AudioFlinger.cpp revision f66b42242342017c26eb97de544dae31dd2537ca 
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      mHardwareStatus(AUDIO_HW_IDLE),
163      mMasterVolume(1.0f),
164      mMasterMute(false),
165      mNextUniqueId(1),
166      mMode(AUDIO_MODE_INVALID),
167      mBtNrecIsOff(false),
168      mIsLowRamDevice(true),
169      mIsDeviceTypeKnown(false),
170      mGlobalEffectEnableTime(0)
171{
172    getpid_cached = getpid();
173    char value[PROPERTY_VALUE_MAX];
174    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
175    if (doLog) {
176        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
177    }
178#ifdef TEE_SINK
179    (void) property_get("ro.debuggable", value, "0");
180    int debuggable = atoi(value);
181    int teeEnabled = 0;
182    if (debuggable) {
183        (void) property_get("af.tee", value, "0");
184        teeEnabled = atoi(value);
185    }
186    // FIXME symbolic constants here
187    if (teeEnabled & 1) {
188        mTeeSinkInputEnabled = true;
189    }
190    if (teeEnabled & 2) {
191        mTeeSinkOutputEnabled = true;
192    }
193    if (teeEnabled & 4) {
194        mTeeSinkTrackEnabled = true;
195    }
196#endif
197}
198
199void AudioFlinger::onFirstRef()
200{
201    int rc = 0;
202
203    Mutex::Autolock _l(mLock);
204
205    /* TODO: move all this work into an Init() function */
206    char val_str[PROPERTY_VALUE_MAX] = { 0 };
207    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
208        uint32_t int_val;
209        if (1 == sscanf(val_str, "%u", &int_val)) {
210            mStandbyTimeInNsecs = milliseconds(int_val);
211            ALOGI("Using %u mSec as standby time.", int_val);
212        } else {
213            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
214            ALOGI("Using default %u mSec as standby time.",
215                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
216        }
217    }
218
219    mMode = AUDIO_MODE_NORMAL;
220}
221
222AudioFlinger::~AudioFlinger()
223{
224    while (!mRecordThreads.isEmpty()) {
225        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
226        closeInput_nonvirtual(mRecordThreads.keyAt(0));
227    }
228    while (!mPlaybackThreads.isEmpty()) {
229        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
230        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
231    }
232
233    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
234        // no mHardwareLock needed, as there are no other references to this
235        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
236        delete mAudioHwDevs.valueAt(i);
237    }
238
239    // Tell media.log service about any old writers that still need to be unregistered
240    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
241    if (binder != 0) {
242        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
243        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
244            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
245            mUnregisteredWriters.pop();
246            mediaLogService->unregisterWriter(iMemory);
247        }
248    }
249
250}
251
252static const char * const audio_interfaces[] = {
253    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
254    AUDIO_HARDWARE_MODULE_ID_A2DP,
255    AUDIO_HARDWARE_MODULE_ID_USB,
256};
257#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
258
259AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
260        audio_module_handle_t module,
261        audio_devices_t devices)
262{
263    // if module is 0, the request comes from an old policy manager and we should load
264    // well known modules
265    if (module == 0) {
266        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
267        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
268            loadHwModule_l(audio_interfaces[i]);
269        }
270        // then try to find a module supporting the requested device.
271        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
272            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
273            audio_hw_device_t *dev = audioHwDevice->hwDevice();
274            if ((dev->get_supported_devices != NULL) &&
275                    (dev->get_supported_devices(dev) & devices) == devices)
276                return audioHwDevice;
277        }
278    } else {
279        // check a match for the requested module handle
280        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
281        if (audioHwDevice != NULL) {
282            return audioHwDevice;
283        }
284    }
285
286    return NULL;
287}
288
289void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
290{
291    const size_t SIZE = 256;
292    char buffer[SIZE];
293    String8 result;
294
295    result.append("Clients:\n");
296    for (size_t i = 0; i < mClients.size(); ++i) {
297        sp<Client> client = mClients.valueAt(i).promote();
298        if (client != 0) {
299            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
300            result.append(buffer);
301        }
302    }
303
304    result.append("Notification Clients:\n");
305    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
306        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
307        result.append(buffer);
308    }
309
310    result.append("Global session refs:\n");
311    result.append("  session   pid count\n");
312    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
313        AudioSessionRef *r = mAudioSessionRefs[i];
314        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
315        result.append(buffer);
316    }
317    write(fd, result.string(), result.size());
318}
319
320
321void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
322{
323    const size_t SIZE = 256;
324    char buffer[SIZE];
325    String8 result;
326    hardware_call_state hardwareStatus = mHardwareStatus;
327
328    snprintf(buffer, SIZE, "Hardware status: %d\n"
329                           "Standby Time mSec: %u\n",
330                            hardwareStatus,
331                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
332    result.append(buffer);
333    write(fd, result.string(), result.size());
334}
335
336void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
337{
338    const size_t SIZE = 256;
339    char buffer[SIZE];
340    String8 result;
341    snprintf(buffer, SIZE, "Permission Denial: "
342            "can't dump AudioFlinger from pid=%d, uid=%d\n",
343            IPCThreadState::self()->getCallingPid(),
344            IPCThreadState::self()->getCallingUid());
345    result.append(buffer);
346    write(fd, result.string(), result.size());
347}
348
349bool AudioFlinger::dumpTryLock(Mutex& mutex)
350{
351    bool locked = false;
352    for (int i = 0; i < kDumpLockRetries; ++i) {
353        if (mutex.tryLock() == NO_ERROR) {
354            locked = true;
355            break;
356        }
357        usleep(kDumpLockSleepUs);
358    }
359    return locked;
360}
361
362status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
363{
364    if (!dumpAllowed()) {
365        dumpPermissionDenial(fd, args);
366    } else {
367        // get state of hardware lock
368        bool hardwareLocked = dumpTryLock(mHardwareLock);
369        if (!hardwareLocked) {
370            String8 result(kHardwareLockedString);
371            write(fd, result.string(), result.size());
372        } else {
373            mHardwareLock.unlock();
374        }
375
376        bool locked = dumpTryLock(mLock);
377
378        // failed to lock - AudioFlinger is probably deadlocked
379        if (!locked) {
380            String8 result(kDeadlockedString);
381            write(fd, result.string(), result.size());
382        }
383
384        dumpClients(fd, args);
385        dumpInternals(fd, args);
386
387        // dump playback threads
388        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
389            mPlaybackThreads.valueAt(i)->dump(fd, args);
390        }
391
392        // dump record threads
393        for (size_t i = 0; i < mRecordThreads.size(); i++) {
394            mRecordThreads.valueAt(i)->dump(fd, args);
395        }
396
397        // dump all hardware devs
398        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
399            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
400            dev->dump(dev, fd);
401        }
402
403#ifdef TEE_SINK
404        // dump the serially shared record tee sink
405        if (mRecordTeeSource != 0) {
406            dumpTee(fd, mRecordTeeSource);
407        }
408#endif
409
410        if (locked) {
411            mLock.unlock();
412        }
413
414        // append a copy of media.log here by forwarding fd to it, but don't attempt
415        // to lookup the service if it's not running, as it will block for a second
416        if (mLogMemoryDealer != 0) {
417            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
418            if (binder != 0) {
419                fdprintf(fd, "\nmedia.log:\n");
420                Vector<String16> args;
421                binder->dump(fd, args);
422            }
423        }
424    }
425    return NO_ERROR;
426}
427
428sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
429{
430    // If pid is already in the mClients wp<> map, then use that entry
431    // (for which promote() is always != 0), otherwise create a new entry and Client.
432    sp<Client> client = mClients.valueFor(pid).promote();
433    if (client == 0) {
434        client = new Client(this, pid);
435        mClients.add(pid, client);
436    }
437
438    return client;
439}
440
441sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
442{
443    // If there is no memory allocated for logs, return a dummy writer that does nothing
444    if (mLogMemoryDealer == 0) {
445        return new NBLog::Writer();
446    }
447    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
448    // Similarly if we can't contact the media.log service, also return a dummy writer
449    if (binder == 0) {
450        return new NBLog::Writer();
451    }
452    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
453    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
454    // If allocation fails, consult the vector of previously unregistered writers
455    // and garbage-collect one or more them until an allocation succeeds
456    if (shared == 0) {
457        Mutex::Autolock _l(mUnregisteredWritersLock);
458        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
459            {
460                // Pick the oldest stale writer to garbage-collect
461                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
462                mUnregisteredWriters.removeAt(0);
463                mediaLogService->unregisterWriter(iMemory);
464                // Now the media.log remote reference to IMemory is gone.  When our last local
465                // reference to IMemory also drops to zero at end of this block,
466                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
467            }
468            // Re-attempt the allocation
469            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
470            if (shared != 0) {
471                goto success;
472            }
473        }
474        // Even after garbage-collecting all old writers, there is still not enough memory,
475        // so return a dummy writer
476        return new NBLog::Writer();
477    }
478success:
479    mediaLogService->registerWriter(shared, size, name);
480    return new NBLog::Writer(size, shared);
481}
482
483void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
484{
485    if (writer == 0) {
486        return;
487    }
488    sp<IMemory> iMemory(writer->getIMemory());
489    if (iMemory == 0) {
490        return;
491    }
492    // Rather than removing the writer immediately, append it to a queue of old writers to
493    // be garbage-collected later.  This allows us to continue to view old logs for a while.
494    Mutex::Autolock _l(mUnregisteredWritersLock);
495    mUnregisteredWriters.push(writer);
496}
497
498// IAudioFlinger interface
499
500
501sp<IAudioTrack> AudioFlinger::createTrack(
502        audio_stream_type_t streamType,
503        uint32_t sampleRate,
504        audio_format_t format,
505        audio_channel_mask_t channelMask,
506        size_t *frameCount,
507        IAudioFlinger::track_flags_t *flags,
508        const sp<IMemory>& sharedBuffer,
509        audio_io_handle_t output,
510        pid_t tid,
511        int *sessionId,
512        String8& name,
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    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
531    // and we don't yet support 8.24 or 32-bit PCM
532    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
533        ALOGE("createTrack() invalid format %#x", format);
534        lStatus = BAD_VALUE;
535        goto Exit;
536    }
537
538    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
539        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
540        lStatus = BAD_VALUE;
541        goto Exit;
542    }
543
544    {
545        Mutex::Autolock _l(mLock);
546        PlaybackThread *thread = checkPlaybackThread_l(output);
547        PlaybackThread *effectThread = NULL;
548        if (thread == NULL) {
549            ALOGE("no playback thread found for output handle %d", output);
550            lStatus = BAD_VALUE;
551            goto Exit;
552        }
553
554        pid_t pid = IPCThreadState::self()->getCallingPid();
555
556        client = registerPid_l(pid);
557
558        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
559        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
560            // check if an effect chain with the same session ID is present on another
561            // output thread and move it here.
562            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
563                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
564                if (mPlaybackThreads.keyAt(i) != output) {
565                    uint32_t sessions = t->hasAudioSession(*sessionId);
566                    if (sessions & PlaybackThread::EFFECT_SESSION) {
567                        effectThread = t.get();
568                        break;
569                    }
570                }
571            }
572            lSessionId = *sessionId;
573        } else {
574            // if no audio session id is provided, create one here
575            lSessionId = nextUniqueId();
576            if (sessionId != NULL) {
577                *sessionId = lSessionId;
578            }
579        }
580        ALOGV("createTrack() lSessionId: %d", lSessionId);
581
582        track = thread->createTrack_l(client, streamType, sampleRate, format,
583                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
584        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
585        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
586
587        // move effect chain to this output thread if an effect on same session was waiting
588        // for a track to be created
589        if (lStatus == NO_ERROR && effectThread != NULL) {
590            // no risk of deadlock because AudioFlinger::mLock is held
591            Mutex::Autolock _dl(thread->mLock);
592            Mutex::Autolock _sl(effectThread->mLock);
593            moveEffectChain_l(lSessionId, effectThread, thread, true);
594        }
595
596        // Look for sync events awaiting for a session to be used.
597        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
598            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
599                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
600                    if (lStatus == NO_ERROR) {
601                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
602                    } else {
603                        mPendingSyncEvents[i]->cancel();
604                    }
605                    mPendingSyncEvents.removeAt(i);
606                    i--;
607                }
608            }
609        }
610
611    }
612
613    if (lStatus == NO_ERROR) {
614        // s for server's pid, n for normal mixer name, f for fast index
615        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
616                track->fastIndex());
617        trackHandle = new TrackHandle(track);
618    } else {
619        // remove local strong reference to Client before deleting the Track so that the Client
620        // destructor is called by the TrackBase destructor with mLock held
621        client.clear();
622        track.clear();
623    }
624
625Exit:
626    *status = lStatus;
627    return trackHandle;
628}
629
630uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
631{
632    Mutex::Autolock _l(mLock);
633    PlaybackThread *thread = checkPlaybackThread_l(output);
634    if (thread == NULL) {
635        ALOGW("sampleRate() unknown thread %d", output);
636        return 0;
637    }
638    return thread->sampleRate();
639}
640
641int AudioFlinger::channelCount(audio_io_handle_t output) const
642{
643    Mutex::Autolock _l(mLock);
644    PlaybackThread *thread = checkPlaybackThread_l(output);
645    if (thread == NULL) {
646        ALOGW("channelCount() unknown thread %d", output);
647        return 0;
648    }
649    return thread->channelCount();
650}
651
652audio_format_t AudioFlinger::format(audio_io_handle_t output) const
653{
654    Mutex::Autolock _l(mLock);
655    PlaybackThread *thread = checkPlaybackThread_l(output);
656    if (thread == NULL) {
657        ALOGW("format() unknown thread %d", output);
658        return AUDIO_FORMAT_INVALID;
659    }
660    return thread->format();
661}
662
663size_t AudioFlinger::frameCount(audio_io_handle_t output) const
664{
665    Mutex::Autolock _l(mLock);
666    PlaybackThread *thread = checkPlaybackThread_l(output);
667    if (thread == NULL) {
668        ALOGW("frameCount() unknown thread %d", output);
669        return 0;
670    }
671    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
672    //       should examine all callers and fix them to handle smaller counts
673    return thread->frameCount();
674}
675
676uint32_t AudioFlinger::latency(audio_io_handle_t output) const
677{
678    Mutex::Autolock _l(mLock);
679    PlaybackThread *thread = checkPlaybackThread_l(output);
680    if (thread == NULL) {
681        ALOGW("latency(): no playback thread found for output handle %d", output);
682        return 0;
683    }
684    return thread->latency();
685}
686
687status_t AudioFlinger::setMasterVolume(float value)
688{
689    status_t ret = initCheck();
690    if (ret != NO_ERROR) {
691        return ret;
692    }
693
694    // check calling permissions
695    if (!settingsAllowed()) {
696        return PERMISSION_DENIED;
697    }
698
699    Mutex::Autolock _l(mLock);
700    mMasterVolume = value;
701
702    // Set master volume in the HALs which support it.
703    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
704        AutoMutex lock(mHardwareLock);
705        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
706
707        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
708        if (dev->canSetMasterVolume()) {
709            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
710        }
711        mHardwareStatus = AUDIO_HW_IDLE;
712    }
713
714    // Now set the master volume in each playback thread.  Playback threads
715    // assigned to HALs which do not have master volume support will apply
716    // master volume during the mix operation.  Threads with HALs which do
717    // support master volume will simply ignore the setting.
718    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
719        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
720
721    return NO_ERROR;
722}
723
724status_t AudioFlinger::setMode(audio_mode_t mode)
725{
726    status_t ret = initCheck();
727    if (ret != NO_ERROR) {
728        return ret;
729    }
730
731    // check calling permissions
732    if (!settingsAllowed()) {
733        return PERMISSION_DENIED;
734    }
735    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
736        ALOGW("Illegal value: setMode(%d)", mode);
737        return BAD_VALUE;
738    }
739
740    { // scope for the lock
741        AutoMutex lock(mHardwareLock);
742        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
743        mHardwareStatus = AUDIO_HW_SET_MODE;
744        ret = dev->set_mode(dev, mode);
745        mHardwareStatus = AUDIO_HW_IDLE;
746    }
747
748    if (NO_ERROR == ret) {
749        Mutex::Autolock _l(mLock);
750        mMode = mode;
751        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
752            mPlaybackThreads.valueAt(i)->setMode(mode);
753    }
754
755    return ret;
756}
757
758status_t AudioFlinger::setMicMute(bool state)
759{
760    status_t ret = initCheck();
761    if (ret != NO_ERROR) {
762        return ret;
763    }
764
765    // check calling permissions
766    if (!settingsAllowed()) {
767        return PERMISSION_DENIED;
768    }
769
770    AutoMutex lock(mHardwareLock);
771    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
772    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
773    ret = dev->set_mic_mute(dev, state);
774    mHardwareStatus = AUDIO_HW_IDLE;
775    return ret;
776}
777
778bool AudioFlinger::getMicMute() const
779{
780    status_t ret = initCheck();
781    if (ret != NO_ERROR) {
782        return false;
783    }
784
785    bool state = AUDIO_MODE_INVALID;
786    AutoMutex lock(mHardwareLock);
787    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
788    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
789    dev->get_mic_mute(dev, &state);
790    mHardwareStatus = AUDIO_HW_IDLE;
791    return state;
792}
793
794status_t AudioFlinger::setMasterMute(bool muted)
795{
796    status_t ret = initCheck();
797    if (ret != NO_ERROR) {
798        return ret;
799    }
800
801    // check calling permissions
802    if (!settingsAllowed()) {
803        return PERMISSION_DENIED;
804    }
805
806    Mutex::Autolock _l(mLock);
807    mMasterMute = muted;
808
809    // Set master mute in the HALs which support it.
810    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
811        AutoMutex lock(mHardwareLock);
812        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
813
814        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
815        if (dev->canSetMasterMute()) {
816            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
817        }
818        mHardwareStatus = AUDIO_HW_IDLE;
819    }
820
821    // Now set the master mute in each playback thread.  Playback threads
822    // assigned to HALs which do not have master mute support will apply master
823    // mute during the mix operation.  Threads with HALs which do support master
824    // mute will simply ignore the setting.
825    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
826        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
827
828    return NO_ERROR;
829}
830
831float AudioFlinger::masterVolume() const
832{
833    Mutex::Autolock _l(mLock);
834    return masterVolume_l();
835}
836
837bool AudioFlinger::masterMute() const
838{
839    Mutex::Autolock _l(mLock);
840    return masterMute_l();
841}
842
843float AudioFlinger::masterVolume_l() const
844{
845    return mMasterVolume;
846}
847
848bool AudioFlinger::masterMute_l() const
849{
850    return mMasterMute;
851}
852
853status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
854        audio_io_handle_t output)
855{
856    // check calling permissions
857    if (!settingsAllowed()) {
858        return PERMISSION_DENIED;
859    }
860
861    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
862        ALOGE("setStreamVolume() invalid stream %d", stream);
863        return BAD_VALUE;
864    }
865
866    AutoMutex lock(mLock);
867    PlaybackThread *thread = NULL;
868    if (output) {
869        thread = checkPlaybackThread_l(output);
870        if (thread == NULL) {
871            return BAD_VALUE;
872        }
873    }
874
875    mStreamTypes[stream].volume = value;
876
877    if (thread == NULL) {
878        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
879            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
880        }
881    } else {
882        thread->setStreamVolume(stream, value);
883    }
884
885    return NO_ERROR;
886}
887
888status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
889{
890    // check calling permissions
891    if (!settingsAllowed()) {
892        return PERMISSION_DENIED;
893    }
894
895    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
896        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
897        ALOGE("setStreamMute() invalid stream %d", stream);
898        return BAD_VALUE;
899    }
900
901    AutoMutex lock(mLock);
902    mStreamTypes[stream].mute = muted;
903    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
904        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
905
906    return NO_ERROR;
907}
908
909float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
910{
911    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
912        return 0.0f;
913    }
914
915    AutoMutex lock(mLock);
916    float volume;
917    if (output) {
918        PlaybackThread *thread = checkPlaybackThread_l(output);
919        if (thread == NULL) {
920            return 0.0f;
921        }
922        volume = thread->streamVolume(stream);
923    } else {
924        volume = streamVolume_l(stream);
925    }
926
927    return volume;
928}
929
930bool AudioFlinger::streamMute(audio_stream_type_t stream) const
931{
932    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
933        return true;
934    }
935
936    AutoMutex lock(mLock);
937    return streamMute_l(stream);
938}
939
940status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
941{
942    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
943            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
944
945    // check calling permissions
946    if (!settingsAllowed()) {
947        return PERMISSION_DENIED;
948    }
949
950    // ioHandle == 0 means the parameters are global to the audio hardware interface
951    if (ioHandle == 0) {
952        Mutex::Autolock _l(mLock);
953        status_t final_result = NO_ERROR;
954        {
955            AutoMutex lock(mHardwareLock);
956            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
957            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
958                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
959                status_t result = dev->set_parameters(dev, keyValuePairs.string());
960                final_result = result ?: final_result;
961            }
962            mHardwareStatus = AUDIO_HW_IDLE;
963        }
964        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
965        AudioParameter param = AudioParameter(keyValuePairs);
966        String8 value;
967        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
968            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
969            if (mBtNrecIsOff != btNrecIsOff) {
970                for (size_t i = 0; i < mRecordThreads.size(); i++) {
971                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
972                    audio_devices_t device = thread->inDevice();
973                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
974                    // collect all of the thread's session IDs
975                    KeyedVector<int, bool> ids = thread->sessionIds();
976                    // suspend effects associated with those session IDs
977                    for (size_t j = 0; j < ids.size(); ++j) {
978                        int sessionId = ids.keyAt(j);
979                        thread->setEffectSuspended(FX_IID_AEC,
980                                                   suspend,
981                                                   sessionId);
982                        thread->setEffectSuspended(FX_IID_NS,
983                                                   suspend,
984                                                   sessionId);
985                    }
986                }
987                mBtNrecIsOff = btNrecIsOff;
988            }
989        }
990        String8 screenState;
991        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
992            bool isOff = screenState == "off";
993            if (isOff != (AudioFlinger::mScreenState & 1)) {
994                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
995            }
996        }
997        return final_result;
998    }
999
1000    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1001    // and the thread is exited once the lock is released
1002    sp<ThreadBase> thread;
1003    {
1004        Mutex::Autolock _l(mLock);
1005        thread = checkPlaybackThread_l(ioHandle);
1006        if (thread == 0) {
1007            thread = checkRecordThread_l(ioHandle);
1008        } else if (thread == primaryPlaybackThread_l()) {
1009            // indicate output device change to all input threads for pre processing
1010            AudioParameter param = AudioParameter(keyValuePairs);
1011            int value;
1012            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1013                    (value != 0)) {
1014                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1015                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1016                }
1017            }
1018        }
1019    }
1020    if (thread != 0) {
1021        return thread->setParameters(keyValuePairs);
1022    }
1023    return BAD_VALUE;
1024}
1025
1026String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1027{
1028    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1029            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1030
1031    Mutex::Autolock _l(mLock);
1032
1033    if (ioHandle == 0) {
1034        String8 out_s8;
1035
1036        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1037            char *s;
1038            {
1039            AutoMutex lock(mHardwareLock);
1040            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1041            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1042            s = dev->get_parameters(dev, keys.string());
1043            mHardwareStatus = AUDIO_HW_IDLE;
1044            }
1045            out_s8 += String8(s ? s : "");
1046            free(s);
1047        }
1048        return out_s8;
1049    }
1050
1051    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1052    if (playbackThread != NULL) {
1053        return playbackThread->getParameters(keys);
1054    }
1055    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1056    if (recordThread != NULL) {
1057        return recordThread->getParameters(keys);
1058    }
1059    return String8("");
1060}
1061
1062size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1063        audio_channel_mask_t channelMask) const
1064{
1065    status_t ret = initCheck();
1066    if (ret != NO_ERROR) {
1067        return 0;
1068    }
1069
1070    AutoMutex lock(mHardwareLock);
1071    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1072    struct audio_config config;
1073    memset(&config, 0, sizeof(config));
1074    config.sample_rate = sampleRate;
1075    config.channel_mask = channelMask;
1076    config.format = format;
1077
1078    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1079    size_t size = dev->get_input_buffer_size(dev, &config);
1080    mHardwareStatus = AUDIO_HW_IDLE;
1081    return size;
1082}
1083
1084uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1085{
1086    Mutex::Autolock _l(mLock);
1087
1088    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1089    if (recordThread != NULL) {
1090        return recordThread->getInputFramesLost();
1091    }
1092    return 0;
1093}
1094
1095status_t AudioFlinger::setVoiceVolume(float value)
1096{
1097    status_t ret = initCheck();
1098    if (ret != NO_ERROR) {
1099        return ret;
1100    }
1101
1102    // check calling permissions
1103    if (!settingsAllowed()) {
1104        return PERMISSION_DENIED;
1105    }
1106
1107    AutoMutex lock(mHardwareLock);
1108    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1109    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1110    ret = dev->set_voice_volume(dev, value);
1111    mHardwareStatus = AUDIO_HW_IDLE;
1112
1113    return ret;
1114}
1115
1116status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1117        audio_io_handle_t output) const
1118{
1119    status_t status;
1120
1121    Mutex::Autolock _l(mLock);
1122
1123    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1124    if (playbackThread != NULL) {
1125        return playbackThread->getRenderPosition(halFrames, dspFrames);
1126    }
1127
1128    return BAD_VALUE;
1129}
1130
1131void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1132{
1133
1134    Mutex::Autolock _l(mLock);
1135
1136    pid_t pid = IPCThreadState::self()->getCallingPid();
1137    if (mNotificationClients.indexOfKey(pid) < 0) {
1138        sp<NotificationClient> notificationClient = new NotificationClient(this,
1139                                                                            client,
1140                                                                            pid);
1141        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1142
1143        mNotificationClients.add(pid, notificationClient);
1144
1145        sp<IBinder> binder = client->asBinder();
1146        binder->linkToDeath(notificationClient);
1147
1148        // the config change is always sent from playback or record threads to avoid deadlock
1149        // with AudioSystem::gLock
1150        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1151            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1152        }
1153
1154        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1155            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1156        }
1157    }
1158}
1159
1160void AudioFlinger::removeNotificationClient(pid_t pid)
1161{
1162    Mutex::Autolock _l(mLock);
1163
1164    mNotificationClients.removeItem(pid);
1165
1166    ALOGV("%d died, releasing its sessions", pid);
1167    size_t num = mAudioSessionRefs.size();
1168    bool removed = false;
1169    for (size_t i = 0; i< num; ) {
1170        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1171        ALOGV(" pid %d @ %d", ref->mPid, i);
1172        if (ref->mPid == pid) {
1173            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1174            mAudioSessionRefs.removeAt(i);
1175            delete ref;
1176            removed = true;
1177            num--;
1178        } else {
1179            i++;
1180        }
1181    }
1182    if (removed) {
1183        purgeStaleEffects_l();
1184    }
1185}
1186
1187// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1188void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1189{
1190    size_t size = mNotificationClients.size();
1191    for (size_t i = 0; i < size; i++) {
1192        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1193                                                                               param2);
1194    }
1195}
1196
1197// removeClient_l() must be called with AudioFlinger::mLock held
1198void AudioFlinger::removeClient_l(pid_t pid)
1199{
1200    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1201            IPCThreadState::self()->getCallingPid());
1202    mClients.removeItem(pid);
1203}
1204
1205// getEffectThread_l() must be called with AudioFlinger::mLock held
1206sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1207{
1208    sp<PlaybackThread> thread;
1209
1210    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1211        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1212            ALOG_ASSERT(thread == 0);
1213            thread = mPlaybackThreads.valueAt(i);
1214        }
1215    }
1216
1217    return thread;
1218}
1219
1220
1221
1222// ----------------------------------------------------------------------------
1223
1224AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1225    :   RefBase(),
1226        mAudioFlinger(audioFlinger),
1227        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1228        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1229        mPid(pid),
1230        mTimedTrackCount(0)
1231{
1232    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1233}
1234
1235// Client destructor must be called with AudioFlinger::mLock held
1236AudioFlinger::Client::~Client()
1237{
1238    mAudioFlinger->removeClient_l(mPid);
1239}
1240
1241sp<MemoryDealer> AudioFlinger::Client::heap() const
1242{
1243    return mMemoryDealer;
1244}
1245
1246// Reserve one of the limited slots for a timed audio track associated
1247// with this client
1248bool AudioFlinger::Client::reserveTimedTrack()
1249{
1250    const int kMaxTimedTracksPerClient = 4;
1251
1252    Mutex::Autolock _l(mTimedTrackLock);
1253
1254    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1255        ALOGW("can not create timed track - pid %d has exceeded the limit",
1256             mPid);
1257        return false;
1258    }
1259
1260    mTimedTrackCount++;
1261    return true;
1262}
1263
1264// Release a slot for a timed audio track
1265void AudioFlinger::Client::releaseTimedTrack()
1266{
1267    Mutex::Autolock _l(mTimedTrackLock);
1268    mTimedTrackCount--;
1269}
1270
1271// ----------------------------------------------------------------------------
1272
1273AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1274                                                     const sp<IAudioFlingerClient>& client,
1275                                                     pid_t pid)
1276    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1277{
1278}
1279
1280AudioFlinger::NotificationClient::~NotificationClient()
1281{
1282}
1283
1284void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1285{
1286    sp<NotificationClient> keep(this);
1287    mAudioFlinger->removeNotificationClient(mPid);
1288}
1289
1290
1291// ----------------------------------------------------------------------------
1292
1293static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1294    return audio_is_remote_submix_device(inDevice);
1295}
1296
1297sp<IAudioRecord> AudioFlinger::openRecord(
1298        audio_io_handle_t input,
1299        uint32_t sampleRate,
1300        audio_format_t format,
1301        audio_channel_mask_t channelMask,
1302        size_t *frameCount,
1303        IAudioFlinger::track_flags_t *flags,
1304        pid_t tid,
1305        int *sessionId,
1306        status_t *status)
1307{
1308    sp<RecordThread::RecordTrack> recordTrack;
1309    sp<RecordHandle> recordHandle;
1310    sp<Client> client;
1311    status_t lStatus;
1312    RecordThread *thread;
1313    size_t inFrameCount;
1314    int lSessionId;
1315
1316    // check calling permissions
1317    if (!recordingAllowed()) {
1318        ALOGE("openRecord() permission denied: recording not allowed");
1319        lStatus = PERMISSION_DENIED;
1320        goto Exit;
1321    }
1322
1323    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1324        ALOGE("openRecord() invalid format %#x", format);
1325        lStatus = BAD_VALUE;
1326        goto Exit;
1327    }
1328
1329    // add client to list
1330    { // scope for mLock
1331        Mutex::Autolock _l(mLock);
1332        thread = checkRecordThread_l(input);
1333        if (thread == NULL) {
1334            ALOGE("openRecord() checkRecordThread_l failed");
1335            lStatus = BAD_VALUE;
1336            goto Exit;
1337        }
1338
1339        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1340                && !captureAudioOutputAllowed()) {
1341            ALOGE("openRecord() permission denied: capture not allowed");
1342            lStatus = PERMISSION_DENIED;
1343            goto Exit;
1344        }
1345
1346        pid_t pid = IPCThreadState::self()->getCallingPid();
1347        client = registerPid_l(pid);
1348
1349        // If no audio session id is provided, create one here
1350        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1351            lSessionId = *sessionId;
1352        } else {
1353            lSessionId = nextUniqueId();
1354            if (sessionId != NULL) {
1355                *sessionId = lSessionId;
1356            }
1357        }
1358        // create new record track.
1359        // The record track uses one track in mHardwareMixerThread by convention.
1360        // TODO: the uid should be passed in as a parameter to openRecord
1361        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1362                                                  frameCount, lSessionId,
1363                                                  IPCThreadState::self()->getCallingUid(),
1364                                                  flags, tid, &lStatus);
1365        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1366    }
1367
1368    if (lStatus != NO_ERROR) {
1369        // remove local strong reference to Client before deleting the RecordTrack so that the
1370        // Client destructor is called by the TrackBase destructor with mLock held
1371        client.clear();
1372        recordTrack.clear();
1373        goto Exit;
1374    }
1375
1376    // return handle to client
1377    recordHandle = new RecordHandle(recordTrack);
1378
1379Exit:
1380    *status = lStatus;
1381    return recordHandle;
1382}
1383
1384
1385
1386// ----------------------------------------------------------------------------
1387
1388audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1389{
1390    if (!settingsAllowed()) {
1391        return 0;
1392    }
1393    Mutex::Autolock _l(mLock);
1394    return loadHwModule_l(name);
1395}
1396
1397// loadHwModule_l() must be called with AudioFlinger::mLock held
1398audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1399{
1400    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1401        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1402            ALOGW("loadHwModule() module %s already loaded", name);
1403            return mAudioHwDevs.keyAt(i);
1404        }
1405    }
1406
1407    audio_hw_device_t *dev;
1408
1409    int rc = load_audio_interface(name, &dev);
1410    if (rc) {
1411        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1412        return 0;
1413    }
1414
1415    mHardwareStatus = AUDIO_HW_INIT;
1416    rc = dev->init_check(dev);
1417    mHardwareStatus = AUDIO_HW_IDLE;
1418    if (rc) {
1419        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1420        return 0;
1421    }
1422
1423    // Check and cache this HAL's level of support for master mute and master
1424    // volume.  If this is the first HAL opened, and it supports the get
1425    // methods, use the initial values provided by the HAL as the current
1426    // master mute and volume settings.
1427
1428    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1429    {  // scope for auto-lock pattern
1430        AutoMutex lock(mHardwareLock);
1431
1432        if (0 == mAudioHwDevs.size()) {
1433            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1434            if (NULL != dev->get_master_volume) {
1435                float mv;
1436                if (OK == dev->get_master_volume(dev, &mv)) {
1437                    mMasterVolume = mv;
1438                }
1439            }
1440
1441            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1442            if (NULL != dev->get_master_mute) {
1443                bool mm;
1444                if (OK == dev->get_master_mute(dev, &mm)) {
1445                    mMasterMute = mm;
1446                }
1447            }
1448        }
1449
1450        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1451        if ((NULL != dev->set_master_volume) &&
1452            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1453            flags = static_cast<AudioHwDevice::Flags>(flags |
1454                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1455        }
1456
1457        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1458        if ((NULL != dev->set_master_mute) &&
1459            (OK == dev->set_master_mute(dev, mMasterMute))) {
1460            flags = static_cast<AudioHwDevice::Flags>(flags |
1461                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1462        }
1463
1464        mHardwareStatus = AUDIO_HW_IDLE;
1465    }
1466
1467    audio_module_handle_t handle = nextUniqueId();
1468    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1469
1470    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1471          name, dev->common.module->name, dev->common.module->id, handle);
1472
1473    return handle;
1474
1475}
1476
1477// ----------------------------------------------------------------------------
1478
1479uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1480{
1481    Mutex::Autolock _l(mLock);
1482    PlaybackThread *thread = primaryPlaybackThread_l();
1483    return thread != NULL ? thread->sampleRate() : 0;
1484}
1485
1486size_t AudioFlinger::getPrimaryOutputFrameCount()
1487{
1488    Mutex::Autolock _l(mLock);
1489    PlaybackThread *thread = primaryPlaybackThread_l();
1490    return thread != NULL ? thread->frameCountHAL() : 0;
1491}
1492
1493// ----------------------------------------------------------------------------
1494
1495status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1496{
1497    uid_t uid = IPCThreadState::self()->getCallingUid();
1498    if (uid != AID_SYSTEM) {
1499        return PERMISSION_DENIED;
1500    }
1501    Mutex::Autolock _l(mLock);
1502    if (mIsDeviceTypeKnown) {
1503        return INVALID_OPERATION;
1504    }
1505    mIsLowRamDevice = isLowRamDevice;
1506    mIsDeviceTypeKnown = true;
1507    return NO_ERROR;
1508}
1509
1510// ----------------------------------------------------------------------------
1511
1512audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1513                                           audio_devices_t *pDevices,
1514                                           uint32_t *pSamplingRate,
1515                                           audio_format_t *pFormat,
1516                                           audio_channel_mask_t *pChannelMask,
1517                                           uint32_t *pLatencyMs,
1518                                           audio_output_flags_t flags,
1519                                           const audio_offload_info_t *offloadInfo)
1520{
1521    struct audio_config config;
1522    memset(&config, 0, sizeof(config));
1523    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1524    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1525    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1526    if (offloadInfo != NULL) {
1527        config.offload_info = *offloadInfo;
1528    }
1529
1530    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1531              module,
1532              (pDevices != NULL) ? *pDevices : 0,
1533              config.sample_rate,
1534              config.format,
1535              config.channel_mask,
1536              flags);
1537    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1538          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1539
1540    if (pDevices == NULL || *pDevices == 0) {
1541        return 0;
1542    }
1543
1544    Mutex::Autolock _l(mLock);
1545
1546    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1547    if (outHwDev == NULL) {
1548        return 0;
1549    }
1550
1551    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1552    audio_io_handle_t id = nextUniqueId();
1553
1554    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1555
1556    audio_stream_out_t *outStream = NULL;
1557    status_t status = hwDevHal->open_output_stream(hwDevHal,
1558                                          id,
1559                                          *pDevices,
1560                                          (audio_output_flags_t)flags,
1561                                          &config,
1562                                          &outStream);
1563
1564    mHardwareStatus = AUDIO_HW_IDLE;
1565    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1566            "Channels %x, status %d",
1567            outStream,
1568            config.sample_rate,
1569            config.format,
1570            config.channel_mask,
1571            status);
1572
1573    if (status == NO_ERROR && outStream != NULL) {
1574        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1575
1576        PlaybackThread *thread;
1577        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1578            thread = new OffloadThread(this, output, id, *pDevices);
1579            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1580        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1581            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1582            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1583            thread = new DirectOutputThread(this, output, id, *pDevices);
1584            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1585        } else {
1586            thread = new MixerThread(this, output, id, *pDevices);
1587            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1588        }
1589        mPlaybackThreads.add(id, thread);
1590
1591        if (pSamplingRate != NULL) {
1592            *pSamplingRate = config.sample_rate;
1593        }
1594        if (pFormat != NULL) {
1595            *pFormat = config.format;
1596        }
1597        if (pChannelMask != NULL) {
1598            *pChannelMask = config.channel_mask;
1599        }
1600        if (pLatencyMs != NULL) {
1601            *pLatencyMs = thread->latency();
1602        }
1603
1604        // notify client processes of the new output creation
1605        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1606
1607        // the first primary output opened designates the primary hw device
1608        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1609            ALOGI("Using module %d has the primary audio interface", module);
1610            mPrimaryHardwareDev = outHwDev;
1611
1612            AutoMutex lock(mHardwareLock);
1613            mHardwareStatus = AUDIO_HW_SET_MODE;
1614            hwDevHal->set_mode(hwDevHal, mMode);
1615            mHardwareStatus = AUDIO_HW_IDLE;
1616        }
1617        return id;
1618    }
1619
1620    return 0;
1621}
1622
1623audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1624        audio_io_handle_t output2)
1625{
1626    Mutex::Autolock _l(mLock);
1627    MixerThread *thread1 = checkMixerThread_l(output1);
1628    MixerThread *thread2 = checkMixerThread_l(output2);
1629
1630    if (thread1 == NULL || thread2 == NULL) {
1631        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1632                output2);
1633        return 0;
1634    }
1635
1636    audio_io_handle_t id = nextUniqueId();
1637    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1638    thread->addOutputTrack(thread2);
1639    mPlaybackThreads.add(id, thread);
1640    // notify client processes of the new output creation
1641    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1642    return id;
1643}
1644
1645status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1646{
1647    return closeOutput_nonvirtual(output);
1648}
1649
1650status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1651{
1652    // keep strong reference on the playback thread so that
1653    // it is not destroyed while exit() is executed
1654    sp<PlaybackThread> thread;
1655    {
1656        Mutex::Autolock _l(mLock);
1657        thread = checkPlaybackThread_l(output);
1658        if (thread == NULL) {
1659            return BAD_VALUE;
1660        }
1661
1662        ALOGV("closeOutput() %d", output);
1663
1664        if (thread->type() == ThreadBase::MIXER) {
1665            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1666                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1667                    DuplicatingThread *dupThread =
1668                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1669                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1670
1671                }
1672            }
1673        }
1674
1675
1676        mPlaybackThreads.removeItem(output);
1677        // save all effects to the default thread
1678        if (mPlaybackThreads.size()) {
1679            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1680            if (dstThread != NULL) {
1681                // audioflinger lock is held here so the acquisition order of thread locks does not
1682                // matter
1683                Mutex::Autolock _dl(dstThread->mLock);
1684                Mutex::Autolock _sl(thread->mLock);
1685                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1686                for (size_t i = 0; i < effectChains.size(); i ++) {
1687                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1688                }
1689            }
1690        }
1691        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1692    }
1693    thread->exit();
1694    // The thread entity (active unit of execution) is no longer running here,
1695    // but the ThreadBase container still exists.
1696
1697    if (thread->type() != ThreadBase::DUPLICATING) {
1698        AudioStreamOut *out = thread->clearOutput();
1699        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1700        // from now on thread->mOutput is NULL
1701        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1702        delete out;
1703    }
1704    return NO_ERROR;
1705}
1706
1707status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1708{
1709    Mutex::Autolock _l(mLock);
1710    PlaybackThread *thread = checkPlaybackThread_l(output);
1711
1712    if (thread == NULL) {
1713        return BAD_VALUE;
1714    }
1715
1716    ALOGV("suspendOutput() %d", output);
1717    thread->suspend();
1718
1719    return NO_ERROR;
1720}
1721
1722status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1723{
1724    Mutex::Autolock _l(mLock);
1725    PlaybackThread *thread = checkPlaybackThread_l(output);
1726
1727    if (thread == NULL) {
1728        return BAD_VALUE;
1729    }
1730
1731    ALOGV("restoreOutput() %d", output);
1732
1733    thread->restore();
1734
1735    return NO_ERROR;
1736}
1737
1738audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1739                                          audio_devices_t *pDevices,
1740                                          uint32_t *pSamplingRate,
1741                                          audio_format_t *pFormat,
1742                                          audio_channel_mask_t *pChannelMask)
1743{
1744    struct audio_config config;
1745    memset(&config, 0, sizeof(config));
1746    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1747    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1748    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1749
1750    uint32_t reqSamplingRate = config.sample_rate;
1751    audio_format_t reqFormat = config.format;
1752    audio_channel_mask_t reqChannelMask = config.channel_mask;
1753
1754    if (pDevices == NULL || *pDevices == 0) {
1755        return 0;
1756    }
1757
1758    Mutex::Autolock _l(mLock);
1759
1760    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1761    if (inHwDev == NULL) {
1762        return 0;
1763    }
1764
1765    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1766    audio_io_handle_t id = nextUniqueId();
1767
1768    audio_stream_in_t *inStream = NULL;
1769    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1770                                        &inStream);
1771    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
1772            "status %d",
1773            inStream,
1774            config.sample_rate,
1775            config.format,
1776            config.channel_mask,
1777            status);
1778
1779    // If the input could not be opened with the requested parameters and we can handle the
1780    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1781    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1782    if (status == BAD_VALUE &&
1783        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1784        (config.sample_rate <= 2 * reqSamplingRate) &&
1785        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1786        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1787        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1788        inStream = NULL;
1789        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1790        // FIXME log this new status; HAL should not propose any further changes
1791    }
1792
1793    if (status == NO_ERROR && inStream != NULL) {
1794
1795#ifdef TEE_SINK
1796        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1797        // or (re-)create if current Pipe is idle and does not match the new format
1798        sp<NBAIO_Sink> teeSink;
1799        enum {
1800            TEE_SINK_NO,    // don't copy input
1801            TEE_SINK_NEW,   // copy input using a new pipe
1802            TEE_SINK_OLD,   // copy input using an existing pipe
1803        } kind;
1804        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1805                                        popcount(inStream->common.get_channels(&inStream->common)));
1806        if (!mTeeSinkInputEnabled) {
1807            kind = TEE_SINK_NO;
1808        } else if (!Format_isValid(format)) {
1809            kind = TEE_SINK_NO;
1810        } else if (mRecordTeeSink == 0) {
1811            kind = TEE_SINK_NEW;
1812        } else if (mRecordTeeSink->getStrongCount() != 1) {
1813            kind = TEE_SINK_NO;
1814        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
1815            kind = TEE_SINK_OLD;
1816        } else {
1817            kind = TEE_SINK_NEW;
1818        }
1819        switch (kind) {
1820        case TEE_SINK_NEW: {
1821            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1822            size_t numCounterOffers = 0;
1823            const NBAIO_Format offers[1] = {format};
1824            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1825            ALOG_ASSERT(index == 0);
1826            PipeReader *pipeReader = new PipeReader(*pipe);
1827            numCounterOffers = 0;
1828            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1829            ALOG_ASSERT(index == 0);
1830            mRecordTeeSink = pipe;
1831            mRecordTeeSource = pipeReader;
1832            teeSink = pipe;
1833            }
1834            break;
1835        case TEE_SINK_OLD:
1836            teeSink = mRecordTeeSink;
1837            break;
1838        case TEE_SINK_NO:
1839        default:
1840            break;
1841        }
1842#endif
1843
1844        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1845
1846        // Start record thread
1847        // RecordThread requires both input and output device indication to forward to audio
1848        // pre processing modules
1849        RecordThread *thread = new RecordThread(this,
1850                                  input,
1851                                  id,
1852                                  primaryOutputDevice_l(),
1853                                  *pDevices
1854#ifdef TEE_SINK
1855                                  , teeSink
1856#endif
1857                                  );
1858        mRecordThreads.add(id, thread);
1859        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1860        if (pSamplingRate != NULL) {
1861            *pSamplingRate = reqSamplingRate;
1862        }
1863        if (pFormat != NULL) {
1864            *pFormat = config.format;
1865        }
1866        if (pChannelMask != NULL) {
1867            *pChannelMask = reqChannelMask;
1868        }
1869
1870        // notify client processes of the new input creation
1871        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1872        return id;
1873    }
1874
1875    return 0;
1876}
1877
1878status_t AudioFlinger::closeInput(audio_io_handle_t input)
1879{
1880    return closeInput_nonvirtual(input);
1881}
1882
1883status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1884{
1885    // keep strong reference on the record thread so that
1886    // it is not destroyed while exit() is executed
1887    sp<RecordThread> thread;
1888    {
1889        Mutex::Autolock _l(mLock);
1890        thread = checkRecordThread_l(input);
1891        if (thread == 0) {
1892            return BAD_VALUE;
1893        }
1894
1895        ALOGV("closeInput() %d", input);
1896        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1897        mRecordThreads.removeItem(input);
1898    }
1899    thread->exit();
1900    // The thread entity (active unit of execution) is no longer running here,
1901    // but the ThreadBase container still exists.
1902
1903    AudioStreamIn *in = thread->clearInput();
1904    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1905    // from now on thread->mInput is NULL
1906    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1907    delete in;
1908
1909    return NO_ERROR;
1910}
1911
1912status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1913{
1914    Mutex::Autolock _l(mLock);
1915    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1916
1917    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1918        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1919        thread->invalidateTracks(stream);
1920    }
1921
1922    return NO_ERROR;
1923}
1924
1925
1926int AudioFlinger::newAudioSessionId()
1927{
1928    return nextUniqueId();
1929}
1930
1931void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
1932{
1933    Mutex::Autolock _l(mLock);
1934    pid_t caller = IPCThreadState::self()->getCallingPid();
1935    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
1936    if (pid != -1 && (caller == getpid_cached)) {
1937        caller = pid;
1938    }
1939
1940    // Ignore requests received from processes not known as notification client. The request
1941    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1942    // called from a different pid leaving a stale session reference.  Also we don't know how
1943    // to clear this reference if the client process dies.
1944    if (mNotificationClients.indexOfKey(caller) < 0) {
1945        ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1946        return;
1947    }
1948
1949    size_t num = mAudioSessionRefs.size();
1950    for (size_t i = 0; i< num; i++) {
1951        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1952        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1953            ref->mCnt++;
1954            ALOGV(" incremented refcount to %d", ref->mCnt);
1955            return;
1956        }
1957    }
1958    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1959    ALOGV(" added new entry for %d", audioSession);
1960}
1961
1962void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
1963{
1964    Mutex::Autolock _l(mLock);
1965    pid_t caller = IPCThreadState::self()->getCallingPid();
1966    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
1967    if (pid != -1 && (caller == getpid_cached)) {
1968        caller = pid;
1969    }
1970    size_t num = mAudioSessionRefs.size();
1971    for (size_t i = 0; i< num; i++) {
1972        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1973        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1974            ref->mCnt--;
1975            ALOGV(" decremented refcount to %d", ref->mCnt);
1976            if (ref->mCnt == 0) {
1977                mAudioSessionRefs.removeAt(i);
1978                delete ref;
1979                purgeStaleEffects_l();
1980            }
1981            return;
1982        }
1983    }
1984    // If the caller is mediaserver it is likely that the session being released was acquired
1985    // on behalf of a process not in notification clients and we ignore the warning.
1986    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
1987}
1988
1989void AudioFlinger::purgeStaleEffects_l() {
1990
1991    ALOGV("purging stale effects");
1992
1993    Vector< sp<EffectChain> > chains;
1994
1995    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1996        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1997        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1998            sp<EffectChain> ec = t->mEffectChains[j];
1999            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2000                chains.push(ec);
2001            }
2002        }
2003    }
2004    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2005        sp<RecordThread> t = mRecordThreads.valueAt(i);
2006        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2007            sp<EffectChain> ec = t->mEffectChains[j];
2008            chains.push(ec);
2009        }
2010    }
2011
2012    for (size_t i = 0; i < chains.size(); i++) {
2013        sp<EffectChain> ec = chains[i];
2014        int sessionid = ec->sessionId();
2015        sp<ThreadBase> t = ec->mThread.promote();
2016        if (t == 0) {
2017            continue;
2018        }
2019        size_t numsessionrefs = mAudioSessionRefs.size();
2020        bool found = false;
2021        for (size_t k = 0; k < numsessionrefs; k++) {
2022            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2023            if (ref->mSessionid == sessionid) {
2024                ALOGV(" session %d still exists for %d with %d refs",
2025                    sessionid, ref->mPid, ref->mCnt);
2026                found = true;
2027                break;
2028            }
2029        }
2030        if (!found) {
2031            Mutex::Autolock _l(t->mLock);
2032            // remove all effects from the chain
2033            while (ec->mEffects.size()) {
2034                sp<EffectModule> effect = ec->mEffects[0];
2035                effect->unPin();
2036                t->removeEffect_l(effect);
2037                if (effect->purgeHandles()) {
2038                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2039                }
2040                AudioSystem::unregisterEffect(effect->id());
2041            }
2042        }
2043    }
2044    return;
2045}
2046
2047// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2048AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2049{
2050    return mPlaybackThreads.valueFor(output).get();
2051}
2052
2053// checkMixerThread_l() must be called with AudioFlinger::mLock held
2054AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2055{
2056    PlaybackThread *thread = checkPlaybackThread_l(output);
2057    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2058}
2059
2060// checkRecordThread_l() must be called with AudioFlinger::mLock held
2061AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2062{
2063    return mRecordThreads.valueFor(input).get();
2064}
2065
2066uint32_t AudioFlinger::nextUniqueId()
2067{
2068    return android_atomic_inc(&mNextUniqueId);
2069}
2070
2071AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2072{
2073    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2074        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2075        AudioStreamOut *output = thread->getOutput();
2076        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2077            return thread;
2078        }
2079    }
2080    return NULL;
2081}
2082
2083audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2084{
2085    PlaybackThread *thread = primaryPlaybackThread_l();
2086
2087    if (thread == NULL) {
2088        return 0;
2089    }
2090
2091    return thread->outDevice();
2092}
2093
2094sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2095                                    int triggerSession,
2096                                    int listenerSession,
2097                                    sync_event_callback_t callBack,
2098                                    void *cookie)
2099{
2100    Mutex::Autolock _l(mLock);
2101
2102    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2103    status_t playStatus = NAME_NOT_FOUND;
2104    status_t recStatus = NAME_NOT_FOUND;
2105    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2106        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2107        if (playStatus == NO_ERROR) {
2108            return event;
2109        }
2110    }
2111    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2112        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2113        if (recStatus == NO_ERROR) {
2114            return event;
2115        }
2116    }
2117    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2118        mPendingSyncEvents.add(event);
2119    } else {
2120        ALOGV("createSyncEvent() invalid event %d", event->type());
2121        event.clear();
2122    }
2123    return event;
2124}
2125
2126// ----------------------------------------------------------------------------
2127//  Effect management
2128// ----------------------------------------------------------------------------
2129
2130
2131status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2132{
2133    Mutex::Autolock _l(mLock);
2134    return EffectQueryNumberEffects(numEffects);
2135}
2136
2137status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2138{
2139    Mutex::Autolock _l(mLock);
2140    return EffectQueryEffect(index, descriptor);
2141}
2142
2143status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2144        effect_descriptor_t *descriptor) const
2145{
2146    Mutex::Autolock _l(mLock);
2147    return EffectGetDescriptor(pUuid, descriptor);
2148}
2149
2150
2151sp<IEffect> AudioFlinger::createEffect(
2152        effect_descriptor_t *pDesc,
2153        const sp<IEffectClient>& effectClient,
2154        int32_t priority,
2155        audio_io_handle_t io,
2156        int sessionId,
2157        status_t *status,
2158        int *id,
2159        int *enabled)
2160{
2161    status_t lStatus = NO_ERROR;
2162    sp<EffectHandle> handle;
2163    effect_descriptor_t desc;
2164
2165    pid_t pid = IPCThreadState::self()->getCallingPid();
2166    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2167            pid, effectClient.get(), priority, sessionId, io);
2168
2169    if (pDesc == NULL) {
2170        lStatus = BAD_VALUE;
2171        goto Exit;
2172    }
2173
2174    // check audio settings permission for global effects
2175    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2176        lStatus = PERMISSION_DENIED;
2177        goto Exit;
2178    }
2179
2180    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2181    // that can only be created by audio policy manager (running in same process)
2182    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2183        lStatus = PERMISSION_DENIED;
2184        goto Exit;
2185    }
2186
2187    {
2188        if (!EffectIsNullUuid(&pDesc->uuid)) {
2189            // if uuid is specified, request effect descriptor
2190            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2191            if (lStatus < 0) {
2192                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2193                goto Exit;
2194            }
2195        } else {
2196            // if uuid is not specified, look for an available implementation
2197            // of the required type in effect factory
2198            if (EffectIsNullUuid(&pDesc->type)) {
2199                ALOGW("createEffect() no effect type");
2200                lStatus = BAD_VALUE;
2201                goto Exit;
2202            }
2203            uint32_t numEffects = 0;
2204            effect_descriptor_t d;
2205            d.flags = 0; // prevent compiler warning
2206            bool found = false;
2207
2208            lStatus = EffectQueryNumberEffects(&numEffects);
2209            if (lStatus < 0) {
2210                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2211                goto Exit;
2212            }
2213            for (uint32_t i = 0; i < numEffects; i++) {
2214                lStatus = EffectQueryEffect(i, &desc);
2215                if (lStatus < 0) {
2216                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2217                    continue;
2218                }
2219                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2220                    // If matching type found save effect descriptor. If the session is
2221                    // 0 and the effect is not auxiliary, continue enumeration in case
2222                    // an auxiliary version of this effect type is available
2223                    found = true;
2224                    d = desc;
2225                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2226                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2227                        break;
2228                    }
2229                }
2230            }
2231            if (!found) {
2232                lStatus = BAD_VALUE;
2233                ALOGW("createEffect() effect not found");
2234                goto Exit;
2235            }
2236            // For same effect type, chose auxiliary version over insert version if
2237            // connect to output mix (Compliance to OpenSL ES)
2238            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2239                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2240                desc = d;
2241            }
2242        }
2243
2244        // Do not allow auxiliary effects on a session different from 0 (output mix)
2245        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2246             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2247            lStatus = INVALID_OPERATION;
2248            goto Exit;
2249        }
2250
2251        // check recording permission for visualizer
2252        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2253            !recordingAllowed()) {
2254            lStatus = PERMISSION_DENIED;
2255            goto Exit;
2256        }
2257
2258        // return effect descriptor
2259        *pDesc = desc;
2260        if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2261            // if the output returned by getOutputForEffect() is removed before we lock the
2262            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2263            // and we will exit safely
2264            io = AudioSystem::getOutputForEffect(&desc);
2265            ALOGV("createEffect got output %d", io);
2266        }
2267
2268        Mutex::Autolock _l(mLock);
2269
2270        // If output is not specified try to find a matching audio session ID in one of the
2271        // output threads.
2272        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2273        // because of code checking output when entering the function.
2274        // Note: io is never 0 when creating an effect on an input
2275        if (io == 0) {
2276            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2277                // output must be specified by AudioPolicyManager when using session
2278                // AUDIO_SESSION_OUTPUT_STAGE
2279                lStatus = BAD_VALUE;
2280                goto Exit;
2281            }
2282            // look for the thread where the specified audio session is present
2283            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2284                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2285                    io = mPlaybackThreads.keyAt(i);
2286                    break;
2287                }
2288            }
2289            if (io == 0) {
2290                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2291                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2292                        io = mRecordThreads.keyAt(i);
2293                        break;
2294                    }
2295                }
2296            }
2297            // If no output thread contains the requested session ID, default to
2298            // first output. The effect chain will be moved to the correct output
2299            // thread when a track with the same session ID is created
2300            if (io == 0 && mPlaybackThreads.size()) {
2301                io = mPlaybackThreads.keyAt(0);
2302            }
2303            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2304        }
2305        ThreadBase *thread = checkRecordThread_l(io);
2306        if (thread == NULL) {
2307            thread = checkPlaybackThread_l(io);
2308            if (thread == NULL) {
2309                ALOGE("createEffect() unknown output thread");
2310                lStatus = BAD_VALUE;
2311                goto Exit;
2312            }
2313        }
2314
2315        sp<Client> client = registerPid_l(pid);
2316
2317        // create effect on selected output thread
2318        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2319                &desc, enabled, &lStatus);
2320        if (handle != 0 && id != NULL) {
2321            *id = handle->id();
2322        }
2323    }
2324
2325Exit:
2326    *status = lStatus;
2327    return handle;
2328}
2329
2330status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2331        audio_io_handle_t dstOutput)
2332{
2333    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2334            sessionId, srcOutput, dstOutput);
2335    Mutex::Autolock _l(mLock);
2336    if (srcOutput == dstOutput) {
2337        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2338        return NO_ERROR;
2339    }
2340    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2341    if (srcThread == NULL) {
2342        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2343        return BAD_VALUE;
2344    }
2345    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2346    if (dstThread == NULL) {
2347        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2348        return BAD_VALUE;
2349    }
2350
2351    Mutex::Autolock _dl(dstThread->mLock);
2352    Mutex::Autolock _sl(srcThread->mLock);
2353    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2354}
2355
2356// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2357status_t AudioFlinger::moveEffectChain_l(int sessionId,
2358                                   AudioFlinger::PlaybackThread *srcThread,
2359                                   AudioFlinger::PlaybackThread *dstThread,
2360                                   bool reRegister)
2361{
2362    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2363            sessionId, srcThread, dstThread);
2364
2365    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2366    if (chain == 0) {
2367        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2368                sessionId, srcThread);
2369        return INVALID_OPERATION;
2370    }
2371
2372    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2373    // so that a new chain is created with correct parameters when first effect is added. This is
2374    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2375    // removed.
2376    srcThread->removeEffectChain_l(chain);
2377
2378    // transfer all effects one by one so that new effect chain is created on new thread with
2379    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2380    sp<EffectChain> dstChain;
2381    uint32_t strategy = 0; // prevent compiler warning
2382    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2383    Vector< sp<EffectModule> > removed;
2384    status_t status = NO_ERROR;
2385    while (effect != 0) {
2386        srcThread->removeEffect_l(effect);
2387        removed.add(effect);
2388        status = dstThread->addEffect_l(effect);
2389        if (status != NO_ERROR) {
2390            break;
2391        }
2392        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2393        if (effect->state() == EffectModule::ACTIVE ||
2394                effect->state() == EffectModule::STOPPING) {
2395            effect->start();
2396        }
2397        // if the move request is not received from audio policy manager, the effect must be
2398        // re-registered with the new strategy and output
2399        if (dstChain == 0) {
2400            dstChain = effect->chain().promote();
2401            if (dstChain == 0) {
2402                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2403                status = NO_INIT;
2404                break;
2405            }
2406            strategy = dstChain->strategy();
2407        }
2408        if (reRegister) {
2409            AudioSystem::unregisterEffect(effect->id());
2410            AudioSystem::registerEffect(&effect->desc(),
2411                                        dstThread->id(),
2412                                        strategy,
2413                                        sessionId,
2414                                        effect->id());
2415            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2416        }
2417        effect = chain->getEffectFromId_l(0);
2418    }
2419
2420    if (status != NO_ERROR) {
2421        for (size_t i = 0; i < removed.size(); i++) {
2422            srcThread->addEffect_l(removed[i]);
2423            if (dstChain != 0 && reRegister) {
2424                AudioSystem::unregisterEffect(removed[i]->id());
2425                AudioSystem::registerEffect(&removed[i]->desc(),
2426                                            srcThread->id(),
2427                                            strategy,
2428                                            sessionId,
2429                                            removed[i]->id());
2430                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2431            }
2432        }
2433    }
2434
2435    return status;
2436}
2437
2438bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2439{
2440    if (mGlobalEffectEnableTime != 0 &&
2441            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2442        return true;
2443    }
2444
2445    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2446        sp<EffectChain> ec =
2447                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2448        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2449            return true;
2450        }
2451    }
2452    return false;
2453}
2454
2455void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2456{
2457    Mutex::Autolock _l(mLock);
2458
2459    mGlobalEffectEnableTime = systemTime();
2460
2461    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2462        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2463        if (t->mType == ThreadBase::OFFLOAD) {
2464            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2465        }
2466    }
2467
2468}
2469
2470struct Entry {
2471#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2472    char mName[MAX_NAME];
2473};
2474
2475int comparEntry(const void *p1, const void *p2)
2476{
2477    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2478}
2479
2480#ifdef TEE_SINK
2481void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2482{
2483    NBAIO_Source *teeSource = source.get();
2484    if (teeSource != NULL) {
2485        // .wav rotation
2486        // There is a benign race condition if 2 threads call this simultaneously.
2487        // They would both traverse the directory, but the result would simply be
2488        // failures at unlink() which are ignored.  It's also unlikely since
2489        // normally dumpsys is only done by bugreport or from the command line.
2490        char teePath[32+256];
2491        strcpy(teePath, "/data/misc/media");
2492        size_t teePathLen = strlen(teePath);
2493        DIR *dir = opendir(teePath);
2494        teePath[teePathLen++] = '/';
2495        if (dir != NULL) {
2496#define MAX_SORT 20 // number of entries to sort
2497#define MAX_KEEP 10 // number of entries to keep
2498            struct Entry entries[MAX_SORT];
2499            size_t entryCount = 0;
2500            while (entryCount < MAX_SORT) {
2501                struct dirent de;
2502                struct dirent *result = NULL;
2503                int rc = readdir_r(dir, &de, &result);
2504                if (rc != 0) {
2505                    ALOGW("readdir_r failed %d", rc);
2506                    break;
2507                }
2508                if (result == NULL) {
2509                    break;
2510                }
2511                if (result != &de) {
2512                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2513                    break;
2514                }
2515                // ignore non .wav file entries
2516                size_t nameLen = strlen(de.d_name);
2517                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2518                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2519                    continue;
2520                }
2521                strcpy(entries[entryCount++].mName, de.d_name);
2522            }
2523            (void) closedir(dir);
2524            if (entryCount > MAX_KEEP) {
2525                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2526                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2527                    strcpy(&teePath[teePathLen], entries[i].mName);
2528                    (void) unlink(teePath);
2529                }
2530            }
2531        } else {
2532            if (fd >= 0) {
2533                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2534            }
2535        }
2536        char teeTime[16];
2537        struct timeval tv;
2538        gettimeofday(&tv, NULL);
2539        struct tm tm;
2540        localtime_r(&tv.tv_sec, &tm);
2541        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2542        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2543        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2544        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2545        if (teeFd >= 0) {
2546            char wavHeader[44];
2547            memcpy(wavHeader,
2548                "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",
2549                sizeof(wavHeader));
2550            NBAIO_Format format = teeSource->format();
2551            unsigned channelCount = Format_channelCount(format);
2552            ALOG_ASSERT(channelCount <= FCC_2);
2553            uint32_t sampleRate = Format_sampleRate(format);
2554            wavHeader[22] = channelCount;       // number of channels
2555            wavHeader[24] = sampleRate;         // sample rate
2556            wavHeader[25] = sampleRate >> 8;
2557            wavHeader[32] = channelCount * 2;   // block alignment
2558            write(teeFd, wavHeader, sizeof(wavHeader));
2559            size_t total = 0;
2560            bool firstRead = true;
2561            for (;;) {
2562#define TEE_SINK_READ 1024
2563                short buffer[TEE_SINK_READ * FCC_2];
2564                size_t count = TEE_SINK_READ;
2565                ssize_t actual = teeSource->read(buffer, count,
2566                        AudioBufferProvider::kInvalidPTS);
2567                bool wasFirstRead = firstRead;
2568                firstRead = false;
2569                if (actual <= 0) {
2570                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2571                        continue;
2572                    }
2573                    break;
2574                }
2575                ALOG_ASSERT(actual <= (ssize_t)count);
2576                write(teeFd, buffer, actual * channelCount * sizeof(short));
2577                total += actual;
2578            }
2579            lseek(teeFd, (off_t) 4, SEEK_SET);
2580            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2581            write(teeFd, &temp, sizeof(temp));
2582            lseek(teeFd, (off_t) 40, SEEK_SET);
2583            temp =  total * channelCount * sizeof(short);
2584            write(teeFd, &temp, sizeof(temp));
2585            close(teeFd);
2586            if (fd >= 0) {
2587                fdprintf(fd, "tee copied to %s\n", teePath);
2588            }
2589        } else {
2590            if (fd >= 0) {
2591                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2592            }
2593        }
2594    }
2595}
2596#endif
2597
2598// ----------------------------------------------------------------------------
2599
2600status_t AudioFlinger::onTransact(
2601        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2602{
2603    return BnAudioFlinger::onTransact(code, data, reply, flags);
2604}
2605
2606}; // namespace android
2607