AudioFlinger.cpp revision 481fb67a595f23c5b7f5be84b06db9b84a41a42f 
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
107static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
108{
109    const hw_module_t *mod;
110    int rc;
111
112    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
113    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
114                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
115    if (rc) {
116        goto out;
117    }
118    rc = audio_hw_device_open(mod, dev);
119    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
120                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
121    if (rc) {
122        goto out;
123    }
124    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
125        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
126        rc = BAD_VALUE;
127        goto out;
128    }
129    return 0;
130
131out:
132    *dev = NULL;
133    return rc;
134}
135
136// ----------------------------------------------------------------------------
137
138AudioFlinger::AudioFlinger()
139    : BnAudioFlinger(),
140      mPrimaryHardwareDev(NULL),
141      mHardwareStatus(AUDIO_HW_IDLE),
142      mMasterVolume(1.0f),
143      mMasterMute(false),
144      mNextUniqueId(1),
145      mMode(AUDIO_MODE_INVALID),
146      mBtNrecIsOff(false),
147      mIsLowRamDevice(true),
148      mIsDeviceTypeKnown(false),
149      mGlobalEffectEnableTime(0)
150{
151    getpid_cached = getpid();
152    char value[PROPERTY_VALUE_MAX];
153    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
154    if (doLog) {
155        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
156    }
157#ifdef TEE_SINK
158    (void) property_get("ro.debuggable", value, "0");
159    int debuggable = atoi(value);
160    int teeEnabled = 0;
161    if (debuggable) {
162        (void) property_get("af.tee", value, "0");
163        teeEnabled = atoi(value);
164    }
165    if (teeEnabled & 1) {
166        mTeeSinkInputEnabled = true;
167    }
168    if (teeEnabled & 2) {
169        mTeeSinkOutputEnabled = true;
170    }
171    if (teeEnabled & 4) {
172        mTeeSinkTrackEnabled = true;
173    }
174#endif
175}
176
177void AudioFlinger::onFirstRef()
178{
179    int rc = 0;
180
181    Mutex::Autolock _l(mLock);
182
183    /* TODO: move all this work into an Init() function */
184    char val_str[PROPERTY_VALUE_MAX] = { 0 };
185    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
186        uint32_t int_val;
187        if (1 == sscanf(val_str, "%u", &int_val)) {
188            mStandbyTimeInNsecs = milliseconds(int_val);
189            ALOGI("Using %u mSec as standby time.", int_val);
190        } else {
191            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
192            ALOGI("Using default %u mSec as standby time.",
193                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
194        }
195    }
196
197    mMode = AUDIO_MODE_NORMAL;
198}
199
200AudioFlinger::~AudioFlinger()
201{
202    while (!mRecordThreads.isEmpty()) {
203        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
204        closeInput_nonvirtual(mRecordThreads.keyAt(0));
205    }
206    while (!mPlaybackThreads.isEmpty()) {
207        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
208        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
209    }
210
211    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
212        // no mHardwareLock needed, as there are no other references to this
213        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
214        delete mAudioHwDevs.valueAt(i);
215    }
216
217    // Tell media.log service about any old writers that still need to be unregistered
218    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
219    if (binder != 0) {
220        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
221        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
222            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
223            mUnregisteredWriters.pop();
224            mediaLogService->unregisterWriter(iMemory);
225        }
226    }
227
228}
229
230static const char * const audio_interfaces[] = {
231    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
232    AUDIO_HARDWARE_MODULE_ID_A2DP,
233    AUDIO_HARDWARE_MODULE_ID_USB,
234};
235#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
236
237AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
238        audio_module_handle_t module,
239        audio_devices_t devices)
240{
241    // if module is 0, the request comes from an old policy manager and we should load
242    // well known modules
243    if (module == 0) {
244        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
245        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
246            loadHwModule_l(audio_interfaces[i]);
247        }
248        // then try to find a module supporting the requested device.
249        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
250            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
251            audio_hw_device_t *dev = audioHwDevice->hwDevice();
252            if ((dev->get_supported_devices != NULL) &&
253                    (dev->get_supported_devices(dev) & devices) == devices)
254                return audioHwDevice;
255        }
256    } else {
257        // check a match for the requested module handle
258        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
259        if (audioHwDevice != NULL) {
260            return audioHwDevice;
261        }
262    }
263
264    return NULL;
265}
266
267void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
268{
269    const size_t SIZE = 256;
270    char buffer[SIZE];
271    String8 result;
272
273    result.append("Clients:\n");
274    for (size_t i = 0; i < mClients.size(); ++i) {
275        sp<Client> client = mClients.valueAt(i).promote();
276        if (client != 0) {
277            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
278            result.append(buffer);
279        }
280    }
281
282    result.append("Notification Clients:\n");
283    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
284        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
285        result.append(buffer);
286    }
287
288    result.append("Global session refs:\n");
289    result.append(" session pid count\n");
290    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
291        AudioSessionRef *r = mAudioSessionRefs[i];
292        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
293        result.append(buffer);
294    }
295    write(fd, result.string(), result.size());
296}
297
298
299void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
300{
301    const size_t SIZE = 256;
302    char buffer[SIZE];
303    String8 result;
304    hardware_call_state hardwareStatus = mHardwareStatus;
305
306    snprintf(buffer, SIZE, "Hardware status: %d\n"
307                           "Standby Time mSec: %u\n",
308                            hardwareStatus,
309                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
310    result.append(buffer);
311    write(fd, result.string(), result.size());
312}
313
314void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
315{
316    const size_t SIZE = 256;
317    char buffer[SIZE];
318    String8 result;
319    snprintf(buffer, SIZE, "Permission Denial: "
320            "can't dump AudioFlinger from pid=%d, uid=%d\n",
321            IPCThreadState::self()->getCallingPid(),
322            IPCThreadState::self()->getCallingUid());
323    result.append(buffer);
324    write(fd, result.string(), result.size());
325}
326
327bool AudioFlinger::dumpTryLock(Mutex& mutex)
328{
329    bool locked = false;
330    for (int i = 0; i < kDumpLockRetries; ++i) {
331        if (mutex.tryLock() == NO_ERROR) {
332            locked = true;
333            break;
334        }
335        usleep(kDumpLockSleepUs);
336    }
337    return locked;
338}
339
340status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
341{
342    if (!dumpAllowed()) {
343        dumpPermissionDenial(fd, args);
344    } else {
345        // get state of hardware lock
346        bool hardwareLocked = dumpTryLock(mHardwareLock);
347        if (!hardwareLocked) {
348            String8 result(kHardwareLockedString);
349            write(fd, result.string(), result.size());
350        } else {
351            mHardwareLock.unlock();
352        }
353
354        bool locked = dumpTryLock(mLock);
355
356        // failed to lock - AudioFlinger is probably deadlocked
357        if (!locked) {
358            String8 result(kDeadlockedString);
359            write(fd, result.string(), result.size());
360        }
361
362        dumpClients(fd, args);
363        dumpInternals(fd, args);
364
365        // dump playback threads
366        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
367            mPlaybackThreads.valueAt(i)->dump(fd, args);
368        }
369
370        // dump record threads
371        for (size_t i = 0; i < mRecordThreads.size(); i++) {
372            mRecordThreads.valueAt(i)->dump(fd, args);
373        }
374
375        // dump all hardware devs
376        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
377            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
378            dev->dump(dev, fd);
379        }
380
381#ifdef TEE_SINK
382        // dump the serially shared record tee sink
383        if (mRecordTeeSource != 0) {
384            dumpTee(fd, mRecordTeeSource);
385        }
386#endif
387
388        if (locked) {
389            mLock.unlock();
390        }
391
392        // append a copy of media.log here by forwarding fd to it, but don't attempt
393        // to lookup the service if it's not running, as it will block for a second
394        if (mLogMemoryDealer != 0) {
395            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
396            if (binder != 0) {
397                fdprintf(fd, "\nmedia.log:\n");
398                Vector<String16> args;
399                binder->dump(fd, args);
400            }
401        }
402    }
403    return NO_ERROR;
404}
405
406sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
407{
408    // If pid is already in the mClients wp<> map, then use that entry
409    // (for which promote() is always != 0), otherwise create a new entry and Client.
410    sp<Client> client = mClients.valueFor(pid).promote();
411    if (client == 0) {
412        client = new Client(this, pid);
413        mClients.add(pid, client);
414    }
415
416    return client;
417}
418
419sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
420{
421    // If there is no memory allocated for logs, return a dummy writer that does nothing
422    if (mLogMemoryDealer == 0) {
423        return new NBLog::Writer();
424    }
425    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
426    // Similarly if we can't contact the media.log service, also return a dummy writer
427    if (binder == 0) {
428        return new NBLog::Writer();
429    }
430    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
431    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
432    // If allocation fails, consult the vector of previously unregistered writers
433    // and garbage-collect one or more them until an allocation succeeds
434    if (shared == 0) {
435        Mutex::Autolock _l(mUnregisteredWritersLock);
436        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
437            {
438                // Pick the oldest stale writer to garbage-collect
439                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
440                mUnregisteredWriters.removeAt(0);
441                mediaLogService->unregisterWriter(iMemory);
442                // Now the media.log remote reference to IMemory is gone.  When our last local
443                // reference to IMemory also drops to zero at end of this block,
444                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
445            }
446            // Re-attempt the allocation
447            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
448            if (shared != 0) {
449                goto success;
450            }
451        }
452        // Even after garbage-collecting all old writers, there is still not enough memory,
453        // so return a dummy writer
454        return new NBLog::Writer();
455    }
456success:
457    mediaLogService->registerWriter(shared, size, name);
458    return new NBLog::Writer(size, shared);
459}
460
461void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
462{
463    if (writer == 0) {
464        return;
465    }
466    sp<IMemory> iMemory(writer->getIMemory());
467    if (iMemory == 0) {
468        return;
469    }
470    // Rather than removing the writer immediately, append it to a queue of old writers to
471    // be garbage-collected later.  This allows us to continue to view old logs for a while.
472    Mutex::Autolock _l(mUnregisteredWritersLock);
473    mUnregisteredWriters.push(writer);
474}
475
476// IAudioFlinger interface
477
478
479sp<IAudioTrack> AudioFlinger::createTrack(
480        audio_stream_type_t streamType,
481        uint32_t sampleRate,
482        audio_format_t format,
483        audio_channel_mask_t channelMask,
484        size_t frameCount,
485        IAudioFlinger::track_flags_t *flags,
486        const sp<IMemory>& sharedBuffer,
487        audio_io_handle_t output,
488        pid_t tid,
489        int *sessionId,
490        String8& name,
491        int clientUid,
492        status_t *status)
493{
494    sp<PlaybackThread::Track> track;
495    sp<TrackHandle> trackHandle;
496    sp<Client> client;
497    status_t lStatus;
498    int lSessionId;
499
500    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
501    // but if someone uses binder directly they could bypass that and cause us to crash
502    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
503        ALOGE("createTrack() invalid stream type %d", streamType);
504        lStatus = BAD_VALUE;
505        goto Exit;
506    }
507
508    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
509    // and we don't yet support 8.24 or 32-bit PCM
510    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
511        ALOGE("createTrack() invalid format %d", format);
512        lStatus = BAD_VALUE;
513        goto Exit;
514    }
515
516    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
517        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
518        lStatus = BAD_VALUE;
519        goto Exit;
520    }
521
522    {
523        Mutex::Autolock _l(mLock);
524        PlaybackThread *thread = checkPlaybackThread_l(output);
525        PlaybackThread *effectThread = NULL;
526        if (thread == NULL) {
527            ALOGE("no playback thread found for output handle %d", output);
528            lStatus = BAD_VALUE;
529            goto Exit;
530        }
531
532        pid_t pid = IPCThreadState::self()->getCallingPid();
533
534        client = registerPid_l(pid);
535
536        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
537        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
538            // check if an effect chain with the same session ID is present on another
539            // output thread and move it here.
540            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
541                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
542                if (mPlaybackThreads.keyAt(i) != output) {
543                    uint32_t sessions = t->hasAudioSession(*sessionId);
544                    if (sessions & PlaybackThread::EFFECT_SESSION) {
545                        effectThread = t.get();
546                        break;
547                    }
548                }
549            }
550            lSessionId = *sessionId;
551        } else {
552            // if no audio session id is provided, create one here
553            lSessionId = nextUniqueId();
554            if (sessionId != NULL) {
555                *sessionId = lSessionId;
556            }
557        }
558        ALOGV("createTrack() lSessionId: %d", lSessionId);
559
560        track = thread->createTrack_l(client, streamType, sampleRate, format,
561                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
562        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
563
564        // move effect chain to this output thread if an effect on same session was waiting
565        // for a track to be created
566        if (lStatus == NO_ERROR && effectThread != NULL) {
567            // no risk of deadlock because AudioFlinger::mLock is held
568            Mutex::Autolock _dl(thread->mLock);
569            Mutex::Autolock _sl(effectThread->mLock);
570            moveEffectChain_l(lSessionId, effectThread, thread, true);
571        }
572
573        // Look for sync events awaiting for a session to be used.
574        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
575            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
576                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
577                    if (lStatus == NO_ERROR) {
578                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
579                    } else {
580                        mPendingSyncEvents[i]->cancel();
581                    }
582                    mPendingSyncEvents.removeAt(i);
583                    i--;
584                }
585            }
586        }
587
588    }
589
590    if (lStatus == NO_ERROR) {
591        // s for server's pid, n for normal mixer name, f for fast index
592        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
593                track->fastIndex());
594        trackHandle = new TrackHandle(track);
595    } else {
596        // remove local strong reference to Client before deleting the Track so that the Client
597        // destructor is called by the TrackBase destructor with mLock held
598        client.clear();
599        track.clear();
600    }
601
602Exit:
603    *status = lStatus;
604    return trackHandle;
605}
606
607uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
608{
609    Mutex::Autolock _l(mLock);
610    PlaybackThread *thread = checkPlaybackThread_l(output);
611    if (thread == NULL) {
612        ALOGW("sampleRate() unknown thread %d", output);
613        return 0;
614    }
615    return thread->sampleRate();
616}
617
618int AudioFlinger::channelCount(audio_io_handle_t output) const
619{
620    Mutex::Autolock _l(mLock);
621    PlaybackThread *thread = checkPlaybackThread_l(output);
622    if (thread == NULL) {
623        ALOGW("channelCount() unknown thread %d", output);
624        return 0;
625    }
626    return thread->channelCount();
627}
628
629audio_format_t AudioFlinger::format(audio_io_handle_t output) const
630{
631    Mutex::Autolock _l(mLock);
632    PlaybackThread *thread = checkPlaybackThread_l(output);
633    if (thread == NULL) {
634        ALOGW("format() unknown thread %d", output);
635        return AUDIO_FORMAT_INVALID;
636    }
637    return thread->format();
638}
639
640size_t AudioFlinger::frameCount(audio_io_handle_t output) const
641{
642    Mutex::Autolock _l(mLock);
643    PlaybackThread *thread = checkPlaybackThread_l(output);
644    if (thread == NULL) {
645        ALOGW("frameCount() unknown thread %d", output);
646        return 0;
647    }
648    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
649    //       should examine all callers and fix them to handle smaller counts
650    return thread->frameCount();
651}
652
653uint32_t AudioFlinger::latency(audio_io_handle_t output) const
654{
655    Mutex::Autolock _l(mLock);
656    PlaybackThread *thread = checkPlaybackThread_l(output);
657    if (thread == NULL) {
658        ALOGW("latency(): no playback thread found for output handle %d", output);
659        return 0;
660    }
661    return thread->latency();
662}
663
664status_t AudioFlinger::setMasterVolume(float value)
665{
666    status_t ret = initCheck();
667    if (ret != NO_ERROR) {
668        return ret;
669    }
670
671    // check calling permissions
672    if (!settingsAllowed()) {
673        return PERMISSION_DENIED;
674    }
675
676    Mutex::Autolock _l(mLock);
677    mMasterVolume = value;
678
679    // Set master volume in the HALs which support it.
680    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
681        AutoMutex lock(mHardwareLock);
682        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
683
684        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
685        if (dev->canSetMasterVolume()) {
686            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
687        }
688        mHardwareStatus = AUDIO_HW_IDLE;
689    }
690
691    // Now set the master volume in each playback thread.  Playback threads
692    // assigned to HALs which do not have master volume support will apply
693    // master volume during the mix operation.  Threads with HALs which do
694    // support master volume will simply ignore the setting.
695    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
696        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
697
698    return NO_ERROR;
699}
700
701status_t AudioFlinger::setMode(audio_mode_t mode)
702{
703    status_t ret = initCheck();
704    if (ret != NO_ERROR) {
705        return ret;
706    }
707
708    // check calling permissions
709    if (!settingsAllowed()) {
710        return PERMISSION_DENIED;
711    }
712    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
713        ALOGW("Illegal value: setMode(%d)", mode);
714        return BAD_VALUE;
715    }
716
717    { // scope for the lock
718        AutoMutex lock(mHardwareLock);
719        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
720        mHardwareStatus = AUDIO_HW_SET_MODE;
721        ret = dev->set_mode(dev, mode);
722        mHardwareStatus = AUDIO_HW_IDLE;
723    }
724
725    if (NO_ERROR == ret) {
726        Mutex::Autolock _l(mLock);
727        mMode = mode;
728        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
729            mPlaybackThreads.valueAt(i)->setMode(mode);
730    }
731
732    return ret;
733}
734
735status_t AudioFlinger::setMicMute(bool state)
736{
737    status_t ret = initCheck();
738    if (ret != NO_ERROR) {
739        return ret;
740    }
741
742    // check calling permissions
743    if (!settingsAllowed()) {
744        return PERMISSION_DENIED;
745    }
746
747    AutoMutex lock(mHardwareLock);
748    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
749    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
750    ret = dev->set_mic_mute(dev, state);
751    mHardwareStatus = AUDIO_HW_IDLE;
752    return ret;
753}
754
755bool AudioFlinger::getMicMute() const
756{
757    status_t ret = initCheck();
758    if (ret != NO_ERROR) {
759        return false;
760    }
761
762    bool state = AUDIO_MODE_INVALID;
763    AutoMutex lock(mHardwareLock);
764    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
765    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
766    dev->get_mic_mute(dev, &state);
767    mHardwareStatus = AUDIO_HW_IDLE;
768    return state;
769}
770
771status_t AudioFlinger::setMasterMute(bool muted)
772{
773    status_t ret = initCheck();
774    if (ret != NO_ERROR) {
775        return ret;
776    }
777
778    // check calling permissions
779    if (!settingsAllowed()) {
780        return PERMISSION_DENIED;
781    }
782
783    Mutex::Autolock _l(mLock);
784    mMasterMute = muted;
785
786    // Set master mute in the HALs which support it.
787    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
788        AutoMutex lock(mHardwareLock);
789        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
790
791        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
792        if (dev->canSetMasterMute()) {
793            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
794        }
795        mHardwareStatus = AUDIO_HW_IDLE;
796    }
797
798    // Now set the master mute in each playback thread.  Playback threads
799    // assigned to HALs which do not have master mute support will apply master
800    // mute during the mix operation.  Threads with HALs which do support master
801    // mute will simply ignore the setting.
802    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
803        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
804
805    return NO_ERROR;
806}
807
808float AudioFlinger::masterVolume() const
809{
810    Mutex::Autolock _l(mLock);
811    return masterVolume_l();
812}
813
814bool AudioFlinger::masterMute() const
815{
816    Mutex::Autolock _l(mLock);
817    return masterMute_l();
818}
819
820float AudioFlinger::masterVolume_l() const
821{
822    return mMasterVolume;
823}
824
825bool AudioFlinger::masterMute_l() const
826{
827    return mMasterMute;
828}
829
830status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
831        audio_io_handle_t output)
832{
833    // check calling permissions
834    if (!settingsAllowed()) {
835        return PERMISSION_DENIED;
836    }
837
838    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
839        ALOGE("setStreamVolume() invalid stream %d", stream);
840        return BAD_VALUE;
841    }
842
843    AutoMutex lock(mLock);
844    PlaybackThread *thread = NULL;
845    if (output) {
846        thread = checkPlaybackThread_l(output);
847        if (thread == NULL) {
848            return BAD_VALUE;
849        }
850    }
851
852    mStreamTypes[stream].volume = value;
853
854    if (thread == NULL) {
855        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
856            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
857        }
858    } else {
859        thread->setStreamVolume(stream, value);
860    }
861
862    return NO_ERROR;
863}
864
865status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
866{
867    // check calling permissions
868    if (!settingsAllowed()) {
869        return PERMISSION_DENIED;
870    }
871
872    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
873        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
874        ALOGE("setStreamMute() invalid stream %d", stream);
875        return BAD_VALUE;
876    }
877
878    AutoMutex lock(mLock);
879    mStreamTypes[stream].mute = muted;
880    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
881        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
882
883    return NO_ERROR;
884}
885
886float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
887{
888    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
889        return 0.0f;
890    }
891
892    AutoMutex lock(mLock);
893    float volume;
894    if (output) {
895        PlaybackThread *thread = checkPlaybackThread_l(output);
896        if (thread == NULL) {
897            return 0.0f;
898        }
899        volume = thread->streamVolume(stream);
900    } else {
901        volume = streamVolume_l(stream);
902    }
903
904    return volume;
905}
906
907bool AudioFlinger::streamMute(audio_stream_type_t stream) const
908{
909    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
910        return true;
911    }
912
913    AutoMutex lock(mLock);
914    return streamMute_l(stream);
915}
916
917status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
918{
919    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
920            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
921
922    // check calling permissions
923    if (!settingsAllowed()) {
924        return PERMISSION_DENIED;
925    }
926
927    // ioHandle == 0 means the parameters are global to the audio hardware interface
928    if (ioHandle == 0) {
929        Mutex::Autolock _l(mLock);
930        status_t final_result = NO_ERROR;
931        {
932            AutoMutex lock(mHardwareLock);
933            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
934            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
935                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
936                status_t result = dev->set_parameters(dev, keyValuePairs.string());
937                final_result = result ?: final_result;
938            }
939            mHardwareStatus = AUDIO_HW_IDLE;
940        }
941        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
942        AudioParameter param = AudioParameter(keyValuePairs);
943        String8 value;
944        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
945            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
946            if (mBtNrecIsOff != btNrecIsOff) {
947                for (size_t i = 0; i < mRecordThreads.size(); i++) {
948                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
949                    audio_devices_t device = thread->inDevice();
950                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
951                    // collect all of the thread's session IDs
952                    KeyedVector<int, bool> ids = thread->sessionIds();
953                    // suspend effects associated with those session IDs
954                    for (size_t j = 0; j < ids.size(); ++j) {
955                        int sessionId = ids.keyAt(j);
956                        thread->setEffectSuspended(FX_IID_AEC,
957                                                   suspend,
958                                                   sessionId);
959                        thread->setEffectSuspended(FX_IID_NS,
960                                                   suspend,
961                                                   sessionId);
962                    }
963                }
964                mBtNrecIsOff = btNrecIsOff;
965            }
966        }
967        String8 screenState;
968        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
969            bool isOff = screenState == "off";
970            if (isOff != (AudioFlinger::mScreenState & 1)) {
971                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
972            }
973        }
974        return final_result;
975    }
976
977    // hold a strong ref on thread in case closeOutput() or closeInput() is called
978    // and the thread is exited once the lock is released
979    sp<ThreadBase> thread;
980    {
981        Mutex::Autolock _l(mLock);
982        thread = checkPlaybackThread_l(ioHandle);
983        if (thread == 0) {
984            thread = checkRecordThread_l(ioHandle);
985        } else if (thread == primaryPlaybackThread_l()) {
986            // indicate output device change to all input threads for pre processing
987            AudioParameter param = AudioParameter(keyValuePairs);
988            int value;
989            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
990                    (value != 0)) {
991                for (size_t i = 0; i < mRecordThreads.size(); i++) {
992                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
993                }
994            }
995        }
996    }
997    if (thread != 0) {
998        return thread->setParameters(keyValuePairs);
999    }
1000    return BAD_VALUE;
1001}
1002
1003String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1004{
1005    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1006            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1007
1008    Mutex::Autolock _l(mLock);
1009
1010    if (ioHandle == 0) {
1011        String8 out_s8;
1012
1013        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1014            char *s;
1015            {
1016            AutoMutex lock(mHardwareLock);
1017            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1018            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1019            s = dev->get_parameters(dev, keys.string());
1020            mHardwareStatus = AUDIO_HW_IDLE;
1021            }
1022            out_s8 += String8(s ? s : "");
1023            free(s);
1024        }
1025        return out_s8;
1026    }
1027
1028    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1029    if (playbackThread != NULL) {
1030        return playbackThread->getParameters(keys);
1031    }
1032    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1033    if (recordThread != NULL) {
1034        return recordThread->getParameters(keys);
1035    }
1036    return String8("");
1037}
1038
1039size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1040        audio_channel_mask_t channelMask) const
1041{
1042    status_t ret = initCheck();
1043    if (ret != NO_ERROR) {
1044        return 0;
1045    }
1046
1047    AutoMutex lock(mHardwareLock);
1048    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1049    struct audio_config config;
1050    memset(&config, 0, sizeof(config));
1051    config.sample_rate = sampleRate;
1052    config.channel_mask = channelMask;
1053    config.format = format;
1054
1055    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1056    size_t size = dev->get_input_buffer_size(dev, &config);
1057    mHardwareStatus = AUDIO_HW_IDLE;
1058    return size;
1059}
1060
1061unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1062{
1063    Mutex::Autolock _l(mLock);
1064
1065    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1066    if (recordThread != NULL) {
1067        return recordThread->getInputFramesLost();
1068    }
1069    return 0;
1070}
1071
1072status_t AudioFlinger::setVoiceVolume(float value)
1073{
1074    status_t ret = initCheck();
1075    if (ret != NO_ERROR) {
1076        return ret;
1077    }
1078
1079    // check calling permissions
1080    if (!settingsAllowed()) {
1081        return PERMISSION_DENIED;
1082    }
1083
1084    AutoMutex lock(mHardwareLock);
1085    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1086    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1087    ret = dev->set_voice_volume(dev, value);
1088    mHardwareStatus = AUDIO_HW_IDLE;
1089
1090    return ret;
1091}
1092
1093status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1094        audio_io_handle_t output) const
1095{
1096    status_t status;
1097
1098    Mutex::Autolock _l(mLock);
1099
1100    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1101    if (playbackThread != NULL) {
1102        return playbackThread->getRenderPosition(halFrames, dspFrames);
1103    }
1104
1105    return BAD_VALUE;
1106}
1107
1108void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1109{
1110
1111    Mutex::Autolock _l(mLock);
1112
1113    pid_t pid = IPCThreadState::self()->getCallingPid();
1114    if (mNotificationClients.indexOfKey(pid) < 0) {
1115        sp<NotificationClient> notificationClient = new NotificationClient(this,
1116                                                                            client,
1117                                                                            pid);
1118        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1119
1120        mNotificationClients.add(pid, notificationClient);
1121
1122        sp<IBinder> binder = client->asBinder();
1123        binder->linkToDeath(notificationClient);
1124
1125        // the config change is always sent from playback or record threads to avoid deadlock
1126        // with AudioSystem::gLock
1127        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1128            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1129        }
1130
1131        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1132            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1133        }
1134    }
1135}
1136
1137void AudioFlinger::removeNotificationClient(pid_t pid)
1138{
1139    Mutex::Autolock _l(mLock);
1140
1141    mNotificationClients.removeItem(pid);
1142
1143    ALOGV("%d died, releasing its sessions", pid);
1144    size_t num = mAudioSessionRefs.size();
1145    bool removed = false;
1146    for (size_t i = 0; i< num; ) {
1147        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1148        ALOGV(" pid %d @ %d", ref->mPid, i);
1149        if (ref->mPid == pid) {
1150            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1151            mAudioSessionRefs.removeAt(i);
1152            delete ref;
1153            removed = true;
1154            num--;
1155        } else {
1156            i++;
1157        }
1158    }
1159    if (removed) {
1160        purgeStaleEffects_l();
1161    }
1162}
1163
1164// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1165void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1166{
1167    size_t size = mNotificationClients.size();
1168    for (size_t i = 0; i < size; i++) {
1169        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1170                                                                               param2);
1171    }
1172}
1173
1174// removeClient_l() must be called with AudioFlinger::mLock held
1175void AudioFlinger::removeClient_l(pid_t pid)
1176{
1177    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1178            IPCThreadState::self()->getCallingPid());
1179    mClients.removeItem(pid);
1180}
1181
1182// getEffectThread_l() must be called with AudioFlinger::mLock held
1183sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1184{
1185    sp<PlaybackThread> thread;
1186
1187    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1188        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1189            ALOG_ASSERT(thread == 0);
1190            thread = mPlaybackThreads.valueAt(i);
1191        }
1192    }
1193
1194    return thread;
1195}
1196
1197
1198
1199// ----------------------------------------------------------------------------
1200
1201AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1202    :   RefBase(),
1203        mAudioFlinger(audioFlinger),
1204        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1205        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1206        mPid(pid),
1207        mTimedTrackCount(0)
1208{
1209    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1210}
1211
1212// Client destructor must be called with AudioFlinger::mLock held
1213AudioFlinger::Client::~Client()
1214{
1215    mAudioFlinger->removeClient_l(mPid);
1216}
1217
1218sp<MemoryDealer> AudioFlinger::Client::heap() const
1219{
1220    return mMemoryDealer;
1221}
1222
1223// Reserve one of the limited slots for a timed audio track associated
1224// with this client
1225bool AudioFlinger::Client::reserveTimedTrack()
1226{
1227    const int kMaxTimedTracksPerClient = 4;
1228
1229    Mutex::Autolock _l(mTimedTrackLock);
1230
1231    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1232        ALOGW("can not create timed track - pid %d has exceeded the limit",
1233             mPid);
1234        return false;
1235    }
1236
1237    mTimedTrackCount++;
1238    return true;
1239}
1240
1241// Release a slot for a timed audio track
1242void AudioFlinger::Client::releaseTimedTrack()
1243{
1244    Mutex::Autolock _l(mTimedTrackLock);
1245    mTimedTrackCount--;
1246}
1247
1248// ----------------------------------------------------------------------------
1249
1250AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1251                                                     const sp<IAudioFlingerClient>& client,
1252                                                     pid_t pid)
1253    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1254{
1255}
1256
1257AudioFlinger::NotificationClient::~NotificationClient()
1258{
1259}
1260
1261void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1262{
1263    sp<NotificationClient> keep(this);
1264    mAudioFlinger->removeNotificationClient(mPid);
1265}
1266
1267
1268// ----------------------------------------------------------------------------
1269
1270static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1271    return audio_is_remote_submix_device(inDevice);
1272}
1273
1274sp<IAudioRecord> AudioFlinger::openRecord(
1275        audio_io_handle_t input,
1276        uint32_t sampleRate,
1277        audio_format_t format,
1278        audio_channel_mask_t channelMask,
1279        size_t frameCount,
1280        IAudioFlinger::track_flags_t *flags,
1281        pid_t tid,
1282        int *sessionId,
1283        status_t *status)
1284{
1285    sp<RecordThread::RecordTrack> recordTrack;
1286    sp<RecordHandle> recordHandle;
1287    sp<Client> client;
1288    status_t lStatus;
1289    RecordThread *thread;
1290    size_t inFrameCount;
1291    int lSessionId;
1292
1293    // check calling permissions
1294    if (!recordingAllowed()) {
1295        ALOGE("openRecord() permission denied: recording not allowed");
1296        lStatus = PERMISSION_DENIED;
1297        goto Exit;
1298    }
1299
1300    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1301        ALOGE("openRecord() invalid format %d", format);
1302        lStatus = BAD_VALUE;
1303        goto Exit;
1304    }
1305
1306    // add client to list
1307    { // scope for mLock
1308        Mutex::Autolock _l(mLock);
1309        thread = checkRecordThread_l(input);
1310        if (thread == NULL) {
1311            ALOGE("openRecord() checkRecordThread_l failed");
1312            lStatus = BAD_VALUE;
1313            goto Exit;
1314        }
1315
1316        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1317                && !captureAudioOutputAllowed()) {
1318            ALOGE("openRecord() permission denied: capture not allowed");
1319            lStatus = PERMISSION_DENIED;
1320            goto Exit;
1321        }
1322
1323        pid_t pid = IPCThreadState::self()->getCallingPid();
1324        client = registerPid_l(pid);
1325
1326        // If no audio session id is provided, create one here
1327        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1328            lSessionId = *sessionId;
1329        } else {
1330            lSessionId = nextUniqueId();
1331            if (sessionId != NULL) {
1332                *sessionId = lSessionId;
1333            }
1334        }
1335        // create new record track.
1336        // The record track uses one track in mHardwareMixerThread by convention.
1337        // TODO: the uid should be passed in as a parameter to openRecord
1338        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1339                                                  frameCount, lSessionId,
1340                                                  IPCThreadState::self()->getCallingUid(),
1341                                                  flags, tid, &lStatus);
1342        LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
1343    }
1344
1345    if (lStatus != NO_ERROR) {
1346        // remove local strong reference to Client before deleting the RecordTrack so that the
1347        // Client destructor is called by the TrackBase destructor with mLock held
1348        client.clear();
1349        recordTrack.clear();
1350        goto Exit;
1351    }
1352
1353    // return handle to client
1354    recordHandle = new RecordHandle(recordTrack);
1355
1356Exit:
1357    *status = lStatus;
1358    return recordHandle;
1359}
1360
1361
1362
1363// ----------------------------------------------------------------------------
1364
1365audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1366{
1367    if (!settingsAllowed()) {
1368        return 0;
1369    }
1370    Mutex::Autolock _l(mLock);
1371    return loadHwModule_l(name);
1372}
1373
1374// loadHwModule_l() must be called with AudioFlinger::mLock held
1375audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1376{
1377    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1378        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1379            ALOGW("loadHwModule() module %s already loaded", name);
1380            return mAudioHwDevs.keyAt(i);
1381        }
1382    }
1383
1384    audio_hw_device_t *dev;
1385
1386    int rc = load_audio_interface(name, &dev);
1387    if (rc) {
1388        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1389        return 0;
1390    }
1391
1392    mHardwareStatus = AUDIO_HW_INIT;
1393    rc = dev->init_check(dev);
1394    mHardwareStatus = AUDIO_HW_IDLE;
1395    if (rc) {
1396        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1397        return 0;
1398    }
1399
1400    // Check and cache this HAL's level of support for master mute and master
1401    // volume.  If this is the first HAL opened, and it supports the get
1402    // methods, use the initial values provided by the HAL as the current
1403    // master mute and volume settings.
1404
1405    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1406    {  // scope for auto-lock pattern
1407        AutoMutex lock(mHardwareLock);
1408
1409        if (0 == mAudioHwDevs.size()) {
1410            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1411            if (NULL != dev->get_master_volume) {
1412                float mv;
1413                if (OK == dev->get_master_volume(dev, &mv)) {
1414                    mMasterVolume = mv;
1415                }
1416            }
1417
1418            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1419            if (NULL != dev->get_master_mute) {
1420                bool mm;
1421                if (OK == dev->get_master_mute(dev, &mm)) {
1422                    mMasterMute = mm;
1423                }
1424            }
1425        }
1426
1427        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1428        if ((NULL != dev->set_master_volume) &&
1429            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1430            flags = static_cast<AudioHwDevice::Flags>(flags |
1431                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1432        }
1433
1434        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1435        if ((NULL != dev->set_master_mute) &&
1436            (OK == dev->set_master_mute(dev, mMasterMute))) {
1437            flags = static_cast<AudioHwDevice::Flags>(flags |
1438                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1439        }
1440
1441        mHardwareStatus = AUDIO_HW_IDLE;
1442    }
1443
1444    audio_module_handle_t handle = nextUniqueId();
1445    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1446
1447    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1448          name, dev->common.module->name, dev->common.module->id, handle);
1449
1450    return handle;
1451
1452}
1453
1454// ----------------------------------------------------------------------------
1455
1456uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1457{
1458    Mutex::Autolock _l(mLock);
1459    PlaybackThread *thread = primaryPlaybackThread_l();
1460    return thread != NULL ? thread->sampleRate() : 0;
1461}
1462
1463size_t AudioFlinger::getPrimaryOutputFrameCount()
1464{
1465    Mutex::Autolock _l(mLock);
1466    PlaybackThread *thread = primaryPlaybackThread_l();
1467    return thread != NULL ? thread->frameCountHAL() : 0;
1468}
1469
1470// ----------------------------------------------------------------------------
1471
1472status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1473{
1474    uid_t uid = IPCThreadState::self()->getCallingUid();
1475    if (uid != AID_SYSTEM) {
1476        return PERMISSION_DENIED;
1477    }
1478    Mutex::Autolock _l(mLock);
1479    if (mIsDeviceTypeKnown) {
1480        return INVALID_OPERATION;
1481    }
1482    mIsLowRamDevice = isLowRamDevice;
1483    mIsDeviceTypeKnown = true;
1484    return NO_ERROR;
1485}
1486
1487// ----------------------------------------------------------------------------
1488
1489audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1490                                           audio_devices_t *pDevices,
1491                                           uint32_t *pSamplingRate,
1492                                           audio_format_t *pFormat,
1493                                           audio_channel_mask_t *pChannelMask,
1494                                           uint32_t *pLatencyMs,
1495                                           audio_output_flags_t flags,
1496                                           const audio_offload_info_t *offloadInfo)
1497{
1498    struct audio_config config;
1499    memset(&config, 0, sizeof(config));
1500    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1501    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1502    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1503    if (offloadInfo != NULL) {
1504        config.offload_info = *offloadInfo;
1505    }
1506
1507    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1508              module,
1509              (pDevices != NULL) ? *pDevices : 0,
1510              config.sample_rate,
1511              config.format,
1512              config.channel_mask,
1513              flags);
1514    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1515          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1516
1517    if (pDevices == NULL || *pDevices == 0) {
1518        return 0;
1519    }
1520
1521    Mutex::Autolock _l(mLock);
1522
1523    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1524    if (outHwDev == NULL) {
1525        return 0;
1526    }
1527
1528    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1529    audio_io_handle_t id = nextUniqueId();
1530
1531    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1532
1533    audio_stream_out_t *outStream = NULL;
1534    status_t status = hwDevHal->open_output_stream(hwDevHal,
1535                                          id,
1536                                          *pDevices,
1537                                          (audio_output_flags_t)flags,
1538                                          &config,
1539                                          &outStream);
1540
1541    mHardwareStatus = AUDIO_HW_IDLE;
1542    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1543            "Channels %x, status %d",
1544            outStream,
1545            config.sample_rate,
1546            config.format,
1547            config.channel_mask,
1548            status);
1549
1550    if (status == NO_ERROR && outStream != NULL) {
1551        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1552
1553        PlaybackThread *thread;
1554        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1555            thread = new OffloadThread(this, output, id, *pDevices);
1556            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1557        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1558            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1559            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1560            thread = new DirectOutputThread(this, output, id, *pDevices);
1561            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1562        } else {
1563            thread = new MixerThread(this, output, id, *pDevices);
1564            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1565        }
1566        mPlaybackThreads.add(id, thread);
1567
1568        if (pSamplingRate != NULL) {
1569            *pSamplingRate = config.sample_rate;
1570        }
1571        if (pFormat != NULL) {
1572            *pFormat = config.format;
1573        }
1574        if (pChannelMask != NULL) {
1575            *pChannelMask = config.channel_mask;
1576        }
1577        if (pLatencyMs != NULL) {
1578            *pLatencyMs = thread->latency();
1579        }
1580
1581        // notify client processes of the new output creation
1582        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1583
1584        // the first primary output opened designates the primary hw device
1585        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1586            ALOGI("Using module %d has the primary audio interface", module);
1587            mPrimaryHardwareDev = outHwDev;
1588
1589            AutoMutex lock(mHardwareLock);
1590            mHardwareStatus = AUDIO_HW_SET_MODE;
1591            hwDevHal->set_mode(hwDevHal, mMode);
1592            mHardwareStatus = AUDIO_HW_IDLE;
1593        }
1594        return id;
1595    }
1596
1597    return 0;
1598}
1599
1600audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1601        audio_io_handle_t output2)
1602{
1603    Mutex::Autolock _l(mLock);
1604    MixerThread *thread1 = checkMixerThread_l(output1);
1605    MixerThread *thread2 = checkMixerThread_l(output2);
1606
1607    if (thread1 == NULL || thread2 == NULL) {
1608        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1609                output2);
1610        return 0;
1611    }
1612
1613    audio_io_handle_t id = nextUniqueId();
1614    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1615    thread->addOutputTrack(thread2);
1616    mPlaybackThreads.add(id, thread);
1617    // notify client processes of the new output creation
1618    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1619    return id;
1620}
1621
1622status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1623{
1624    return closeOutput_nonvirtual(output);
1625}
1626
1627status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1628{
1629    // keep strong reference on the playback thread so that
1630    // it is not destroyed while exit() is executed
1631    sp<PlaybackThread> thread;
1632    {
1633        Mutex::Autolock _l(mLock);
1634        thread = checkPlaybackThread_l(output);
1635        if (thread == NULL) {
1636            return BAD_VALUE;
1637        }
1638
1639        ALOGV("closeOutput() %d", output);
1640
1641        if (thread->type() == ThreadBase::MIXER) {
1642            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1643                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1644                    DuplicatingThread *dupThread =
1645                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1646                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1647
1648                }
1649            }
1650        }
1651
1652
1653        mPlaybackThreads.removeItem(output);
1654        // save all effects to the default thread
1655        if (mPlaybackThreads.size()) {
1656            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1657            if (dstThread != NULL) {
1658                // audioflinger lock is held here so the acquisition order of thread locks does not
1659                // matter
1660                Mutex::Autolock _dl(dstThread->mLock);
1661                Mutex::Autolock _sl(thread->mLock);
1662                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1663                for (size_t i = 0; i < effectChains.size(); i ++) {
1664                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1665                }
1666            }
1667        }
1668        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1669    }
1670    thread->exit();
1671    // The thread entity (active unit of execution) is no longer running here,
1672    // but the ThreadBase container still exists.
1673
1674    if (thread->type() != ThreadBase::DUPLICATING) {
1675        AudioStreamOut *out = thread->clearOutput();
1676        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1677        // from now on thread->mOutput is NULL
1678        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1679        delete out;
1680    }
1681    return NO_ERROR;
1682}
1683
1684status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1685{
1686    Mutex::Autolock _l(mLock);
1687    PlaybackThread *thread = checkPlaybackThread_l(output);
1688
1689    if (thread == NULL) {
1690        return BAD_VALUE;
1691    }
1692
1693    ALOGV("suspendOutput() %d", output);
1694    thread->suspend();
1695
1696    return NO_ERROR;
1697}
1698
1699status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1700{
1701    Mutex::Autolock _l(mLock);
1702    PlaybackThread *thread = checkPlaybackThread_l(output);
1703
1704    if (thread == NULL) {
1705        return BAD_VALUE;
1706    }
1707
1708    ALOGV("restoreOutput() %d", output);
1709
1710    thread->restore();
1711
1712    return NO_ERROR;
1713}
1714
1715audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1716                                          audio_devices_t *pDevices,
1717                                          uint32_t *pSamplingRate,
1718                                          audio_format_t *pFormat,
1719                                          audio_channel_mask_t *pChannelMask)
1720{
1721    struct audio_config config;
1722    memset(&config, 0, sizeof(config));
1723    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1724    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1725    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1726
1727    uint32_t reqSamplingRate = config.sample_rate;
1728    audio_format_t reqFormat = config.format;
1729    audio_channel_mask_t reqChannelMask = config.channel_mask;
1730
1731    if (pDevices == NULL || *pDevices == 0) {
1732        return 0;
1733    }
1734
1735    Mutex::Autolock _l(mLock);
1736
1737    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1738    if (inHwDev == NULL) {
1739        return 0;
1740    }
1741
1742    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1743    audio_io_handle_t id = nextUniqueId();
1744
1745    audio_stream_in_t *inStream = NULL;
1746    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1747                                        &inStream);
1748    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1749            "status %d",
1750            inStream,
1751            config.sample_rate,
1752            config.format,
1753            config.channel_mask,
1754            status);
1755
1756    // If the input could not be opened with the requested parameters and we can handle the
1757    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1758    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1759    if (status == BAD_VALUE &&
1760        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1761        (config.sample_rate <= 2 * reqSamplingRate) &&
1762        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1763        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1764        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1765        inStream = NULL;
1766        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1767        // FIXME log this new status; HAL should not propose any further changes
1768    }
1769
1770    if (status == NO_ERROR && inStream != NULL) {
1771
1772#ifdef TEE_SINK
1773        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1774        // or (re-)create if current Pipe is idle and does not match the new format
1775        sp<NBAIO_Sink> teeSink;
1776        enum {
1777            TEE_SINK_NO,    // don't copy input
1778            TEE_SINK_NEW,   // copy input using a new pipe
1779            TEE_SINK_OLD,   // copy input using an existing pipe
1780        } kind;
1781        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1782                                        popcount(inStream->common.get_channels(&inStream->common)));
1783        if (!mTeeSinkInputEnabled) {
1784            kind = TEE_SINK_NO;
1785        } else if (format == Format_Invalid) {
1786            kind = TEE_SINK_NO;
1787        } else if (mRecordTeeSink == 0) {
1788            kind = TEE_SINK_NEW;
1789        } else if (mRecordTeeSink->getStrongCount() != 1) {
1790            kind = TEE_SINK_NO;
1791        } else if (format == mRecordTeeSink->format()) {
1792            kind = TEE_SINK_OLD;
1793        } else {
1794            kind = TEE_SINK_NEW;
1795        }
1796        switch (kind) {
1797        case TEE_SINK_NEW: {
1798            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1799            size_t numCounterOffers = 0;
1800            const NBAIO_Format offers[1] = {format};
1801            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1802            ALOG_ASSERT(index == 0);
1803            PipeReader *pipeReader = new PipeReader(*pipe);
1804            numCounterOffers = 0;
1805            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1806            ALOG_ASSERT(index == 0);
1807            mRecordTeeSink = pipe;
1808            mRecordTeeSource = pipeReader;
1809            teeSink = pipe;
1810            }
1811            break;
1812        case TEE_SINK_OLD:
1813            teeSink = mRecordTeeSink;
1814            break;
1815        case TEE_SINK_NO:
1816        default:
1817            break;
1818        }
1819#endif
1820
1821        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1822
1823        // Start record thread
1824        // RecordThread requires both input and output device indication to forward to audio
1825        // pre processing modules
1826        RecordThread *thread = new RecordThread(this,
1827                                  input,
1828                                  reqSamplingRate,
1829                                  reqChannelMask,
1830                                  id,
1831                                  primaryOutputDevice_l(),
1832                                  *pDevices
1833#ifdef TEE_SINK
1834                                  , teeSink
1835#endif
1836                                  );
1837        mRecordThreads.add(id, thread);
1838        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1839        if (pSamplingRate != NULL) {
1840            *pSamplingRate = reqSamplingRate;
1841        }
1842        if (pFormat != NULL) {
1843            *pFormat = config.format;
1844        }
1845        if (pChannelMask != NULL) {
1846            *pChannelMask = reqChannelMask;
1847        }
1848
1849        // notify client processes of the new input creation
1850        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1851        return id;
1852    }
1853
1854    return 0;
1855}
1856
1857status_t AudioFlinger::closeInput(audio_io_handle_t input)
1858{
1859    return closeInput_nonvirtual(input);
1860}
1861
1862status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1863{
1864    // keep strong reference on the record thread so that
1865    // it is not destroyed while exit() is executed
1866    sp<RecordThread> thread;
1867    {
1868        Mutex::Autolock _l(mLock);
1869        thread = checkRecordThread_l(input);
1870        if (thread == 0) {
1871            return BAD_VALUE;
1872        }
1873
1874        ALOGV("closeInput() %d", input);
1875        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1876        mRecordThreads.removeItem(input);
1877    }
1878    thread->exit();
1879    // The thread entity (active unit of execution) is no longer running here,
1880    // but the ThreadBase container still exists.
1881
1882    AudioStreamIn *in = thread->clearInput();
1883    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1884    // from now on thread->mInput is NULL
1885    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1886    delete in;
1887
1888    return NO_ERROR;
1889}
1890
1891status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1892{
1893    Mutex::Autolock _l(mLock);
1894    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1895
1896    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1897        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1898        thread->invalidateTracks(stream);
1899    }
1900
1901    return NO_ERROR;
1902}
1903
1904
1905int AudioFlinger::newAudioSessionId()
1906{
1907    return nextUniqueId();
1908}
1909
1910void AudioFlinger::acquireAudioSessionId(int audioSession)
1911{
1912    Mutex::Autolock _l(mLock);
1913    pid_t caller = IPCThreadState::self()->getCallingPid();
1914    ALOGV("acquiring %d from %d", audioSession, caller);
1915
1916    // Ignore requests received from processes not known as notification client. The request
1917    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1918    // called from a different pid leaving a stale session reference.  Also we don't know how
1919    // to clear this reference if the client process dies.
1920    if (mNotificationClients.indexOfKey(caller) < 0) {
1921        ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1922        return;
1923    }
1924
1925    size_t num = mAudioSessionRefs.size();
1926    for (size_t i = 0; i< num; i++) {
1927        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1928        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1929            ref->mCnt++;
1930            ALOGV(" incremented refcount to %d", ref->mCnt);
1931            return;
1932        }
1933    }
1934    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1935    ALOGV(" added new entry for %d", audioSession);
1936}
1937
1938void AudioFlinger::releaseAudioSessionId(int audioSession)
1939{
1940    Mutex::Autolock _l(mLock);
1941    pid_t caller = IPCThreadState::self()->getCallingPid();
1942    ALOGV("releasing %d from %d", audioSession, caller);
1943    size_t num = mAudioSessionRefs.size();
1944    for (size_t i = 0; i< num; i++) {
1945        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1946        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1947            ref->mCnt--;
1948            ALOGV(" decremented refcount to %d", ref->mCnt);
1949            if (ref->mCnt == 0) {
1950                mAudioSessionRefs.removeAt(i);
1951                delete ref;
1952                purgeStaleEffects_l();
1953            }
1954            return;
1955        }
1956    }
1957    // If the caller is mediaserver it is likely that the session being released was acquired
1958    // on behalf of a process not in notification clients and we ignore the warning.
1959    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
1960}
1961
1962void AudioFlinger::purgeStaleEffects_l() {
1963
1964    ALOGV("purging stale effects");
1965
1966    Vector< sp<EffectChain> > chains;
1967
1968    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1969        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1970        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1971            sp<EffectChain> ec = t->mEffectChains[j];
1972            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1973                chains.push(ec);
1974            }
1975        }
1976    }
1977    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1978        sp<RecordThread> t = mRecordThreads.valueAt(i);
1979        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1980            sp<EffectChain> ec = t->mEffectChains[j];
1981            chains.push(ec);
1982        }
1983    }
1984
1985    for (size_t i = 0; i < chains.size(); i++) {
1986        sp<EffectChain> ec = chains[i];
1987        int sessionid = ec->sessionId();
1988        sp<ThreadBase> t = ec->mThread.promote();
1989        if (t == 0) {
1990            continue;
1991        }
1992        size_t numsessionrefs = mAudioSessionRefs.size();
1993        bool found = false;
1994        for (size_t k = 0; k < numsessionrefs; k++) {
1995            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1996            if (ref->mSessionid == sessionid) {
1997                ALOGV(" session %d still exists for %d with %d refs",
1998                    sessionid, ref->mPid, ref->mCnt);
1999                found = true;
2000                break;
2001            }
2002        }
2003        if (!found) {
2004            Mutex::Autolock _l(t->mLock);
2005            // remove all effects from the chain
2006            while (ec->mEffects.size()) {
2007                sp<EffectModule> effect = ec->mEffects[0];
2008                effect->unPin();
2009                t->removeEffect_l(effect);
2010                if (effect->purgeHandles()) {
2011                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2012                }
2013                AudioSystem::unregisterEffect(effect->id());
2014            }
2015        }
2016    }
2017    return;
2018}
2019
2020// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2021AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2022{
2023    return mPlaybackThreads.valueFor(output).get();
2024}
2025
2026// checkMixerThread_l() must be called with AudioFlinger::mLock held
2027AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2028{
2029    PlaybackThread *thread = checkPlaybackThread_l(output);
2030    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2031}
2032
2033// checkRecordThread_l() must be called with AudioFlinger::mLock held
2034AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2035{
2036    return mRecordThreads.valueFor(input).get();
2037}
2038
2039uint32_t AudioFlinger::nextUniqueId()
2040{
2041    return android_atomic_inc(&mNextUniqueId);
2042}
2043
2044AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2045{
2046    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2047        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2048        AudioStreamOut *output = thread->getOutput();
2049        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2050            return thread;
2051        }
2052    }
2053    return NULL;
2054}
2055
2056audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2057{
2058    PlaybackThread *thread = primaryPlaybackThread_l();
2059
2060    if (thread == NULL) {
2061        return 0;
2062    }
2063
2064    return thread->outDevice();
2065}
2066
2067sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2068                                    int triggerSession,
2069                                    int listenerSession,
2070                                    sync_event_callback_t callBack,
2071                                    void *cookie)
2072{
2073    Mutex::Autolock _l(mLock);
2074
2075    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2076    status_t playStatus = NAME_NOT_FOUND;
2077    status_t recStatus = NAME_NOT_FOUND;
2078    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2079        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2080        if (playStatus == NO_ERROR) {
2081            return event;
2082        }
2083    }
2084    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2085        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2086        if (recStatus == NO_ERROR) {
2087            return event;
2088        }
2089    }
2090    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2091        mPendingSyncEvents.add(event);
2092    } else {
2093        ALOGV("createSyncEvent() invalid event %d", event->type());
2094        event.clear();
2095    }
2096    return event;
2097}
2098
2099// ----------------------------------------------------------------------------
2100//  Effect management
2101// ----------------------------------------------------------------------------
2102
2103
2104status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2105{
2106    Mutex::Autolock _l(mLock);
2107    return EffectQueryNumberEffects(numEffects);
2108}
2109
2110status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2111{
2112    Mutex::Autolock _l(mLock);
2113    return EffectQueryEffect(index, descriptor);
2114}
2115
2116status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2117        effect_descriptor_t *descriptor) const
2118{
2119    Mutex::Autolock _l(mLock);
2120    return EffectGetDescriptor(pUuid, descriptor);
2121}
2122
2123
2124sp<IEffect> AudioFlinger::createEffect(
2125        effect_descriptor_t *pDesc,
2126        const sp<IEffectClient>& effectClient,
2127        int32_t priority,
2128        audio_io_handle_t io,
2129        int sessionId,
2130        status_t *status,
2131        int *id,
2132        int *enabled)
2133{
2134    status_t lStatus = NO_ERROR;
2135    sp<EffectHandle> handle;
2136    effect_descriptor_t desc;
2137
2138    pid_t pid = IPCThreadState::self()->getCallingPid();
2139    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2140            pid, effectClient.get(), priority, sessionId, io);
2141
2142    if (pDesc == NULL) {
2143        lStatus = BAD_VALUE;
2144        goto Exit;
2145    }
2146
2147    // check audio settings permission for global effects
2148    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2149        lStatus = PERMISSION_DENIED;
2150        goto Exit;
2151    }
2152
2153    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2154    // that can only be created by audio policy manager (running in same process)
2155    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2156        lStatus = PERMISSION_DENIED;
2157        goto Exit;
2158    }
2159
2160    {
2161        if (!EffectIsNullUuid(&pDesc->uuid)) {
2162            // if uuid is specified, request effect descriptor
2163            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2164            if (lStatus < 0) {
2165                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2166                goto Exit;
2167            }
2168        } else {
2169            // if uuid is not specified, look for an available implementation
2170            // of the required type in effect factory
2171            if (EffectIsNullUuid(&pDesc->type)) {
2172                ALOGW("createEffect() no effect type");
2173                lStatus = BAD_VALUE;
2174                goto Exit;
2175            }
2176            uint32_t numEffects = 0;
2177            effect_descriptor_t d;
2178            d.flags = 0; // prevent compiler warning
2179            bool found = false;
2180
2181            lStatus = EffectQueryNumberEffects(&numEffects);
2182            if (lStatus < 0) {
2183                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2184                goto Exit;
2185            }
2186            for (uint32_t i = 0; i < numEffects; i++) {
2187                lStatus = EffectQueryEffect(i, &desc);
2188                if (lStatus < 0) {
2189                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2190                    continue;
2191                }
2192                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2193                    // If matching type found save effect descriptor. If the session is
2194                    // 0 and the effect is not auxiliary, continue enumeration in case
2195                    // an auxiliary version of this effect type is available
2196                    found = true;
2197                    d = desc;
2198                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2199                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2200                        break;
2201                    }
2202                }
2203            }
2204            if (!found) {
2205                lStatus = BAD_VALUE;
2206                ALOGW("createEffect() effect not found");
2207                goto Exit;
2208            }
2209            // For same effect type, chose auxiliary version over insert version if
2210            // connect to output mix (Compliance to OpenSL ES)
2211            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2212                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2213                desc = d;
2214            }
2215        }
2216
2217        // Do not allow auxiliary effects on a session different from 0 (output mix)
2218        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2219             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2220            lStatus = INVALID_OPERATION;
2221            goto Exit;
2222        }
2223
2224        // check recording permission for visualizer
2225        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2226            !recordingAllowed()) {
2227            lStatus = PERMISSION_DENIED;
2228            goto Exit;
2229        }
2230
2231        // return effect descriptor
2232        *pDesc = desc;
2233        if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2234            // if the output returned by getOutputForEffect() is removed before we lock the
2235            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2236            // and we will exit safely
2237            io = AudioSystem::getOutputForEffect(&desc);
2238            ALOGV("createEffect got output %d", io);
2239        }
2240
2241        Mutex::Autolock _l(mLock);
2242
2243        // If output is not specified try to find a matching audio session ID in one of the
2244        // output threads.
2245        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2246        // because of code checking output when entering the function.
2247        // Note: io is never 0 when creating an effect on an input
2248        if (io == 0) {
2249            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2250                // output must be specified by AudioPolicyManager when using session
2251                // AUDIO_SESSION_OUTPUT_STAGE
2252                lStatus = BAD_VALUE;
2253                goto Exit;
2254            }
2255            // look for the thread where the specified audio session is present
2256            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2257                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2258                    io = mPlaybackThreads.keyAt(i);
2259                    break;
2260                }
2261            }
2262            if (io == 0) {
2263                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2264                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2265                        io = mRecordThreads.keyAt(i);
2266                        break;
2267                    }
2268                }
2269            }
2270            // If no output thread contains the requested session ID, default to
2271            // first output. The effect chain will be moved to the correct output
2272            // thread when a track with the same session ID is created
2273            if (io == 0 && mPlaybackThreads.size()) {
2274                io = mPlaybackThreads.keyAt(0);
2275            }
2276            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2277        }
2278        ThreadBase *thread = checkRecordThread_l(io);
2279        if (thread == NULL) {
2280            thread = checkPlaybackThread_l(io);
2281            if (thread == NULL) {
2282                ALOGE("createEffect() unknown output thread");
2283                lStatus = BAD_VALUE;
2284                goto Exit;
2285            }
2286        }
2287
2288        sp<Client> client = registerPid_l(pid);
2289
2290        // create effect on selected output thread
2291        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2292                &desc, enabled, &lStatus);
2293        if (handle != 0 && id != NULL) {
2294            *id = handle->id();
2295        }
2296    }
2297
2298Exit:
2299    *status = lStatus;
2300    return handle;
2301}
2302
2303status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2304        audio_io_handle_t dstOutput)
2305{
2306    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2307            sessionId, srcOutput, dstOutput);
2308    Mutex::Autolock _l(mLock);
2309    if (srcOutput == dstOutput) {
2310        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2311        return NO_ERROR;
2312    }
2313    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2314    if (srcThread == NULL) {
2315        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2316        return BAD_VALUE;
2317    }
2318    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2319    if (dstThread == NULL) {
2320        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2321        return BAD_VALUE;
2322    }
2323
2324    Mutex::Autolock _dl(dstThread->mLock);
2325    Mutex::Autolock _sl(srcThread->mLock);
2326    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2327}
2328
2329// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2330status_t AudioFlinger::moveEffectChain_l(int sessionId,
2331                                   AudioFlinger::PlaybackThread *srcThread,
2332                                   AudioFlinger::PlaybackThread *dstThread,
2333                                   bool reRegister)
2334{
2335    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2336            sessionId, srcThread, dstThread);
2337
2338    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2339    if (chain == 0) {
2340        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2341                sessionId, srcThread);
2342        return INVALID_OPERATION;
2343    }
2344
2345    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2346    // so that a new chain is created with correct parameters when first effect is added. This is
2347    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2348    // removed.
2349    srcThread->removeEffectChain_l(chain);
2350
2351    // transfer all effects one by one so that new effect chain is created on new thread with
2352    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2353    sp<EffectChain> dstChain;
2354    uint32_t strategy = 0; // prevent compiler warning
2355    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2356    Vector< sp<EffectModule> > removed;
2357    status_t status = NO_ERROR;
2358    while (effect != 0) {
2359        srcThread->removeEffect_l(effect);
2360        removed.add(effect);
2361        status = dstThread->addEffect_l(effect);
2362        if (status != NO_ERROR) {
2363            break;
2364        }
2365        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2366        if (effect->state() == EffectModule::ACTIVE ||
2367                effect->state() == EffectModule::STOPPING) {
2368            effect->start();
2369        }
2370        // if the move request is not received from audio policy manager, the effect must be
2371        // re-registered with the new strategy and output
2372        if (dstChain == 0) {
2373            dstChain = effect->chain().promote();
2374            if (dstChain == 0) {
2375                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2376                status = NO_INIT;
2377                break;
2378            }
2379            strategy = dstChain->strategy();
2380        }
2381        if (reRegister) {
2382            AudioSystem::unregisterEffect(effect->id());
2383            AudioSystem::registerEffect(&effect->desc(),
2384                                        dstThread->id(),
2385                                        strategy,
2386                                        sessionId,
2387                                        effect->id());
2388            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2389        }
2390        effect = chain->getEffectFromId_l(0);
2391    }
2392
2393    if (status != NO_ERROR) {
2394        for (size_t i = 0; i < removed.size(); i++) {
2395            srcThread->addEffect_l(removed[i]);
2396            if (dstChain != 0 && reRegister) {
2397                AudioSystem::unregisterEffect(removed[i]->id());
2398                AudioSystem::registerEffect(&removed[i]->desc(),
2399                                            srcThread->id(),
2400                                            strategy,
2401                                            sessionId,
2402                                            removed[i]->id());
2403                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2404            }
2405        }
2406    }
2407
2408    return status;
2409}
2410
2411bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2412{
2413    if (mGlobalEffectEnableTime != 0 &&
2414            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2415        return true;
2416    }
2417
2418    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2419        sp<EffectChain> ec =
2420                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2421        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2422            return true;
2423        }
2424    }
2425    return false;
2426}
2427
2428void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2429{
2430    Mutex::Autolock _l(mLock);
2431
2432    mGlobalEffectEnableTime = systemTime();
2433
2434    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2435        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2436        if (t->mType == ThreadBase::OFFLOAD) {
2437            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2438        }
2439    }
2440
2441}
2442
2443struct Entry {
2444#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2445    char mName[MAX_NAME];
2446};
2447
2448int comparEntry(const void *p1, const void *p2)
2449{
2450    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2451}
2452
2453#ifdef TEE_SINK
2454void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2455{
2456    NBAIO_Source *teeSource = source.get();
2457    if (teeSource != NULL) {
2458        // .wav rotation
2459        // There is a benign race condition if 2 threads call this simultaneously.
2460        // They would both traverse the directory, but the result would simply be
2461        // failures at unlink() which are ignored.  It's also unlikely since
2462        // normally dumpsys is only done by bugreport or from the command line.
2463        char teePath[32+256];
2464        strcpy(teePath, "/data/misc/media");
2465        size_t teePathLen = strlen(teePath);
2466        DIR *dir = opendir(teePath);
2467        teePath[teePathLen++] = '/';
2468        if (dir != NULL) {
2469#define MAX_SORT 20 // number of entries to sort
2470#define MAX_KEEP 10 // number of entries to keep
2471            struct Entry entries[MAX_SORT];
2472            size_t entryCount = 0;
2473            while (entryCount < MAX_SORT) {
2474                struct dirent de;
2475                struct dirent *result = NULL;
2476                int rc = readdir_r(dir, &de, &result);
2477                if (rc != 0) {
2478                    ALOGW("readdir_r failed %d", rc);
2479                    break;
2480                }
2481                if (result == NULL) {
2482                    break;
2483                }
2484                if (result != &de) {
2485                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2486                    break;
2487                }
2488                // ignore non .wav file entries
2489                size_t nameLen = strlen(de.d_name);
2490                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2491                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2492                    continue;
2493                }
2494                strcpy(entries[entryCount++].mName, de.d_name);
2495            }
2496            (void) closedir(dir);
2497            if (entryCount > MAX_KEEP) {
2498                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2499                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2500                    strcpy(&teePath[teePathLen], entries[i].mName);
2501                    (void) unlink(teePath);
2502                }
2503            }
2504        } else {
2505            if (fd >= 0) {
2506                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2507            }
2508        }
2509        char teeTime[16];
2510        struct timeval tv;
2511        gettimeofday(&tv, NULL);
2512        struct tm tm;
2513        localtime_r(&tv.tv_sec, &tm);
2514        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2515        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2516        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2517        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2518        if (teeFd >= 0) {
2519            char wavHeader[44];
2520            memcpy(wavHeader,
2521                "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",
2522                sizeof(wavHeader));
2523            NBAIO_Format format = teeSource->format();
2524            unsigned channelCount = Format_channelCount(format);
2525            ALOG_ASSERT(channelCount <= FCC_2);
2526            uint32_t sampleRate = Format_sampleRate(format);
2527            wavHeader[22] = channelCount;       // number of channels
2528            wavHeader[24] = sampleRate;         // sample rate
2529            wavHeader[25] = sampleRate >> 8;
2530            wavHeader[32] = channelCount * 2;   // block alignment
2531            write(teeFd, wavHeader, sizeof(wavHeader));
2532            size_t total = 0;
2533            bool firstRead = true;
2534            for (;;) {
2535#define TEE_SINK_READ 1024
2536                short buffer[TEE_SINK_READ * FCC_2];
2537                size_t count = TEE_SINK_READ;
2538                ssize_t actual = teeSource->read(buffer, count,
2539                        AudioBufferProvider::kInvalidPTS);
2540                bool wasFirstRead = firstRead;
2541                firstRead = false;
2542                if (actual <= 0) {
2543                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2544                        continue;
2545                    }
2546                    break;
2547                }
2548                ALOG_ASSERT(actual <= (ssize_t)count);
2549                write(teeFd, buffer, actual * channelCount * sizeof(short));
2550                total += actual;
2551            }
2552            lseek(teeFd, (off_t) 4, SEEK_SET);
2553            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2554            write(teeFd, &temp, sizeof(temp));
2555            lseek(teeFd, (off_t) 40, SEEK_SET);
2556            temp =  total * channelCount * sizeof(short);
2557            write(teeFd, &temp, sizeof(temp));
2558            close(teeFd);
2559            if (fd >= 0) {
2560                fdprintf(fd, "tee copied to %s\n", teePath);
2561            }
2562        } else {
2563            if (fd >= 0) {
2564                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2565            }
2566        }
2567    }
2568}
2569#endif
2570
2571// ----------------------------------------------------------------------------
2572
2573status_t AudioFlinger::onTransact(
2574        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2575{
2576    return BnAudioFlinger::onTransact(code, data, reply, flags);
2577}
2578
2579}; // namespace android
2580