AudioFlinger.cpp revision d776ac63ce9c013c9626226e43f7db606e035838 
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(int event, audio_io_handle_t ioHandle, const void *param2)
1199{
1200    size_t size = mNotificationClients.size();
1201    for (size_t i = 0; i < size; i++) {
1202        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1203                                                                               param2);
1204    }
1205}
1206
1207// removeClient_l() must be called with AudioFlinger::mLock held
1208void AudioFlinger::removeClient_l(pid_t pid)
1209{
1210    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1211            IPCThreadState::self()->getCallingPid());
1212    mClients.removeItem(pid);
1213}
1214
1215// getEffectThread_l() must be called with AudioFlinger::mLock held
1216sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1217{
1218    sp<PlaybackThread> thread;
1219
1220    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1221        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1222            ALOG_ASSERT(thread == 0);
1223            thread = mPlaybackThreads.valueAt(i);
1224        }
1225    }
1226
1227    return thread;
1228}
1229
1230
1231
1232// ----------------------------------------------------------------------------
1233
1234AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1235    :   RefBase(),
1236        mAudioFlinger(audioFlinger),
1237        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1238        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1239        mPid(pid),
1240        mTimedTrackCount(0)
1241{
1242    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1243}
1244
1245// Client destructor must be called with AudioFlinger::mLock held
1246AudioFlinger::Client::~Client()
1247{
1248    mAudioFlinger->removeClient_l(mPid);
1249}
1250
1251sp<MemoryDealer> AudioFlinger::Client::heap() const
1252{
1253    return mMemoryDealer;
1254}
1255
1256// Reserve one of the limited slots for a timed audio track associated
1257// with this client
1258bool AudioFlinger::Client::reserveTimedTrack()
1259{
1260    const int kMaxTimedTracksPerClient = 4;
1261
1262    Mutex::Autolock _l(mTimedTrackLock);
1263
1264    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1265        ALOGW("can not create timed track - pid %d has exceeded the limit",
1266             mPid);
1267        return false;
1268    }
1269
1270    mTimedTrackCount++;
1271    return true;
1272}
1273
1274// Release a slot for a timed audio track
1275void AudioFlinger::Client::releaseTimedTrack()
1276{
1277    Mutex::Autolock _l(mTimedTrackLock);
1278    mTimedTrackCount--;
1279}
1280
1281// ----------------------------------------------------------------------------
1282
1283AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1284                                                     const sp<IAudioFlingerClient>& client,
1285                                                     pid_t pid)
1286    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1287{
1288}
1289
1290AudioFlinger::NotificationClient::~NotificationClient()
1291{
1292}
1293
1294void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1295{
1296    sp<NotificationClient> keep(this);
1297    mAudioFlinger->removeNotificationClient(mPid);
1298}
1299
1300
1301// ----------------------------------------------------------------------------
1302
1303static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1304    return audio_is_remote_submix_device(inDevice);
1305}
1306
1307sp<IAudioRecord> AudioFlinger::openRecord(
1308        audio_io_handle_t input,
1309        uint32_t sampleRate,
1310        audio_format_t format,
1311        audio_channel_mask_t channelMask,
1312        size_t *frameCount,
1313        IAudioFlinger::track_flags_t *flags,
1314        pid_t tid,
1315        int *sessionId,
1316        sp<IMemory>& cblk,
1317        sp<IMemory>& buffers,
1318        status_t *status)
1319{
1320    sp<RecordThread::RecordTrack> recordTrack;
1321    sp<RecordHandle> recordHandle;
1322    sp<Client> client;
1323    status_t lStatus;
1324    int lSessionId;
1325
1326    cblk.clear();
1327    buffers.clear();
1328
1329    // check calling permissions
1330    if (!recordingAllowed()) {
1331        ALOGE("openRecord() permission denied: recording not allowed");
1332        lStatus = PERMISSION_DENIED;
1333        goto Exit;
1334    }
1335
1336    // further sample rate checks are performed by createRecordTrack_l()
1337    if (sampleRate == 0) {
1338        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1339        lStatus = BAD_VALUE;
1340        goto Exit;
1341    }
1342
1343    // we don't yet support anything other than 16-bit PCM
1344    if (!(audio_is_valid_format(format) &&
1345            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
1346        ALOGE("openRecord() invalid format %#x", format);
1347        lStatus = BAD_VALUE;
1348        goto Exit;
1349    }
1350
1351    // further channel mask checks are performed by createRecordTrack_l()
1352    if (!audio_is_input_channel(channelMask)) {
1353        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1354        lStatus = BAD_VALUE;
1355        goto Exit;
1356    }
1357
1358    {
1359        Mutex::Autolock _l(mLock);
1360        RecordThread *thread = checkRecordThread_l(input);
1361        if (thread == NULL) {
1362            ALOGE("openRecord() checkRecordThread_l failed");
1363            lStatus = BAD_VALUE;
1364            goto Exit;
1365        }
1366
1367        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1368                && !captureAudioOutputAllowed()) {
1369            ALOGE("openRecord() permission denied: capture not allowed");
1370            lStatus = PERMISSION_DENIED;
1371            goto Exit;
1372        }
1373
1374        pid_t pid = IPCThreadState::self()->getCallingPid();
1375        client = registerPid_l(pid);
1376
1377        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1378            lSessionId = *sessionId;
1379        } else {
1380            // if no audio session id is provided, create one here
1381            lSessionId = nextUniqueId();
1382            if (sessionId != NULL) {
1383                *sessionId = lSessionId;
1384            }
1385        }
1386        ALOGV("openRecord() lSessionId: %d", lSessionId);
1387
1388        // TODO: the uid should be passed in as a parameter to openRecord
1389        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1390                                                  frameCount, lSessionId,
1391                                                  IPCThreadState::self()->getCallingUid(),
1392                                                  flags, tid, &lStatus);
1393        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1394    }
1395
1396    if (lStatus != NO_ERROR) {
1397        // remove local strong reference to Client before deleting the RecordTrack so that the
1398        // Client destructor is called by the TrackBase destructor with mLock held
1399        client.clear();
1400        recordTrack.clear();
1401        goto Exit;
1402    }
1403
1404    cblk = recordTrack->getCblk();
1405    buffers = recordTrack->getBuffers();
1406
1407    // return handle to client
1408    recordHandle = new RecordHandle(recordTrack);
1409
1410Exit:
1411    *status = lStatus;
1412    return recordHandle;
1413}
1414
1415
1416
1417// ----------------------------------------------------------------------------
1418
1419audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1420{
1421    if (!settingsAllowed()) {
1422        return 0;
1423    }
1424    Mutex::Autolock _l(mLock);
1425    return loadHwModule_l(name);
1426}
1427
1428// loadHwModule_l() must be called with AudioFlinger::mLock held
1429audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1430{
1431    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1432        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1433            ALOGW("loadHwModule() module %s already loaded", name);
1434            return mAudioHwDevs.keyAt(i);
1435        }
1436    }
1437
1438    audio_hw_device_t *dev;
1439
1440    int rc = load_audio_interface(name, &dev);
1441    if (rc) {
1442        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1443        return 0;
1444    }
1445
1446    mHardwareStatus = AUDIO_HW_INIT;
1447    rc = dev->init_check(dev);
1448    mHardwareStatus = AUDIO_HW_IDLE;
1449    if (rc) {
1450        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1451        return 0;
1452    }
1453
1454    // Check and cache this HAL's level of support for master mute and master
1455    // volume.  If this is the first HAL opened, and it supports the get
1456    // methods, use the initial values provided by the HAL as the current
1457    // master mute and volume settings.
1458
1459    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1460    {  // scope for auto-lock pattern
1461        AutoMutex lock(mHardwareLock);
1462
1463        if (0 == mAudioHwDevs.size()) {
1464            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1465            if (NULL != dev->get_master_volume) {
1466                float mv;
1467                if (OK == dev->get_master_volume(dev, &mv)) {
1468                    mMasterVolume = mv;
1469                }
1470            }
1471
1472            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1473            if (NULL != dev->get_master_mute) {
1474                bool mm;
1475                if (OK == dev->get_master_mute(dev, &mm)) {
1476                    mMasterMute = mm;
1477                }
1478            }
1479        }
1480
1481        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1482        if ((NULL != dev->set_master_volume) &&
1483            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1484            flags = static_cast<AudioHwDevice::Flags>(flags |
1485                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1486        }
1487
1488        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1489        if ((NULL != dev->set_master_mute) &&
1490            (OK == dev->set_master_mute(dev, mMasterMute))) {
1491            flags = static_cast<AudioHwDevice::Flags>(flags |
1492                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1493        }
1494
1495        mHardwareStatus = AUDIO_HW_IDLE;
1496    }
1497
1498    audio_module_handle_t handle = nextUniqueId();
1499    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1500
1501    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1502          name, dev->common.module->name, dev->common.module->id, handle);
1503
1504    return handle;
1505
1506}
1507
1508// ----------------------------------------------------------------------------
1509
1510uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1511{
1512    Mutex::Autolock _l(mLock);
1513    PlaybackThread *thread = primaryPlaybackThread_l();
1514    return thread != NULL ? thread->sampleRate() : 0;
1515}
1516
1517size_t AudioFlinger::getPrimaryOutputFrameCount()
1518{
1519    Mutex::Autolock _l(mLock);
1520    PlaybackThread *thread = primaryPlaybackThread_l();
1521    return thread != NULL ? thread->frameCountHAL() : 0;
1522}
1523
1524// ----------------------------------------------------------------------------
1525
1526status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1527{
1528    uid_t uid = IPCThreadState::self()->getCallingUid();
1529    if (uid != AID_SYSTEM) {
1530        return PERMISSION_DENIED;
1531    }
1532    Mutex::Autolock _l(mLock);
1533    if (mIsDeviceTypeKnown) {
1534        return INVALID_OPERATION;
1535    }
1536    mIsLowRamDevice = isLowRamDevice;
1537    mIsDeviceTypeKnown = true;
1538    return NO_ERROR;
1539}
1540
1541// ----------------------------------------------------------------------------
1542
1543audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1544                                           audio_devices_t *pDevices,
1545                                           uint32_t *pSamplingRate,
1546                                           audio_format_t *pFormat,
1547                                           audio_channel_mask_t *pChannelMask,
1548                                           uint32_t *pLatencyMs,
1549                                           audio_output_flags_t flags,
1550                                           const audio_offload_info_t *offloadInfo)
1551{
1552    struct audio_config config;
1553    memset(&config, 0, sizeof(config));
1554    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1555    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1556    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1557    if (offloadInfo != NULL) {
1558        config.offload_info = *offloadInfo;
1559    }
1560
1561    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1562              module,
1563              (pDevices != NULL) ? *pDevices : 0,
1564              config.sample_rate,
1565              config.format,
1566              config.channel_mask,
1567              flags);
1568    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1569          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1570
1571    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1572        return AUDIO_IO_HANDLE_NONE;
1573    }
1574
1575    Mutex::Autolock _l(mLock);
1576
1577    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1578    if (outHwDev == NULL) {
1579        return AUDIO_IO_HANDLE_NONE;
1580    }
1581
1582    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1583    audio_io_handle_t id = nextUniqueId();
1584
1585    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1586
1587    audio_stream_out_t *outStream = NULL;
1588    status_t status = hwDevHal->open_output_stream(hwDevHal,
1589                                          id,
1590                                          *pDevices,
1591                                          (audio_output_flags_t)flags,
1592                                          &config,
1593                                          &outStream);
1594
1595    mHardwareStatus = AUDIO_HW_IDLE;
1596    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1597            "Channels %x, status %d",
1598            outStream,
1599            config.sample_rate,
1600            config.format,
1601            config.channel_mask,
1602            status);
1603
1604    if (status == NO_ERROR && outStream != NULL) {
1605        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1606
1607        PlaybackThread *thread;
1608        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1609            thread = new OffloadThread(this, output, id, *pDevices);
1610            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1611        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1612            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1613            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1614            thread = new DirectOutputThread(this, output, id, *pDevices);
1615            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1616        } else {
1617            thread = new MixerThread(this, output, id, *pDevices);
1618            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1619        }
1620        mPlaybackThreads.add(id, thread);
1621
1622        if (pSamplingRate != NULL) {
1623            *pSamplingRate = config.sample_rate;
1624        }
1625        if (pFormat != NULL) {
1626            *pFormat = config.format;
1627        }
1628        if (pChannelMask != NULL) {
1629            *pChannelMask = config.channel_mask;
1630        }
1631        if (pLatencyMs != NULL) {
1632            *pLatencyMs = thread->latency();
1633        }
1634
1635        // notify client processes of the new output creation
1636        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1637
1638        // the first primary output opened designates the primary hw device
1639        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1640            ALOGI("Using module %d has the primary audio interface", module);
1641            mPrimaryHardwareDev = outHwDev;
1642
1643            AutoMutex lock(mHardwareLock);
1644            mHardwareStatus = AUDIO_HW_SET_MODE;
1645            hwDevHal->set_mode(hwDevHal, mMode);
1646            mHardwareStatus = AUDIO_HW_IDLE;
1647        }
1648        return id;
1649    }
1650
1651    return AUDIO_IO_HANDLE_NONE;
1652}
1653
1654audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1655        audio_io_handle_t output2)
1656{
1657    Mutex::Autolock _l(mLock);
1658    MixerThread *thread1 = checkMixerThread_l(output1);
1659    MixerThread *thread2 = checkMixerThread_l(output2);
1660
1661    if (thread1 == NULL || thread2 == NULL) {
1662        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1663                output2);
1664        return AUDIO_IO_HANDLE_NONE;
1665    }
1666
1667    audio_io_handle_t id = nextUniqueId();
1668    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1669    thread->addOutputTrack(thread2);
1670    mPlaybackThreads.add(id, thread);
1671    // notify client processes of the new output creation
1672    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1673    return id;
1674}
1675
1676status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1677{
1678    return closeOutput_nonvirtual(output);
1679}
1680
1681status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1682{
1683    // keep strong reference on the playback thread so that
1684    // it is not destroyed while exit() is executed
1685    sp<PlaybackThread> thread;
1686    {
1687        Mutex::Autolock _l(mLock);
1688        thread = checkPlaybackThread_l(output);
1689        if (thread == NULL) {
1690            return BAD_VALUE;
1691        }
1692
1693        ALOGV("closeOutput() %d", output);
1694
1695        if (thread->type() == ThreadBase::MIXER) {
1696            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1697                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1698                    DuplicatingThread *dupThread =
1699                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1700                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1701
1702                }
1703            }
1704        }
1705
1706
1707        mPlaybackThreads.removeItem(output);
1708        // save all effects to the default thread
1709        if (mPlaybackThreads.size()) {
1710            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1711            if (dstThread != NULL) {
1712                // audioflinger lock is held here so the acquisition order of thread locks does not
1713                // matter
1714                Mutex::Autolock _dl(dstThread->mLock);
1715                Mutex::Autolock _sl(thread->mLock);
1716                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1717                for (size_t i = 0; i < effectChains.size(); i ++) {
1718                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1719                }
1720            }
1721        }
1722        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1723    }
1724    thread->exit();
1725    // The thread entity (active unit of execution) is no longer running here,
1726    // but the ThreadBase container still exists.
1727
1728    if (thread->type() != ThreadBase::DUPLICATING) {
1729        AudioStreamOut *out = thread->clearOutput();
1730        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1731        // from now on thread->mOutput is NULL
1732        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1733        delete out;
1734    }
1735    return NO_ERROR;
1736}
1737
1738status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1739{
1740    Mutex::Autolock _l(mLock);
1741    PlaybackThread *thread = checkPlaybackThread_l(output);
1742
1743    if (thread == NULL) {
1744        return BAD_VALUE;
1745    }
1746
1747    ALOGV("suspendOutput() %d", output);
1748    thread->suspend();
1749
1750    return NO_ERROR;
1751}
1752
1753status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1754{
1755    Mutex::Autolock _l(mLock);
1756    PlaybackThread *thread = checkPlaybackThread_l(output);
1757
1758    if (thread == NULL) {
1759        return BAD_VALUE;
1760    }
1761
1762    ALOGV("restoreOutput() %d", output);
1763
1764    thread->restore();
1765
1766    return NO_ERROR;
1767}
1768
1769audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1770                                          audio_devices_t *pDevices,
1771                                          uint32_t *pSamplingRate,
1772                                          audio_format_t *pFormat,
1773                                          audio_channel_mask_t *pChannelMask)
1774{
1775    struct audio_config config;
1776    memset(&config, 0, sizeof(config));
1777    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1778    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1779    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1780
1781    uint32_t reqSamplingRate = config.sample_rate;
1782    audio_format_t reqFormat = config.format;
1783    audio_channel_mask_t reqChannelMask = config.channel_mask;
1784
1785    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1786        return 0;
1787    }
1788
1789    Mutex::Autolock _l(mLock);
1790
1791    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1792    if (inHwDev == NULL) {
1793        return 0;
1794    }
1795
1796    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1797    audio_io_handle_t id = nextUniqueId();
1798
1799    audio_stream_in_t *inStream = NULL;
1800    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1801                                        &inStream);
1802    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
1803            "status %d",
1804            inStream,
1805            config.sample_rate,
1806            config.format,
1807            config.channel_mask,
1808            status);
1809
1810    // If the input could not be opened with the requested parameters and we can handle the
1811    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1812    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1813    if (status == BAD_VALUE &&
1814        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1815        (config.sample_rate <= 2 * reqSamplingRate) &&
1816        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1817        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1818        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1819        inStream = NULL;
1820        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1821        // FIXME log this new status; HAL should not propose any further changes
1822    }
1823
1824    if (status == NO_ERROR && inStream != NULL) {
1825
1826#ifdef TEE_SINK
1827        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1828        // or (re-)create if current Pipe is idle and does not match the new format
1829        sp<NBAIO_Sink> teeSink;
1830        enum {
1831            TEE_SINK_NO,    // don't copy input
1832            TEE_SINK_NEW,   // copy input using a new pipe
1833            TEE_SINK_OLD,   // copy input using an existing pipe
1834        } kind;
1835        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1836                                        popcount(inStream->common.get_channels(&inStream->common)));
1837        if (!mTeeSinkInputEnabled) {
1838            kind = TEE_SINK_NO;
1839        } else if (!Format_isValid(format)) {
1840            kind = TEE_SINK_NO;
1841        } else if (mRecordTeeSink == 0) {
1842            kind = TEE_SINK_NEW;
1843        } else if (mRecordTeeSink->getStrongCount() != 1) {
1844            kind = TEE_SINK_NO;
1845        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
1846            kind = TEE_SINK_OLD;
1847        } else {
1848            kind = TEE_SINK_NEW;
1849        }
1850        switch (kind) {
1851        case TEE_SINK_NEW: {
1852            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1853            size_t numCounterOffers = 0;
1854            const NBAIO_Format offers[1] = {format};
1855            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1856            ALOG_ASSERT(index == 0);
1857            PipeReader *pipeReader = new PipeReader(*pipe);
1858            numCounterOffers = 0;
1859            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1860            ALOG_ASSERT(index == 0);
1861            mRecordTeeSink = pipe;
1862            mRecordTeeSource = pipeReader;
1863            teeSink = pipe;
1864            }
1865            break;
1866        case TEE_SINK_OLD:
1867            teeSink = mRecordTeeSink;
1868            break;
1869        case TEE_SINK_NO:
1870        default:
1871            break;
1872        }
1873#endif
1874
1875        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1876
1877        // Start record thread
1878        // RecordThread requires both input and output device indication to forward to audio
1879        // pre processing modules
1880        RecordThread *thread = new RecordThread(this,
1881                                  input,
1882                                  id,
1883                                  primaryOutputDevice_l(),
1884                                  *pDevices
1885#ifdef TEE_SINK
1886                                  , teeSink
1887#endif
1888                                  );
1889        mRecordThreads.add(id, thread);
1890        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1891        if (pSamplingRate != NULL) {
1892            *pSamplingRate = reqSamplingRate;
1893        }
1894        if (pFormat != NULL) {
1895            *pFormat = config.format;
1896        }
1897        if (pChannelMask != NULL) {
1898            *pChannelMask = reqChannelMask;
1899        }
1900
1901        // notify client processes of the new input creation
1902        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1903        return id;
1904    }
1905
1906    return 0;
1907}
1908
1909status_t AudioFlinger::closeInput(audio_io_handle_t input)
1910{
1911    return closeInput_nonvirtual(input);
1912}
1913
1914status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1915{
1916    // keep strong reference on the record thread so that
1917    // it is not destroyed while exit() is executed
1918    sp<RecordThread> thread;
1919    {
1920        Mutex::Autolock _l(mLock);
1921        thread = checkRecordThread_l(input);
1922        if (thread == 0) {
1923            return BAD_VALUE;
1924        }
1925
1926        ALOGV("closeInput() %d", input);
1927        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1928        mRecordThreads.removeItem(input);
1929    }
1930    thread->exit();
1931    // The thread entity (active unit of execution) is no longer running here,
1932    // but the ThreadBase container still exists.
1933
1934    AudioStreamIn *in = thread->clearInput();
1935    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1936    // from now on thread->mInput is NULL
1937    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1938    delete in;
1939
1940    return NO_ERROR;
1941}
1942
1943status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
1944{
1945    Mutex::Autolock _l(mLock);
1946    ALOGV("invalidateStream() stream %d", stream);
1947
1948    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1949        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1950        thread->invalidateTracks(stream);
1951    }
1952
1953    return NO_ERROR;
1954}
1955
1956
1957int AudioFlinger::newAudioSessionId()
1958{
1959    return nextUniqueId();
1960}
1961
1962void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
1963{
1964    Mutex::Autolock _l(mLock);
1965    pid_t caller = IPCThreadState::self()->getCallingPid();
1966    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
1967    if (pid != -1 && (caller == getpid_cached)) {
1968        caller = pid;
1969    }
1970
1971    // Ignore requests received from processes not known as notification client. The request
1972    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1973    // called from a different pid leaving a stale session reference.  Also we don't know how
1974    // to clear this reference if the client process dies.
1975    if (mNotificationClients.indexOfKey(caller) < 0) {
1976        ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1977        return;
1978    }
1979
1980    size_t num = mAudioSessionRefs.size();
1981    for (size_t i = 0; i< num; i++) {
1982        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1983        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1984            ref->mCnt++;
1985            ALOGV(" incremented refcount to %d", ref->mCnt);
1986            return;
1987        }
1988    }
1989    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1990    ALOGV(" added new entry for %d", audioSession);
1991}
1992
1993void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
1994{
1995    Mutex::Autolock _l(mLock);
1996    pid_t caller = IPCThreadState::self()->getCallingPid();
1997    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
1998    if (pid != -1 && (caller == getpid_cached)) {
1999        caller = pid;
2000    }
2001    size_t num = mAudioSessionRefs.size();
2002    for (size_t i = 0; i< num; i++) {
2003        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2004        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2005            ref->mCnt--;
2006            ALOGV(" decremented refcount to %d", ref->mCnt);
2007            if (ref->mCnt == 0) {
2008                mAudioSessionRefs.removeAt(i);
2009                delete ref;
2010                purgeStaleEffects_l();
2011            }
2012            return;
2013        }
2014    }
2015    // If the caller is mediaserver it is likely that the session being released was acquired
2016    // on behalf of a process not in notification clients and we ignore the warning.
2017    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2018}
2019
2020void AudioFlinger::purgeStaleEffects_l() {
2021
2022    ALOGV("purging stale effects");
2023
2024    Vector< sp<EffectChain> > chains;
2025
2026    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2027        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2028        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2029            sp<EffectChain> ec = t->mEffectChains[j];
2030            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2031                chains.push(ec);
2032            }
2033        }
2034    }
2035    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2036        sp<RecordThread> t = mRecordThreads.valueAt(i);
2037        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2038            sp<EffectChain> ec = t->mEffectChains[j];
2039            chains.push(ec);
2040        }
2041    }
2042
2043    for (size_t i = 0; i < chains.size(); i++) {
2044        sp<EffectChain> ec = chains[i];
2045        int sessionid = ec->sessionId();
2046        sp<ThreadBase> t = ec->mThread.promote();
2047        if (t == 0) {
2048            continue;
2049        }
2050        size_t numsessionrefs = mAudioSessionRefs.size();
2051        bool found = false;
2052        for (size_t k = 0; k < numsessionrefs; k++) {
2053            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2054            if (ref->mSessionid == sessionid) {
2055                ALOGV(" session %d still exists for %d with %d refs",
2056                    sessionid, ref->mPid, ref->mCnt);
2057                found = true;
2058                break;
2059            }
2060        }
2061        if (!found) {
2062            Mutex::Autolock _l(t->mLock);
2063            // remove all effects from the chain
2064            while (ec->mEffects.size()) {
2065                sp<EffectModule> effect = ec->mEffects[0];
2066                effect->unPin();
2067                t->removeEffect_l(effect);
2068                if (effect->purgeHandles()) {
2069                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2070                }
2071                AudioSystem::unregisterEffect(effect->id());
2072            }
2073        }
2074    }
2075    return;
2076}
2077
2078// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2079AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2080{
2081    return mPlaybackThreads.valueFor(output).get();
2082}
2083
2084// checkMixerThread_l() must be called with AudioFlinger::mLock held
2085AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2086{
2087    PlaybackThread *thread = checkPlaybackThread_l(output);
2088    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2089}
2090
2091// checkRecordThread_l() must be called with AudioFlinger::mLock held
2092AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2093{
2094    return mRecordThreads.valueFor(input).get();
2095}
2096
2097uint32_t AudioFlinger::nextUniqueId()
2098{
2099    return (uint32_t) android_atomic_inc(&mNextUniqueId);
2100}
2101
2102AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2103{
2104    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2105        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2106        AudioStreamOut *output = thread->getOutput();
2107        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2108            return thread;
2109        }
2110    }
2111    return NULL;
2112}
2113
2114audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2115{
2116    PlaybackThread *thread = primaryPlaybackThread_l();
2117
2118    if (thread == NULL) {
2119        return 0;
2120    }
2121
2122    return thread->outDevice();
2123}
2124
2125sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2126                                    int triggerSession,
2127                                    int listenerSession,
2128                                    sync_event_callback_t callBack,
2129                                    wp<RefBase> cookie)
2130{
2131    Mutex::Autolock _l(mLock);
2132
2133    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2134    status_t playStatus = NAME_NOT_FOUND;
2135    status_t recStatus = NAME_NOT_FOUND;
2136    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2137        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2138        if (playStatus == NO_ERROR) {
2139            return event;
2140        }
2141    }
2142    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2143        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2144        if (recStatus == NO_ERROR) {
2145            return event;
2146        }
2147    }
2148    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2149        mPendingSyncEvents.add(event);
2150    } else {
2151        ALOGV("createSyncEvent() invalid event %d", event->type());
2152        event.clear();
2153    }
2154    return event;
2155}
2156
2157// ----------------------------------------------------------------------------
2158//  Effect management
2159// ----------------------------------------------------------------------------
2160
2161
2162status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2163{
2164    Mutex::Autolock _l(mLock);
2165    return EffectQueryNumberEffects(numEffects);
2166}
2167
2168status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2169{
2170    Mutex::Autolock _l(mLock);
2171    return EffectQueryEffect(index, descriptor);
2172}
2173
2174status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2175        effect_descriptor_t *descriptor) const
2176{
2177    Mutex::Autolock _l(mLock);
2178    return EffectGetDescriptor(pUuid, descriptor);
2179}
2180
2181
2182sp<IEffect> AudioFlinger::createEffect(
2183        effect_descriptor_t *pDesc,
2184        const sp<IEffectClient>& effectClient,
2185        int32_t priority,
2186        audio_io_handle_t io,
2187        int sessionId,
2188        status_t *status,
2189        int *id,
2190        int *enabled)
2191{
2192    status_t lStatus = NO_ERROR;
2193    sp<EffectHandle> handle;
2194    effect_descriptor_t desc;
2195
2196    pid_t pid = IPCThreadState::self()->getCallingPid();
2197    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2198            pid, effectClient.get(), priority, sessionId, io);
2199
2200    if (pDesc == NULL) {
2201        lStatus = BAD_VALUE;
2202        goto Exit;
2203    }
2204
2205    // check audio settings permission for global effects
2206    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2207        lStatus = PERMISSION_DENIED;
2208        goto Exit;
2209    }
2210
2211    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2212    // that can only be created by audio policy manager (running in same process)
2213    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2214        lStatus = PERMISSION_DENIED;
2215        goto Exit;
2216    }
2217
2218    {
2219        if (!EffectIsNullUuid(&pDesc->uuid)) {
2220            // if uuid is specified, request effect descriptor
2221            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2222            if (lStatus < 0) {
2223                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2224                goto Exit;
2225            }
2226        } else {
2227            // if uuid is not specified, look for an available implementation
2228            // of the required type in effect factory
2229            if (EffectIsNullUuid(&pDesc->type)) {
2230                ALOGW("createEffect() no effect type");
2231                lStatus = BAD_VALUE;
2232                goto Exit;
2233            }
2234            uint32_t numEffects = 0;
2235            effect_descriptor_t d;
2236            d.flags = 0; // prevent compiler warning
2237            bool found = false;
2238
2239            lStatus = EffectQueryNumberEffects(&numEffects);
2240            if (lStatus < 0) {
2241                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2242                goto Exit;
2243            }
2244            for (uint32_t i = 0; i < numEffects; i++) {
2245                lStatus = EffectQueryEffect(i, &desc);
2246                if (lStatus < 0) {
2247                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2248                    continue;
2249                }
2250                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2251                    // If matching type found save effect descriptor. If the session is
2252                    // 0 and the effect is not auxiliary, continue enumeration in case
2253                    // an auxiliary version of this effect type is available
2254                    found = true;
2255                    d = desc;
2256                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2257                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2258                        break;
2259                    }
2260                }
2261            }
2262            if (!found) {
2263                lStatus = BAD_VALUE;
2264                ALOGW("createEffect() effect not found");
2265                goto Exit;
2266            }
2267            // For same effect type, chose auxiliary version over insert version if
2268            // connect to output mix (Compliance to OpenSL ES)
2269            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2270                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2271                desc = d;
2272            }
2273        }
2274
2275        // Do not allow auxiliary effects on a session different from 0 (output mix)
2276        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2277             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2278            lStatus = INVALID_OPERATION;
2279            goto Exit;
2280        }
2281
2282        // check recording permission for visualizer
2283        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2284            !recordingAllowed()) {
2285            lStatus = PERMISSION_DENIED;
2286            goto Exit;
2287        }
2288
2289        // return effect descriptor
2290        *pDesc = desc;
2291        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2292            // if the output returned by getOutputForEffect() is removed before we lock the
2293            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2294            // and we will exit safely
2295            io = AudioSystem::getOutputForEffect(&desc);
2296            ALOGV("createEffect got output %d", io);
2297        }
2298
2299        Mutex::Autolock _l(mLock);
2300
2301        // If output is not specified try to find a matching audio session ID in one of the
2302        // output threads.
2303        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2304        // because of code checking output when entering the function.
2305        // Note: io is never 0 when creating an effect on an input
2306        if (io == AUDIO_IO_HANDLE_NONE) {
2307            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2308                // output must be specified by AudioPolicyManager when using session
2309                // AUDIO_SESSION_OUTPUT_STAGE
2310                lStatus = BAD_VALUE;
2311                goto Exit;
2312            }
2313            // look for the thread where the specified audio session is present
2314            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2315                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2316                    io = mPlaybackThreads.keyAt(i);
2317                    break;
2318                }
2319            }
2320            if (io == 0) {
2321                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2322                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2323                        io = mRecordThreads.keyAt(i);
2324                        break;
2325                    }
2326                }
2327            }
2328            // If no output thread contains the requested session ID, default to
2329            // first output. The effect chain will be moved to the correct output
2330            // thread when a track with the same session ID is created
2331            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2332                io = mPlaybackThreads.keyAt(0);
2333            }
2334            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2335        }
2336        ThreadBase *thread = checkRecordThread_l(io);
2337        if (thread == NULL) {
2338            thread = checkPlaybackThread_l(io);
2339            if (thread == NULL) {
2340                ALOGE("createEffect() unknown output thread");
2341                lStatus = BAD_VALUE;
2342                goto Exit;
2343            }
2344        }
2345
2346        sp<Client> client = registerPid_l(pid);
2347
2348        // create effect on selected output thread
2349        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2350                &desc, enabled, &lStatus);
2351        if (handle != 0 && id != NULL) {
2352            *id = handle->id();
2353        }
2354    }
2355
2356Exit:
2357    *status = lStatus;
2358    return handle;
2359}
2360
2361status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2362        audio_io_handle_t dstOutput)
2363{
2364    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2365            sessionId, srcOutput, dstOutput);
2366    Mutex::Autolock _l(mLock);
2367    if (srcOutput == dstOutput) {
2368        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2369        return NO_ERROR;
2370    }
2371    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2372    if (srcThread == NULL) {
2373        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2374        return BAD_VALUE;
2375    }
2376    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2377    if (dstThread == NULL) {
2378        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2379        return BAD_VALUE;
2380    }
2381
2382    Mutex::Autolock _dl(dstThread->mLock);
2383    Mutex::Autolock _sl(srcThread->mLock);
2384    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2385}
2386
2387// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2388status_t AudioFlinger::moveEffectChain_l(int sessionId,
2389                                   AudioFlinger::PlaybackThread *srcThread,
2390                                   AudioFlinger::PlaybackThread *dstThread,
2391                                   bool reRegister)
2392{
2393    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2394            sessionId, srcThread, dstThread);
2395
2396    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2397    if (chain == 0) {
2398        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2399                sessionId, srcThread);
2400        return INVALID_OPERATION;
2401    }
2402
2403    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2404    // so that a new chain is created with correct parameters when first effect is added. This is
2405    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2406    // removed.
2407    srcThread->removeEffectChain_l(chain);
2408
2409    // transfer all effects one by one so that new effect chain is created on new thread with
2410    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2411    sp<EffectChain> dstChain;
2412    uint32_t strategy = 0; // prevent compiler warning
2413    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2414    Vector< sp<EffectModule> > removed;
2415    status_t status = NO_ERROR;
2416    while (effect != 0) {
2417        srcThread->removeEffect_l(effect);
2418        removed.add(effect);
2419        status = dstThread->addEffect_l(effect);
2420        if (status != NO_ERROR) {
2421            break;
2422        }
2423        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2424        if (effect->state() == EffectModule::ACTIVE ||
2425                effect->state() == EffectModule::STOPPING) {
2426            effect->start();
2427        }
2428        // if the move request is not received from audio policy manager, the effect must be
2429        // re-registered with the new strategy and output
2430        if (dstChain == 0) {
2431            dstChain = effect->chain().promote();
2432            if (dstChain == 0) {
2433                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2434                status = NO_INIT;
2435                break;
2436            }
2437            strategy = dstChain->strategy();
2438        }
2439        if (reRegister) {
2440            AudioSystem::unregisterEffect(effect->id());
2441            AudioSystem::registerEffect(&effect->desc(),
2442                                        dstThread->id(),
2443                                        strategy,
2444                                        sessionId,
2445                                        effect->id());
2446            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2447        }
2448        effect = chain->getEffectFromId_l(0);
2449    }
2450
2451    if (status != NO_ERROR) {
2452        for (size_t i = 0; i < removed.size(); i++) {
2453            srcThread->addEffect_l(removed[i]);
2454            if (dstChain != 0 && reRegister) {
2455                AudioSystem::unregisterEffect(removed[i]->id());
2456                AudioSystem::registerEffect(&removed[i]->desc(),
2457                                            srcThread->id(),
2458                                            strategy,
2459                                            sessionId,
2460                                            removed[i]->id());
2461                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2462            }
2463        }
2464    }
2465
2466    return status;
2467}
2468
2469bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2470{
2471    if (mGlobalEffectEnableTime != 0 &&
2472            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2473        return true;
2474    }
2475
2476    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2477        sp<EffectChain> ec =
2478                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2479        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2480            return true;
2481        }
2482    }
2483    return false;
2484}
2485
2486void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2487{
2488    Mutex::Autolock _l(mLock);
2489
2490    mGlobalEffectEnableTime = systemTime();
2491
2492    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2493        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2494        if (t->mType == ThreadBase::OFFLOAD) {
2495            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2496        }
2497    }
2498
2499}
2500
2501struct Entry {
2502#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2503    char mName[MAX_NAME];
2504};
2505
2506int comparEntry(const void *p1, const void *p2)
2507{
2508    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2509}
2510
2511#ifdef TEE_SINK
2512void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2513{
2514    NBAIO_Source *teeSource = source.get();
2515    if (teeSource != NULL) {
2516        // .wav rotation
2517        // There is a benign race condition if 2 threads call this simultaneously.
2518        // They would both traverse the directory, but the result would simply be
2519        // failures at unlink() which are ignored.  It's also unlikely since
2520        // normally dumpsys is only done by bugreport or from the command line.
2521        char teePath[32+256];
2522        strcpy(teePath, "/data/misc/media");
2523        size_t teePathLen = strlen(teePath);
2524        DIR *dir = opendir(teePath);
2525        teePath[teePathLen++] = '/';
2526        if (dir != NULL) {
2527#define MAX_SORT 20 // number of entries to sort
2528#define MAX_KEEP 10 // number of entries to keep
2529            struct Entry entries[MAX_SORT];
2530            size_t entryCount = 0;
2531            while (entryCount < MAX_SORT) {
2532                struct dirent de;
2533                struct dirent *result = NULL;
2534                int rc = readdir_r(dir, &de, &result);
2535                if (rc != 0) {
2536                    ALOGW("readdir_r failed %d", rc);
2537                    break;
2538                }
2539                if (result == NULL) {
2540                    break;
2541                }
2542                if (result != &de) {
2543                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2544                    break;
2545                }
2546                // ignore non .wav file entries
2547                size_t nameLen = strlen(de.d_name);
2548                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2549                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2550                    continue;
2551                }
2552                strcpy(entries[entryCount++].mName, de.d_name);
2553            }
2554            (void) closedir(dir);
2555            if (entryCount > MAX_KEEP) {
2556                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2557                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2558                    strcpy(&teePath[teePathLen], entries[i].mName);
2559                    (void) unlink(teePath);
2560                }
2561            }
2562        } else {
2563            if (fd >= 0) {
2564                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2565            }
2566        }
2567        char teeTime[16];
2568        struct timeval tv;
2569        gettimeofday(&tv, NULL);
2570        struct tm tm;
2571        localtime_r(&tv.tv_sec, &tm);
2572        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2573        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2574        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2575        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2576        if (teeFd >= 0) {
2577            char wavHeader[44];
2578            memcpy(wavHeader,
2579                "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",
2580                sizeof(wavHeader));
2581            NBAIO_Format format = teeSource->format();
2582            unsigned channelCount = Format_channelCount(format);
2583            ALOG_ASSERT(channelCount <= FCC_2);
2584            uint32_t sampleRate = Format_sampleRate(format);
2585            wavHeader[22] = channelCount;       // number of channels
2586            wavHeader[24] = sampleRate;         // sample rate
2587            wavHeader[25] = sampleRate >> 8;
2588            wavHeader[32] = channelCount * 2;   // block alignment
2589            write(teeFd, wavHeader, sizeof(wavHeader));
2590            size_t total = 0;
2591            bool firstRead = true;
2592            for (;;) {
2593#define TEE_SINK_READ 1024
2594                short buffer[TEE_SINK_READ * FCC_2];
2595                size_t count = TEE_SINK_READ;
2596                ssize_t actual = teeSource->read(buffer, count,
2597                        AudioBufferProvider::kInvalidPTS);
2598                bool wasFirstRead = firstRead;
2599                firstRead = false;
2600                if (actual <= 0) {
2601                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2602                        continue;
2603                    }
2604                    break;
2605                }
2606                ALOG_ASSERT(actual <= (ssize_t)count);
2607                write(teeFd, buffer, actual * channelCount * sizeof(short));
2608                total += actual;
2609            }
2610            lseek(teeFd, (off_t) 4, SEEK_SET);
2611            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2612            write(teeFd, &temp, sizeof(temp));
2613            lseek(teeFd, (off_t) 40, SEEK_SET);
2614            temp =  total * channelCount * sizeof(short);
2615            write(teeFd, &temp, sizeof(temp));
2616            close(teeFd);
2617            if (fd >= 0) {
2618                fdprintf(fd, "tee copied to %s\n", teePath);
2619            }
2620        } else {
2621            if (fd >= 0) {
2622                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2623            }
2624        }
2625    }
2626}
2627#endif
2628
2629// ----------------------------------------------------------------------------
2630
2631status_t AudioFlinger::onTransact(
2632        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2633{
2634    return BnAudioFlinger::onTransact(code, data, reply, flags);
2635}
2636
2637}; // namespace android
2638