AudioFlinger.cpp revision 4cb668392ee0433462251afbee109405c6efacc8 
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/AudioResamplerPublic.h>
49
50#include <media/EffectsFactoryApi.h>
51#include <audio_effects/effect_visualizer.h>
52#include <audio_effects/effect_ns.h>
53#include <audio_effects/effect_aec.h>
54
55#include <audio_utils/primitives.h>
56
57#include <powermanager/PowerManager.h>
58
59#include <common_time/cc_helper.h>
60
61#include <media/IMediaLogService.h>
62
63#include <media/nbaio/Pipe.h>
64#include <media/nbaio/PipeReader.h>
65#include <media/AudioParameter.h>
66#include <private/android_filesystem_config.h>
67
68// ----------------------------------------------------------------------------
69
70// Note: the following macro is used for extremely verbose logging message.  In
71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
72// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
73// are so verbose that we want to suppress them even when we have ALOG_ASSERT
74// turned on.  Do not uncomment the #def below unless you really know what you
75// are doing and want to see all of the extremely verbose messages.
76//#define VERY_VERY_VERBOSE_LOGGING
77#ifdef VERY_VERY_VERBOSE_LOGGING
78#define ALOGVV ALOGV
79#else
80#define ALOGVV(a...) do { } while(0)
81#endif
82
83namespace android {
84
85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
86static const char kHardwareLockedString[] = "Hardware lock is taken\n";
87static const char kClientLockedString[] = "Client lock is taken\n";
88
89
90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
91
92uint32_t AudioFlinger::mScreenState;
93
94#ifdef TEE_SINK
95bool AudioFlinger::mTeeSinkInputEnabled = false;
96bool AudioFlinger::mTeeSinkOutputEnabled = false;
97bool AudioFlinger::mTeeSinkTrackEnabled = false;
98
99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
102#endif
103
104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
105// we define a minimum time during which a global effect is considered enabled.
106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
107
108// ----------------------------------------------------------------------------
109
110const char *formatToString(audio_format_t format) {
111    switch (format & AUDIO_FORMAT_MAIN_MASK) {
112    case AUDIO_FORMAT_PCM:
113        switch (format) {
114        case AUDIO_FORMAT_PCM_16_BIT: return "pcm16";
115        case AUDIO_FORMAT_PCM_8_BIT: return "pcm8";
116        case AUDIO_FORMAT_PCM_32_BIT: return "pcm32";
117        case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24";
118        case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat";
119        case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24";
120        default:
121            break;
122        }
123        break;
124    case AUDIO_FORMAT_MP3: return "mp3";
125    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
126    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
127    case AUDIO_FORMAT_AAC: return "aac";
128    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
129    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
130    case AUDIO_FORMAT_VORBIS: return "vorbis";
131    case AUDIO_FORMAT_OPUS: return "opus";
132    case AUDIO_FORMAT_AC3: return "ac-3";
133    case AUDIO_FORMAT_E_AC3: return "e-ac-3";
134    default:
135        break;
136    }
137    return "unknown";
138}
139
140static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
141{
142    const hw_module_t *mod;
143    int rc;
144
145    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
146    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
147                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
148    if (rc) {
149        goto out;
150    }
151    rc = audio_hw_device_open(mod, dev);
152    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
153                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
154    if (rc) {
155        goto out;
156    }
157    if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
158        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
159        rc = BAD_VALUE;
160        goto out;
161    }
162    return 0;
163
164out:
165    *dev = NULL;
166    return rc;
167}
168
169// ----------------------------------------------------------------------------
170
171AudioFlinger::AudioFlinger()
172    : BnAudioFlinger(),
173      mPrimaryHardwareDev(NULL),
174      mAudioHwDevs(NULL),
175      mHardwareStatus(AUDIO_HW_IDLE),
176      mMasterVolume(1.0f),
177      mMasterMute(false),
178      mNextUniqueId(1),
179      mMode(AUDIO_MODE_INVALID),
180      mBtNrecIsOff(false),
181      mIsLowRamDevice(true),
182      mIsDeviceTypeKnown(false),
183      mGlobalEffectEnableTime(0),
184      mPrimaryOutputSampleRate(0)
185{
186    getpid_cached = getpid();
187    char value[PROPERTY_VALUE_MAX];
188    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
189    if (doLog) {
190        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
191                MemoryHeapBase::READ_ONLY);
192    }
193
194#ifdef TEE_SINK
195    (void) property_get("ro.debuggable", value, "0");
196    int debuggable = atoi(value);
197    int teeEnabled = 0;
198    if (debuggable) {
199        (void) property_get("af.tee", value, "0");
200        teeEnabled = atoi(value);
201    }
202    // FIXME symbolic constants here
203    if (teeEnabled & 1) {
204        mTeeSinkInputEnabled = true;
205    }
206    if (teeEnabled & 2) {
207        mTeeSinkOutputEnabled = true;
208    }
209    if (teeEnabled & 4) {
210        mTeeSinkTrackEnabled = true;
211    }
212#endif
213}
214
215void AudioFlinger::onFirstRef()
216{
217    int rc = 0;
218
219    Mutex::Autolock _l(mLock);
220
221    /* TODO: move all this work into an Init() function */
222    char val_str[PROPERTY_VALUE_MAX] = { 0 };
223    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
224        uint32_t int_val;
225        if (1 == sscanf(val_str, "%u", &int_val)) {
226            mStandbyTimeInNsecs = milliseconds(int_val);
227            ALOGI("Using %u mSec as standby time.", int_val);
228        } else {
229            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
230            ALOGI("Using default %u mSec as standby time.",
231                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
232        }
233    }
234
235    mPatchPanel = new PatchPanel(this);
236
237    mMode = AUDIO_MODE_NORMAL;
238}
239
240AudioFlinger::~AudioFlinger()
241{
242    while (!mRecordThreads.isEmpty()) {
243        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
244        closeInput_nonvirtual(mRecordThreads.keyAt(0));
245    }
246    while (!mPlaybackThreads.isEmpty()) {
247        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
248        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
249    }
250
251    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
252        // no mHardwareLock needed, as there are no other references to this
253        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
254        delete mAudioHwDevs.valueAt(i);
255    }
256
257    // Tell media.log service about any old writers that still need to be unregistered
258    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
259    if (binder != 0) {
260        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
261        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
262            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
263            mUnregisteredWriters.pop();
264            mediaLogService->unregisterWriter(iMemory);
265        }
266    }
267
268}
269
270static const char * const audio_interfaces[] = {
271    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
272    AUDIO_HARDWARE_MODULE_ID_A2DP,
273    AUDIO_HARDWARE_MODULE_ID_USB,
274};
275#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
276
277AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
278        audio_module_handle_t module,
279        audio_devices_t devices)
280{
281    // if module is 0, the request comes from an old policy manager and we should load
282    // well known modules
283    if (module == 0) {
284        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
285        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
286            loadHwModule_l(audio_interfaces[i]);
287        }
288        // then try to find a module supporting the requested device.
289        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
290            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
291            audio_hw_device_t *dev = audioHwDevice->hwDevice();
292            if ((dev->get_supported_devices != NULL) &&
293                    (dev->get_supported_devices(dev) & devices) == devices)
294                return audioHwDevice;
295        }
296    } else {
297        // check a match for the requested module handle
298        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
299        if (audioHwDevice != NULL) {
300            return audioHwDevice;
301        }
302    }
303
304    return NULL;
305}
306
307void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
308{
309    const size_t SIZE = 256;
310    char buffer[SIZE];
311    String8 result;
312
313    result.append("Clients:\n");
314    for (size_t i = 0; i < mClients.size(); ++i) {
315        sp<Client> client = mClients.valueAt(i).promote();
316        if (client != 0) {
317            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
318            result.append(buffer);
319        }
320    }
321
322    result.append("Notification Clients:\n");
323    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
324        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
325        result.append(buffer);
326    }
327
328    result.append("Global session refs:\n");
329    result.append("  session   pid count\n");
330    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
331        AudioSessionRef *r = mAudioSessionRefs[i];
332        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
333        result.append(buffer);
334    }
335    write(fd, result.string(), result.size());
336}
337
338
339void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
340{
341    const size_t SIZE = 256;
342    char buffer[SIZE];
343    String8 result;
344    hardware_call_state hardwareStatus = mHardwareStatus;
345
346    snprintf(buffer, SIZE, "Hardware status: %d\n"
347                           "Standby Time mSec: %u\n",
348                            hardwareStatus,
349                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
350    result.append(buffer);
351    write(fd, result.string(), result.size());
352}
353
354void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
355{
356    const size_t SIZE = 256;
357    char buffer[SIZE];
358    String8 result;
359    snprintf(buffer, SIZE, "Permission Denial: "
360            "can't dump AudioFlinger from pid=%d, uid=%d\n",
361            IPCThreadState::self()->getCallingPid(),
362            IPCThreadState::self()->getCallingUid());
363    result.append(buffer);
364    write(fd, result.string(), result.size());
365}
366
367bool AudioFlinger::dumpTryLock(Mutex& mutex)
368{
369    bool locked = false;
370    for (int i = 0; i < kDumpLockRetries; ++i) {
371        if (mutex.tryLock() == NO_ERROR) {
372            locked = true;
373            break;
374        }
375        usleep(kDumpLockSleepUs);
376    }
377    return locked;
378}
379
380status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
381{
382    if (!dumpAllowed()) {
383        dumpPermissionDenial(fd, args);
384    } else {
385        // get state of hardware lock
386        bool hardwareLocked = dumpTryLock(mHardwareLock);
387        if (!hardwareLocked) {
388            String8 result(kHardwareLockedString);
389            write(fd, result.string(), result.size());
390        } else {
391            mHardwareLock.unlock();
392        }
393
394        bool locked = dumpTryLock(mLock);
395
396        // failed to lock - AudioFlinger is probably deadlocked
397        if (!locked) {
398            String8 result(kDeadlockedString);
399            write(fd, result.string(), result.size());
400        }
401
402        bool clientLocked = dumpTryLock(mClientLock);
403        if (!clientLocked) {
404            String8 result(kClientLockedString);
405            write(fd, result.string(), result.size());
406        }
407
408        EffectDumpEffects(fd);
409
410        dumpClients(fd, args);
411        if (clientLocked) {
412            mClientLock.unlock();
413        }
414
415        dumpInternals(fd, args);
416
417        // dump playback threads
418        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
419            mPlaybackThreads.valueAt(i)->dump(fd, args);
420        }
421
422        // dump record threads
423        for (size_t i = 0; i < mRecordThreads.size(); i++) {
424            mRecordThreads.valueAt(i)->dump(fd, args);
425        }
426
427        // dump orphan effect chains
428        if (mOrphanEffectChains.size() != 0) {
429            write(fd, "  Orphan Effect Chains\n", strlen("  Orphan Effect Chains\n"));
430            for (size_t i = 0; i < mOrphanEffectChains.size(); i++) {
431                mOrphanEffectChains.valueAt(i)->dump(fd, args);
432            }
433        }
434        // dump all hardware devs
435        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
436            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
437            dev->dump(dev, fd);
438        }
439
440#ifdef TEE_SINK
441        // dump the serially shared record tee sink
442        if (mRecordTeeSource != 0) {
443            dumpTee(fd, mRecordTeeSource);
444        }
445#endif
446
447        if (locked) {
448            mLock.unlock();
449        }
450
451        // append a copy of media.log here by forwarding fd to it, but don't attempt
452        // to lookup the service if it's not running, as it will block for a second
453        if (mLogMemoryDealer != 0) {
454            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
455            if (binder != 0) {
456                dprintf(fd, "\nmedia.log:\n");
457                Vector<String16> args;
458                binder->dump(fd, args);
459            }
460        }
461    }
462    return NO_ERROR;
463}
464
465sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
466{
467    Mutex::Autolock _cl(mClientLock);
468    // If pid is already in the mClients wp<> map, then use that entry
469    // (for which promote() is always != 0), otherwise create a new entry and Client.
470    sp<Client> client = mClients.valueFor(pid).promote();
471    if (client == 0) {
472        client = new Client(this, pid);
473        mClients.add(pid, client);
474    }
475
476    return client;
477}
478
479sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
480{
481    // If there is no memory allocated for logs, return a dummy writer that does nothing
482    if (mLogMemoryDealer == 0) {
483        return new NBLog::Writer();
484    }
485    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
486    // Similarly if we can't contact the media.log service, also return a dummy writer
487    if (binder == 0) {
488        return new NBLog::Writer();
489    }
490    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
491    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
492    // If allocation fails, consult the vector of previously unregistered writers
493    // and garbage-collect one or more them until an allocation succeeds
494    if (shared == 0) {
495        Mutex::Autolock _l(mUnregisteredWritersLock);
496        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
497            {
498                // Pick the oldest stale writer to garbage-collect
499                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
500                mUnregisteredWriters.removeAt(0);
501                mediaLogService->unregisterWriter(iMemory);
502                // Now the media.log remote reference to IMemory is gone.  When our last local
503                // reference to IMemory also drops to zero at end of this block,
504                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
505            }
506            // Re-attempt the allocation
507            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
508            if (shared != 0) {
509                goto success;
510            }
511        }
512        // Even after garbage-collecting all old writers, there is still not enough memory,
513        // so return a dummy writer
514        return new NBLog::Writer();
515    }
516success:
517    mediaLogService->registerWriter(shared, size, name);
518    return new NBLog::Writer(size, shared);
519}
520
521void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
522{
523    if (writer == 0) {
524        return;
525    }
526    sp<IMemory> iMemory(writer->getIMemory());
527    if (iMemory == 0) {
528        return;
529    }
530    // Rather than removing the writer immediately, append it to a queue of old writers to
531    // be garbage-collected later.  This allows us to continue to view old logs for a while.
532    Mutex::Autolock _l(mUnregisteredWritersLock);
533    mUnregisteredWriters.push(writer);
534}
535
536// IAudioFlinger interface
537
538
539sp<IAudioTrack> AudioFlinger::createTrack(
540        audio_stream_type_t streamType,
541        uint32_t sampleRate,
542        audio_format_t format,
543        audio_channel_mask_t channelMask,
544        size_t *frameCount,
545        IAudioFlinger::track_flags_t *flags,
546        const sp<IMemory>& sharedBuffer,
547        audio_io_handle_t output,
548        pid_t tid,
549        int *sessionId,
550        int clientUid,
551        status_t *status)
552{
553    sp<PlaybackThread::Track> track;
554    sp<TrackHandle> trackHandle;
555    sp<Client> client;
556    status_t lStatus;
557    int lSessionId;
558
559    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
560    // but if someone uses binder directly they could bypass that and cause us to crash
561    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
562        ALOGE("createTrack() invalid stream type %d", streamType);
563        lStatus = BAD_VALUE;
564        goto Exit;
565    }
566
567    // further sample rate checks are performed by createTrack_l() depending on the thread type
568    if (sampleRate == 0) {
569        ALOGE("createTrack() invalid sample rate %u", sampleRate);
570        lStatus = BAD_VALUE;
571        goto Exit;
572    }
573
574    // further channel mask checks are performed by createTrack_l() depending on the thread type
575    if (!audio_is_output_channel(channelMask)) {
576        ALOGE("createTrack() invalid channel mask %#x", channelMask);
577        lStatus = BAD_VALUE;
578        goto Exit;
579    }
580
581    // further format checks are performed by createTrack_l() depending on the thread type
582    if (!audio_is_valid_format(format)) {
583        ALOGE("createTrack() invalid format %#x", format);
584        lStatus = BAD_VALUE;
585        goto Exit;
586    }
587
588    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
589        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
590        lStatus = BAD_VALUE;
591        goto Exit;
592    }
593
594    {
595        Mutex::Autolock _l(mLock);
596        PlaybackThread *thread = checkPlaybackThread_l(output);
597        if (thread == NULL) {
598            ALOGE("no playback thread found for output handle %d", output);
599            lStatus = BAD_VALUE;
600            goto Exit;
601        }
602
603        pid_t pid = IPCThreadState::self()->getCallingPid();
604        client = registerPid(pid);
605
606        PlaybackThread *effectThread = NULL;
607        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
608            lSessionId = *sessionId;
609            // check if an effect chain with the same session ID is present on another
610            // output thread and move it here.
611            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
612                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
613                if (mPlaybackThreads.keyAt(i) != output) {
614                    uint32_t sessions = t->hasAudioSession(lSessionId);
615                    if (sessions & PlaybackThread::EFFECT_SESSION) {
616                        effectThread = t.get();
617                        break;
618                    }
619                }
620            }
621        } else {
622            // if no audio session id is provided, create one here
623            lSessionId = nextUniqueId();
624            if (sessionId != NULL) {
625                *sessionId = lSessionId;
626            }
627        }
628        ALOGV("createTrack() lSessionId: %d", lSessionId);
629
630        track = thread->createTrack_l(client, streamType, sampleRate, format,
631                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
632        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
633        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
634
635        // move effect chain to this output thread if an effect on same session was waiting
636        // for a track to be created
637        if (lStatus == NO_ERROR && effectThread != NULL) {
638            // no risk of deadlock because AudioFlinger::mLock is held
639            Mutex::Autolock _dl(thread->mLock);
640            Mutex::Autolock _sl(effectThread->mLock);
641            moveEffectChain_l(lSessionId, effectThread, thread, true);
642        }
643
644        // Look for sync events awaiting for a session to be used.
645        for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
646            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
647                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
648                    if (lStatus == NO_ERROR) {
649                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
650                    } else {
651                        mPendingSyncEvents[i]->cancel();
652                    }
653                    mPendingSyncEvents.removeAt(i);
654                    i--;
655                }
656            }
657        }
658
659        setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId);
660    }
661
662    if (lStatus != NO_ERROR) {
663        // remove local strong reference to Client before deleting the Track so that the
664        // Client destructor is called by the TrackBase destructor with mClientLock held
665        // Don't hold mClientLock when releasing the reference on the track as the
666        // destructor will acquire it.
667        {
668            Mutex::Autolock _cl(mClientLock);
669            client.clear();
670        }
671        track.clear();
672        goto Exit;
673    }
674
675    // return handle to client
676    trackHandle = new TrackHandle(track);
677
678Exit:
679    *status = lStatus;
680    return trackHandle;
681}
682
683uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
684{
685    Mutex::Autolock _l(mLock);
686    PlaybackThread *thread = checkPlaybackThread_l(output);
687    if (thread == NULL) {
688        ALOGW("sampleRate() unknown thread %d", output);
689        return 0;
690    }
691    return thread->sampleRate();
692}
693
694audio_format_t AudioFlinger::format(audio_io_handle_t output) const
695{
696    Mutex::Autolock _l(mLock);
697    PlaybackThread *thread = checkPlaybackThread_l(output);
698    if (thread == NULL) {
699        ALOGW("format() unknown thread %d", output);
700        return AUDIO_FORMAT_INVALID;
701    }
702    return thread->format();
703}
704
705size_t AudioFlinger::frameCount(audio_io_handle_t output) const
706{
707    Mutex::Autolock _l(mLock);
708    PlaybackThread *thread = checkPlaybackThread_l(output);
709    if (thread == NULL) {
710        ALOGW("frameCount() unknown thread %d", output);
711        return 0;
712    }
713    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
714    //       should examine all callers and fix them to handle smaller counts
715    return thread->frameCount();
716}
717
718uint32_t AudioFlinger::latency(audio_io_handle_t output) const
719{
720    Mutex::Autolock _l(mLock);
721    PlaybackThread *thread = checkPlaybackThread_l(output);
722    if (thread == NULL) {
723        ALOGW("latency(): no playback thread found for output handle %d", output);
724        return 0;
725    }
726    return thread->latency();
727}
728
729status_t AudioFlinger::setMasterVolume(float value)
730{
731    status_t ret = initCheck();
732    if (ret != NO_ERROR) {
733        return ret;
734    }
735
736    // check calling permissions
737    if (!settingsAllowed()) {
738        return PERMISSION_DENIED;
739    }
740
741    Mutex::Autolock _l(mLock);
742    mMasterVolume = value;
743
744    // Set master volume in the HALs which support it.
745    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
746        AutoMutex lock(mHardwareLock);
747        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
748
749        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
750        if (dev->canSetMasterVolume()) {
751            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
752        }
753        mHardwareStatus = AUDIO_HW_IDLE;
754    }
755
756    // Now set the master volume in each playback thread.  Playback threads
757    // assigned to HALs which do not have master volume support will apply
758    // master volume during the mix operation.  Threads with HALs which do
759    // support master volume will simply ignore the setting.
760    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
761        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
762
763    return NO_ERROR;
764}
765
766status_t AudioFlinger::setMode(audio_mode_t mode)
767{
768    status_t ret = initCheck();
769    if (ret != NO_ERROR) {
770        return ret;
771    }
772
773    // check calling permissions
774    if (!settingsAllowed()) {
775        return PERMISSION_DENIED;
776    }
777    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
778        ALOGW("Illegal value: setMode(%d)", mode);
779        return BAD_VALUE;
780    }
781
782    { // scope for the lock
783        AutoMutex lock(mHardwareLock);
784        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
785        mHardwareStatus = AUDIO_HW_SET_MODE;
786        ret = dev->set_mode(dev, mode);
787        mHardwareStatus = AUDIO_HW_IDLE;
788    }
789
790    if (NO_ERROR == ret) {
791        Mutex::Autolock _l(mLock);
792        mMode = mode;
793        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
794            mPlaybackThreads.valueAt(i)->setMode(mode);
795    }
796
797    return ret;
798}
799
800status_t AudioFlinger::setMicMute(bool state)
801{
802    status_t ret = initCheck();
803    if (ret != NO_ERROR) {
804        return ret;
805    }
806
807    // check calling permissions
808    if (!settingsAllowed()) {
809        return PERMISSION_DENIED;
810    }
811
812    AutoMutex lock(mHardwareLock);
813    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
814    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
815        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
816        status_t result = dev->set_mic_mute(dev, state);
817        if (result != NO_ERROR) {
818            ret = result;
819        }
820    }
821    mHardwareStatus = AUDIO_HW_IDLE;
822    return ret;
823}
824
825bool AudioFlinger::getMicMute() const
826{
827    status_t ret = initCheck();
828    if (ret != NO_ERROR) {
829        return false;
830    }
831    bool mute = true;
832    bool state = AUDIO_MODE_INVALID;
833    AutoMutex lock(mHardwareLock);
834    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
835    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
836        audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
837        status_t result = dev->get_mic_mute(dev, &state);
838        if (result == NO_ERROR) {
839            mute = mute && state;
840        }
841    }
842    mHardwareStatus = AUDIO_HW_IDLE;
843
844    return mute;
845}
846
847status_t AudioFlinger::setMasterMute(bool muted)
848{
849    status_t ret = initCheck();
850    if (ret != NO_ERROR) {
851        return ret;
852    }
853
854    // check calling permissions
855    if (!settingsAllowed()) {
856        return PERMISSION_DENIED;
857    }
858
859    Mutex::Autolock _l(mLock);
860    mMasterMute = muted;
861
862    // Set master mute in the HALs which support it.
863    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
864        AutoMutex lock(mHardwareLock);
865        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
866
867        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
868        if (dev->canSetMasterMute()) {
869            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
870        }
871        mHardwareStatus = AUDIO_HW_IDLE;
872    }
873
874    // Now set the master mute in each playback thread.  Playback threads
875    // assigned to HALs which do not have master mute support will apply master
876    // mute during the mix operation.  Threads with HALs which do support master
877    // mute will simply ignore the setting.
878    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
879        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
880
881    return NO_ERROR;
882}
883
884float AudioFlinger::masterVolume() const
885{
886    Mutex::Autolock _l(mLock);
887    return masterVolume_l();
888}
889
890bool AudioFlinger::masterMute() const
891{
892    Mutex::Autolock _l(mLock);
893    return masterMute_l();
894}
895
896float AudioFlinger::masterVolume_l() const
897{
898    return mMasterVolume;
899}
900
901bool AudioFlinger::masterMute_l() const
902{
903    return mMasterMute;
904}
905
906status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const
907{
908    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
909        ALOGW("setStreamVolume() invalid stream %d", stream);
910        return BAD_VALUE;
911    }
912    pid_t caller = IPCThreadState::self()->getCallingPid();
913    if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) {
914        ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream);
915        return PERMISSION_DENIED;
916    }
917
918    return NO_ERROR;
919}
920
921status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
922        audio_io_handle_t output)
923{
924    // check calling permissions
925    if (!settingsAllowed()) {
926        return PERMISSION_DENIED;
927    }
928
929    status_t status = checkStreamType(stream);
930    if (status != NO_ERROR) {
931        return status;
932    }
933    ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume");
934
935    AutoMutex lock(mLock);
936    PlaybackThread *thread = NULL;
937    if (output != AUDIO_IO_HANDLE_NONE) {
938        thread = checkPlaybackThread_l(output);
939        if (thread == NULL) {
940            return BAD_VALUE;
941        }
942    }
943
944    mStreamTypes[stream].volume = value;
945
946    if (thread == NULL) {
947        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
948            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
949        }
950    } else {
951        thread->setStreamVolume(stream, value);
952    }
953
954    return NO_ERROR;
955}
956
957status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
958{
959    // check calling permissions
960    if (!settingsAllowed()) {
961        return PERMISSION_DENIED;
962    }
963
964    status_t status = checkStreamType(stream);
965    if (status != NO_ERROR) {
966        return status;
967    }
968    ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH");
969
970    if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
971        ALOGE("setStreamMute() invalid stream %d", stream);
972        return BAD_VALUE;
973    }
974
975    AutoMutex lock(mLock);
976    mStreamTypes[stream].mute = muted;
977    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
978        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
979
980    return NO_ERROR;
981}
982
983float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
984{
985    status_t status = checkStreamType(stream);
986    if (status != NO_ERROR) {
987        return 0.0f;
988    }
989
990    AutoMutex lock(mLock);
991    float volume;
992    if (output != AUDIO_IO_HANDLE_NONE) {
993        PlaybackThread *thread = checkPlaybackThread_l(output);
994        if (thread == NULL) {
995            return 0.0f;
996        }
997        volume = thread->streamVolume(stream);
998    } else {
999        volume = streamVolume_l(stream);
1000    }
1001
1002    return volume;
1003}
1004
1005bool AudioFlinger::streamMute(audio_stream_type_t stream) const
1006{
1007    status_t status = checkStreamType(stream);
1008    if (status != NO_ERROR) {
1009        return true;
1010    }
1011
1012    AutoMutex lock(mLock);
1013    return streamMute_l(stream);
1014}
1015
1016
1017void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs)
1018{
1019    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1020        mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1021    }
1022}
1023
1024status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
1025{
1026    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
1027            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
1028
1029    // check calling permissions
1030    if (!settingsAllowed()) {
1031        return PERMISSION_DENIED;
1032    }
1033
1034    // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
1035    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1036        Mutex::Autolock _l(mLock);
1037        status_t final_result = NO_ERROR;
1038        {
1039            AutoMutex lock(mHardwareLock);
1040            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
1041            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1042                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1043                status_t result = dev->set_parameters(dev, keyValuePairs.string());
1044                final_result = result ?: final_result;
1045            }
1046            mHardwareStatus = AUDIO_HW_IDLE;
1047        }
1048        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
1049        AudioParameter param = AudioParameter(keyValuePairs);
1050        String8 value;
1051        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
1052            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
1053            if (mBtNrecIsOff != btNrecIsOff) {
1054                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1055                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
1056                    audio_devices_t device = thread->inDevice();
1057                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
1058                    // collect all of the thread's session IDs
1059                    KeyedVector<int, bool> ids = thread->sessionIds();
1060                    // suspend effects associated with those session IDs
1061                    for (size_t j = 0; j < ids.size(); ++j) {
1062                        int sessionId = ids.keyAt(j);
1063                        thread->setEffectSuspended(FX_IID_AEC,
1064                                                   suspend,
1065                                                   sessionId);
1066                        thread->setEffectSuspended(FX_IID_NS,
1067                                                   suspend,
1068                                                   sessionId);
1069                    }
1070                }
1071                mBtNrecIsOff = btNrecIsOff;
1072            }
1073        }
1074        String8 screenState;
1075        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1076            bool isOff = screenState == "off";
1077            if (isOff != (AudioFlinger::mScreenState & 1)) {
1078                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1079            }
1080        }
1081        return final_result;
1082    }
1083
1084    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1085    // and the thread is exited once the lock is released
1086    sp<ThreadBase> thread;
1087    {
1088        Mutex::Autolock _l(mLock);
1089        thread = checkPlaybackThread_l(ioHandle);
1090        if (thread == 0) {
1091            thread = checkRecordThread_l(ioHandle);
1092        } else if (thread == primaryPlaybackThread_l()) {
1093            // indicate output device change to all input threads for pre processing
1094            AudioParameter param = AudioParameter(keyValuePairs);
1095            int value;
1096            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1097                    (value != 0)) {
1098                broacastParametersToRecordThreads_l(keyValuePairs);
1099            }
1100        }
1101    }
1102    if (thread != 0) {
1103        return thread->setParameters(keyValuePairs);
1104    }
1105    return BAD_VALUE;
1106}
1107
1108String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1109{
1110    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1111            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1112
1113    Mutex::Autolock _l(mLock);
1114
1115    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1116        String8 out_s8;
1117
1118        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1119            char *s;
1120            {
1121            AutoMutex lock(mHardwareLock);
1122            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1123            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1124            s = dev->get_parameters(dev, keys.string());
1125            mHardwareStatus = AUDIO_HW_IDLE;
1126            }
1127            out_s8 += String8(s ? s : "");
1128            free(s);
1129        }
1130        return out_s8;
1131    }
1132
1133    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1134    if (playbackThread != NULL) {
1135        return playbackThread->getParameters(keys);
1136    }
1137    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1138    if (recordThread != NULL) {
1139        return recordThread->getParameters(keys);
1140    }
1141    return String8("");
1142}
1143
1144size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1145        audio_channel_mask_t channelMask) const
1146{
1147    status_t ret = initCheck();
1148    if (ret != NO_ERROR) {
1149        return 0;
1150    }
1151    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
1152        return 0;
1153    }
1154
1155    AutoMutex lock(mHardwareLock);
1156    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1157    audio_config_t config, proposed;
1158    memset(&proposed, 0, sizeof(proposed));
1159    proposed.sample_rate = sampleRate;
1160    proposed.channel_mask = channelMask;
1161    proposed.format = format;
1162
1163    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1164    size_t frames;
1165    for (;;) {
1166        // Note: config is currently a const parameter for get_input_buffer_size()
1167        // but we use a copy from proposed in case config changes from the call.
1168        config = proposed;
1169        frames = dev->get_input_buffer_size(dev, &config);
1170        if (frames != 0) {
1171            break; // hal success, config is the result
1172        }
1173        // change one parameter of the configuration each iteration to a more "common" value
1174        // to see if the device will support it.
1175        if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
1176            proposed.format = AUDIO_FORMAT_PCM_16_BIT;
1177        } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
1178            proposed.sample_rate = 44100;           // legacy AudioRecord.java. TODO: Query hw?
1179        } else {
1180            ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
1181                    "format %#x, channelMask 0x%X",
1182                    sampleRate, format, channelMask);
1183            break; // retries failed, break out of loop with frames == 0.
1184        }
1185    }
1186    mHardwareStatus = AUDIO_HW_IDLE;
1187    if (frames > 0 && config.sample_rate != sampleRate) {
1188        frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
1189    }
1190    return frames; // may be converted to bytes at the Java level.
1191}
1192
1193uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1194{
1195    Mutex::Autolock _l(mLock);
1196
1197    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1198    if (recordThread != NULL) {
1199        return recordThread->getInputFramesLost();
1200    }
1201    return 0;
1202}
1203
1204status_t AudioFlinger::setVoiceVolume(float value)
1205{
1206    status_t ret = initCheck();
1207    if (ret != NO_ERROR) {
1208        return ret;
1209    }
1210
1211    // check calling permissions
1212    if (!settingsAllowed()) {
1213        return PERMISSION_DENIED;
1214    }
1215
1216    AutoMutex lock(mHardwareLock);
1217    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1218    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1219    ret = dev->set_voice_volume(dev, value);
1220    mHardwareStatus = AUDIO_HW_IDLE;
1221
1222    return ret;
1223}
1224
1225status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1226        audio_io_handle_t output) const
1227{
1228    status_t status;
1229
1230    Mutex::Autolock _l(mLock);
1231
1232    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1233    if (playbackThread != NULL) {
1234        return playbackThread->getRenderPosition(halFrames, dspFrames);
1235    }
1236
1237    return BAD_VALUE;
1238}
1239
1240void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1241{
1242    Mutex::Autolock _l(mLock);
1243    if (client == 0) {
1244        return;
1245    }
1246    bool clientAdded = false;
1247    {
1248        Mutex::Autolock _cl(mClientLock);
1249
1250        pid_t pid = IPCThreadState::self()->getCallingPid();
1251        if (mNotificationClients.indexOfKey(pid) < 0) {
1252            sp<NotificationClient> notificationClient = new NotificationClient(this,
1253                                                                                client,
1254                                                                                pid);
1255            ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1256
1257            mNotificationClients.add(pid, notificationClient);
1258
1259            sp<IBinder> binder = IInterface::asBinder(client);
1260            binder->linkToDeath(notificationClient);
1261            clientAdded = true;
1262        }
1263    }
1264
1265    // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
1266    // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock.
1267    if (clientAdded) {
1268        // the config change is always sent from playback or record threads to avoid deadlock
1269        // with AudioSystem::gLock
1270        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1271            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1272        }
1273
1274        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1275            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1276        }
1277    }
1278}
1279
1280void AudioFlinger::removeNotificationClient(pid_t pid)
1281{
1282    Mutex::Autolock _l(mLock);
1283    {
1284        Mutex::Autolock _cl(mClientLock);
1285        mNotificationClients.removeItem(pid);
1286    }
1287
1288    ALOGV("%d died, releasing its sessions", pid);
1289    size_t num = mAudioSessionRefs.size();
1290    bool removed = false;
1291    for (size_t i = 0; i< num; ) {
1292        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1293        ALOGV(" pid %d @ %d", ref->mPid, i);
1294        if (ref->mPid == pid) {
1295            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1296            mAudioSessionRefs.removeAt(i);
1297            delete ref;
1298            removed = true;
1299            num--;
1300        } else {
1301            i++;
1302        }
1303    }
1304    if (removed) {
1305        purgeStaleEffects_l();
1306    }
1307}
1308
1309void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2)
1310{
1311    Mutex::Autolock _l(mClientLock);
1312    size_t size = mNotificationClients.size();
1313    for (size_t i = 0; i < size; i++) {
1314        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event,
1315                                                                              ioHandle,
1316                                                                              param2);
1317    }
1318}
1319
1320// removeClient_l() must be called with AudioFlinger::mClientLock held
1321void AudioFlinger::removeClient_l(pid_t pid)
1322{
1323    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1324            IPCThreadState::self()->getCallingPid());
1325    mClients.removeItem(pid);
1326}
1327
1328// getEffectThread_l() must be called with AudioFlinger::mLock held
1329sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1330{
1331    sp<PlaybackThread> thread;
1332
1333    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1334        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1335            ALOG_ASSERT(thread == 0);
1336            thread = mPlaybackThreads.valueAt(i);
1337        }
1338    }
1339
1340    return thread;
1341}
1342
1343
1344
1345// ----------------------------------------------------------------------------
1346
1347AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1348    :   RefBase(),
1349        mAudioFlinger(audioFlinger),
1350        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1351        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1352        mPid(pid),
1353        mTimedTrackCount(0)
1354{
1355    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1356}
1357
1358// Client destructor must be called with AudioFlinger::mClientLock held
1359AudioFlinger::Client::~Client()
1360{
1361    mAudioFlinger->removeClient_l(mPid);
1362}
1363
1364sp<MemoryDealer> AudioFlinger::Client::heap() const
1365{
1366    return mMemoryDealer;
1367}
1368
1369// Reserve one of the limited slots for a timed audio track associated
1370// with this client
1371bool AudioFlinger::Client::reserveTimedTrack()
1372{
1373    const int kMaxTimedTracksPerClient = 4;
1374
1375    Mutex::Autolock _l(mTimedTrackLock);
1376
1377    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1378        ALOGW("can not create timed track - pid %d has exceeded the limit",
1379             mPid);
1380        return false;
1381    }
1382
1383    mTimedTrackCount++;
1384    return true;
1385}
1386
1387// Release a slot for a timed audio track
1388void AudioFlinger::Client::releaseTimedTrack()
1389{
1390    Mutex::Autolock _l(mTimedTrackLock);
1391    mTimedTrackCount--;
1392}
1393
1394// ----------------------------------------------------------------------------
1395
1396AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1397                                                     const sp<IAudioFlingerClient>& client,
1398                                                     pid_t pid)
1399    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1400{
1401}
1402
1403AudioFlinger::NotificationClient::~NotificationClient()
1404{
1405}
1406
1407void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1408{
1409    sp<NotificationClient> keep(this);
1410    mAudioFlinger->removeNotificationClient(mPid);
1411}
1412
1413
1414// ----------------------------------------------------------------------------
1415
1416static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1417    return audio_is_remote_submix_device(inDevice);
1418}
1419
1420sp<IAudioRecord> AudioFlinger::openRecord(
1421        audio_io_handle_t input,
1422        uint32_t sampleRate,
1423        audio_format_t format,
1424        audio_channel_mask_t channelMask,
1425        const String16& opPackageName,
1426        size_t *frameCount,
1427        IAudioFlinger::track_flags_t *flags,
1428        pid_t tid,
1429        int clientUid,
1430        int *sessionId,
1431        size_t *notificationFrames,
1432        sp<IMemory>& cblk,
1433        sp<IMemory>& buffers,
1434        status_t *status)
1435{
1436    sp<RecordThread::RecordTrack> recordTrack;
1437    sp<RecordHandle> recordHandle;
1438    sp<Client> client;
1439    status_t lStatus;
1440    int lSessionId;
1441
1442    cblk.clear();
1443    buffers.clear();
1444
1445    // check calling permissions
1446    if (!recordingAllowed(opPackageName)) {
1447        ALOGE("openRecord() permission denied: recording not allowed");
1448        lStatus = PERMISSION_DENIED;
1449        goto Exit;
1450    }
1451
1452    // further sample rate checks are performed by createRecordTrack_l()
1453    if (sampleRate == 0) {
1454        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1455        lStatus = BAD_VALUE;
1456        goto Exit;
1457    }
1458
1459    // we don't yet support anything other than linear PCM
1460    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
1461        ALOGE("openRecord() invalid format %#x", format);
1462        lStatus = BAD_VALUE;
1463        goto Exit;
1464    }
1465
1466    // further channel mask checks are performed by createRecordTrack_l()
1467    if (!audio_is_input_channel(channelMask)) {
1468        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1469        lStatus = BAD_VALUE;
1470        goto Exit;
1471    }
1472
1473    {
1474        Mutex::Autolock _l(mLock);
1475        RecordThread *thread = checkRecordThread_l(input);
1476        if (thread == NULL) {
1477            ALOGE("openRecord() checkRecordThread_l failed");
1478            lStatus = BAD_VALUE;
1479            goto Exit;
1480        }
1481
1482        pid_t pid = IPCThreadState::self()->getCallingPid();
1483        client = registerPid(pid);
1484
1485        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1486            lSessionId = *sessionId;
1487        } else {
1488            // if no audio session id is provided, create one here
1489            lSessionId = nextUniqueId();
1490            if (sessionId != NULL) {
1491                *sessionId = lSessionId;
1492            }
1493        }
1494        ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input);
1495
1496        // TODO: the uid should be passed in as a parameter to openRecord
1497        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1498                                                  frameCount, lSessionId, notificationFrames,
1499                                                  clientUid, flags, tid, &lStatus);
1500        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1501
1502        if (lStatus == NO_ERROR) {
1503            // Check if one effect chain was awaiting for an AudioRecord to be created on this
1504            // session and move it to this thread.
1505            sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId);
1506            if (chain != 0) {
1507                Mutex::Autolock _l(thread->mLock);
1508                thread->addEffectChain_l(chain);
1509            }
1510        }
1511    }
1512
1513    if (lStatus != NO_ERROR) {
1514        // remove local strong reference to Client before deleting the RecordTrack so that the
1515        // Client destructor is called by the TrackBase destructor with mClientLock held
1516        // Don't hold mClientLock when releasing the reference on the track as the
1517        // destructor will acquire it.
1518        {
1519            Mutex::Autolock _cl(mClientLock);
1520            client.clear();
1521        }
1522        recordTrack.clear();
1523        goto Exit;
1524    }
1525
1526    cblk = recordTrack->getCblk();
1527    buffers = recordTrack->getBuffers();
1528
1529    // return handle to client
1530    recordHandle = new RecordHandle(recordTrack);
1531
1532Exit:
1533    *status = lStatus;
1534    return recordHandle;
1535}
1536
1537
1538
1539// ----------------------------------------------------------------------------
1540
1541audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1542{
1543    if (name == NULL) {
1544        return 0;
1545    }
1546    if (!settingsAllowed()) {
1547        return 0;
1548    }
1549    Mutex::Autolock _l(mLock);
1550    return loadHwModule_l(name);
1551}
1552
1553// loadHwModule_l() must be called with AudioFlinger::mLock held
1554audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1555{
1556    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1557        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1558            ALOGW("loadHwModule() module %s already loaded", name);
1559            return mAudioHwDevs.keyAt(i);
1560        }
1561    }
1562
1563    audio_hw_device_t *dev;
1564
1565    int rc = load_audio_interface(name, &dev);
1566    if (rc) {
1567        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1568        return 0;
1569    }
1570
1571    mHardwareStatus = AUDIO_HW_INIT;
1572    rc = dev->init_check(dev);
1573    mHardwareStatus = AUDIO_HW_IDLE;
1574    if (rc) {
1575        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1576        return 0;
1577    }
1578
1579    // Check and cache this HAL's level of support for master mute and master
1580    // volume.  If this is the first HAL opened, and it supports the get
1581    // methods, use the initial values provided by the HAL as the current
1582    // master mute and volume settings.
1583
1584    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1585    {  // scope for auto-lock pattern
1586        AutoMutex lock(mHardwareLock);
1587
1588        if (0 == mAudioHwDevs.size()) {
1589            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1590            if (NULL != dev->get_master_volume) {
1591                float mv;
1592                if (OK == dev->get_master_volume(dev, &mv)) {
1593                    mMasterVolume = mv;
1594                }
1595            }
1596
1597            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1598            if (NULL != dev->get_master_mute) {
1599                bool mm;
1600                if (OK == dev->get_master_mute(dev, &mm)) {
1601                    mMasterMute = mm;
1602                }
1603            }
1604        }
1605
1606        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1607        if ((NULL != dev->set_master_volume) &&
1608            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1609            flags = static_cast<AudioHwDevice::Flags>(flags |
1610                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1611        }
1612
1613        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1614        if ((NULL != dev->set_master_mute) &&
1615            (OK == dev->set_master_mute(dev, mMasterMute))) {
1616            flags = static_cast<AudioHwDevice::Flags>(flags |
1617                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1618        }
1619
1620        mHardwareStatus = AUDIO_HW_IDLE;
1621    }
1622
1623    audio_module_handle_t handle = nextUniqueId();
1624    mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags));
1625
1626    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1627          name, dev->common.module->name, dev->common.module->id, handle);
1628
1629    return handle;
1630
1631}
1632
1633// ----------------------------------------------------------------------------
1634
1635uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1636{
1637    Mutex::Autolock _l(mLock);
1638    PlaybackThread *thread = primaryPlaybackThread_l();
1639    return thread != NULL ? thread->sampleRate() : 0;
1640}
1641
1642size_t AudioFlinger::getPrimaryOutputFrameCount()
1643{
1644    Mutex::Autolock _l(mLock);
1645    PlaybackThread *thread = primaryPlaybackThread_l();
1646    return thread != NULL ? thread->frameCountHAL() : 0;
1647}
1648
1649// ----------------------------------------------------------------------------
1650
1651status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1652{
1653    uid_t uid = IPCThreadState::self()->getCallingUid();
1654    if (uid != AID_SYSTEM) {
1655        return PERMISSION_DENIED;
1656    }
1657    Mutex::Autolock _l(mLock);
1658    if (mIsDeviceTypeKnown) {
1659        return INVALID_OPERATION;
1660    }
1661    mIsLowRamDevice = isLowRamDevice;
1662    mIsDeviceTypeKnown = true;
1663    return NO_ERROR;
1664}
1665
1666audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId)
1667{
1668    Mutex::Autolock _l(mLock);
1669
1670    ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
1671    if (index >= 0) {
1672        ALOGV("getAudioHwSyncForSession found ID %d for session %d",
1673              mHwAvSyncIds.valueAt(index), sessionId);
1674        return mHwAvSyncIds.valueAt(index);
1675    }
1676
1677    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1678    if (dev == NULL) {
1679        return AUDIO_HW_SYNC_INVALID;
1680    }
1681    char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC);
1682    AudioParameter param = AudioParameter(String8(reply));
1683    free(reply);
1684
1685    int value;
1686    if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) {
1687        ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId);
1688        return AUDIO_HW_SYNC_INVALID;
1689    }
1690
1691    // allow only one session for a given HW A/V sync ID.
1692    for (size_t i = 0; i < mHwAvSyncIds.size(); i++) {
1693        if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) {
1694            ALOGV("getAudioHwSyncForSession removing ID %d for session %d",
1695                  value, mHwAvSyncIds.keyAt(i));
1696            mHwAvSyncIds.removeItemsAt(i);
1697            break;
1698        }
1699    }
1700
1701    mHwAvSyncIds.add(sessionId, value);
1702
1703    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1704        sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i);
1705        uint32_t sessions = thread->hasAudioSession(sessionId);
1706        if (sessions & PlaybackThread::TRACK_SESSION) {
1707            AudioParameter param = AudioParameter();
1708            param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value);
1709            thread->setParameters(param.toString());
1710            break;
1711        }
1712    }
1713
1714    ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId);
1715    return (audio_hw_sync_t)value;
1716}
1717
1718// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held
1719void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId)
1720{
1721    ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
1722    if (index >= 0) {
1723        audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index);
1724        ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId);
1725        AudioParameter param = AudioParameter();
1726        param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId);
1727        thread->setParameters(param.toString());
1728    }
1729}
1730
1731
1732// ----------------------------------------------------------------------------
1733
1734
1735sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module,
1736                                                            audio_io_handle_t *output,
1737                                                            audio_config_t *config,
1738                                                            audio_devices_t devices,
1739                                                            const String8& address,
1740                                                            audio_output_flags_t flags)
1741{
1742    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices);
1743    if (outHwDev == NULL) {
1744        return 0;
1745    }
1746
1747    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1748    if (*output == AUDIO_IO_HANDLE_NONE) {
1749        *output = nextUniqueId();
1750    }
1751
1752    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1753
1754    // FOR TESTING ONLY:
1755    // This if statement allows overriding the audio policy settings
1756    // and forcing a specific format or channel mask to the HAL/Sink device for testing.
1757    if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
1758        // Check only for Normal Mixing mode
1759        if (kEnableExtendedPrecision) {
1760            // Specify format (uncomment one below to choose)
1761            //config->format = AUDIO_FORMAT_PCM_FLOAT;
1762            //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
1763            //config->format = AUDIO_FORMAT_PCM_32_BIT;
1764            //config->format = AUDIO_FORMAT_PCM_8_24_BIT;
1765            // ALOGV("openOutput_l() upgrading format to %#08x", config->format);
1766        }
1767        if (kEnableExtendedChannels) {
1768            // Specify channel mask (uncomment one below to choose)
1769            //config->channel_mask = audio_channel_out_mask_from_count(4);  // for USB 4ch
1770            //config->channel_mask = audio_channel_mask_from_representation_and_bits(
1771            //        AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1);  // another 4ch example
1772        }
1773    }
1774
1775    AudioStreamOut *outputStream = NULL;
1776    status_t status = outHwDev->openOutputStream(
1777            &outputStream,
1778            *output,
1779            devices,
1780            flags,
1781            config,
1782            address.string());
1783
1784    mHardwareStatus = AUDIO_HW_IDLE;
1785
1786    if (status == NO_ERROR) {
1787
1788        PlaybackThread *thread;
1789        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1790            thread = new OffloadThread(this, outputStream, *output, devices);
1791            ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread);
1792        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
1793                || !isValidPcmSinkFormat(config->format)
1794                || !isValidPcmSinkChannelMask(config->channel_mask)) {
1795            thread = new DirectOutputThread(this, outputStream, *output, devices);
1796            ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread);
1797        } else {
1798            thread = new MixerThread(this, outputStream, *output, devices);
1799            ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread);
1800        }
1801        mPlaybackThreads.add(*output, thread);
1802        return thread;
1803    }
1804
1805    return 0;
1806}
1807
1808status_t AudioFlinger::openOutput(audio_module_handle_t module,
1809                                  audio_io_handle_t *output,
1810                                  audio_config_t *config,
1811                                  audio_devices_t *devices,
1812                                  const String8& address,
1813                                  uint32_t *latencyMs,
1814                                  audio_output_flags_t flags)
1815{
1816    ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1817              module,
1818              (devices != NULL) ? *devices : 0,
1819              config->sample_rate,
1820              config->format,
1821              config->channel_mask,
1822              flags);
1823
1824    if (*devices == AUDIO_DEVICE_NONE) {
1825        return BAD_VALUE;
1826    }
1827
1828    Mutex::Autolock _l(mLock);
1829
1830    sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags);
1831    if (thread != 0) {
1832        *latencyMs = thread->latency();
1833
1834        // notify client processes of the new output creation
1835        thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED);
1836
1837        // the first primary output opened designates the primary hw device
1838        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1839            ALOGI("Using module %d has the primary audio interface", module);
1840            mPrimaryHardwareDev = thread->getOutput()->audioHwDev;
1841
1842            AutoMutex lock(mHardwareLock);
1843            mHardwareStatus = AUDIO_HW_SET_MODE;
1844            mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode);
1845            mHardwareStatus = AUDIO_HW_IDLE;
1846
1847            mPrimaryOutputSampleRate = config->sample_rate;
1848        }
1849        return NO_ERROR;
1850    }
1851
1852    return NO_INIT;
1853}
1854
1855audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1856        audio_io_handle_t output2)
1857{
1858    Mutex::Autolock _l(mLock);
1859    MixerThread *thread1 = checkMixerThread_l(output1);
1860    MixerThread *thread2 = checkMixerThread_l(output2);
1861
1862    if (thread1 == NULL || thread2 == NULL) {
1863        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1864                output2);
1865        return AUDIO_IO_HANDLE_NONE;
1866    }
1867
1868    audio_io_handle_t id = nextUniqueId();
1869    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1870    thread->addOutputTrack(thread2);
1871    mPlaybackThreads.add(id, thread);
1872    // notify client processes of the new output creation
1873    thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED);
1874    return id;
1875}
1876
1877status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1878{
1879    return closeOutput_nonvirtual(output);
1880}
1881
1882status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1883{
1884    // keep strong reference on the playback thread so that
1885    // it is not destroyed while exit() is executed
1886    sp<PlaybackThread> thread;
1887    {
1888        Mutex::Autolock _l(mLock);
1889        thread = checkPlaybackThread_l(output);
1890        if (thread == NULL) {
1891            return BAD_VALUE;
1892        }
1893
1894        ALOGV("closeOutput() %d", output);
1895
1896        if (thread->type() == ThreadBase::MIXER) {
1897            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1898                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1899                    DuplicatingThread *dupThread =
1900                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1901                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1902
1903                }
1904            }
1905        }
1906
1907
1908        mPlaybackThreads.removeItem(output);
1909        // save all effects to the default thread
1910        if (mPlaybackThreads.size()) {
1911            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1912            if (dstThread != NULL) {
1913                // audioflinger lock is held here so the acquisition order of thread locks does not
1914                // matter
1915                Mutex::Autolock _dl(dstThread->mLock);
1916                Mutex::Autolock _sl(thread->mLock);
1917                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1918                for (size_t i = 0; i < effectChains.size(); i ++) {
1919                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1920                }
1921            }
1922        }
1923        audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL);
1924    }
1925    thread->exit();
1926    // The thread entity (active unit of execution) is no longer running here,
1927    // but the ThreadBase container still exists.
1928
1929    if (thread->type() != ThreadBase::DUPLICATING) {
1930        closeOutputFinish(thread);
1931    }
1932
1933    return NO_ERROR;
1934}
1935
1936void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread)
1937{
1938    AudioStreamOut *out = thread->clearOutput();
1939    ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1940    // from now on thread->mOutput is NULL
1941    out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1942    delete out;
1943}
1944
1945void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread)
1946{
1947    mPlaybackThreads.removeItem(thread->mId);
1948    thread->exit();
1949    closeOutputFinish(thread);
1950}
1951
1952status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1953{
1954    Mutex::Autolock _l(mLock);
1955    PlaybackThread *thread = checkPlaybackThread_l(output);
1956
1957    if (thread == NULL) {
1958        return BAD_VALUE;
1959    }
1960
1961    ALOGV("suspendOutput() %d", output);
1962    thread->suspend();
1963
1964    return NO_ERROR;
1965}
1966
1967status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1968{
1969    Mutex::Autolock _l(mLock);
1970    PlaybackThread *thread = checkPlaybackThread_l(output);
1971
1972    if (thread == NULL) {
1973        return BAD_VALUE;
1974    }
1975
1976    ALOGV("restoreOutput() %d", output);
1977
1978    thread->restore();
1979
1980    return NO_ERROR;
1981}
1982
1983status_t AudioFlinger::openInput(audio_module_handle_t module,
1984                                          audio_io_handle_t *input,
1985                                          audio_config_t *config,
1986                                          audio_devices_t *devices,
1987                                          const String8& address,
1988                                          audio_source_t source,
1989                                          audio_input_flags_t flags)
1990{
1991    Mutex::Autolock _l(mLock);
1992
1993    if (*devices == AUDIO_DEVICE_NONE) {
1994        return BAD_VALUE;
1995    }
1996
1997    sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags);
1998
1999    if (thread != 0) {
2000        // notify client processes of the new input creation
2001        thread->audioConfigChanged(AudioSystem::INPUT_OPENED);
2002        return NO_ERROR;
2003    }
2004    return NO_INIT;
2005}
2006
2007sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module,
2008                                                         audio_io_handle_t *input,
2009                                                         audio_config_t *config,
2010                                                         audio_devices_t devices,
2011                                                         const String8& address,
2012                                                         audio_source_t source,
2013                                                         audio_input_flags_t flags)
2014{
2015    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
2016    if (inHwDev == NULL) {
2017        *input = AUDIO_IO_HANDLE_NONE;
2018        return 0;
2019    }
2020
2021    if (*input == AUDIO_IO_HANDLE_NONE) {
2022        *input = nextUniqueId();
2023    }
2024
2025    audio_config_t halconfig = *config;
2026    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
2027    audio_stream_in_t *inStream = NULL;
2028    status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
2029                                        &inStream, flags, address.string(), source);
2030    ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d"
2031           ", Format %#x, Channels %x, flags %#x, status %d addr %s",
2032            inStream,
2033            halconfig.sample_rate,
2034            halconfig.format,
2035            halconfig.channel_mask,
2036            flags,
2037            status, address.string());
2038
2039    // If the input could not be opened with the requested parameters and we can handle the
2040    // conversion internally, try to open again with the proposed parameters.
2041    if (status == BAD_VALUE &&
2042        audio_is_linear_pcm(config->format) &&
2043        audio_is_linear_pcm(halconfig.format) &&
2044        (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
2045        (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) &&
2046        (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) {
2047        // FIXME describe the change proposed by HAL (save old values so we can log them here)
2048        ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
2049        inStream = NULL;
2050        status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
2051                                            &inStream, flags, address.string(), source);
2052        // FIXME log this new status; HAL should not propose any further changes
2053    }
2054
2055    if (status == NO_ERROR && inStream != NULL) {
2056
2057#ifdef TEE_SINK
2058        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
2059        // or (re-)create if current Pipe is idle and does not match the new format
2060        sp<NBAIO_Sink> teeSink;
2061        enum {
2062            TEE_SINK_NO,    // don't copy input
2063            TEE_SINK_NEW,   // copy input using a new pipe
2064            TEE_SINK_OLD,   // copy input using an existing pipe
2065        } kind;
2066        NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate,
2067                audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format);
2068        if (!mTeeSinkInputEnabled) {
2069            kind = TEE_SINK_NO;
2070        } else if (!Format_isValid(format)) {
2071            kind = TEE_SINK_NO;
2072        } else if (mRecordTeeSink == 0) {
2073            kind = TEE_SINK_NEW;
2074        } else if (mRecordTeeSink->getStrongCount() != 1) {
2075            kind = TEE_SINK_NO;
2076        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
2077            kind = TEE_SINK_OLD;
2078        } else {
2079            kind = TEE_SINK_NEW;
2080        }
2081        switch (kind) {
2082        case TEE_SINK_NEW: {
2083            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
2084            size_t numCounterOffers = 0;
2085            const NBAIO_Format offers[1] = {format};
2086            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
2087            ALOG_ASSERT(index == 0);
2088            PipeReader *pipeReader = new PipeReader(*pipe);
2089            numCounterOffers = 0;
2090            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
2091            ALOG_ASSERT(index == 0);
2092            mRecordTeeSink = pipe;
2093            mRecordTeeSource = pipeReader;
2094            teeSink = pipe;
2095            }
2096            break;
2097        case TEE_SINK_OLD:
2098            teeSink = mRecordTeeSink;
2099            break;
2100        case TEE_SINK_NO:
2101        default:
2102            break;
2103        }
2104#endif
2105
2106        AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream);
2107
2108        // Start record thread
2109        // RecordThread requires both input and output device indication to forward to audio
2110        // pre processing modules
2111        sp<RecordThread> thread = new RecordThread(this,
2112                                  inputStream,
2113                                  *input,
2114                                  primaryOutputDevice_l(),
2115                                  devices
2116#ifdef TEE_SINK
2117                                  , teeSink
2118#endif
2119                                  );
2120        mRecordThreads.add(*input, thread);
2121        ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
2122        return thread;
2123    }
2124
2125    *input = AUDIO_IO_HANDLE_NONE;
2126    return 0;
2127}
2128
2129status_t AudioFlinger::closeInput(audio_io_handle_t input)
2130{
2131    return closeInput_nonvirtual(input);
2132}
2133
2134status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
2135{
2136    // keep strong reference on the record thread so that
2137    // it is not destroyed while exit() is executed
2138    sp<RecordThread> thread;
2139    {
2140        Mutex::Autolock _l(mLock);
2141        thread = checkRecordThread_l(input);
2142        if (thread == 0) {
2143            return BAD_VALUE;
2144        }
2145
2146        ALOGV("closeInput() %d", input);
2147
2148        // If we still have effect chains, it means that a client still holds a handle
2149        // on at least one effect. We must either move the chain to an existing thread with the
2150        // same session ID or put it aside in case a new record thread is opened for a
2151        // new capture on the same session
2152        sp<EffectChain> chain;
2153        {
2154            Mutex::Autolock _sl(thread->mLock);
2155            Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
2156            // Note: maximum one chain per record thread
2157            if (effectChains.size() != 0) {
2158                chain = effectChains[0];
2159            }
2160        }
2161        if (chain != 0) {
2162            // first check if a record thread is already opened with a client on the same session.
2163            // This should only happen in case of overlap between one thread tear down and the
2164            // creation of its replacement
2165            size_t i;
2166            for (i = 0; i < mRecordThreads.size(); i++) {
2167                sp<RecordThread> t = mRecordThreads.valueAt(i);
2168                if (t == thread) {
2169                    continue;
2170                }
2171                if (t->hasAudioSession(chain->sessionId()) != 0) {
2172                    Mutex::Autolock _l(t->mLock);
2173                    ALOGV("closeInput() found thread %d for effect session %d",
2174                          t->id(), chain->sessionId());
2175                    t->addEffectChain_l(chain);
2176                    break;
2177                }
2178            }
2179            // put the chain aside if we could not find a record thread with the same session id.
2180            if (i == mRecordThreads.size()) {
2181                putOrphanEffectChain_l(chain);
2182            }
2183        }
2184        audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL);
2185        mRecordThreads.removeItem(input);
2186    }
2187    // FIXME: calling thread->exit() without mLock held should not be needed anymore now that
2188    // we have a different lock for notification client
2189    closeInputFinish(thread);
2190    return NO_ERROR;
2191}
2192
2193void AudioFlinger::closeInputFinish(sp<RecordThread> thread)
2194{
2195    thread->exit();
2196    AudioStreamIn *in = thread->clearInput();
2197    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
2198    // from now on thread->mInput is NULL
2199    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
2200    delete in;
2201}
2202
2203void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread)
2204{
2205    mRecordThreads.removeItem(thread->mId);
2206    closeInputFinish(thread);
2207}
2208
2209status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
2210{
2211    Mutex::Autolock _l(mLock);
2212    ALOGV("invalidateStream() stream %d", stream);
2213
2214    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2215        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2216        thread->invalidateTracks(stream);
2217    }
2218
2219    return NO_ERROR;
2220}
2221
2222
2223audio_unique_id_t AudioFlinger::newAudioUniqueId()
2224{
2225    return nextUniqueId();
2226}
2227
2228void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
2229{
2230    Mutex::Autolock _l(mLock);
2231    pid_t caller = IPCThreadState::self()->getCallingPid();
2232    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
2233    if (pid != -1 && (caller == getpid_cached)) {
2234        caller = pid;
2235    }
2236
2237    {
2238        Mutex::Autolock _cl(mClientLock);
2239        // Ignore requests received from processes not known as notification client. The request
2240        // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
2241        // called from a different pid leaving a stale session reference.  Also we don't know how
2242        // to clear this reference if the client process dies.
2243        if (mNotificationClients.indexOfKey(caller) < 0) {
2244            ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
2245            return;
2246        }
2247    }
2248
2249    size_t num = mAudioSessionRefs.size();
2250    for (size_t i = 0; i< num; i++) {
2251        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
2252        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2253            ref->mCnt++;
2254            ALOGV(" incremented refcount to %d", ref->mCnt);
2255            return;
2256        }
2257    }
2258    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
2259    ALOGV(" added new entry for %d", audioSession);
2260}
2261
2262void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
2263{
2264    Mutex::Autolock _l(mLock);
2265    pid_t caller = IPCThreadState::self()->getCallingPid();
2266    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
2267    if (pid != -1 && (caller == getpid_cached)) {
2268        caller = pid;
2269    }
2270    size_t num = mAudioSessionRefs.size();
2271    for (size_t i = 0; i< num; i++) {
2272        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2273        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2274            ref->mCnt--;
2275            ALOGV(" decremented refcount to %d", ref->mCnt);
2276            if (ref->mCnt == 0) {
2277                mAudioSessionRefs.removeAt(i);
2278                delete ref;
2279                purgeStaleEffects_l();
2280            }
2281            return;
2282        }
2283    }
2284    // If the caller is mediaserver it is likely that the session being released was acquired
2285    // on behalf of a process not in notification clients and we ignore the warning.
2286    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2287}
2288
2289void AudioFlinger::purgeStaleEffects_l() {
2290
2291    ALOGV("purging stale effects");
2292
2293    Vector< sp<EffectChain> > chains;
2294
2295    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2296        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2297        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2298            sp<EffectChain> ec = t->mEffectChains[j];
2299            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2300                chains.push(ec);
2301            }
2302        }
2303    }
2304    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2305        sp<RecordThread> t = mRecordThreads.valueAt(i);
2306        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2307            sp<EffectChain> ec = t->mEffectChains[j];
2308            chains.push(ec);
2309        }
2310    }
2311
2312    for (size_t i = 0; i < chains.size(); i++) {
2313        sp<EffectChain> ec = chains[i];
2314        int sessionid = ec->sessionId();
2315        sp<ThreadBase> t = ec->mThread.promote();
2316        if (t == 0) {
2317            continue;
2318        }
2319        size_t numsessionrefs = mAudioSessionRefs.size();
2320        bool found = false;
2321        for (size_t k = 0; k < numsessionrefs; k++) {
2322            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2323            if (ref->mSessionid == sessionid) {
2324                ALOGV(" session %d still exists for %d with %d refs",
2325                    sessionid, ref->mPid, ref->mCnt);
2326                found = true;
2327                break;
2328            }
2329        }
2330        if (!found) {
2331            Mutex::Autolock _l(t->mLock);
2332            // remove all effects from the chain
2333            while (ec->mEffects.size()) {
2334                sp<EffectModule> effect = ec->mEffects[0];
2335                effect->unPin();
2336                t->removeEffect_l(effect);
2337                if (effect->purgeHandles()) {
2338                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2339                }
2340                AudioSystem::unregisterEffect(effect->id());
2341            }
2342        }
2343    }
2344    return;
2345}
2346
2347// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2348AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2349{
2350    return mPlaybackThreads.valueFor(output).get();
2351}
2352
2353// checkMixerThread_l() must be called with AudioFlinger::mLock held
2354AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2355{
2356    PlaybackThread *thread = checkPlaybackThread_l(output);
2357    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2358}
2359
2360// checkRecordThread_l() must be called with AudioFlinger::mLock held
2361AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2362{
2363    return mRecordThreads.valueFor(input).get();
2364}
2365
2366uint32_t AudioFlinger::nextUniqueId()
2367{
2368    return (uint32_t) android_atomic_inc(&mNextUniqueId);
2369}
2370
2371AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2372{
2373    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2374        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2375        AudioStreamOut *output = thread->getOutput();
2376        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2377            return thread;
2378        }
2379    }
2380    return NULL;
2381}
2382
2383audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2384{
2385    PlaybackThread *thread = primaryPlaybackThread_l();
2386
2387    if (thread == NULL) {
2388        return 0;
2389    }
2390
2391    return thread->outDevice();
2392}
2393
2394sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2395                                    int triggerSession,
2396                                    int listenerSession,
2397                                    sync_event_callback_t callBack,
2398                                    wp<RefBase> cookie)
2399{
2400    Mutex::Autolock _l(mLock);
2401
2402    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2403    status_t playStatus = NAME_NOT_FOUND;
2404    status_t recStatus = NAME_NOT_FOUND;
2405    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2406        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2407        if (playStatus == NO_ERROR) {
2408            return event;
2409        }
2410    }
2411    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2412        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2413        if (recStatus == NO_ERROR) {
2414            return event;
2415        }
2416    }
2417    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2418        mPendingSyncEvents.add(event);
2419    } else {
2420        ALOGV("createSyncEvent() invalid event %d", event->type());
2421        event.clear();
2422    }
2423    return event;
2424}
2425
2426// ----------------------------------------------------------------------------
2427//  Effect management
2428// ----------------------------------------------------------------------------
2429
2430
2431status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2432{
2433    Mutex::Autolock _l(mLock);
2434    return EffectQueryNumberEffects(numEffects);
2435}
2436
2437status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2438{
2439    Mutex::Autolock _l(mLock);
2440    return EffectQueryEffect(index, descriptor);
2441}
2442
2443status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2444        effect_descriptor_t *descriptor) const
2445{
2446    Mutex::Autolock _l(mLock);
2447    return EffectGetDescriptor(pUuid, descriptor);
2448}
2449
2450
2451sp<IEffect> AudioFlinger::createEffect(
2452        effect_descriptor_t *pDesc,
2453        const sp<IEffectClient>& effectClient,
2454        int32_t priority,
2455        audio_io_handle_t io,
2456        int sessionId,
2457        const String16& opPackageName,
2458        status_t *status,
2459        int *id,
2460        int *enabled)
2461{
2462    status_t lStatus = NO_ERROR;
2463    sp<EffectHandle> handle;
2464    effect_descriptor_t desc;
2465
2466    pid_t pid = IPCThreadState::self()->getCallingPid();
2467    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2468            pid, effectClient.get(), priority, sessionId, io);
2469
2470    if (pDesc == NULL) {
2471        lStatus = BAD_VALUE;
2472        goto Exit;
2473    }
2474
2475    // check audio settings permission for global effects
2476    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2477        lStatus = PERMISSION_DENIED;
2478        goto Exit;
2479    }
2480
2481    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2482    // that can only be created by audio policy manager (running in same process)
2483    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2484        lStatus = PERMISSION_DENIED;
2485        goto Exit;
2486    }
2487
2488    {
2489        if (!EffectIsNullUuid(&pDesc->uuid)) {
2490            // if uuid is specified, request effect descriptor
2491            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2492            if (lStatus < 0) {
2493                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2494                goto Exit;
2495            }
2496        } else {
2497            // if uuid is not specified, look for an available implementation
2498            // of the required type in effect factory
2499            if (EffectIsNullUuid(&pDesc->type)) {
2500                ALOGW("createEffect() no effect type");
2501                lStatus = BAD_VALUE;
2502                goto Exit;
2503            }
2504            uint32_t numEffects = 0;
2505            effect_descriptor_t d;
2506            d.flags = 0; // prevent compiler warning
2507            bool found = false;
2508
2509            lStatus = EffectQueryNumberEffects(&numEffects);
2510            if (lStatus < 0) {
2511                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2512                goto Exit;
2513            }
2514            for (uint32_t i = 0; i < numEffects; i++) {
2515                lStatus = EffectQueryEffect(i, &desc);
2516                if (lStatus < 0) {
2517                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2518                    continue;
2519                }
2520                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2521                    // If matching type found save effect descriptor. If the session is
2522                    // 0 and the effect is not auxiliary, continue enumeration in case
2523                    // an auxiliary version of this effect type is available
2524                    found = true;
2525                    d = desc;
2526                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2527                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2528                        break;
2529                    }
2530                }
2531            }
2532            if (!found) {
2533                lStatus = BAD_VALUE;
2534                ALOGW("createEffect() effect not found");
2535                goto Exit;
2536            }
2537            // For same effect type, chose auxiliary version over insert version if
2538            // connect to output mix (Compliance to OpenSL ES)
2539            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2540                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2541                desc = d;
2542            }
2543        }
2544
2545        // Do not allow auxiliary effects on a session different from 0 (output mix)
2546        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2547             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2548            lStatus = INVALID_OPERATION;
2549            goto Exit;
2550        }
2551
2552        // check recording permission for visualizer
2553        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2554            !recordingAllowed(opPackageName)) {
2555            lStatus = PERMISSION_DENIED;
2556            goto Exit;
2557        }
2558
2559        // return effect descriptor
2560        *pDesc = desc;
2561        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2562            // if the output returned by getOutputForEffect() is removed before we lock the
2563            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2564            // and we will exit safely
2565            io = AudioSystem::getOutputForEffect(&desc);
2566            ALOGV("createEffect got output %d", io);
2567        }
2568
2569        Mutex::Autolock _l(mLock);
2570
2571        // If output is not specified try to find a matching audio session ID in one of the
2572        // output threads.
2573        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2574        // because of code checking output when entering the function.
2575        // Note: io is never 0 when creating an effect on an input
2576        if (io == AUDIO_IO_HANDLE_NONE) {
2577            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2578                // output must be specified by AudioPolicyManager when using session
2579                // AUDIO_SESSION_OUTPUT_STAGE
2580                lStatus = BAD_VALUE;
2581                goto Exit;
2582            }
2583            // look for the thread where the specified audio session is present
2584            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2585                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2586                    io = mPlaybackThreads.keyAt(i);
2587                    break;
2588                }
2589            }
2590            if (io == 0) {
2591                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2592                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2593                        io = mRecordThreads.keyAt(i);
2594                        break;
2595                    }
2596                }
2597            }
2598            // If no output thread contains the requested session ID, default to
2599            // first output. The effect chain will be moved to the correct output
2600            // thread when a track with the same session ID is created
2601            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2602                io = mPlaybackThreads.keyAt(0);
2603            }
2604            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2605        }
2606        ThreadBase *thread = checkRecordThread_l(io);
2607        if (thread == NULL) {
2608            thread = checkPlaybackThread_l(io);
2609            if (thread == NULL) {
2610                ALOGE("createEffect() unknown output thread");
2611                lStatus = BAD_VALUE;
2612                goto Exit;
2613            }
2614        } else {
2615            // Check if one effect chain was awaiting for an effect to be created on this
2616            // session and used it instead of creating a new one.
2617            sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId);
2618            if (chain != 0) {
2619                Mutex::Autolock _l(thread->mLock);
2620                thread->addEffectChain_l(chain);
2621            }
2622        }
2623
2624        sp<Client> client = registerPid(pid);
2625
2626        // create effect on selected output thread
2627        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2628                &desc, enabled, &lStatus);
2629        if (handle != 0 && id != NULL) {
2630            *id = handle->id();
2631        }
2632        if (handle == 0) {
2633            // remove local strong reference to Client with mClientLock held
2634            Mutex::Autolock _cl(mClientLock);
2635            client.clear();
2636        }
2637    }
2638
2639Exit:
2640    *status = lStatus;
2641    return handle;
2642}
2643
2644status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2645        audio_io_handle_t dstOutput)
2646{
2647    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2648            sessionId, srcOutput, dstOutput);
2649    Mutex::Autolock _l(mLock);
2650    if (srcOutput == dstOutput) {
2651        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2652        return NO_ERROR;
2653    }
2654    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2655    if (srcThread == NULL) {
2656        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2657        return BAD_VALUE;
2658    }
2659    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2660    if (dstThread == NULL) {
2661        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2662        return BAD_VALUE;
2663    }
2664
2665    Mutex::Autolock _dl(dstThread->mLock);
2666    Mutex::Autolock _sl(srcThread->mLock);
2667    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2668}
2669
2670// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2671status_t AudioFlinger::moveEffectChain_l(int sessionId,
2672                                   AudioFlinger::PlaybackThread *srcThread,
2673                                   AudioFlinger::PlaybackThread *dstThread,
2674                                   bool reRegister)
2675{
2676    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2677            sessionId, srcThread, dstThread);
2678
2679    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2680    if (chain == 0) {
2681        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2682                sessionId, srcThread);
2683        return INVALID_OPERATION;
2684    }
2685
2686    // Check whether the destination thread has a channel count of FCC_2, which is
2687    // currently required for (most) effects. Prevent moving the effect chain here rather
2688    // than disabling the addEffect_l() call in dstThread below.
2689    if ((dstThread->type() == ThreadBase::MIXER || dstThread->type() == ThreadBase::DUPLICATING) &&
2690            dstThread->mChannelCount != FCC_2) {
2691        ALOGW("moveEffectChain_l() effect chain failed because"
2692                " destination thread %p channel count(%u) != %u",
2693                dstThread, dstThread->mChannelCount, FCC_2);
2694        return INVALID_OPERATION;
2695    }
2696
2697    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2698    // so that a new chain is created with correct parameters when first effect is added. This is
2699    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2700    // removed.
2701    srcThread->removeEffectChain_l(chain);
2702
2703    // transfer all effects one by one so that new effect chain is created on new thread with
2704    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2705    sp<EffectChain> dstChain;
2706    uint32_t strategy = 0; // prevent compiler warning
2707    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2708    Vector< sp<EffectModule> > removed;
2709    status_t status = NO_ERROR;
2710    while (effect != 0) {
2711        srcThread->removeEffect_l(effect);
2712        removed.add(effect);
2713        status = dstThread->addEffect_l(effect);
2714        if (status != NO_ERROR) {
2715            break;
2716        }
2717        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2718        if (effect->state() == EffectModule::ACTIVE ||
2719                effect->state() == EffectModule::STOPPING) {
2720            effect->start();
2721        }
2722        // if the move request is not received from audio policy manager, the effect must be
2723        // re-registered with the new strategy and output
2724        if (dstChain == 0) {
2725            dstChain = effect->chain().promote();
2726            if (dstChain == 0) {
2727                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2728                status = NO_INIT;
2729                break;
2730            }
2731            strategy = dstChain->strategy();
2732        }
2733        if (reRegister) {
2734            AudioSystem::unregisterEffect(effect->id());
2735            AudioSystem::registerEffect(&effect->desc(),
2736                                        dstThread->id(),
2737                                        strategy,
2738                                        sessionId,
2739                                        effect->id());
2740            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2741        }
2742        effect = chain->getEffectFromId_l(0);
2743    }
2744
2745    if (status != NO_ERROR) {
2746        for (size_t i = 0; i < removed.size(); i++) {
2747            srcThread->addEffect_l(removed[i]);
2748            if (dstChain != 0 && reRegister) {
2749                AudioSystem::unregisterEffect(removed[i]->id());
2750                AudioSystem::registerEffect(&removed[i]->desc(),
2751                                            srcThread->id(),
2752                                            strategy,
2753                                            sessionId,
2754                                            removed[i]->id());
2755                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2756            }
2757        }
2758    }
2759
2760    return status;
2761}
2762
2763bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2764{
2765    if (mGlobalEffectEnableTime != 0 &&
2766            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2767        return true;
2768    }
2769
2770    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2771        sp<EffectChain> ec =
2772                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2773        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2774            return true;
2775        }
2776    }
2777    return false;
2778}
2779
2780void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2781{
2782    Mutex::Autolock _l(mLock);
2783
2784    mGlobalEffectEnableTime = systemTime();
2785
2786    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2787        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2788        if (t->mType == ThreadBase::OFFLOAD) {
2789            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2790        }
2791    }
2792
2793}
2794
2795status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain)
2796{
2797    audio_session_t session = (audio_session_t)chain->sessionId();
2798    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2799    ALOGV("putOrphanEffectChain_l session %d index %d", session, index);
2800    if (index >= 0) {
2801        ALOGW("putOrphanEffectChain_l chain for session %d already present", session);
2802        return ALREADY_EXISTS;
2803    }
2804    mOrphanEffectChains.add(session, chain);
2805    return NO_ERROR;
2806}
2807
2808sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session)
2809{
2810    sp<EffectChain> chain;
2811    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2812    ALOGV("getOrphanEffectChain_l session %d index %d", session, index);
2813    if (index >= 0) {
2814        chain = mOrphanEffectChains.valueAt(index);
2815        mOrphanEffectChains.removeItemsAt(index);
2816    }
2817    return chain;
2818}
2819
2820bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect)
2821{
2822    Mutex::Autolock _l(mLock);
2823    audio_session_t session = (audio_session_t)effect->sessionId();
2824    ssize_t index = mOrphanEffectChains.indexOfKey(session);
2825    ALOGV("updateOrphanEffectChains session %d index %d", session, index);
2826    if (index >= 0) {
2827        sp<EffectChain> chain = mOrphanEffectChains.valueAt(index);
2828        if (chain->removeEffect_l(effect) == 0) {
2829            ALOGV("updateOrphanEffectChains removing effect chain at index %d", index);
2830            mOrphanEffectChains.removeItemsAt(index);
2831        }
2832        return true;
2833    }
2834    return false;
2835}
2836
2837
2838struct Entry {
2839#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21
2840    char mFileName[TEE_MAX_FILENAME];
2841};
2842
2843int comparEntry(const void *p1, const void *p2)
2844{
2845    return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName);
2846}
2847
2848#ifdef TEE_SINK
2849void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2850{
2851    NBAIO_Source *teeSource = source.get();
2852    if (teeSource != NULL) {
2853        // .wav rotation
2854        // There is a benign race condition if 2 threads call this simultaneously.
2855        // They would both traverse the directory, but the result would simply be
2856        // failures at unlink() which are ignored.  It's also unlikely since
2857        // normally dumpsys is only done by bugreport or from the command line.
2858        char teePath[32+256];
2859        strcpy(teePath, "/data/misc/media");
2860        size_t teePathLen = strlen(teePath);
2861        DIR *dir = opendir(teePath);
2862        teePath[teePathLen++] = '/';
2863        if (dir != NULL) {
2864#define TEE_MAX_SORT 20 // number of entries to sort
2865#define TEE_MAX_KEEP 10 // number of entries to keep
2866            struct Entry entries[TEE_MAX_SORT];
2867            size_t entryCount = 0;
2868            while (entryCount < TEE_MAX_SORT) {
2869                struct dirent de;
2870                struct dirent *result = NULL;
2871                int rc = readdir_r(dir, &de, &result);
2872                if (rc != 0) {
2873                    ALOGW("readdir_r failed %d", rc);
2874                    break;
2875                }
2876                if (result == NULL) {
2877                    break;
2878                }
2879                if (result != &de) {
2880                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2881                    break;
2882                }
2883                // ignore non .wav file entries
2884                size_t nameLen = strlen(de.d_name);
2885                if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME ||
2886                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2887                    continue;
2888                }
2889                strcpy(entries[entryCount++].mFileName, de.d_name);
2890            }
2891            (void) closedir(dir);
2892            if (entryCount > TEE_MAX_KEEP) {
2893                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2894                for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) {
2895                    strcpy(&teePath[teePathLen], entries[i].mFileName);
2896                    (void) unlink(teePath);
2897                }
2898            }
2899        } else {
2900            if (fd >= 0) {
2901                dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2902            }
2903        }
2904        char teeTime[16];
2905        struct timeval tv;
2906        gettimeofday(&tv, NULL);
2907        struct tm tm;
2908        localtime_r(&tv.tv_sec, &tm);
2909        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2910        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2911        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2912        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2913        if (teeFd >= 0) {
2914            // FIXME use libsndfile
2915            char wavHeader[44];
2916            memcpy(wavHeader,
2917                "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",
2918                sizeof(wavHeader));
2919            NBAIO_Format format = teeSource->format();
2920            unsigned channelCount = Format_channelCount(format);
2921            uint32_t sampleRate = Format_sampleRate(format);
2922            size_t frameSize = Format_frameSize(format);
2923            wavHeader[22] = channelCount;       // number of channels
2924            wavHeader[24] = sampleRate;         // sample rate
2925            wavHeader[25] = sampleRate >> 8;
2926            wavHeader[32] = frameSize;          // block alignment
2927            wavHeader[33] = frameSize >> 8;
2928            write(teeFd, wavHeader, sizeof(wavHeader));
2929            size_t total = 0;
2930            bool firstRead = true;
2931#define TEE_SINK_READ 1024                      // frames per I/O operation
2932            void *buffer = malloc(TEE_SINK_READ * frameSize);
2933            for (;;) {
2934                size_t count = TEE_SINK_READ;
2935                ssize_t actual = teeSource->read(buffer, count,
2936                        AudioBufferProvider::kInvalidPTS);
2937                bool wasFirstRead = firstRead;
2938                firstRead = false;
2939                if (actual <= 0) {
2940                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2941                        continue;
2942                    }
2943                    break;
2944                }
2945                ALOG_ASSERT(actual <= (ssize_t)count);
2946                write(teeFd, buffer, actual * frameSize);
2947                total += actual;
2948            }
2949            free(buffer);
2950            lseek(teeFd, (off_t) 4, SEEK_SET);
2951            uint32_t temp = 44 + total * frameSize - 8;
2952            // FIXME not big-endian safe
2953            write(teeFd, &temp, sizeof(temp));
2954            lseek(teeFd, (off_t) 40, SEEK_SET);
2955            temp =  total * frameSize;
2956            // FIXME not big-endian safe
2957            write(teeFd, &temp, sizeof(temp));
2958            close(teeFd);
2959            if (fd >= 0) {
2960                dprintf(fd, "tee copied to %s\n", teePath);
2961            }
2962        } else {
2963            if (fd >= 0) {
2964                dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2965            }
2966        }
2967    }
2968}
2969#endif
2970
2971// ----------------------------------------------------------------------------
2972
2973status_t AudioFlinger::onTransact(
2974        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2975{
2976    return BnAudioFlinger::onTransact(code, data, reply, flags);
2977}
2978
2979} // namespace android
2980