AudioFlinger.cpp revision 6e0d67d7b496ce17c0970a4ffd3a6f808860949c 
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_ALLOCATE) {
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        LOG_ALWAYS_FATAL_IF((track != 0) != (lStatus == NO_ERROR));
563        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
564
565        // move effect chain to this output thread if an effect on same session was waiting
566        // for a track to be created
567        if (lStatus == NO_ERROR && effectThread != NULL) {
568            // no risk of deadlock because AudioFlinger::mLock is held
569            Mutex::Autolock _dl(thread->mLock);
570            Mutex::Autolock _sl(effectThread->mLock);
571            moveEffectChain_l(lSessionId, effectThread, thread, true);
572        }
573
574        // Look for sync events awaiting for a session to be used.
575        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
576            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
577                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
578                    if (lStatus == NO_ERROR) {
579                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
580                    } else {
581                        mPendingSyncEvents[i]->cancel();
582                    }
583                    mPendingSyncEvents.removeAt(i);
584                    i--;
585                }
586            }
587        }
588
589    }
590
591    if (lStatus == NO_ERROR) {
592        // s for server's pid, n for normal mixer name, f for fast index
593        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
594                track->fastIndex());
595        trackHandle = new TrackHandle(track);
596    } else {
597        // remove local strong reference to Client before deleting the Track so that the Client
598        // destructor is called by the TrackBase destructor with mLock held
599        client.clear();
600        track.clear();
601    }
602
603Exit:
604    *status = lStatus;
605    return trackHandle;
606}
607
608uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
609{
610    Mutex::Autolock _l(mLock);
611    PlaybackThread *thread = checkPlaybackThread_l(output);
612    if (thread == NULL) {
613        ALOGW("sampleRate() unknown thread %d", output);
614        return 0;
615    }
616    return thread->sampleRate();
617}
618
619int AudioFlinger::channelCount(audio_io_handle_t output) const
620{
621    Mutex::Autolock _l(mLock);
622    PlaybackThread *thread = checkPlaybackThread_l(output);
623    if (thread == NULL) {
624        ALOGW("channelCount() unknown thread %d", output);
625        return 0;
626    }
627    return thread->channelCount();
628}
629
630audio_format_t AudioFlinger::format(audio_io_handle_t output) const
631{
632    Mutex::Autolock _l(mLock);
633    PlaybackThread *thread = checkPlaybackThread_l(output);
634    if (thread == NULL) {
635        ALOGW("format() unknown thread %d", output);
636        return AUDIO_FORMAT_INVALID;
637    }
638    return thread->format();
639}
640
641size_t AudioFlinger::frameCount(audio_io_handle_t output) const
642{
643    Mutex::Autolock _l(mLock);
644    PlaybackThread *thread = checkPlaybackThread_l(output);
645    if (thread == NULL) {
646        ALOGW("frameCount() unknown thread %d", output);
647        return 0;
648    }
649    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
650    //       should examine all callers and fix them to handle smaller counts
651    return thread->frameCount();
652}
653
654uint32_t AudioFlinger::latency(audio_io_handle_t output) const
655{
656    Mutex::Autolock _l(mLock);
657    PlaybackThread *thread = checkPlaybackThread_l(output);
658    if (thread == NULL) {
659        ALOGW("latency(): no playback thread found for output handle %d", output);
660        return 0;
661    }
662    return thread->latency();
663}
664
665status_t AudioFlinger::setMasterVolume(float value)
666{
667    status_t ret = initCheck();
668    if (ret != NO_ERROR) {
669        return ret;
670    }
671
672    // check calling permissions
673    if (!settingsAllowed()) {
674        return PERMISSION_DENIED;
675    }
676
677    Mutex::Autolock _l(mLock);
678    mMasterVolume = value;
679
680    // Set master volume in the HALs which support it.
681    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
682        AutoMutex lock(mHardwareLock);
683        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
684
685        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
686        if (dev->canSetMasterVolume()) {
687            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
688        }
689        mHardwareStatus = AUDIO_HW_IDLE;
690    }
691
692    // Now set the master volume in each playback thread.  Playback threads
693    // assigned to HALs which do not have master volume support will apply
694    // master volume during the mix operation.  Threads with HALs which do
695    // support master volume will simply ignore the setting.
696    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
697        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
698
699    return NO_ERROR;
700}
701
702status_t AudioFlinger::setMode(audio_mode_t mode)
703{
704    status_t ret = initCheck();
705    if (ret != NO_ERROR) {
706        return ret;
707    }
708
709    // check calling permissions
710    if (!settingsAllowed()) {
711        return PERMISSION_DENIED;
712    }
713    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
714        ALOGW("Illegal value: setMode(%d)", mode);
715        return BAD_VALUE;
716    }
717
718    { // scope for the lock
719        AutoMutex lock(mHardwareLock);
720        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
721        mHardwareStatus = AUDIO_HW_SET_MODE;
722        ret = dev->set_mode(dev, mode);
723        mHardwareStatus = AUDIO_HW_IDLE;
724    }
725
726    if (NO_ERROR == ret) {
727        Mutex::Autolock _l(mLock);
728        mMode = mode;
729        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
730            mPlaybackThreads.valueAt(i)->setMode(mode);
731    }
732
733    return ret;
734}
735
736status_t AudioFlinger::setMicMute(bool state)
737{
738    status_t ret = initCheck();
739    if (ret != NO_ERROR) {
740        return ret;
741    }
742
743    // check calling permissions
744    if (!settingsAllowed()) {
745        return PERMISSION_DENIED;
746    }
747
748    AutoMutex lock(mHardwareLock);
749    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
750    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
751    ret = dev->set_mic_mute(dev, state);
752    mHardwareStatus = AUDIO_HW_IDLE;
753    return ret;
754}
755
756bool AudioFlinger::getMicMute() const
757{
758    status_t ret = initCheck();
759    if (ret != NO_ERROR) {
760        return false;
761    }
762
763    bool state = AUDIO_MODE_INVALID;
764    AutoMutex lock(mHardwareLock);
765    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
766    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
767    dev->get_mic_mute(dev, &state);
768    mHardwareStatus = AUDIO_HW_IDLE;
769    return state;
770}
771
772status_t AudioFlinger::setMasterMute(bool muted)
773{
774    status_t ret = initCheck();
775    if (ret != NO_ERROR) {
776        return ret;
777    }
778
779    // check calling permissions
780    if (!settingsAllowed()) {
781        return PERMISSION_DENIED;
782    }
783
784    Mutex::Autolock _l(mLock);
785    mMasterMute = muted;
786
787    // Set master mute in the HALs which support it.
788    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
789        AutoMutex lock(mHardwareLock);
790        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
791
792        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
793        if (dev->canSetMasterMute()) {
794            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
795        }
796        mHardwareStatus = AUDIO_HW_IDLE;
797    }
798
799    // Now set the master mute in each playback thread.  Playback threads
800    // assigned to HALs which do not have master mute support will apply master
801    // mute during the mix operation.  Threads with HALs which do support master
802    // mute will simply ignore the setting.
803    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
804        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
805
806    return NO_ERROR;
807}
808
809float AudioFlinger::masterVolume() const
810{
811    Mutex::Autolock _l(mLock);
812    return masterVolume_l();
813}
814
815bool AudioFlinger::masterMute() const
816{
817    Mutex::Autolock _l(mLock);
818    return masterMute_l();
819}
820
821float AudioFlinger::masterVolume_l() const
822{
823    return mMasterVolume;
824}
825
826bool AudioFlinger::masterMute_l() const
827{
828    return mMasterMute;
829}
830
831status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
832        audio_io_handle_t output)
833{
834    // check calling permissions
835    if (!settingsAllowed()) {
836        return PERMISSION_DENIED;
837    }
838
839    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
840        ALOGE("setStreamVolume() invalid stream %d", stream);
841        return BAD_VALUE;
842    }
843
844    AutoMutex lock(mLock);
845    PlaybackThread *thread = NULL;
846    if (output) {
847        thread = checkPlaybackThread_l(output);
848        if (thread == NULL) {
849            return BAD_VALUE;
850        }
851    }
852
853    mStreamTypes[stream].volume = value;
854
855    if (thread == NULL) {
856        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
857            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
858        }
859    } else {
860        thread->setStreamVolume(stream, value);
861    }
862
863    return NO_ERROR;
864}
865
866status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
867{
868    // check calling permissions
869    if (!settingsAllowed()) {
870        return PERMISSION_DENIED;
871    }
872
873    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
874        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
875        ALOGE("setStreamMute() invalid stream %d", stream);
876        return BAD_VALUE;
877    }
878
879    AutoMutex lock(mLock);
880    mStreamTypes[stream].mute = muted;
881    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
882        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
883
884    return NO_ERROR;
885}
886
887float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
888{
889    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
890        return 0.0f;
891    }
892
893    AutoMutex lock(mLock);
894    float volume;
895    if (output) {
896        PlaybackThread *thread = checkPlaybackThread_l(output);
897        if (thread == NULL) {
898            return 0.0f;
899        }
900        volume = thread->streamVolume(stream);
901    } else {
902        volume = streamVolume_l(stream);
903    }
904
905    return volume;
906}
907
908bool AudioFlinger::streamMute(audio_stream_type_t stream) const
909{
910    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
911        return true;
912    }
913
914    AutoMutex lock(mLock);
915    return streamMute_l(stream);
916}
917
918status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
919{
920    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
921            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
922
923    // check calling permissions
924    if (!settingsAllowed()) {
925        return PERMISSION_DENIED;
926    }
927
928    // ioHandle == 0 means the parameters are global to the audio hardware interface
929    if (ioHandle == 0) {
930        Mutex::Autolock _l(mLock);
931        status_t final_result = NO_ERROR;
932        {
933            AutoMutex lock(mHardwareLock);
934            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
935            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
936                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
937                status_t result = dev->set_parameters(dev, keyValuePairs.string());
938                final_result = result ?: final_result;
939            }
940            mHardwareStatus = AUDIO_HW_IDLE;
941        }
942        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
943        AudioParameter param = AudioParameter(keyValuePairs);
944        String8 value;
945        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
946            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
947            if (mBtNrecIsOff != btNrecIsOff) {
948                for (size_t i = 0; i < mRecordThreads.size(); i++) {
949                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
950                    audio_devices_t device = thread->inDevice();
951                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
952                    // collect all of the thread's session IDs
953                    KeyedVector<int, bool> ids = thread->sessionIds();
954                    // suspend effects associated with those session IDs
955                    for (size_t j = 0; j < ids.size(); ++j) {
956                        int sessionId = ids.keyAt(j);
957                        thread->setEffectSuspended(FX_IID_AEC,
958                                                   suspend,
959                                                   sessionId);
960                        thread->setEffectSuspended(FX_IID_NS,
961                                                   suspend,
962                                                   sessionId);
963                    }
964                }
965                mBtNrecIsOff = btNrecIsOff;
966            }
967        }
968        String8 screenState;
969        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
970            bool isOff = screenState == "off";
971            if (isOff != (AudioFlinger::mScreenState & 1)) {
972                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
973            }
974        }
975        return final_result;
976    }
977
978    // hold a strong ref on thread in case closeOutput() or closeInput() is called
979    // and the thread is exited once the lock is released
980    sp<ThreadBase> thread;
981    {
982        Mutex::Autolock _l(mLock);
983        thread = checkPlaybackThread_l(ioHandle);
984        if (thread == 0) {
985            thread = checkRecordThread_l(ioHandle);
986        } else if (thread == primaryPlaybackThread_l()) {
987            // indicate output device change to all input threads for pre processing
988            AudioParameter param = AudioParameter(keyValuePairs);
989            int value;
990            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
991                    (value != 0)) {
992                for (size_t i = 0; i < mRecordThreads.size(); i++) {
993                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
994                }
995            }
996        }
997    }
998    if (thread != 0) {
999        return thread->setParameters(keyValuePairs);
1000    }
1001    return BAD_VALUE;
1002}
1003
1004String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1005{
1006    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1007            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1008
1009    Mutex::Autolock _l(mLock);
1010
1011    if (ioHandle == 0) {
1012        String8 out_s8;
1013
1014        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1015            char *s;
1016            {
1017            AutoMutex lock(mHardwareLock);
1018            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1019            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1020            s = dev->get_parameters(dev, keys.string());
1021            mHardwareStatus = AUDIO_HW_IDLE;
1022            }
1023            out_s8 += String8(s ? s : "");
1024            free(s);
1025        }
1026        return out_s8;
1027    }
1028
1029    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1030    if (playbackThread != NULL) {
1031        return playbackThread->getParameters(keys);
1032    }
1033    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1034    if (recordThread != NULL) {
1035        return recordThread->getParameters(keys);
1036    }
1037    return String8("");
1038}
1039
1040size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1041        audio_channel_mask_t channelMask) const
1042{
1043    status_t ret = initCheck();
1044    if (ret != NO_ERROR) {
1045        return 0;
1046    }
1047
1048    AutoMutex lock(mHardwareLock);
1049    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1050    struct audio_config config;
1051    memset(&config, 0, sizeof(config));
1052    config.sample_rate = sampleRate;
1053    config.channel_mask = channelMask;
1054    config.format = format;
1055
1056    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1057    size_t size = dev->get_input_buffer_size(dev, &config);
1058    mHardwareStatus = AUDIO_HW_IDLE;
1059    return size;
1060}
1061
1062uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1063{
1064    Mutex::Autolock _l(mLock);
1065
1066    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1067    if (recordThread != NULL) {
1068        return recordThread->getInputFramesLost();
1069    }
1070    return 0;
1071}
1072
1073status_t AudioFlinger::setVoiceVolume(float value)
1074{
1075    status_t ret = initCheck();
1076    if (ret != NO_ERROR) {
1077        return ret;
1078    }
1079
1080    // check calling permissions
1081    if (!settingsAllowed()) {
1082        return PERMISSION_DENIED;
1083    }
1084
1085    AutoMutex lock(mHardwareLock);
1086    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1087    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1088    ret = dev->set_voice_volume(dev, value);
1089    mHardwareStatus = AUDIO_HW_IDLE;
1090
1091    return ret;
1092}
1093
1094status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1095        audio_io_handle_t output) const
1096{
1097    status_t status;
1098
1099    Mutex::Autolock _l(mLock);
1100
1101    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1102    if (playbackThread != NULL) {
1103        return playbackThread->getRenderPosition(halFrames, dspFrames);
1104    }
1105
1106    return BAD_VALUE;
1107}
1108
1109void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1110{
1111
1112    Mutex::Autolock _l(mLock);
1113
1114    pid_t pid = IPCThreadState::self()->getCallingPid();
1115    if (mNotificationClients.indexOfKey(pid) < 0) {
1116        sp<NotificationClient> notificationClient = new NotificationClient(this,
1117                                                                            client,
1118                                                                            pid);
1119        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1120
1121        mNotificationClients.add(pid, notificationClient);
1122
1123        sp<IBinder> binder = client->asBinder();
1124        binder->linkToDeath(notificationClient);
1125
1126        // the config change is always sent from playback or record threads to avoid deadlock
1127        // with AudioSystem::gLock
1128        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1129            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1130        }
1131
1132        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1133            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1134        }
1135    }
1136}
1137
1138void AudioFlinger::removeNotificationClient(pid_t pid)
1139{
1140    Mutex::Autolock _l(mLock);
1141
1142    mNotificationClients.removeItem(pid);
1143
1144    ALOGV("%d died, releasing its sessions", pid);
1145    size_t num = mAudioSessionRefs.size();
1146    bool removed = false;
1147    for (size_t i = 0; i< num; ) {
1148        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1149        ALOGV(" pid %d @ %d", ref->mPid, i);
1150        if (ref->mPid == pid) {
1151            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1152            mAudioSessionRefs.removeAt(i);
1153            delete ref;
1154            removed = true;
1155            num--;
1156        } else {
1157            i++;
1158        }
1159    }
1160    if (removed) {
1161        purgeStaleEffects_l();
1162    }
1163}
1164
1165// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1166void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1167{
1168    size_t size = mNotificationClients.size();
1169    for (size_t i = 0; i < size; i++) {
1170        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1171                                                                               param2);
1172    }
1173}
1174
1175// removeClient_l() must be called with AudioFlinger::mLock held
1176void AudioFlinger::removeClient_l(pid_t pid)
1177{
1178    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1179            IPCThreadState::self()->getCallingPid());
1180    mClients.removeItem(pid);
1181}
1182
1183// getEffectThread_l() must be called with AudioFlinger::mLock held
1184sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1185{
1186    sp<PlaybackThread> thread;
1187
1188    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1189        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1190            ALOG_ASSERT(thread == 0);
1191            thread = mPlaybackThreads.valueAt(i);
1192        }
1193    }
1194
1195    return thread;
1196}
1197
1198
1199
1200// ----------------------------------------------------------------------------
1201
1202AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1203    :   RefBase(),
1204        mAudioFlinger(audioFlinger),
1205        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1206        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1207        mPid(pid),
1208        mTimedTrackCount(0)
1209{
1210    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1211}
1212
1213// Client destructor must be called with AudioFlinger::mLock held
1214AudioFlinger::Client::~Client()
1215{
1216    mAudioFlinger->removeClient_l(mPid);
1217}
1218
1219sp<MemoryDealer> AudioFlinger::Client::heap() const
1220{
1221    return mMemoryDealer;
1222}
1223
1224// Reserve one of the limited slots for a timed audio track associated
1225// with this client
1226bool AudioFlinger::Client::reserveTimedTrack()
1227{
1228    const int kMaxTimedTracksPerClient = 4;
1229
1230    Mutex::Autolock _l(mTimedTrackLock);
1231
1232    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1233        ALOGW("can not create timed track - pid %d has exceeded the limit",
1234             mPid);
1235        return false;
1236    }
1237
1238    mTimedTrackCount++;
1239    return true;
1240}
1241
1242// Release a slot for a timed audio track
1243void AudioFlinger::Client::releaseTimedTrack()
1244{
1245    Mutex::Autolock _l(mTimedTrackLock);
1246    mTimedTrackCount--;
1247}
1248
1249// ----------------------------------------------------------------------------
1250
1251AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1252                                                     const sp<IAudioFlingerClient>& client,
1253                                                     pid_t pid)
1254    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1255{
1256}
1257
1258AudioFlinger::NotificationClient::~NotificationClient()
1259{
1260}
1261
1262void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1263{
1264    sp<NotificationClient> keep(this);
1265    mAudioFlinger->removeNotificationClient(mPid);
1266}
1267
1268
1269// ----------------------------------------------------------------------------
1270
1271static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1272    return audio_is_remote_submix_device(inDevice);
1273}
1274
1275sp<IAudioRecord> AudioFlinger::openRecord(
1276        audio_io_handle_t input,
1277        uint32_t sampleRate,
1278        audio_format_t format,
1279        audio_channel_mask_t channelMask,
1280        size_t *frameCount,
1281        IAudioFlinger::track_flags_t *flags,
1282        pid_t tid,
1283        int *sessionId,
1284        status_t *status)
1285{
1286    sp<RecordThread::RecordTrack> recordTrack;
1287    sp<RecordHandle> recordHandle;
1288    sp<Client> client;
1289    status_t lStatus;
1290    RecordThread *thread;
1291    size_t inFrameCount;
1292    int lSessionId;
1293
1294    // check calling permissions
1295    if (!recordingAllowed()) {
1296        ALOGE("openRecord() permission denied: recording not allowed");
1297        lStatus = PERMISSION_DENIED;
1298        goto Exit;
1299    }
1300
1301    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1302        ALOGE("openRecord() invalid format %d", format);
1303        lStatus = BAD_VALUE;
1304        goto Exit;
1305    }
1306
1307    // add client to list
1308    { // scope for mLock
1309        Mutex::Autolock _l(mLock);
1310        thread = checkRecordThread_l(input);
1311        if (thread == NULL) {
1312            ALOGE("openRecord() checkRecordThread_l failed");
1313            lStatus = BAD_VALUE;
1314            goto Exit;
1315        }
1316
1317        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1318                && !captureAudioOutputAllowed()) {
1319            ALOGE("openRecord() permission denied: capture not allowed");
1320            lStatus = PERMISSION_DENIED;
1321            goto Exit;
1322        }
1323
1324        pid_t pid = IPCThreadState::self()->getCallingPid();
1325        client = registerPid_l(pid);
1326
1327        // If no audio session id is provided, create one here
1328        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1329            lSessionId = *sessionId;
1330        } else {
1331            lSessionId = nextUniqueId();
1332            if (sessionId != NULL) {
1333                *sessionId = lSessionId;
1334            }
1335        }
1336        // create new record track.
1337        // The record track uses one track in mHardwareMixerThread by convention.
1338        // TODO: the uid should be passed in as a parameter to openRecord
1339        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1340                                                  frameCount, lSessionId,
1341                                                  IPCThreadState::self()->getCallingUid(),
1342                                                  flags, tid, &lStatus);
1343        LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
1344    }
1345
1346    if (lStatus != NO_ERROR) {
1347        // remove local strong reference to Client before deleting the RecordTrack so that the
1348        // Client destructor is called by the TrackBase destructor with mLock held
1349        client.clear();
1350        recordTrack.clear();
1351        goto Exit;
1352    }
1353
1354    // return handle to client
1355    recordHandle = new RecordHandle(recordTrack);
1356
1357Exit:
1358    *status = lStatus;
1359    return recordHandle;
1360}
1361
1362
1363
1364// ----------------------------------------------------------------------------
1365
1366audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1367{
1368    if (!settingsAllowed()) {
1369        return 0;
1370    }
1371    Mutex::Autolock _l(mLock);
1372    return loadHwModule_l(name);
1373}
1374
1375// loadHwModule_l() must be called with AudioFlinger::mLock held
1376audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1377{
1378    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1379        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1380            ALOGW("loadHwModule() module %s already loaded", name);
1381            return mAudioHwDevs.keyAt(i);
1382        }
1383    }
1384
1385    audio_hw_device_t *dev;
1386
1387    int rc = load_audio_interface(name, &dev);
1388    if (rc) {
1389        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1390        return 0;
1391    }
1392
1393    mHardwareStatus = AUDIO_HW_INIT;
1394    rc = dev->init_check(dev);
1395    mHardwareStatus = AUDIO_HW_IDLE;
1396    if (rc) {
1397        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1398        return 0;
1399    }
1400
1401    // Check and cache this HAL's level of support for master mute and master
1402    // volume.  If this is the first HAL opened, and it supports the get
1403    // methods, use the initial values provided by the HAL as the current
1404    // master mute and volume settings.
1405
1406    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1407    {  // scope for auto-lock pattern
1408        AutoMutex lock(mHardwareLock);
1409
1410        if (0 == mAudioHwDevs.size()) {
1411            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1412            if (NULL != dev->get_master_volume) {
1413                float mv;
1414                if (OK == dev->get_master_volume(dev, &mv)) {
1415                    mMasterVolume = mv;
1416                }
1417            }
1418
1419            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1420            if (NULL != dev->get_master_mute) {
1421                bool mm;
1422                if (OK == dev->get_master_mute(dev, &mm)) {
1423                    mMasterMute = mm;
1424                }
1425            }
1426        }
1427
1428        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1429        if ((NULL != dev->set_master_volume) &&
1430            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1431            flags = static_cast<AudioHwDevice::Flags>(flags |
1432                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1433        }
1434
1435        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1436        if ((NULL != dev->set_master_mute) &&
1437            (OK == dev->set_master_mute(dev, mMasterMute))) {
1438            flags = static_cast<AudioHwDevice::Flags>(flags |
1439                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1440        }
1441
1442        mHardwareStatus = AUDIO_HW_IDLE;
1443    }
1444
1445    audio_module_handle_t handle = nextUniqueId();
1446    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1447
1448    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1449          name, dev->common.module->name, dev->common.module->id, handle);
1450
1451    return handle;
1452
1453}
1454
1455// ----------------------------------------------------------------------------
1456
1457uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1458{
1459    Mutex::Autolock _l(mLock);
1460    PlaybackThread *thread = primaryPlaybackThread_l();
1461    return thread != NULL ? thread->sampleRate() : 0;
1462}
1463
1464size_t AudioFlinger::getPrimaryOutputFrameCount()
1465{
1466    Mutex::Autolock _l(mLock);
1467    PlaybackThread *thread = primaryPlaybackThread_l();
1468    return thread != NULL ? thread->frameCountHAL() : 0;
1469}
1470
1471// ----------------------------------------------------------------------------
1472
1473status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1474{
1475    uid_t uid = IPCThreadState::self()->getCallingUid();
1476    if (uid != AID_SYSTEM) {
1477        return PERMISSION_DENIED;
1478    }
1479    Mutex::Autolock _l(mLock);
1480    if (mIsDeviceTypeKnown) {
1481        return INVALID_OPERATION;
1482    }
1483    mIsLowRamDevice = isLowRamDevice;
1484    mIsDeviceTypeKnown = true;
1485    return NO_ERROR;
1486}
1487
1488// ----------------------------------------------------------------------------
1489
1490audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1491                                           audio_devices_t *pDevices,
1492                                           uint32_t *pSamplingRate,
1493                                           audio_format_t *pFormat,
1494                                           audio_channel_mask_t *pChannelMask,
1495                                           uint32_t *pLatencyMs,
1496                                           audio_output_flags_t flags,
1497                                           const audio_offload_info_t *offloadInfo)
1498{
1499    struct audio_config config;
1500    memset(&config, 0, sizeof(config));
1501    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1502    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1503    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1504    if (offloadInfo != NULL) {
1505        config.offload_info = *offloadInfo;
1506    }
1507
1508    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1509              module,
1510              (pDevices != NULL) ? *pDevices : 0,
1511              config.sample_rate,
1512              config.format,
1513              config.channel_mask,
1514              flags);
1515    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1516          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1517
1518    if (pDevices == NULL || *pDevices == 0) {
1519        return 0;
1520    }
1521
1522    Mutex::Autolock _l(mLock);
1523
1524    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1525    if (outHwDev == NULL) {
1526        return 0;
1527    }
1528
1529    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1530    audio_io_handle_t id = nextUniqueId();
1531
1532    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1533
1534    audio_stream_out_t *outStream = NULL;
1535    status_t status = hwDevHal->open_output_stream(hwDevHal,
1536                                          id,
1537                                          *pDevices,
1538                                          (audio_output_flags_t)flags,
1539                                          &config,
1540                                          &outStream);
1541
1542    mHardwareStatus = AUDIO_HW_IDLE;
1543    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1544            "Channels %x, status %d",
1545            outStream,
1546            config.sample_rate,
1547            config.format,
1548            config.channel_mask,
1549            status);
1550
1551    if (status == NO_ERROR && outStream != NULL) {
1552        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1553
1554        PlaybackThread *thread;
1555        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1556            thread = new OffloadThread(this, output, id, *pDevices);
1557            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1558        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1559            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1560            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1561            thread = new DirectOutputThread(this, output, id, *pDevices);
1562            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1563        } else {
1564            thread = new MixerThread(this, output, id, *pDevices);
1565            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1566        }
1567        mPlaybackThreads.add(id, thread);
1568
1569        if (pSamplingRate != NULL) {
1570            *pSamplingRate = config.sample_rate;
1571        }
1572        if (pFormat != NULL) {
1573            *pFormat = config.format;
1574        }
1575        if (pChannelMask != NULL) {
1576            *pChannelMask = config.channel_mask;
1577        }
1578        if (pLatencyMs != NULL) {
1579            *pLatencyMs = thread->latency();
1580        }
1581
1582        // notify client processes of the new output creation
1583        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1584
1585        // the first primary output opened designates the primary hw device
1586        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1587            ALOGI("Using module %d has the primary audio interface", module);
1588            mPrimaryHardwareDev = outHwDev;
1589
1590            AutoMutex lock(mHardwareLock);
1591            mHardwareStatus = AUDIO_HW_SET_MODE;
1592            hwDevHal->set_mode(hwDevHal, mMode);
1593            mHardwareStatus = AUDIO_HW_IDLE;
1594        }
1595        return id;
1596    }
1597
1598    return 0;
1599}
1600
1601audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1602        audio_io_handle_t output2)
1603{
1604    Mutex::Autolock _l(mLock);
1605    MixerThread *thread1 = checkMixerThread_l(output1);
1606    MixerThread *thread2 = checkMixerThread_l(output2);
1607
1608    if (thread1 == NULL || thread2 == NULL) {
1609        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1610                output2);
1611        return 0;
1612    }
1613
1614    audio_io_handle_t id = nextUniqueId();
1615    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1616    thread->addOutputTrack(thread2);
1617    mPlaybackThreads.add(id, thread);
1618    // notify client processes of the new output creation
1619    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1620    return id;
1621}
1622
1623status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1624{
1625    return closeOutput_nonvirtual(output);
1626}
1627
1628status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1629{
1630    // keep strong reference on the playback thread so that
1631    // it is not destroyed while exit() is executed
1632    sp<PlaybackThread> thread;
1633    {
1634        Mutex::Autolock _l(mLock);
1635        thread = checkPlaybackThread_l(output);
1636        if (thread == NULL) {
1637            return BAD_VALUE;
1638        }
1639
1640        ALOGV("closeOutput() %d", output);
1641
1642        if (thread->type() == ThreadBase::MIXER) {
1643            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1644                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1645                    DuplicatingThread *dupThread =
1646                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1647                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1648
1649                }
1650            }
1651        }
1652
1653
1654        mPlaybackThreads.removeItem(output);
1655        // save all effects to the default thread
1656        if (mPlaybackThreads.size()) {
1657            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1658            if (dstThread != NULL) {
1659                // audioflinger lock is held here so the acquisition order of thread locks does not
1660                // matter
1661                Mutex::Autolock _dl(dstThread->mLock);
1662                Mutex::Autolock _sl(thread->mLock);
1663                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1664                for (size_t i = 0; i < effectChains.size(); i ++) {
1665                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1666                }
1667            }
1668        }
1669        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1670    }
1671    thread->exit();
1672    // The thread entity (active unit of execution) is no longer running here,
1673    // but the ThreadBase container still exists.
1674
1675    if (thread->type() != ThreadBase::DUPLICATING) {
1676        AudioStreamOut *out = thread->clearOutput();
1677        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1678        // from now on thread->mOutput is NULL
1679        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1680        delete out;
1681    }
1682    return NO_ERROR;
1683}
1684
1685status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1686{
1687    Mutex::Autolock _l(mLock);
1688    PlaybackThread *thread = checkPlaybackThread_l(output);
1689
1690    if (thread == NULL) {
1691        return BAD_VALUE;
1692    }
1693
1694    ALOGV("suspendOutput() %d", output);
1695    thread->suspend();
1696
1697    return NO_ERROR;
1698}
1699
1700status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1701{
1702    Mutex::Autolock _l(mLock);
1703    PlaybackThread *thread = checkPlaybackThread_l(output);
1704
1705    if (thread == NULL) {
1706        return BAD_VALUE;
1707    }
1708
1709    ALOGV("restoreOutput() %d", output);
1710
1711    thread->restore();
1712
1713    return NO_ERROR;
1714}
1715
1716audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1717                                          audio_devices_t *pDevices,
1718                                          uint32_t *pSamplingRate,
1719                                          audio_format_t *pFormat,
1720                                          audio_channel_mask_t *pChannelMask)
1721{
1722    struct audio_config config;
1723    memset(&config, 0, sizeof(config));
1724    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1725    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1726    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1727
1728    uint32_t reqSamplingRate = config.sample_rate;
1729    audio_format_t reqFormat = config.format;
1730    audio_channel_mask_t reqChannelMask = config.channel_mask;
1731
1732    if (pDevices == NULL || *pDevices == 0) {
1733        return 0;
1734    }
1735
1736    Mutex::Autolock _l(mLock);
1737
1738    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1739    if (inHwDev == NULL) {
1740        return 0;
1741    }
1742
1743    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1744    audio_io_handle_t id = nextUniqueId();
1745
1746    audio_stream_in_t *inStream = NULL;
1747    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1748                                        &inStream);
1749    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1750            "status %d",
1751            inStream,
1752            config.sample_rate,
1753            config.format,
1754            config.channel_mask,
1755            status);
1756
1757    // If the input could not be opened with the requested parameters and we can handle the
1758    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1759    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1760    if (status == BAD_VALUE &&
1761        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1762        (config.sample_rate <= 2 * reqSamplingRate) &&
1763        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1764        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1765        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1766        inStream = NULL;
1767        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1768        // FIXME log this new status; HAL should not propose any further changes
1769    }
1770
1771    if (status == NO_ERROR && inStream != NULL) {
1772
1773#ifdef TEE_SINK
1774        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1775        // or (re-)create if current Pipe is idle and does not match the new format
1776        sp<NBAIO_Sink> teeSink;
1777        enum {
1778            TEE_SINK_NO,    // don't copy input
1779            TEE_SINK_NEW,   // copy input using a new pipe
1780            TEE_SINK_OLD,   // copy input using an existing pipe
1781        } kind;
1782        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1783                                        popcount(inStream->common.get_channels(&inStream->common)));
1784        if (!mTeeSinkInputEnabled) {
1785            kind = TEE_SINK_NO;
1786        } else if (!Format_isValid(format)) {
1787            kind = TEE_SINK_NO;
1788        } else if (mRecordTeeSink == 0) {
1789            kind = TEE_SINK_NEW;
1790        } else if (mRecordTeeSink->getStrongCount() != 1) {
1791            kind = TEE_SINK_NO;
1792        } else if (format == mRecordTeeSink->format()) {
1793            kind = TEE_SINK_OLD;
1794        } else {
1795            kind = TEE_SINK_NEW;
1796        }
1797        switch (kind) {
1798        case TEE_SINK_NEW: {
1799            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1800            size_t numCounterOffers = 0;
1801            const NBAIO_Format offers[1] = {format};
1802            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1803            ALOG_ASSERT(index == 0);
1804            PipeReader *pipeReader = new PipeReader(*pipe);
1805            numCounterOffers = 0;
1806            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1807            ALOG_ASSERT(index == 0);
1808            mRecordTeeSink = pipe;
1809            mRecordTeeSource = pipeReader;
1810            teeSink = pipe;
1811            }
1812            break;
1813        case TEE_SINK_OLD:
1814            teeSink = mRecordTeeSink;
1815            break;
1816        case TEE_SINK_NO:
1817        default:
1818            break;
1819        }
1820#endif
1821
1822        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1823
1824        // Start record thread
1825        // RecordThread requires both input and output device indication to forward to audio
1826        // pre processing modules
1827        RecordThread *thread = new RecordThread(this,
1828                                  input,
1829                                  reqSamplingRate,
1830                                  reqChannelMask,
1831                                  id,
1832                                  primaryOutputDevice_l(),
1833                                  *pDevices
1834#ifdef TEE_SINK
1835                                  , teeSink
1836#endif
1837                                  );
1838        mRecordThreads.add(id, thread);
1839        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1840        if (pSamplingRate != NULL) {
1841            *pSamplingRate = reqSamplingRate;
1842        }
1843        if (pFormat != NULL) {
1844            *pFormat = config.format;
1845        }
1846        if (pChannelMask != NULL) {
1847            *pChannelMask = reqChannelMask;
1848        }
1849
1850        // notify client processes of the new input creation
1851        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1852        return id;
1853    }
1854
1855    return 0;
1856}
1857
1858status_t AudioFlinger::closeInput(audio_io_handle_t input)
1859{
1860    return closeInput_nonvirtual(input);
1861}
1862
1863status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1864{
1865    // keep strong reference on the record thread so that
1866    // it is not destroyed while exit() is executed
1867    sp<RecordThread> thread;
1868    {
1869        Mutex::Autolock _l(mLock);
1870        thread = checkRecordThread_l(input);
1871        if (thread == 0) {
1872            return BAD_VALUE;
1873        }
1874
1875        ALOGV("closeInput() %d", input);
1876        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1877        mRecordThreads.removeItem(input);
1878    }
1879    thread->exit();
1880    // The thread entity (active unit of execution) is no longer running here,
1881    // but the ThreadBase container still exists.
1882
1883    AudioStreamIn *in = thread->clearInput();
1884    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1885    // from now on thread->mInput is NULL
1886    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1887    delete in;
1888
1889    return NO_ERROR;
1890}
1891
1892status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1893{
1894    Mutex::Autolock _l(mLock);
1895    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1896
1897    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1898        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1899        thread->invalidateTracks(stream);
1900    }
1901
1902    return NO_ERROR;
1903}
1904
1905
1906int AudioFlinger::newAudioSessionId()
1907{
1908    return nextUniqueId();
1909}
1910
1911void AudioFlinger::acquireAudioSessionId(int audioSession)
1912{
1913    Mutex::Autolock _l(mLock);
1914    pid_t caller = IPCThreadState::self()->getCallingPid();
1915    ALOGV("acquiring %d from %d", audioSession, caller);
1916
1917    // Ignore requests received from processes not known as notification client. The request
1918    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1919    // called from a different pid leaving a stale session reference.  Also we don't know how
1920    // to clear this reference if the client process dies.
1921    if (mNotificationClients.indexOfKey(caller) < 0) {
1922        ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1923        return;
1924    }
1925
1926    size_t num = mAudioSessionRefs.size();
1927    for (size_t i = 0; i< num; i++) {
1928        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1929        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1930            ref->mCnt++;
1931            ALOGV(" incremented refcount to %d", ref->mCnt);
1932            return;
1933        }
1934    }
1935    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1936    ALOGV(" added new entry for %d", audioSession);
1937}
1938
1939void AudioFlinger::releaseAudioSessionId(int audioSession)
1940{
1941    Mutex::Autolock _l(mLock);
1942    pid_t caller = IPCThreadState::self()->getCallingPid();
1943    ALOGV("releasing %d from %d", audioSession, caller);
1944    size_t num = mAudioSessionRefs.size();
1945    for (size_t i = 0; i< num; i++) {
1946        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1947        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1948            ref->mCnt--;
1949            ALOGV(" decremented refcount to %d", ref->mCnt);
1950            if (ref->mCnt == 0) {
1951                mAudioSessionRefs.removeAt(i);
1952                delete ref;
1953                purgeStaleEffects_l();
1954            }
1955            return;
1956        }
1957    }
1958    // If the caller is mediaserver it is likely that the session being released was acquired
1959    // on behalf of a process not in notification clients and we ignore the warning.
1960    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
1961}
1962
1963void AudioFlinger::purgeStaleEffects_l() {
1964
1965    ALOGV("purging stale effects");
1966
1967    Vector< sp<EffectChain> > chains;
1968
1969    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1970        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1971        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1972            sp<EffectChain> ec = t->mEffectChains[j];
1973            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1974                chains.push(ec);
1975            }
1976        }
1977    }
1978    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1979        sp<RecordThread> t = mRecordThreads.valueAt(i);
1980        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1981            sp<EffectChain> ec = t->mEffectChains[j];
1982            chains.push(ec);
1983        }
1984    }
1985
1986    for (size_t i = 0; i < chains.size(); i++) {
1987        sp<EffectChain> ec = chains[i];
1988        int sessionid = ec->sessionId();
1989        sp<ThreadBase> t = ec->mThread.promote();
1990        if (t == 0) {
1991            continue;
1992        }
1993        size_t numsessionrefs = mAudioSessionRefs.size();
1994        bool found = false;
1995        for (size_t k = 0; k < numsessionrefs; k++) {
1996            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1997            if (ref->mSessionid == sessionid) {
1998                ALOGV(" session %d still exists for %d with %d refs",
1999                    sessionid, ref->mPid, ref->mCnt);
2000                found = true;
2001                break;
2002            }
2003        }
2004        if (!found) {
2005            Mutex::Autolock _l(t->mLock);
2006            // remove all effects from the chain
2007            while (ec->mEffects.size()) {
2008                sp<EffectModule> effect = ec->mEffects[0];
2009                effect->unPin();
2010                t->removeEffect_l(effect);
2011                if (effect->purgeHandles()) {
2012                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2013                }
2014                AudioSystem::unregisterEffect(effect->id());
2015            }
2016        }
2017    }
2018    return;
2019}
2020
2021// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2022AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2023{
2024    return mPlaybackThreads.valueFor(output).get();
2025}
2026
2027// checkMixerThread_l() must be called with AudioFlinger::mLock held
2028AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2029{
2030    PlaybackThread *thread = checkPlaybackThread_l(output);
2031    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2032}
2033
2034// checkRecordThread_l() must be called with AudioFlinger::mLock held
2035AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2036{
2037    return mRecordThreads.valueFor(input).get();
2038}
2039
2040uint32_t AudioFlinger::nextUniqueId()
2041{
2042    return android_atomic_inc(&mNextUniqueId);
2043}
2044
2045AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2046{
2047    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2048        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2049        AudioStreamOut *output = thread->getOutput();
2050        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2051            return thread;
2052        }
2053    }
2054    return NULL;
2055}
2056
2057audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2058{
2059    PlaybackThread *thread = primaryPlaybackThread_l();
2060
2061    if (thread == NULL) {
2062        return 0;
2063    }
2064
2065    return thread->outDevice();
2066}
2067
2068sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2069                                    int triggerSession,
2070                                    int listenerSession,
2071                                    sync_event_callback_t callBack,
2072                                    void *cookie)
2073{
2074    Mutex::Autolock _l(mLock);
2075
2076    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2077    status_t playStatus = NAME_NOT_FOUND;
2078    status_t recStatus = NAME_NOT_FOUND;
2079    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2080        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2081        if (playStatus == NO_ERROR) {
2082            return event;
2083        }
2084    }
2085    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2086        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2087        if (recStatus == NO_ERROR) {
2088            return event;
2089        }
2090    }
2091    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2092        mPendingSyncEvents.add(event);
2093    } else {
2094        ALOGV("createSyncEvent() invalid event %d", event->type());
2095        event.clear();
2096    }
2097    return event;
2098}
2099
2100// ----------------------------------------------------------------------------
2101//  Effect management
2102// ----------------------------------------------------------------------------
2103
2104
2105status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2106{
2107    Mutex::Autolock _l(mLock);
2108    return EffectQueryNumberEffects(numEffects);
2109}
2110
2111status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2112{
2113    Mutex::Autolock _l(mLock);
2114    return EffectQueryEffect(index, descriptor);
2115}
2116
2117status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2118        effect_descriptor_t *descriptor) const
2119{
2120    Mutex::Autolock _l(mLock);
2121    return EffectGetDescriptor(pUuid, descriptor);
2122}
2123
2124
2125sp<IEffect> AudioFlinger::createEffect(
2126        effect_descriptor_t *pDesc,
2127        const sp<IEffectClient>& effectClient,
2128        int32_t priority,
2129        audio_io_handle_t io,
2130        int sessionId,
2131        status_t *status,
2132        int *id,
2133        int *enabled)
2134{
2135    status_t lStatus = NO_ERROR;
2136    sp<EffectHandle> handle;
2137    effect_descriptor_t desc;
2138
2139    pid_t pid = IPCThreadState::self()->getCallingPid();
2140    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2141            pid, effectClient.get(), priority, sessionId, io);
2142
2143    if (pDesc == NULL) {
2144        lStatus = BAD_VALUE;
2145        goto Exit;
2146    }
2147
2148    // check audio settings permission for global effects
2149    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2150        lStatus = PERMISSION_DENIED;
2151        goto Exit;
2152    }
2153
2154    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2155    // that can only be created by audio policy manager (running in same process)
2156    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2157        lStatus = PERMISSION_DENIED;
2158        goto Exit;
2159    }
2160
2161    {
2162        if (!EffectIsNullUuid(&pDesc->uuid)) {
2163            // if uuid is specified, request effect descriptor
2164            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2165            if (lStatus < 0) {
2166                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2167                goto Exit;
2168            }
2169        } else {
2170            // if uuid is not specified, look for an available implementation
2171            // of the required type in effect factory
2172            if (EffectIsNullUuid(&pDesc->type)) {
2173                ALOGW("createEffect() no effect type");
2174                lStatus = BAD_VALUE;
2175                goto Exit;
2176            }
2177            uint32_t numEffects = 0;
2178            effect_descriptor_t d;
2179            d.flags = 0; // prevent compiler warning
2180            bool found = false;
2181
2182            lStatus = EffectQueryNumberEffects(&numEffects);
2183            if (lStatus < 0) {
2184                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2185                goto Exit;
2186            }
2187            for (uint32_t i = 0; i < numEffects; i++) {
2188                lStatus = EffectQueryEffect(i, &desc);
2189                if (lStatus < 0) {
2190                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2191                    continue;
2192                }
2193                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2194                    // If matching type found save effect descriptor. If the session is
2195                    // 0 and the effect is not auxiliary, continue enumeration in case
2196                    // an auxiliary version of this effect type is available
2197                    found = true;
2198                    d = desc;
2199                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2200                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2201                        break;
2202                    }
2203                }
2204            }
2205            if (!found) {
2206                lStatus = BAD_VALUE;
2207                ALOGW("createEffect() effect not found");
2208                goto Exit;
2209            }
2210            // For same effect type, chose auxiliary version over insert version if
2211            // connect to output mix (Compliance to OpenSL ES)
2212            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2213                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2214                desc = d;
2215            }
2216        }
2217
2218        // Do not allow auxiliary effects on a session different from 0 (output mix)
2219        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2220             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2221            lStatus = INVALID_OPERATION;
2222            goto Exit;
2223        }
2224
2225        // check recording permission for visualizer
2226        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2227            !recordingAllowed()) {
2228            lStatus = PERMISSION_DENIED;
2229            goto Exit;
2230        }
2231
2232        // return effect descriptor
2233        *pDesc = desc;
2234        if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2235            // if the output returned by getOutputForEffect() is removed before we lock the
2236            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2237            // and we will exit safely
2238            io = AudioSystem::getOutputForEffect(&desc);
2239            ALOGV("createEffect got output %d", io);
2240        }
2241
2242        Mutex::Autolock _l(mLock);
2243
2244        // If output is not specified try to find a matching audio session ID in one of the
2245        // output threads.
2246        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2247        // because of code checking output when entering the function.
2248        // Note: io is never 0 when creating an effect on an input
2249        if (io == 0) {
2250            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2251                // output must be specified by AudioPolicyManager when using session
2252                // AUDIO_SESSION_OUTPUT_STAGE
2253                lStatus = BAD_VALUE;
2254                goto Exit;
2255            }
2256            // look for the thread where the specified audio session is present
2257            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2258                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2259                    io = mPlaybackThreads.keyAt(i);
2260                    break;
2261                }
2262            }
2263            if (io == 0) {
2264                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2265                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2266                        io = mRecordThreads.keyAt(i);
2267                        break;
2268                    }
2269                }
2270            }
2271            // If no output thread contains the requested session ID, default to
2272            // first output. The effect chain will be moved to the correct output
2273            // thread when a track with the same session ID is created
2274            if (io == 0 && mPlaybackThreads.size()) {
2275                io = mPlaybackThreads.keyAt(0);
2276            }
2277            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2278        }
2279        ThreadBase *thread = checkRecordThread_l(io);
2280        if (thread == NULL) {
2281            thread = checkPlaybackThread_l(io);
2282            if (thread == NULL) {
2283                ALOGE("createEffect() unknown output thread");
2284                lStatus = BAD_VALUE;
2285                goto Exit;
2286            }
2287        }
2288
2289        sp<Client> client = registerPid_l(pid);
2290
2291        // create effect on selected output thread
2292        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2293                &desc, enabled, &lStatus);
2294        if (handle != 0 && id != NULL) {
2295            *id = handle->id();
2296        }
2297    }
2298
2299Exit:
2300    *status = lStatus;
2301    return handle;
2302}
2303
2304status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2305        audio_io_handle_t dstOutput)
2306{
2307    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2308            sessionId, srcOutput, dstOutput);
2309    Mutex::Autolock _l(mLock);
2310    if (srcOutput == dstOutput) {
2311        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2312        return NO_ERROR;
2313    }
2314    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2315    if (srcThread == NULL) {
2316        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2317        return BAD_VALUE;
2318    }
2319    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2320    if (dstThread == NULL) {
2321        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2322        return BAD_VALUE;
2323    }
2324
2325    Mutex::Autolock _dl(dstThread->mLock);
2326    Mutex::Autolock _sl(srcThread->mLock);
2327    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2328}
2329
2330// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2331status_t AudioFlinger::moveEffectChain_l(int sessionId,
2332                                   AudioFlinger::PlaybackThread *srcThread,
2333                                   AudioFlinger::PlaybackThread *dstThread,
2334                                   bool reRegister)
2335{
2336    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2337            sessionId, srcThread, dstThread);
2338
2339    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2340    if (chain == 0) {
2341        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2342                sessionId, srcThread);
2343        return INVALID_OPERATION;
2344    }
2345
2346    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2347    // so that a new chain is created with correct parameters when first effect is added. This is
2348    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2349    // removed.
2350    srcThread->removeEffectChain_l(chain);
2351
2352    // transfer all effects one by one so that new effect chain is created on new thread with
2353    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2354    sp<EffectChain> dstChain;
2355    uint32_t strategy = 0; // prevent compiler warning
2356    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2357    Vector< sp<EffectModule> > removed;
2358    status_t status = NO_ERROR;
2359    while (effect != 0) {
2360        srcThread->removeEffect_l(effect);
2361        removed.add(effect);
2362        status = dstThread->addEffect_l(effect);
2363        if (status != NO_ERROR) {
2364            break;
2365        }
2366        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2367        if (effect->state() == EffectModule::ACTIVE ||
2368                effect->state() == EffectModule::STOPPING) {
2369            effect->start();
2370        }
2371        // if the move request is not received from audio policy manager, the effect must be
2372        // re-registered with the new strategy and output
2373        if (dstChain == 0) {
2374            dstChain = effect->chain().promote();
2375            if (dstChain == 0) {
2376                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2377                status = NO_INIT;
2378                break;
2379            }
2380            strategy = dstChain->strategy();
2381        }
2382        if (reRegister) {
2383            AudioSystem::unregisterEffect(effect->id());
2384            AudioSystem::registerEffect(&effect->desc(),
2385                                        dstThread->id(),
2386                                        strategy,
2387                                        sessionId,
2388                                        effect->id());
2389            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2390        }
2391        effect = chain->getEffectFromId_l(0);
2392    }
2393
2394    if (status != NO_ERROR) {
2395        for (size_t i = 0; i < removed.size(); i++) {
2396            srcThread->addEffect_l(removed[i]);
2397            if (dstChain != 0 && reRegister) {
2398                AudioSystem::unregisterEffect(removed[i]->id());
2399                AudioSystem::registerEffect(&removed[i]->desc(),
2400                                            srcThread->id(),
2401                                            strategy,
2402                                            sessionId,
2403                                            removed[i]->id());
2404                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2405            }
2406        }
2407    }
2408
2409    return status;
2410}
2411
2412bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2413{
2414    if (mGlobalEffectEnableTime != 0 &&
2415            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2416        return true;
2417    }
2418
2419    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2420        sp<EffectChain> ec =
2421                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2422        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2423            return true;
2424        }
2425    }
2426    return false;
2427}
2428
2429void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2430{
2431    Mutex::Autolock _l(mLock);
2432
2433    mGlobalEffectEnableTime = systemTime();
2434
2435    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2436        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2437        if (t->mType == ThreadBase::OFFLOAD) {
2438            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2439        }
2440    }
2441
2442}
2443
2444struct Entry {
2445#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2446    char mName[MAX_NAME];
2447};
2448
2449int comparEntry(const void *p1, const void *p2)
2450{
2451    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2452}
2453
2454#ifdef TEE_SINK
2455void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2456{
2457    NBAIO_Source *teeSource = source.get();
2458    if (teeSource != NULL) {
2459        // .wav rotation
2460        // There is a benign race condition if 2 threads call this simultaneously.
2461        // They would both traverse the directory, but the result would simply be
2462        // failures at unlink() which are ignored.  It's also unlikely since
2463        // normally dumpsys is only done by bugreport or from the command line.
2464        char teePath[32+256];
2465        strcpy(teePath, "/data/misc/media");
2466        size_t teePathLen = strlen(teePath);
2467        DIR *dir = opendir(teePath);
2468        teePath[teePathLen++] = '/';
2469        if (dir != NULL) {
2470#define MAX_SORT 20 // number of entries to sort
2471#define MAX_KEEP 10 // number of entries to keep
2472            struct Entry entries[MAX_SORT];
2473            size_t entryCount = 0;
2474            while (entryCount < MAX_SORT) {
2475                struct dirent de;
2476                struct dirent *result = NULL;
2477                int rc = readdir_r(dir, &de, &result);
2478                if (rc != 0) {
2479                    ALOGW("readdir_r failed %d", rc);
2480                    break;
2481                }
2482                if (result == NULL) {
2483                    break;
2484                }
2485                if (result != &de) {
2486                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2487                    break;
2488                }
2489                // ignore non .wav file entries
2490                size_t nameLen = strlen(de.d_name);
2491                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2492                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2493                    continue;
2494                }
2495                strcpy(entries[entryCount++].mName, de.d_name);
2496            }
2497            (void) closedir(dir);
2498            if (entryCount > MAX_KEEP) {
2499                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2500                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2501                    strcpy(&teePath[teePathLen], entries[i].mName);
2502                    (void) unlink(teePath);
2503                }
2504            }
2505        } else {
2506            if (fd >= 0) {
2507                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2508            }
2509        }
2510        char teeTime[16];
2511        struct timeval tv;
2512        gettimeofday(&tv, NULL);
2513        struct tm tm;
2514        localtime_r(&tv.tv_sec, &tm);
2515        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2516        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2517        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2518        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2519        if (teeFd >= 0) {
2520            char wavHeader[44];
2521            memcpy(wavHeader,
2522                "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",
2523                sizeof(wavHeader));
2524            NBAIO_Format format = teeSource->format();
2525            unsigned channelCount = Format_channelCount(format);
2526            ALOG_ASSERT(channelCount <= FCC_2);
2527            uint32_t sampleRate = Format_sampleRate(format);
2528            wavHeader[22] = channelCount;       // number of channels
2529            wavHeader[24] = sampleRate;         // sample rate
2530            wavHeader[25] = sampleRate >> 8;
2531            wavHeader[32] = channelCount * 2;   // block alignment
2532            write(teeFd, wavHeader, sizeof(wavHeader));
2533            size_t total = 0;
2534            bool firstRead = true;
2535            for (;;) {
2536#define TEE_SINK_READ 1024
2537                short buffer[TEE_SINK_READ * FCC_2];
2538                size_t count = TEE_SINK_READ;
2539                ssize_t actual = teeSource->read(buffer, count,
2540                        AudioBufferProvider::kInvalidPTS);
2541                bool wasFirstRead = firstRead;
2542                firstRead = false;
2543                if (actual <= 0) {
2544                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2545                        continue;
2546                    }
2547                    break;
2548                }
2549                ALOG_ASSERT(actual <= (ssize_t)count);
2550                write(teeFd, buffer, actual * channelCount * sizeof(short));
2551                total += actual;
2552            }
2553            lseek(teeFd, (off_t) 4, SEEK_SET);
2554            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2555            write(teeFd, &temp, sizeof(temp));
2556            lseek(teeFd, (off_t) 40, SEEK_SET);
2557            temp =  total * channelCount * sizeof(short);
2558            write(teeFd, &temp, sizeof(temp));
2559            close(teeFd);
2560            if (fd >= 0) {
2561                fdprintf(fd, "tee copied to %s\n", teePath);
2562            }
2563        } else {
2564            if (fd >= 0) {
2565                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2566            }
2567        }
2568    }
2569}
2570#endif
2571
2572// ----------------------------------------------------------------------------
2573
2574status_t AudioFlinger::onTransact(
2575        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2576{
2577    return BnAudioFlinger::onTransact(code, data, reply, flags);
2578}
2579
2580}; // namespace android
2581