            IPCThreadState::self()->getCallingPid(),
345            IPCThreadState::self()->getCallingUid());
346    result.append(buffer);
347    write(fd, result.string(), result.size());
348}
349
350bool AudioFlinger::dumpTryLock(Mutex& mutex)
351{
352    bool locked = false;
353    for (int i = 0; i < kDumpLockRetries; ++i) {
354        if (mutex.tryLock() == NO_ERROR) {
355            locked = true;
356            break;
357        }
358        usleep(kDumpLockSleepUs);
359    }
360    return locked;
361}
362
363status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
364{
365    if (!dumpAllowed()) {
366        dumpPermissionDenial(fd, args);
367    } else {
368        // get state of hardware lock
369        bool hardwareLocked = dumpTryLock(mHardwareLock);
370        if (!hardwareLocked) {
371            String8 result(kHardwareLockedString);
372            write(fd, result.string(), result.size());
373        } else {
374            mHardwareLock.unlock();
375        }
376
377        bool locked = dumpTryLock(mLock);
378
379        // failed to lock - AudioFlinger is probably deadlocked
380        if (!locked) {
381            String8 result(kDeadlockedString);
382            write(fd, result.string(), result.size());
383        }
384
385        dumpClients(fd, args);
386        dumpInternals(fd, args);
387
388        // dump playback threads
389        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
390            mPlaybackThreads.valueAt(i)->dump(fd, args);
391        }
392
393        // dump record threads
394        for (size_t i = 0; i < mRecordThreads.size(); i++) {
395            mRecordThreads.valueAt(i)->dump(fd, args);
396        }
397
398        // dump all hardware devs
399        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
400            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
401            dev->dump(dev, fd);
402        }
403
404#ifdef TEE_SINK
405        // dump the serially shared record tee sink
406        if (mRecordTeeSource != 0) {
407            dumpTee(fd, mRecordTeeSource);
408        }
409#endif
410
411        if (locked) {
412            mLock.unlock();
413        }
414
415        // append a copy of media.log here by forwarding fd to it, but don't attempt
416        // to lookup the service if it's not running, as it will block for a second
417        if (mLogMemoryDealer != 0) {
418            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
419            if (binder != 0) {
420                fdprintf(fd, "\nmedia.log:\n");
421                Vector<String16> args;
422                binder->dump(fd, args);
423            }
424        }
425    }
426    return NO_ERROR;
427}
428
429sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
430{
431    // If pid is already in the mClients wp<> map, then use that entry
432    // (for which promote() is always != 0), otherwise create a new entry and Client.
433    sp<Client> client = mClients.valueFor(pid).promote();
434    if (client == 0) {
435        client = new Client(this, pid);
436        mClients.add(pid, client);
437    }
438
439    return client;
440}
441
442sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
443{
444    // If there is no memory allocated for logs, return a dummy writer that does nothing
445    if (mLogMemoryDealer == 0) {
446        return new NBLog::Writer();
447    }
448    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
449    // Similarly if we can't contact the media.log service, also return a dummy writer
450    if (binder == 0) {
451        return new NBLog::Writer();
452    }
453    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
454    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
455    // If allocation fails, consult the vector of previously unregistered writers
456    // and garbage-collect one or more them until an allocation succeeds
457    if (shared == 0) {
458        Mutex::Autolock _l(mUnregisteredWritersLock);
459        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
460            {
461                // Pick the oldest stale writer to garbage-collect
462                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
463                mUnregisteredWriters.removeAt(0);
464                mediaLogService->unregisterWriter(iMemory);
465                // Now the media.log remote reference to IMemory is gone.  When our last local
466                // reference to IMemory also drops to zero at end of this block,
467                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
468            }
469            // Re-attempt the allocation
470            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
471            if (shared != 0) {
472                goto success;
473            }
474        }
475        // Even after garbage-collecting all old writers, there is still not enough memory,
476        // so return a dummy writer
477        return new NBLog::Writer();
478    }
479success:
480    mediaLogService->registerWriter(shared, size, name);
481    return new NBLog::Writer(size, shared);
482}
483
484void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
485{
486    if (writer == 0) {
487        return;
488    }
489    sp<IMemory> iMemory(writer->getIMemory());
490    if (iMemory == 0) {
491        return;
492    }
493    // Rather than removing the writer immediately, append it to a queue of old writers to
494    // be garbage-collected later.  This allows us to continue to view old logs for a while.
495    Mutex::Autolock _l(mUnregisteredWritersLock);
496    mUnregisteredWriters.push(writer);
497}
498
499// IAudioFlinger interface
500
501
502sp<IAudioTrack> AudioFlinger::createTrack(
503        audio_stream_type_t streamType,
504        uint32_t sampleRate,
505        audio_format_t format,
506        audio_channel_mask_t channelMask,
507        size_t *frameCount,
508        IAudioFlinger::track_flags_t *flags,
509        const sp<IMemory>& sharedBuffer,
510        audio_io_handle_t output,
511        pid_t tid,
512        int *sessionId,
513        int clientUid,
514        status_t *status)
515{
516    sp<PlaybackThread::Track> track;
517    sp<TrackHandle> trackHandle;
518    sp<Client> client;
519    status_t lStatus;
520    int lSessionId;
521
522    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
523    // but if someone uses binder directly they could bypass that and cause us to crash
524    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
525        ALOGE("createTrack() invalid stream type %d", streamType);
526        lStatus = BAD_VALUE;
527        goto Exit;
528    }
529
530    // further sample rate checks are performed by createTrack_l() depending on the thread type
531    if (sampleRate == 0) {
532        ALOGE("createTrack() invalid sample rate %u", sampleRate);
533        lStatus = BAD_VALUE;
534        goto Exit;
535    }
536
537    // further channel mask checks are performed by createTrack_l() depending on the thread type
538    if (!audio_is_output_channel(channelMask)) {
539        ALOGE("createTrack() invalid channel mask %#x", channelMask);
540        lStatus = BAD_VALUE;
541        goto Exit;
542    }
543
544    // further format checks are performed by createTrack_l() depending on the thread type
545    if (!audio_is_valid_format(format)) {
546        ALOGE("createTrack() invalid format %#x", format);
547        lStatus = BAD_VALUE;
548        goto Exit;
549    }
550
551    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
552        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
553        lStatus = BAD_VALUE;
554        goto Exit;
555    }
556
557    {
558        Mutex::Autolock _l(mLock);
559        PlaybackThread *thread = checkPlaybackThread_l(output);
560        if (thread == NULL) {
561            ALOGE("no playback thread found for output handle %d", output);
562            lStatus = BAD_VALUE;
563            goto Exit;
564        }
565
566        pid_t pid = IPCThreadState::self()->getCallingPid();
567        client = registerPid_l(pid);
568
569        PlaybackThread *effectThread = NULL;
570        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
571            lSessionId = *sessionId;
572            // check if an effect chain with the same session ID is present on another
573            // output thread and move it here.
574            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
575                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
576                if (mPlaybackThreads.keyAt(i) != output) {
577                    uint32_t sessions = t->hasAudioSession(lSessionId);
578                    if (sessions & PlaybackThread::EFFECT_SESSION) {
579                        effectThread = t.get();
580                        break;
581                    }
582                }
583            }
584        } else {
585            // if no audio session id is provided, create one here
586            lSessionId = nextUniqueId();
587            if (sessionId != NULL) {
588                *sessionId = lSessionId;
589            }
590        }
591        ALOGV("createTrack() lSessionId: %d", lSessionId);
592
593        track = thread->createTrack_l(client, streamType, sampleRate, format,
594                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
595        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
596        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
597
598        // move effect chain to this output thread if an effect on same session was waiting
599        // for a track to be created
600        if (lStatus == NO_ERROR && effectThread != NULL) {
601            // no risk of deadlock because AudioFlinger::mLock is held
602            Mutex::Autolock _dl(thread->mLock);
603            Mutex::Autolock _sl(effectThread->mLock);
604            moveEffectChain_l(lSessionId, effectThread, thread, true);
605        }
606
607        // Look for sync events awaiting for a session to be used.
608        for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
609            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
610                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
611                    if (lStatus == NO_ERROR) {
612                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
613                    } else {
614                        mPendingSyncEvents[i]->cancel();
615                    }
616                    mPendingSyncEvents.removeAt(i);
617                    i--;
618                }
619            }
620        }
621
622    }
623
624    if (lStatus != NO_ERROR) {
625        // remove local strong reference to Client before deleting the Track so that the
626        // Client destructor is called by the TrackBase destructor with mLock held
627        client.clear();
628        track.clear();
629        goto Exit;
630    }
631
632    // return handle to client
633    trackHandle = new TrackHandle(track);
634
635Exit:
636    *status = lStatus;
637    return trackHandle;
638}
639
640uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
641{
642    Mutex::Autolock _l(mLock);
643    PlaybackThread *thread = checkPlaybackThread_l(output);
644    if (thread == NULL) {
645        ALOGW("sampleRate() unknown thread %d", output);
646        return 0;
647    }
648    return thread->sampleRate();
649}
650
651int AudioFlinger::channelCount(audio_io_handle_t output) const
652{
653    Mutex::Autolock _l(mLock);
654    PlaybackThread *thread = checkPlaybackThread_l(output);
655    if (thread == NULL) {
656        ALOGW("channelCount() unknown thread %d", output);
657        return 0;
658    }
659    return thread->channelCount();
660}
661
662audio_format_t AudioFlinger::format(audio_io_handle_t output) const
663{
664    Mutex::Autolock _l(mLock);
665    PlaybackThread *thread = checkPlaybackThread_l(output);
666    if (thread == NULL) {
667        ALOGW("format() unknown thread %d", output);
668        return AUDIO_FORMAT_INVALID;
669    }
670    return thread->format();
671}
672
673size_t AudioFlinger::frameCount(audio_io_handle_t output) const
674{
675    Mutex::Autolock _l(mLock);
676    PlaybackThread *thread = checkPlaybackThread_l(output);
677    if (thread == NULL) {
678        ALOGW("frameCount() unknown thread %d", output);
679        return 0;
680    }
681    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
682    //       should examine all callers and fix them to handle smaller counts
683    return thread->frameCount();
684}
685
686uint32_t AudioFlinger::latency(audio_io_handle_t output) const
687{
688    Mutex::Autolock _l(mLock);
689    PlaybackThread *thread = checkPlaybackThread_l(output);
690    if (thread == NULL) {
691        ALOGW("latency(): no playback thread found for output handle %d", output);
692        return 0;
693    }
694    return thread->latency();
695}
696
697status_t AudioFlinger::setMasterVolume(float value)
698{
699    status_t ret = initCheck();
700    if (ret != NO_ERROR) {
701        return ret;
702    }
703
704    // check calling permissions
705    if (!settingsAllowed()) {
706        return PERMISSION_DENIED;
707    }
708
709    Mutex::Autolock _l(mLock);
710    mMasterVolume = value;
711
712    // Set master volume in the HALs which support it.
713    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
714        AutoMutex lock(mHardwareLock);
715        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
716
717        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
718        if (dev->canSetMasterVolume()) {
719            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
720        }
721        mHardwareStatus = AUDIO_HW_IDLE;
722    }
723
724    // Now set the master volume in each playback thread.  Playback threads
725    // assigned to HALs which do not have master volume support will apply
726    // master volume during the mix operation.  Threads with HALs which do
727    // support master volume will simply ignore the setting.
728    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
729        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
730
731    return NO_ERROR;
732}
733
734status_t AudioFlinger::setMode(audio_mode_t mode)
735{
736    status_t ret = initCheck();
737    if (ret != NO_ERROR) {
738        return ret;
739    }
740
741    // check calling permissions
742    if (!settingsAllowed()) {
743        return PERMISSION_DENIED;
744    }
745    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
746        ALOGW("Illegal value: setMode(%d)", mode);
747        return BAD_VALUE;
748    }
749
750    { // scope for the lock
751        AutoMutex lock(mHardwareLock);
752        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
753        mHardwareStatus = AUDIO_HW_SET_MODE;
754        ret = dev->set_mode(dev, mode);
755        mHardwareStatus = AUDIO_HW_IDLE;
756    }
757
758    if (NO_ERROR == ret) {
759        Mutex::Autolock _l(mLock);
760        mMode = mode;
761        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
762            mPlaybackThreads.valueAt(i)->setMode(mode);
763    }
764
765    return ret;
766}
767
768status_t AudioFlinger::setMicMute(bool state)
769{
770    status_t ret = initCheck();
771    if (ret != NO_ERROR) {
772        return ret;
773    }
774
775    // check calling permissions
776    if (!settingsAllowed()) {
777        return PERMISSION_DENIED;
778    }
779
780    AutoMutex lock(mHardwareLock);
781    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
782    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
783    ret = dev->set_mic_mute(dev, state);
784    mHardwareStatus = AUDIO_HW_IDLE;
785    return ret;
786}
787
788bool AudioFlinger::getMicMute() const
789{
790    status_t ret = initCheck();
791    if (ret != NO_ERROR) {
792        return false;
793    }
794
795    bool state = AUDIO_MODE_INVALID;
796    AutoMutex lock(mHardwareLock);
797    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
798    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
799    dev->get_mic_mute(dev, &state);
800    mHardwareStatus = AUDIO_HW_IDLE;
801    return state;
802}
803
804status_t AudioFlinger::setMasterMute(bool muted)
805{
806    status_t ret = initCheck();
807    if (ret != NO_ERROR) {
808        return ret;
809    }
810
811    // check calling permissions
812    if (!settingsAllowed()) {
813        return PERMISSION_DENIED;
814    }
815
816    Mutex::Autolock _l(mLock);
817    mMasterMute = muted;
818
819    // Set master mute in the HALs which support it.
820    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
821        AutoMutex lock(mHardwareLock);
822        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
823
824        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
825        if (dev->canSetMasterMute()) {
826            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
827        }
828        mHardwareStatus = AUDIO_HW_IDLE;
829    }
830
831    // Now set the master mute in each playback thread.  Playback threads
832    // assigned to HALs which do not have master mute support will apply master
833    // mute during the mix operation.  Threads with HALs which do support master
834    // mute will simply ignore the setting.
835    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
836        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
837
838    return NO_ERROR;
839}
840
841float AudioFlinger::masterVolume() const
842{
843    Mutex::Autolock _l(mLock);
844    return masterVolume_l();
845}
846
847bool AudioFlinger::masterMute() const
848{
849    Mutex::Autolock _l(mLock);
850    return masterMute_l();
851}
852
853float AudioFlinger::masterVolume_l() const
854{
855    return mMasterVolume;
856}
857
858bool AudioFlinger::masterMute_l() const
859{
860    return mMasterMute;
861}
862
863status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
864        audio_io_handle_t output)
865{
866    // check calling permissions
867    if (!settingsAllowed()) {
868        return PERMISSION_DENIED;
869    }
870
871    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
872        ALOGE("setStreamVolume() invalid stream %d", stream);
873        return BAD_VALUE;
874    }
875
876    AutoMutex lock(mLock);
877    PlaybackThread *thread = NULL;
878    if (output != AUDIO_IO_HANDLE_NONE) {
879        thread = checkPlaybackThread_l(output);
880        if (thread == NULL) {
881            return BAD_VALUE;
882        }
883    }
884
885    mStreamTypes[stream].volume = value;
886
887    if (thread == NULL) {
888        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
889            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
890        }
891    } else {
892        thread->setStreamVolume(stream, value);
893    }
894
895    return NO_ERROR;
896}
897
898status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
899{
900    // check calling permissions
901    if (!settingsAllowed()) {
902        return PERMISSION_DENIED;
903    }
904
905    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
906        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
907        ALOGE("setStreamMute() invalid stream %d", stream);
908        return BAD_VALUE;
909    }
910
911    AutoMutex lock(mLock);
912    mStreamTypes[stream].mute = muted;
913    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
914        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
915
916    return NO_ERROR;
917}
918
919float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
920{
921    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
922        return 0.0f;
923    }
924
925    AutoMutex lock(mLock);
926    float volume;
927    if (output != AUDIO_IO_HANDLE_NONE) {
928        PlaybackThread *thread = checkPlaybackThread_l(output);
929        if (thread == NULL) {
930            return 0.0f;
931        }
932        volume = thread->streamVolume(stream);
933    } else {
934        volume = streamVolume_l(stream);
935    }
936
937    return volume;
938}
939
940bool AudioFlinger::streamMute(audio_stream_type_t stream) const
941{
942    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
943        return true;
944    }
945
946    AutoMutex lock(mLock);
947    return streamMute_l(stream);
948}
949
950status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
951{
952    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
953            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
954
955    // check calling permissions
956    if (!settingsAllowed()) {
957        return PERMISSION_DENIED;
958    }
959
960    // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
961    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
962        Mutex::Autolock _l(mLock);
963        status_t final_result = NO_ERROR;
964        {
965            AutoMutex lock(mHardwareLock);
966            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
967            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
968                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
969                status_t result = dev->set_parameters(dev, keyValuePairs.string());
970                final_result = result ?: final_result;
971            }
972            mHardwareStatus = AUDIO_HW_IDLE;
973        }
974        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
975        AudioParameter param = AudioParameter(keyValuePairs);
976        String8 value;
977        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
978            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
979            if (mBtNrecIsOff != btNrecIsOff) {
980                for (size_t i = 0; i < mRecordThreads.size(); i++) {
981                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
982                    audio_devices_t device = thread->inDevice();
983                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
984                    // collect all of the thread's session IDs
985                    KeyedVector<int, bool> ids = thread->sessionIds();
986                    // suspend effects associated with those session IDs
987                    for (size_t j = 0; j < ids.size(); ++j) {
988                        int sessionId = ids.keyAt(j);
989                        thread->setEffectSuspended(FX_IID_AEC,
990                                                   suspend,
991                                                   sessionId);
992                        thread->setEffectSuspended(FX_IID_NS,
993                                                   suspend,
994                                                   sessionId);
995                    }
996                }
997                mBtNrecIsOff = btNrecIsOff;
998            }
999        }
1000        String8 screenState;
1001        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1002            bool isOff = screenState == "off";
1003            if (isOff != (AudioFlinger::mScreenState & 1)) {
1004                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1005            }
1006        }
1007        return final_result;
1008    }
1009
1010    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1011    // and the thread is exited once the lock is released
1012    sp<ThreadBase> thread;
1013    {
1014        Mutex::Autolock _l(mLock);
1015        thread = checkPlaybackThread_l(ioHandle);
1016        if (thread == 0) {
1017            thread = checkRecordThread_l(ioHandle);
1018        } else if (thread == primaryPlaybackThread_l()) {
1019            // indicate output device change to all input threads for pre processing
1020            AudioParameter param = AudioParameter(keyValuePairs);
1021            int value;
1022            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1023                    (value != 0)) {
1024                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1025                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1026                }
1027            }
1028        }
1029    }
1030    if (thread != 0) {
1031        return thread->setParameters(keyValuePairs);
1032    }
1033    return BAD_VALUE;
1034}
1035
1036String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1037{
1038    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1039            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1040
1041    Mutex::Autolock _l(mLock);
1042
1043    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1044        String8 out_s8;
1045
1046        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1047            char *s;
1048            {
1049            AutoMutex lock(mHardwareLock);
1050            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1051            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1052            s = dev->get_parameters(dev, keys.string());
1053            mHardwareStatus = AUDIO_HW_IDLE;
1054            }
1055            out_s8 += String8(s ? s : "");
1056            free(s);
1057        }
1058        return out_s8;
1059    }
1060
1061    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1062    if (playbackThread != NULL) {
1063        return playbackThread->getParameters(keys);
1064    }
1065    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1066    if (recordThread != NULL) {
1067        return recordThread->getParameters(keys);
1068    }
1069    return String8("");
1070}
1071
1072size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1073        audio_channel_mask_t channelMask) const
1074{
1075    status_t ret = initCheck();
1076    if (ret != NO_ERROR) {
1077        return 0;
1078    }
1079
1080    AutoMutex lock(mHardwareLock);
1081    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1082    struct audio_config config;
1083    memset(&config, 0, sizeof(config));
1084    config.sample_rate = sampleRate;
1085    config.channel_mask = channelMask;
1086    config.format = format;
1087
1088    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1089    size_t size = dev->get_input_buffer_size(dev, &config);
1090    mHardwareStatus = AUDIO_HW_IDLE;
1091    return size;
1092}
1093
1094uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1095{
1096    Mutex::Autolock _l(mLock);
1097
1098    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1099    if (recordThread != NULL) {
1100        return recordThread->getInputFramesLost();
1101    }
1102    return 0;
1103}
1104
1105status_t AudioFlinger::setVoiceVolume(float value)
1106{
1107    status_t ret = initCheck();
1108    if (ret != NO_ERROR) {
1109        return ret;
1110    }
1111
1112    // check calling permissions
1113    if (!settingsAllowed()) {
1114        return PERMISSION_DENIED;
1115    }
1116
1117    AutoMutex lock(mHardwareLock);
1118    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1119    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1120    ret = dev->set_voice_volume(dev, value);
1121    mHardwareStatus = AUDIO_HW_IDLE;
1122
1123    return ret;
1124}
1125
1126status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1127        audio_io_handle_t output) const
1128{
1129    status_t status;
1130
1131    Mutex::Autolock _l(mLock);
1132
1133    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1134    if (playbackThread != NULL) {
1135        return playbackThread->getRenderPosition(halFrames, dspFrames);
1136    }
1137
1138    return BAD_VALUE;
1139}
1140
1141void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1142{
1143
1144    Mutex::Autolock _l(mLock);
1145
1146    pid_t pid = IPCThreadState::self()->getCallingPid();
1147    if (mNotificationClients.indexOfKey(pid) < 0) {
1148        sp<NotificationClient> notificationClient = new NotificationClient(this,
1149                                                                            client,
1150                                                                            pid);
1151        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1152
1153        mNotificationClients.add(pid, notificationClient);
1154
1155        sp<IBinder> binder = client->asBinder();
1156        binder->linkToDeath(notificationClient);
1157
1158        // the config change is always sent from playback or record threads to avoid deadlock
1159        // with AudioSystem::gLock
1160        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1161            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1162        }
1163
1164        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1165            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1166        }
1167    }
1168}
1169
1170void AudioFlinger::removeNotificationClient(pid_t pid)
1171{
1172    Mutex::Autolock _l(mLock);
1173
1174    mNotificationClients.removeItem(pid);
1175
1176    ALOGV("%d died, releasing its sessions", pid);
1177    size_t num = mAudioSessionRefs.size();
1178    bool removed = false;
1179    for (size_t i = 0; i< num; ) {
1180        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1181        ALOGV(" pid %d @ %d", ref->mPid, i);
1182        if (ref->mPid == pid) {
1183            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1184            mAudioSessionRefs.removeAt(i);
1185            delete ref;
1186            removed = true;
1187            num--;
1188        } else {
1189            i++;
1190        }
1191    }
1192    if (removed) {
1193        purgeStaleEffects_l();
1194    }
1195}
1196
1197// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1198void AudioFlinger::audioConfigChanged_l(
1199                    const DefaultKeyedVector< pid_t,sp<NotificationClient> >& notificationClients,
1200                    int event,
1201                    audio_io_handle_t ioHandle,
1202                    const void *param2)
1203{
1204    size_t size = notificationClients.size();
1205    for (size_t i = 0; i < size; i++) {
1206        notificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event,
1207                                                                              ioHandle,
1208                                                                              param2);
1209    }
1210}
1211
1212// removeClient_l() must be called with AudioFlinger::mLock held
1213void AudioFlinger::removeClient_l(pid_t pid)
1214{
1215    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1216            IPCThreadState::self()->getCallingPid());
1217    mClients.removeItem(pid);
1218}
1219
1220// getEffectThread_l() must be called with AudioFlinger::mLock held
1221sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1222{
1223    sp<PlaybackThread> thread;
1224
1225    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1226        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1227            ALOG_ASSERT(thread == 0);
1228            thread = mPlaybackThreads.valueAt(i);
1229        }
1230    }
1231
1232    return thread;
1233}
1234
1235
1236
1237// ----------------------------------------------------------------------------
1238
1239AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1240    :   RefBase(),
1241        mAudioFlinger(audioFlinger),
1242        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1243        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1244        mPid(pid),
1245        mTimedTrackCount(0)
1246{
1247    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1248}
1249
1250// Client destructor must be called with AudioFlinger::mLock held
1251AudioFlinger::Client::~Client()
1252{
1253    mAudioFlinger->removeClient_l(mPid);
1254}
1255
1256sp<MemoryDealer> AudioFlinger::Client::heap() const
1257{
1258    return mMemoryDealer;
1259}
1260
1261// Reserve one of the limited slots for a timed audio track associated
1262// with this client
1263bool AudioFlinger::Client::reserveTimedTrack()
1264{
1265    const int kMaxTimedTracksPerClient = 4;
1266
1267    Mutex::Autolock _l(mTimedTrackLock);
1268
1269    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1270        ALOGW("can not create timed track - pid %d has exceeded the limit",
1271             mPid);
1272        return false;
1273    }
1274
1275    mTimedTrackCount++;
1276    return true;
1277}
1278
1279// Release a slot for a timed audio track
1280void AudioFlinger::Client::releaseTimedTrack()
1281{
1282    Mutex::Autolock _l(mTimedTrackLock);
1283    mTimedTrackCount--;
1284}
1285
1286// ----------------------------------------------------------------------------
1287
1288AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1289                                                     const sp<IAudioFlingerClient>& client,
1290                                                     pid_t pid)
1291    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1292{
1293}
1294
1295AudioFlinger::NotificationClient::~NotificationClient()
1296{
1297}
1298
1299void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1300{
1301    sp<NotificationClient> keep(this);
1302    mAudioFlinger->removeNotificationClient(mPid);
1303}
1304
1305
1306// ----------------------------------------------------------------------------
1307
1308static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1309    return audio_is_remote_submix_device(inDevice);
1310}
1311
1312sp<IAudioRecord> AudioFlinger::openRecord(
1313        audio_io_handle_t input,
1314        uint32_t sampleRate,
1315        audio_format_t format,
1316        audio_channel_mask_t channelMask,
1317        size_t *frameCount,
1318        IAudioFlinger::track_flags_t *flags,
1319        pid_t tid,
1320        int *sessionId,
1321        sp<IMemory>& cblk,
1322        sp<IMemory>& buffers,
1323        status_t *status)
1324{
1325    sp<RecordThread::RecordTrack> recordTrack;
1326    sp<RecordHandle> recordHandle;
1327    sp<Client> client;
1328    status_t lStatus;
1329    int lSessionId;
1330
1331    cblk.clear();
1332    buffers.clear();
1333
1334    // check calling permissions
1335    if (!recordingAllowed()) {
1336        ALOGE("openRecord() permission denied: recording not allowed");
1337        lStatus = PERMISSION_DENIED;
1338        goto Exit;
1339    }
1340
1341    // further sample rate checks are performed by createRecordTrack_l()
1342    if (sampleRate == 0) {
1343        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1344        lStatus = BAD_VALUE;
1345        goto Exit;
1346    }
1347
1348    // we don't yet support anything other than 16-bit PCM
1349    if (!(audio_is_valid_format(format) &&
1350            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
1351        ALOGE("openRecord() invalid format %#x", format);
1352        lStatus = BAD_VALUE;
1353        goto Exit;
1354    }
1355
1356    // further channel mask checks are performed by createRecordTrack_l()
1357    if (!audio_is_input_channel(channelMask)) {
1358        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1359        lStatus = BAD_VALUE;
1360        goto Exit;
1361    }
1362
1363    {
1364        Mutex::Autolock _l(mLock);
1365        RecordThread *thread = checkRecordThread_l(input);
1366        if (thread == NULL) {
1367            ALOGE("openRecord() checkRecordThread_l failed");
1368            lStatus = BAD_VALUE;
1369            goto Exit;
1370        }
1371
1372        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1373                && !captureAudioOutputAllowed()) {
1374            ALOGE("openRecord() permission denied: capture not allowed");
1375            lStatus = PERMISSION_DENIED;
1376            goto Exit;
1377        }
1378
1379        pid_t pid = IPCThreadState::self()->getCallingPid();
1380        client = registerPid_l(pid);
1381
1382        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1383            lSessionId = *sessionId;
1384        } else {
1385            // if no audio session id is provided, create one here
1386            lSessionId = nextUniqueId();
1387            if (sessionId != NULL) {
1388                *sessionId = lSessionId;
1389            }
1390        }
1391        ALOGV("openRecord() lSessionId: %d", lSessionId);
1392
1393        // TODO: the uid should be passed in as a parameter to openRecord
1394        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1395                                                  frameCount, lSessionId,
1396                                                  IPCThreadState::self()->getCallingUid(),
1397                                                  flags, tid, &lStatus);
1398        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1399    }
1400
1401    if (lStatus != NO_ERROR) {
1402        // remove local strong reference to Client before deleting the RecordTrack so that the
1403        // Client destructor is called by the TrackBase destructor with mLock held
1404        client.clear();
1405        recordTrack.clear();
1406        goto Exit;
1407    }
1408
1409    cblk = recordTrack->getCblk();
1410    buffers = recordTrack->getBuffers();
1411
1412    // return handle to client
1413    recordHandle = new RecordHandle(recordTrack);
1414
1415Exit:
1416    *status = lStatus;
1417    return recordHandle;
1418}
1419
1420
1421
1422// ----------------------------------------------------------------------------
1423
1424audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1425{
1426    if (!settingsAllowed()) {
1427        return 0;
1428    }
1429    Mutex::Autolock _l(mLock);
1430    return loadHwModule_l(name);
1431}
1432
1433// loadHwModule_l() must be called with AudioFlinger::mLock held
1434audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1435{
1436    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1437        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1438            ALOGW("loadHwModule() module %s already loaded", name);
1439            return mAudioHwDevs.keyAt(i);
1440        }
1441    }
1442
1443    audio_hw_device_t *dev;
1444
1445    int rc = load_audio_interface(name, &dev);
1446    if (rc) {
1447        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1448        return 0;
1449    }
1450
1451    mHardwareStatus = AUDIO_HW_INIT;
1452    rc = dev->init_check(dev);
1453    mHardwareStatus = AUDIO_HW_IDLE;
1454    if (rc) {
1455        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1456        return 0;
1457    }
1458
1459    // Check and cache this HAL's level of support for master mute and master
1460    // volume.  If this is the first HAL opened, and it supports the get
1461    // methods, use the initial values provided by the HAL as the current
1462    // master mute and volume settings.
1463
1464    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1465    {  // scope for auto-lock pattern
1466        AutoMutex lock(mHardwareLock);
1467
1468        if (0 == mAudioHwDevs.size()) {
1469            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1470            if (NULL != dev->get_master_volume) {
1471                float mv;
1472                if (OK == dev->get_master_volume(dev, &mv)) {
1473                    mMasterVolume = mv;
1474                }
1475            }
1476
1477            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1478            if (NULL != dev->get_master_mute) {
1479                bool mm;
1480                if (OK == dev->get_master_mute(dev, &mm)) {
1481                    mMasterMute = mm;
1482                }
1483            }
1484        }
1485
1486        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1487        if ((NULL != dev->set_master_volume) &&
1488            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1489            flags = static_cast<AudioHwDevice::Flags>(flags |
1490                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1491        }
1492
1493        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1494        if ((NULL != dev->set_master_mute) &&
1495            (OK == dev->set_master_mute(dev, mMasterMute))) {
1496            flags = static_cast<AudioHwDevice::Flags>(flags |
1497                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1498        }
1499
1500        mHardwareStatus = AUDIO_HW_IDLE;
1501    }
1502
1503    audio_module_handle_t handle = nextUniqueId();
1504    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1505
1506    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1507          name, dev->common.module->name, dev->common.module->id, handle);
1508
1509    return handle;
1510
1511}
1512
1513// ----------------------------------------------------------------------------
1514
1515uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1516{
1517    Mutex::Autolock _l(mLock);
1518    PlaybackThread *thread = primaryPlaybackThread_l();
1519    return thread != NULL ? thread->sampleRate() : 0;
1520}
1521
1522size_t AudioFlinger::getPrimaryOutputFrameCount()
1523{
1524    Mutex::Autolock _l(mLock);
1525    PlaybackThread *thread = primaryPlaybackThread_l();
1526    return thread != NULL ? thread->frameCountHAL() : 0;
1527}
1528
1529// ----------------------------------------------------------------------------
1530
1531status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1532{
1533    uid_t uid = IPCThreadState::self()->getCallingUid();
1534    if (uid != AID_SYSTEM) {
1535        return PERMISSION_DENIED;
1536    }
1537    Mutex::Autolock _l(mLock);
1538    if (mIsDeviceTypeKnown) {
1539        return INVALID_OPERATION;
1540    }
1541    mIsLowRamDevice = isLowRamDevice;
1542    mIsDeviceTypeKnown = true;
1543    return NO_ERROR;
1544}
1545
1546// ----------------------------------------------------------------------------
1547
1548audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1549                                           audio_devices_t *pDevices,
1550                                           uint32_t *pSamplingRate,
1551                                           audio_format_t *pFormat,
1552                                           audio_channel_mask_t *pChannelMask,
1553                                           uint32_t *pLatencyMs,
1554                                           audio_output_flags_t flags,
1555                                           const audio_offload_info_t *offloadInfo)
1556{
1557    struct audio_config config;
1558    memset(&config, 0, sizeof(config));
1559    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1560    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1561    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1562    if (offloadInfo != NULL) {
1563        config.offload_info = *offloadInfo;
1564    }
1565
1566    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1567              module,
1568              (pDevices != NULL) ? *pDevices : 0,
1569              config.sample_rate,
1570              config.format,
1571              config.channel_mask,
1572              flags);
1573    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1574          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1575
1576    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1577        return AUDIO_IO_HANDLE_NONE;
1578    }
1579
1580    Mutex::Autolock _l(mLock);
1581
1582    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1583    if (outHwDev == NULL) {
1584        return AUDIO_IO_HANDLE_NONE;
1585    }
1586
1587    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1588    audio_io_handle_t id = nextUniqueId();
1589
1590    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1591
1592    audio_stream_out_t *outStream = NULL;
1593    status_t status = hwDevHal->open_output_stream(hwDevHal,
1594                                          id,
1595                                          *pDevices,
1596                                          (audio_output_flags_t)flags,
1597                                          &config,
1598                                          &outStream);
1599
1600    mHardwareStatus = AUDIO_HW_IDLE;
1601    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1602            "Channels %x, status %d",
1603            outStream,
1604            config.sample_rate,
1605            config.format,
1606            config.channel_mask,
1607            status);
1608
1609    if (status == NO_ERROR && outStream != NULL) {
1610        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1611
1612        PlaybackThread *thread;
1613        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1614            thread = new OffloadThread(this, output, id, *pDevices);
1615            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1616        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1617            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1618            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1619            thread = new DirectOutputThread(this, output, id, *pDevices);
1620            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1621        } else {
1622            thread = new MixerThread(this, output, id, *pDevices);
1623            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1624        }
1625        mPlaybackThreads.add(id, thread);
1626
1627        if (pSamplingRate != NULL) {
1628            *pSamplingRate = config.sample_rate;
1629        }
1630        if (pFormat != NULL) {
1631            *pFormat = config.format;
1632        }
1633        if (pChannelMask != NULL) {
1634            *pChannelMask = config.channel_mask;
1635        }
1636        if (pLatencyMs != NULL) {
1637            *pLatencyMs = thread->latency();
1638        }
1639
1640        // notify client processes of the new output creation
1641        thread->audioConfigChanged_l(mNotificationClients, AudioSystem::OUTPUT_OPENED);
1642
1643        // the first primary output opened designates the primary hw device
1644        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1645            ALOGI("Using module %d has the primary audio interface", module);
1646            mPrimaryHardwareDev = outHwDev;
1647
1648            AutoMutex lock(mHardwareLock);
1649            mHardwareStatus = AUDIO_HW_SET_MODE;
1650            hwDevHal->set_mode(hwDevHal, mMode);
1651            mHardwareStatus = AUDIO_HW_IDLE;
1652        }
1653        return id;
1654    }
1655
1656    return AUDIO_IO_HANDLE_NONE;
1657}
1658
1659audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1660        audio_io_handle_t output2)
1661{
1662    Mutex::Autolock _l(mLock);
1663    MixerThread *thread1 = checkMixerThread_l(output1);
1664    MixerThread *thread2 = checkMixerThread_l(output2);
1665
1666    if (thread1 == NULL || thread2 == NULL) {
1667        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1668                output2);
1669        return AUDIO_IO_HANDLE_NONE;
1670    }
1671
1672    audio_io_handle_t id = nextUniqueId();
1673    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1674    thread->addOutputTrack(thread2);
1675    mPlaybackThreads.add(id, thread);
1676    // notify client processes of the new output creation
1677    thread->audioConfigChanged_l(mNotificationClients, AudioSystem::OUTPUT_OPENED);
1678    return id;
1679}
1680
1681status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1682{
1683    return closeOutput_nonvirtual(output);
1684}
1685
1686status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1687{
1688    // keep strong reference on the playback thread so that
1689    // it is not destroyed while exit() is executed
1690    sp<PlaybackThread> thread;
1691    {
1692        Mutex::Autolock _l(mLock);
1693        thread = checkPlaybackThread_l(output);
1694        if (thread == NULL) {
1695            return BAD_VALUE;
1696        }
1697
1698        ALOGV("closeOutput() %d", output);
1699
1700        if (thread->type() == ThreadBase::MIXER) {
1701            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1702                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1703                    DuplicatingThread *dupThread =
1704                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1705                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1706
1707                }
1708            }
1709        }
1710
1711
1712        mPlaybackThreads.removeItem(output);
1713        // save all effects to the default thread
1714        if (mPlaybackThreads.size()) {
1715            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1716            if (dstThread != NULL) {
1717                // audioflinger lock is held here so the acquisition order of thread locks does not
1718                // matter
1719                Mutex::Autolock _dl(dstThread->mLock);
1720                Mutex::Autolock _sl(thread->mLock);
1721                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1722                for (size_t i = 0; i < effectChains.size(); i ++) {
1723                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1724                }
1725            }
1726        }
1727        audioConfigChanged_l(mNotificationClients, AudioSystem::OUTPUT_CLOSED, output, NULL);
1728    }
1729    thread->exit();
1730    // The thread entity (active unit of execution) is no longer running here,
1731    // but the ThreadBase container still exists.
1732
1733    if (thread->type() != ThreadBase::DUPLICATING) {
1734        AudioStreamOut *out = thread->clearOutput();
1735        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1736        // from now on thread->mOutput is NULL
1737        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1738        delete out;
1739    }
1740    return NO_ERROR;
1741}
1742
1743status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1744{
1745    Mutex::Autolock _l(mLock);
1746    PlaybackThread *thread = checkPlaybackThread_l(output);
1747
1748    if (thread == NULL) {
1749        return BAD_VALUE;
1750    }
1751
1752    ALOGV("suspendOutput() %d", output);
1753    thread->suspend();
1754
1755    return NO_ERROR;
1756}
1757
1758status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1759{
1760    Mutex::Autolock _l(mLock);
1761    PlaybackThread *thread = checkPlaybackThread_l(output);
1762
1763    if (thread == NULL) {
1764        return BAD_VALUE;
1765    }
1766
1767    ALOGV("restoreOutput() %d", output);
1768
1769    thread->restore();
1770
1771    return NO_ERROR;
1772}
1773
1774audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1775                                          audio_devices_t *pDevices,
1776                                          uint32_t *pSamplingRate,
1777                                          audio_format_t *pFormat,
1778                                          audio_channel_mask_t *pChannelMask)
1779{
1780    struct audio_config config;
1781    memset(&config, 0, sizeof(config));
1782    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1783    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1784    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1785
1786    uint32_t reqSamplingRate = config.sample_rate;
1787    audio_format_t reqFormat = config.format;
1788    audio_channel_mask_t reqChannelMask = config.channel_mask;
1789
1790    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1791        return 0;
1792    }
1793
1794    Mutex::Autolock _l(mLock);
1795
1796    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1797    if (inHwDev == NULL) {
1798        return 0;
1799    }
1800
1801    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1802    audio_io_handle_t id = nextUniqueId();
1803
1804    audio_stream_in_t *inStream = NULL;
1805    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1806                                        &inStream);
1807    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
1808            "status %d",
1809            inStream,
1810            config.sample_rate,
1811            config.format,
1812            config.channel_mask,
1813            status);
1814
1815    // If the input could not be opened with the requested parameters and we can handle the
1816    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1817    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1818    if (status == BAD_VALUE &&
1819        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1820        (config.sample_rate <= 2 * reqSamplingRate) &&
1821        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1822        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1823        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1824        inStream = NULL;
1825        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1826        // FIXME log this new status; HAL should not propose any further changes
1827    }
1828
1829    if (status == NO_ERROR && inStream != NULL) {
1830
1831#ifdef TEE_SINK
1832        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1833        // or (re-)create if current Pipe is idle and does not match the new format
1834        sp<NBAIO_Sink> teeSink;
1835        enum {
1836            TEE_SINK_NO,    // don't copy input
1837            TEE_SINK_NEW,   // copy input using a new pipe
1838            TEE_SINK_OLD,   // copy input using an existing pipe
1839        } kind;
1840        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1841                                        popcount(inStream->common.get_channels(&inStream->common)));
1842        if (!mTeeSinkInputEnabled) {
1843            kind = TEE_SINK_NO;
1844        } else if (!Format_isValid(format)) {
1845            kind = TEE_SINK_NO;
1846        } else if (mRecordTeeSink == 0) {
1847            kind = TEE_SINK_NEW;
1848        } else if (mRecordTeeSink->getStrongCount() != 1) {
1849            kind = TEE_SINK_NO;
1850        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
1851            kind = TEE_SINK_OLD;
1852        } else {
1853            kind = TEE_SINK_NEW;
1854        }
1855        switch (kind) {
1856        case TEE_SINK_NEW: {
1857            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1858            size_t numCounterOffers = 0;
1859            const NBAIO_Format offers[1] = {format};
1860            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1861            ALOG_ASSERT(index == 0);
1862            PipeReader *pipeReader = new PipeReader(*pipe);
1863            numCounterOffers = 0;
1864            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1865            ALOG_ASSERT(index == 0);
1866            mRecordTeeSink = pipe;
1867            mRecordTeeSource = pipeReader;
1868            teeSink = pipe;
1869            }
1870            break;
1871        case TEE_SINK_OLD:
1872            teeSink = mRecordTeeSink;
1873            break;
1874        case TEE_SINK_NO:
1875        default:
1876            break;
1877        }
1878#endif
1879
1880        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1881
1882        // Start record thread
1883        // RecordThread requires both input and output device indication to forward to audio
1884        // pre processing modules
1885        RecordThread *thread = new RecordThread(this,
1886                                  input,
1887                                  id,
1888                                  primaryOutputDevice_l(),
1889                                  *pDevices
1890#ifdef TEE_SINK
1891                                  , teeSink
1892#endif
1893                                  );
1894        mRecordThreads.add(id, thread);
1895        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1896        if (pSamplingRate != NULL) {
1897            *pSamplingRate = reqSamplingRate;
1898        }
1899        if (pFormat != NULL) {
1900            *pFormat = config.format;
1901        }
1902        if (pChannelMask != NULL) {
1903            *pChannelMask = reqChannelMask;
1904        }
1905
1906        // notify client processes of the new input creation
1907        thread->audioConfigChanged_l(mNotificationClients, AudioSystem::INPUT_OPENED);
1908        return id;
1909    }
1910
1911    return 0;
1912}
1913
1914status_t AudioFlinger::closeInput(audio_io_handle_t input)
1915{
1916    return closeInput_nonvirtual(input);
1917}
1918
1919status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1920{
1921    // keep strong reference on the record thread so that
1922    // it is not destroyed while exit() is executed
1923    sp<RecordThread> thread;
1924    {
1925        Mutex::Autolock _l(mLock);
1926        thread = checkRecordThread_l(input);
1927        if (thread == 0) {
1928            return BAD_VALUE;
1929        }
1930
1931        ALOGV("closeInput() %d", input);
1932        audioConfigChanged_l(mNotificationClients, AudioSystem::INPUT_CLOSED, input, NULL);
1933        mRecordThreads.removeItem(input);
1934    }
1935    thread->exit();
1936    // The thread entity (active unit of execution) is no longer running here,
1937    // but the ThreadBase container still exists.
1938
1939    AudioStreamIn *in = thread->clearInput();
1940    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1941    // from now on thread->mInput is NULL
1942    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1943    delete in;
1944
1945    return NO_ERROR;
1946}
1947
1948status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
1949{
1950    Mutex::Autolock _l(mLock);
1951    ALOGV("invalidateStream() stream %d", stream);
1952
1953    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1954        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1955        thread->invalidateTracks(stream);
1956    }
1957
1958    return NO_ERROR;
1959}
1960
1961
1962int AudioFlinger::newAudioSessionId()
1963{
1964    return nextUniqueId();
1965}
1966
1967void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
1968{
1969    Mutex::Autolock _l(mLock);
1970    pid_t caller = IPCThreadState::self()->getCallingPid();
1971    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
1972    if (pid != -1 && (caller == getpid_cached)) {
1973        caller = pid;
1974    }
1975
1976    // Ignore requests received from processes not known as notification client. The request
1977    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1978    // called from a different pid leaving a stale session reference.  Also we don't know how
1979    // to clear this reference if the client process dies.
1980    if (mNotificationClients.indexOfKey(caller) < 0) {
1981        ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1982        return;
1983    }
1984
1985    size_t num = mAudioSessionRefs.size();
1986    for (size_t i = 0; i< num; i++) {
1987        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1988        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1989            ref->mCnt++;
1990            ALOGV(" incremented refcount to %d", ref->mCnt);
1991            return;
1992        }
1993    }
1994    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1995    ALOGV(" added new entry for %d", audioSession);
1996}
1997
1998void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
1999{
2000    Mutex::Autolock _l(mLock);
2001    pid_t caller = IPCThreadState::self()->getCallingPid();
2002    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
2003    if (pid != -1 && (caller == getpid_cached)) {
2004        caller = pid;
2005    }
2006    size_t num = mAudioSessionRefs.size();
2007    for (size_t i = 0; i< num; i++) {
2008        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2009        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2010            ref->mCnt--;
2011            ALOGV(" decremented refcount to %d", ref->mCnt);
2012            if (ref->mCnt == 0) {
2013                mAudioSessionRefs.removeAt(i);
2014                delete ref;
2015                purgeStaleEffects_l();
2016            }
2017            return;
2018        }
2019    }
2020    // If the caller is mediaserver it is likely that the session being released was acquired
2021    // on behalf of a process not in notification clients and we ignore the warning.
2022    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2023}
2024
2025void AudioFlinger::purgeStaleEffects_l() {
2026
2027    ALOGV("purging stale effects");
2028
2029    Vector< sp<EffectChain> > chains;
2030
2031    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2032        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2033        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2034            sp<EffectChain> ec = t->mEffectChains[j];
2035            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2036                chains.push(ec);
2037            }
2038        }
2039    }
2040    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2041        sp<RecordThread> t = mRecordThreads.valueAt(i);
2042        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2043            sp<EffectChain> ec = t->mEffectChains[j];
2044            chains.push(ec);
2045        }
2046    }
2047
2048    for (size_t i = 0; i < chains.size(); i++) {
2049        sp<EffectChain> ec = chains[i];
2050        int sessionid = ec->sessionId();
2051        sp<ThreadBase> t = ec->mThread.promote();
2052        if (t == 0) {
2053            continue;
2054        }
2055        size_t numsessionrefs = mAudioSessionRefs.size();
2056        bool found = false;
2057        for (size_t k = 0; k < numsessionrefs; k++) {
2058            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2059            if (ref->mSessionid == sessionid) {
2060                ALOGV(" session %d still exists for %d with %d refs",
2061                    sessionid, ref->mPid, ref->mCnt);
2062                found = true;
2063                break;
2064            }
2065        }
2066        if (!found) {
2067            Mutex::Autolock _l(t->mLock);
2068            // remove all effects from the chain
2069            while (ec->mEffects.size()) {
2070                sp<EffectModule> effect = ec->mEffects[0];
2071                effect->unPin();
2072                t->removeEffect_l(effect);
2073                if (effect->purgeHandles()) {
2074                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2075                }
2076                AudioSystem::unregisterEffect(effect->id());
2077            }
2078        }
2079    }
2080    return;
2081}
2082
2083// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2084AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2085{
2086    return mPlaybackThreads.valueFor(output).get();
2087}
2088
2089// checkMixerThread_l() must be called with AudioFlinger::mLock held
2090AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2091{
2092    PlaybackThread *thread = checkPlaybackThread_l(output);
2093    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2094}
2095
2096// checkRecordThread_l() must be called with AudioFlinger::mLock held
2097AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2098{
2099    return mRecordThreads.valueFor(input).get();
2100}
2101
2102uint32_t AudioFlinger::nextUniqueId()
2103{
2104    return (uint32_t) android_atomic_inc(&mNextUniqueId);
2105}
2106
2107AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2108{
2109    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2110        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2111        AudioStreamOut *output = thread->getOutput();
2112        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2113            return thread;
2114        }
2115    }
2116    return NULL;
2117}
2118
2119audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2120{
2121    PlaybackThread *thread = primaryPlaybackThread_l();
2122
2123    if (thread == NULL) {
2124        return 0;
2125    }
2126
2127    return thread->outDevice();
2128}
2129
2130sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2131                                    int triggerSession,
2132                                    int listenerSession,
2133                                    sync_event_callback_t callBack,
2134                                    wp<RefBase> cookie)
2135{
2136    Mutex::Autolock _l(mLock);
2137
2138    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2139    status_t playStatus = NAME_NOT_FOUND;
2140    status_t recStatus = NAME_NOT_FOUND;
2141    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2142        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2143        if (playStatus == NO_ERROR) {
2144            return event;
2145        }
2146    }
2147    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2148        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2149        if (recStatus == NO_ERROR) {
2150            return event;
2151        }
2152    }
2153    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2154        mPendingSyncEvents.add(event);
2155    } else {
2156        ALOGV("createSyncEvent() invalid event %d", event->type());
2157        event.clear();
2158    }
2159    return event;
2160}
2161
2162// ----------------------------------------------------------------------------
2163//  Effect management
2164// ----------------------------------------------------------------------------
2165
2166
2167status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2168{
2169    Mutex::Autolock _l(mLock);
2170    return EffectQueryNumberEffects(numEffects);
2171}
2172
2173status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2174{
2175    Mutex::Autolock _l(mLock);
2176    return EffectQueryEffect(index, descriptor);
2177}
2178
2179status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2180        effect_descriptor_t *descriptor) const
2181{
2182    Mutex::Autolock _l(mLock);
2183    return EffectGetDescriptor(pUuid, descriptor);
2184}
2185
2186
2187sp<IEffect> AudioFlinger::createEffect(
2188        effect_descriptor_t *pDesc,
2189        const sp<IEffectClient>& effectClient,
2190        int32_t priority,
2191        audio_io_handle_t io,
2192        int sessionId,
2193        status_t *status,
2194        int *id,
2195        int *enabled)
2196{
2197    status_t lStatus = NO_ERROR;
2198    sp<EffectHandle> handle;
2199    effect_descriptor_t desc;
2200
2201    pid_t pid = IPCThreadState::self()->getCallingPid();
2202    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2203            pid, effectClient.get(), priority, sessionId, io);
2204
2205    if (pDesc == NULL) {
2206        lStatus = BAD_VALUE;
2207        goto Exit;
2208    }
2209
2210    // check audio settings permission for global effects
2211    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2212        lStatus = PERMISSION_DENIED;
2213        goto Exit;
2214    }
2215
2216    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2217    // that can only be created by audio policy manager (running in same process)
2218    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2219        lStatus = PERMISSION_DENIED;
2220        goto Exit;
2221    }
2222
2223    {
2224        if (!EffectIsNullUuid(&pDesc->uuid)) {
2225            // if uuid is specified, request effect descriptor
2226            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2227            if (lStatus < 0) {
2228                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2229                goto Exit;
2230            }
2231        } else {
2232            // if uuid is not specified, look for an available implementation
2233            // of the required type in effect factory
2234            if (EffectIsNullUuid(&pDesc->type)) {
2235                ALOGW("createEffect() no effect type");
2236                lStatus = BAD_VALUE;
2237                goto Exit;
2238            }
2239            uint32_t numEffects = 0;
2240            effect_descriptor_t d;
2241            d.flags = 0; // prevent compiler warning
2242            bool found = false;
2243
2244            lStatus = EffectQueryNumberEffects(&numEffects);
2245            if (lStatus < 0) {
2246                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2247                goto Exit;
2248            }
2249            for (uint32_t i = 0; i < numEffects; i++) {
2250                lStatus = EffectQueryEffect(i, &desc);
2251                if (lStatus < 0) {
2252                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2253                    continue;
2254                }
2255                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2256                    // If matching type found save effect descriptor. If the session is
2257                    // 0 and the effect is not auxiliary, continue enumeration in case
2258                    // an auxiliary version of this effect type is available
2259                    found = true;
2260                    d = desc;
2261                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2262                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2263                        break;
2264                    }
2265                }
2266            }
2267            if (!found) {
2268                lStatus = BAD_VALUE;
2269                ALOGW("createEffect() effect not found");
2270                goto Exit;
2271            }
2272            // For same effect type, chose auxiliary version over insert version if
2273            // connect to output mix (Compliance to OpenSL ES)
2274            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2275                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2276                desc = d;
2277            }
2278        }
2279
2280        // Do not allow auxiliary effects on a session different from 0 (output mix)
2281        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2282             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2283            lStatus = INVALID_OPERATION;
2284            goto Exit;
2285        }
2286
2287        // check recording permission for visualizer
2288        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2289            !recordingAllowed()) {
2290            lStatus = PERMISSION_DENIED;
2291            goto Exit;
2292        }
2293
2294        // return effect descriptor
2295        *pDesc = desc;
2296        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2297            // if the output returned by getOutputForEffect() is removed before we lock the
2298            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2299            // and we will exit safely
2300            io = AudioSystem::getOutputForEffect(&desc);
2301            ALOGV("createEffect got output %d", io);
2302        }
2303
2304        Mutex::Autolock _l(mLock);
2305
2306        // If output is not specified try to find a matching audio session ID in one of the
2307        // output threads.
2308        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2309        // because of code checking output when entering the function.
2310        // Note: io is never 0 when creating an effect on an input
2311        if (io == AUDIO_IO_HANDLE_NONE) {
2312            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2313                // output must be specified by AudioPolicyManager when using session
2314                // AUDIO_SESSION_OUTPUT_STAGE
2315                lStatus = BAD_VALUE;
2316                goto Exit;
2317            }
2318            // look for the thread where the specified audio session is present
2319            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2320                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2321                    io = mPlaybackThreads.keyAt(i);
2322                    break;
2323                }
2324            }
2325            if (io == 0) {
2326                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2327                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2328                        io = mRecordThreads.keyAt(i);
2329                        break;
2330                    }
2331                }
2332            }
2333            // If no output thread contains the requested session ID, default to
2334            // first output. The effect chain will be moved to the correct output
2335            // thread when a track with the same session ID is created
2336            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2337                io = mPlaybackThreads.keyAt(0);
2338            }
2339            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2340        }
2341        ThreadBase *thread = checkRecordThread_l(io);
2342        if (thread == NULL) {
2343            thread = checkPlaybackThread_l(io);
2344            if (thread == NULL) {
2345                ALOGE("createEffect() unknown output thread");
2346                lStatus = BAD_VALUE;
2347                goto Exit;
2348            }
2349        }
2350
2351        sp<Client> client = registerPid_l(pid);
2352
2353        // create effect on selected output thread
2354        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2355                &desc, enabled, &lStatus);
2356        if (handle != 0 && id != NULL) {
2357            *id = handle->id();
2358        }
2359    }
2360
2361Exit:
2362    *status = lStatus;
2363    return handle;
2364}
2365
2366status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2367        audio_io_handle_t dstOutput)
2368{
2369    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2370            sessionId, srcOutput, dstOutput);
2371    Mutex::Autolock _l(mLock);
2372    if (srcOutput == dstOutput) {
2373        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2374        return NO_ERROR;
2375    }
2376    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2377    if (srcThread == NULL) {
2378        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2379        return BAD_VALUE;
2380    }
2381    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2382    if (dstThread == NULL) {
2383        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2384        return BAD_VALUE;
2385    }
2386
2387    Mutex::Autolock _dl(dstThread->mLock);
2388    Mutex::Autolock _sl(srcThread->mLock);
2389    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2390}
2391
2392// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2393status_t AudioFlinger::moveEffectChain_l(int sessionId,
2394                                   AudioFlinger::PlaybackThread *srcThread,
2395                                   AudioFlinger::PlaybackThread *dstThread,
2396                                   bool reRegister)
2397{
2398    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2399            sessionId, srcThread, dstThread);
2400
2401    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2402    if (chain == 0) {
2403        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2404                sessionId, srcThread);
2405        return INVALID_OPERATION;
2406    }
2407
2408    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2409    // so that a new chain is created with correct parameters when first effect is added. This is
2410    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2411    // removed.
2412    srcThread->removeEffectChain_l(chain);
2413
2414    // transfer all effects one by one so that new effect chain is created on new thread with
2415    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2416    sp<EffectChain> dstChain;
2417    uint32_t strategy = 0; // prevent compiler warning
2418    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2419    Vector< sp<EffectModule> > removed;
2420    status_t status = NO_ERROR;
2421    while (effect != 0) {
2422        srcThread->removeEffect_l(effect);
2423        removed.add(effect);
2424        status = dstThread->addEffect_l(effect);
2425        if (status != NO_ERROR) {
2426            break;
2427        }
2428        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2429        if (effect->state() == EffectModule::ACTIVE ||
2430                effect->state() == EffectModule::STOPPING) {
2431            effect->start();
2432        }
2433        // if the move request is not received from audio policy manager, the effect must be
2434        // re-registered with the new strategy and output
2435        if (dstChain == 0) {
2436            dstChain = effect->chain().promote();
2437            if (dstChain == 0) {
2438                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2439                status = NO_INIT;
2440                break;
2441            }
2442            strategy = dstChain->strategy();
2443        }
2444        if (reRegister) {
2445            AudioSystem::unregisterEffect(effect->id());
2446            AudioSystem::registerEffect(&effect->desc(),
2447                                        dstThread->id(),
2448                                        strategy,
2449                                        sessionId,
2450                                        effect->id());
2451            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2452        }
2453        effect = chain->getEffectFromId_l(0);
2454    }
2455
2456    if (status != NO_ERROR) {
2457        for (size_t i = 0; i < removed.size(); i++) {
2458            srcThread->addEffect_l(removed[i]);
2459            if (dstChain != 0 && reRegister) {
2460                AudioSystem::unregisterEffect(removed[i]->id());
2461                AudioSystem::registerEffect(&removed[i]->desc(),
2462                                            srcThread->id(),
2463                                            strategy,
2464                                            sessionId,
2465                                            removed[i]->id());
2466                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2467            }
2468        }
2469    }
2470
2471    return status;
2472}
2473
2474bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2475{
2476    if (mGlobalEffectEnableTime != 0 &&
2477            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2478        return true;
2479    }
2480
2481    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2482        sp<EffectChain> ec =
2483                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2484        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2485            return true;
2486        }
2487    }
2488    return false;
2489}
2490
2491void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2492{
2493    Mutex::Autolock _l(mLock);
2494
2495    mGlobalEffectEnableTime = systemTime();
2496
2497    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2498        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2499        if (t->mType == ThreadBase::OFFLOAD) {
2500            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2501        }
2502    }
2503
2504}
2505
2506struct Entry {
2507#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2508    char mName[MAX_NAME];
2509};
2510
2511int comparEntry(const void *p1, const void *p2)
2512{
2513    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2514}
2515
2516#ifdef TEE_SINK
2517void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2518{
2519    NBAIO_Source *teeSource = source.get();
2520    if (teeSource != NULL) {
2521        // .wav rotation
2522        // There is a benign race condition if 2 threads call this simultaneously.
2523        // They would both traverse the directory, but the result would simply be
2524        // failures at unlink() which are ignored.  It's also unlikely since
2525        // normally dumpsys is only done by bugreport or from the command line.
2526        char teePath[32+256];
2527        strcpy(teePath, "/data/misc/media");
2528        size_t teePathLen = strlen(teePath);
2529        DIR *dir = opendir(teePath);
2530        teePath[teePathLen++] = '/';
2531        if (dir != NULL) {
2532#define MAX_SORT 20 // number of entries to sort
2533#define MAX_KEEP 10 // number of entries to keep
2534            struct Entry entries[MAX_SORT];
2535            size_t entryCount = 0;
2536            while (entryCount < MAX_SORT) {
2537                struct dirent de;
2538                struct dirent *result = NULL;
2539                int rc = readdir_r(dir, &de, &result);
2540                if (rc != 0) {
2541                    ALOGW("readdir_r failed %d", rc);
2542                    break;
2543                }
2544                if (result == NULL) {
2545                    break;
2546                }
2547                if (result != &de) {
2548                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2549                    break;
2550                }
2551                // ignore non .wav file entries
2552                size_t nameLen = strlen(de.d_name);
2553                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2554                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2555                    continue;
2556                }
2557                strcpy(entries[entryCount++].mName, de.d_name);
2558            }
2559            (void) closedir(dir);
2560            if (entryCount > MAX_KEEP) {
2561                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2562                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2563                    strcpy(&teePath[teePathLen], entries[i].mName);
2564                    (void) unlink(teePath);
2565                }
2566            }
2567        } else {
2568            if (fd >= 0) {
2569                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2570            }
2571        }
2572        char teeTime[16];
2573        struct timeval tv;
2574        gettimeofday(&tv, NULL);
2575        struct tm tm;
2576        localtime_r(&tv.tv_sec, &tm);
2577        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2578        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2579        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2580        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2581        if (teeFd >= 0) {
2582            char wavHeader[44];
2583            memcpy(wavHeader,
2584                "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",
2585                sizeof(wavHeader));
2586            NBAIO_Format format = teeSource->format();
2587            unsigned channelCount = Format_channelCount(format);
2588            ALOG_ASSERT(channelCount <= FCC_2);
2589            uint32_t sampleRate = Format_sampleRate(format);
2590            wavHeader[22] = channelCount;       // number of channels
2591            wavHeader[24] = sampleRate;         // sample rate
2592            wavHeader[25] = sampleRate >> 8;
2593            wavHeader[32] = channelCount * 2;   // block alignment
2594            write(teeFd, wavHeader, sizeof(wavHeader));
2595            size_t total = 0;
2596            bool firstRead = true;
2597            for (;;) {
2598#define TEE_SINK_READ 1024
2599                short buffer[TEE_SINK_READ * FCC_2];
2600                size_t count = TEE_SINK_READ;
2601                ssize_t actual = teeSource->read(buffer, count,
2602                        AudioBufferProvider::kInvalidPTS);
2603                bool wasFirstRead = firstRead;
2604                firstRead = false;
2605                if (actual <= 0) {
2606                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2607                        continue;
2608                    }
2609                    break;
2610                }
2611                ALOG_ASSERT(actual <= (ssize_t)count);
2612                write(teeFd, buffer, actual * channelCount * sizeof(short));
2613                total += actual;
2614            }
2615            lseek(teeFd, (off_t) 4, SEEK_SET);
2616            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2617            write(teeFd, &temp, sizeof(temp));
2618            lseek(teeFd, (off_t) 40, SEEK_SET);
2619            temp =  total * channelCount * sizeof(short);
2620            write(teeFd, &temp, sizeof(temp));
2621            close(teeFd);
2622            if (fd >= 0) {
2623                fdprintf(fd, "tee copied to %s\n", teePath);
2624            }
2625        } else {
2626            if (fd >= 0) {
2627                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2628            }
2629        }
2630    }
2631}
2632#endif
2633
2634// ----------------------------------------------------------------------------
2635
2636status_t AudioFlinger::onTransact(
2637        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2638{
2639    return BnAudioFlinger::onTransact(code, data, reply, flags);
2640}
2641
2642}; // namespace android
2643