AudioFlinger.cpp revision 951f455566775e5f01e67c5ee26863d7d19209d7 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <dirent.h>
24#include <math.h>
25#include <signal.h>
26#include <sys/time.h>
27#include <sys/resource.h>
28
29#include <binder/IPCThreadState.h>
30#include <binder/IServiceManager.h>
31#include <utils/Log.h>
32#include <utils/Trace.h>
33#include <binder/Parcel.h>
34#include <utils/String16.h>
35#include <utils/threads.h>
36#include <utils/Atomic.h>
37
38#include <cutils/bitops.h>
39#include <cutils/properties.h>
40
41#include <system/audio.h>
42#include <hardware/audio.h>
43
44#include "AudioMixer.h"
45#include "AudioFlinger.h"
46#include "ServiceUtilities.h"
47
48#include <media/EffectsFactoryApi.h>
49#include <audio_effects/effect_visualizer.h>
50#include <audio_effects/effect_ns.h>
51#include <audio_effects/effect_aec.h>
52
53#include <audio_utils/primitives.h>
54
55#include <powermanager/PowerManager.h>
56
57#include <common_time/cc_helper.h>
58
59#include <media/IMediaLogService.h>
60
61#include <media/nbaio/Pipe.h>
62#include <media/nbaio/PipeReader.h>
63#include <media/AudioParameter.h>
64#include <private/android_filesystem_config.h>
65
66// ----------------------------------------------------------------------------
67
68// Note: the following macro is used for extremely verbose logging message.  In
69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
70// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
71// are so verbose that we want to suppress them even when we have ALOG_ASSERT
72// turned on.  Do not uncomment the #def below unless you really know what you
73// are doing and want to see all of the extremely verbose messages.
74//#define VERY_VERY_VERBOSE_LOGGING
75#ifdef VERY_VERY_VERBOSE_LOGGING
76#define ALOGVV ALOGV
77#else
78#define ALOGVV(a...) do { } while(0)
79#endif
80
81namespace android {
82
83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
84static const char kHardwareLockedString[] = "Hardware lock is taken\n";
85static const char kClientLockedString[] = "Client lock is taken\n";
86
87
88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
89
90uint32_t AudioFlinger::mScreenState;
91
92#ifdef TEE_SINK
93bool AudioFlinger::mTeeSinkInputEnabled = false;
94bool AudioFlinger::mTeeSinkOutputEnabled = false;
95bool AudioFlinger::mTeeSinkTrackEnabled = false;
96
97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
100#endif
101
102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
103// we define a minimum time during which a global effect is considered enabled.
104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
105
106// ----------------------------------------------------------------------------
107
108const char *formatToString(audio_format_t format) {
109    switch(format) {
110    case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8";
111    case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16";
112    case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32";
113    case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24";
114    case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24";
115    case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat";
116    case AUDIO_FORMAT_MP3: return "mp3";
117    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
118    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
119    case AUDIO_FORMAT_AAC: return "aac";
120    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
121    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
122    case AUDIO_FORMAT_VORBIS: return "vorbis";
123    default:
124        break;
125    }
126    return "unknown";
127}
128
129static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
130{
131    const hw_module_t *mod;
132    int rc;
133
134    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
135    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
136                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
137    if (rc) {
138        goto out;
139    }
140    rc = audio_hw_device_open(mod, dev);
141    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
142                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
143    if (rc) {
144        goto out;
145    }
146    if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
147        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
148        rc = BAD_VALUE;
149        goto out;
150    }
151    return 0;
152
153out:
154    *dev = NULL;
155    return rc;
156}
157
158// ----------------------------------------------------------------------------
159
160AudioFlinger::AudioFlinger()
161    : BnAudioFlinger(),
162      mPrimaryHardwareDev(NULL),
163      mAudioHwDevs(NULL),
164      mHardwareStatus(AUDIO_HW_IDLE),
165      mMasterVolume(1.0f),
166      mMasterMute(false),
167      mNextUniqueId(1),
168      mMode(AUDIO_MODE_INVALID),
169      mBtNrecIsOff(false),
170      mIsLowRamDevice(true),
171      mIsDeviceTypeKnown(false),
172      mGlobalEffectEnableTime(0)
173{
174    getpid_cached = getpid();
175    char value[PROPERTY_VALUE_MAX];
176    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
177    if (doLog) {
178        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY);
179    }
180
181#ifdef TEE_SINK
182    (void) property_get("ro.debuggable", value, "0");
183    int debuggable = atoi(value);
184    int teeEnabled = 0;
185    if (debuggable) {
186        (void) property_get("af.tee", value, "0");
187        teeEnabled = atoi(value);
188    }
189    // FIXME symbolic constants here
190    if (teeEnabled & 1) {
191        mTeeSinkInputEnabled = true;
192    }
193    if (teeEnabled & 2) {
194        mTeeSinkOutputEnabled = true;
195    }
196    if (teeEnabled & 4) {
197        mTeeSinkTrackEnabled = true;
198    }
199#endif
200}
201
202void AudioFlinger::onFirstRef()
203{
204    int rc = 0;
205
206    Mutex::Autolock _l(mLock);
207
208    /* TODO: move all this work into an Init() function */
209    char val_str[PROPERTY_VALUE_MAX] = { 0 };
210    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
211        uint32_t int_val;
212        if (1 == sscanf(val_str, "%u", &int_val)) {
213            mStandbyTimeInNsecs = milliseconds(int_val);
214            ALOGI("Using %u mSec as standby time.", int_val);
215        } else {
216            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
217            ALOGI("Using default %u mSec as standby time.",
218                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
219        }
220    }
221
222    mPatchPanel = new PatchPanel(this);
223
224    mMode = AUDIO_MODE_NORMAL;
225}
226
227AudioFlinger::~AudioFlinger()
228{
229    while (!mRecordThreads.isEmpty()) {
230        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
231        closeInput_nonvirtual(mRecordThreads.keyAt(0));
232    }
233    while (!mPlaybackThreads.isEmpty()) {
234        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
235        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
236    }
237
238    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
239        // no mHardwareLock needed, as there are no other references to this
240        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
241        delete mAudioHwDevs.valueAt(i);
242    }
243
244    // Tell media.log service about any old writers that still need to be unregistered
245    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
246    if (binder != 0) {
247        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
248        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
249            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
250            mUnregisteredWriters.pop();
251            mediaLogService->unregisterWriter(iMemory);
252        }
253    }
254
255}
256
257static const char * const audio_interfaces[] = {
258    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
259    AUDIO_HARDWARE_MODULE_ID_A2DP,
260    AUDIO_HARDWARE_MODULE_ID_USB,
261};
262#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
263
264AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
265        audio_module_handle_t module,
266        audio_devices_t devices)
267{
268    // if module is 0, the request comes from an old policy manager and we should load
269    // well known modules
270    if (module == 0) {
271        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
272        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
273            loadHwModule_l(audio_interfaces[i]);
274        }
275        // then try to find a module supporting the requested device.
276        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
277            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
278            audio_hw_device_t *dev = audioHwDevice->hwDevice();
279            if ((dev->get_supported_devices != NULL) &&
280                    (dev->get_supported_devices(dev) & devices) == devices)
281                return audioHwDevice;
282        }
283    } else {
284        // check a match for the requested module handle
285        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
286        if (audioHwDevice != NULL) {
287            return audioHwDevice;
288        }
289    }
290
291    return NULL;
292}
293
294void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
295{
296    const size_t SIZE = 256;
297    char buffer[SIZE];
298    String8 result;
299
300    result.append("Clients:\n");
301    for (size_t i = 0; i < mClients.size(); ++i) {
302        sp<Client> client = mClients.valueAt(i).promote();
303        if (client != 0) {
304            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
305            result.append(buffer);
306        }
307    }
308
309    result.append("Notification Clients:\n");
310    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
311        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
312        result.append(buffer);
313    }
314
315    result.append("Global session refs:\n");
316    result.append("  session   pid count\n");
317    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
318        AudioSessionRef *r = mAudioSessionRefs[i];
319        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
320        result.append(buffer);
321    }
322    write(fd, result.string(), result.size());
323}
324
325
326void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
327{
328    const size_t SIZE = 256;
329    char buffer[SIZE];
330    String8 result;
331    hardware_call_state hardwareStatus = mHardwareStatus;
332
333    snprintf(buffer, SIZE, "Hardware status: %d\n"
334                           "Standby Time mSec: %u\n",
335                            hardwareStatus,
336                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
337    result.append(buffer);
338    write(fd, result.string(), result.size());
339}
340
341void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
342{
343    const size_t SIZE = 256;
344    char buffer[SIZE];
345    String8 result;
346    snprintf(buffer, SIZE, "Permission Denial: "
347            "can't dump AudioFlinger from pid=%d, uid=%d\n",
348            IPCThreadState::self()->getCallingPid(),
349            IPCThreadState::self()->getCallingUid());
350    result.append(buffer);
351    write(fd, result.string(), result.size());
352}
353
354bool AudioFlinger::dumpTryLock(Mutex& mutex)
355{
356    bool locked = false;
357    for (int i = 0; i < kDumpLockRetries; ++i) {
358        if (mutex.tryLock() == NO_ERROR) {
359            locked = true;
360            break;
361        }
362        usleep(kDumpLockSleepUs);
363    }
364    return locked;
365}
366
367status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
368{
369    if (!dumpAllowed()) {
370        dumpPermissionDenial(fd, args);
371    } else {
372        // get state of hardware lock
373        bool hardwareLocked = dumpTryLock(mHardwareLock);
374        if (!hardwareLocked) {
375            String8 result(kHardwareLockedString);
376            write(fd, result.string(), result.size());
377        } else {
378            mHardwareLock.unlock();
379        }
380
381        bool locked = dumpTryLock(mLock);
382
383        // failed to lock - AudioFlinger is probably deadlocked
384        if (!locked) {
385            String8 result(kDeadlockedString);
386            write(fd, result.string(), result.size());
387        }
388
389        bool clientLocked = dumpTryLock(mClientLock);
390        if (!clientLocked) {
391            String8 result(kClientLockedString);
392            write(fd, result.string(), result.size());
393        }
394        dumpClients(fd, args);
395        if (clientLocked) {
396            mClientLock.unlock();
397        }
398
399        dumpInternals(fd, args);
400
401        // dump playback threads
402        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
403            mPlaybackThreads.valueAt(i)->dump(fd, args);
404        }
405
406        // dump record threads
407        for (size_t i = 0; i < mRecordThreads.size(); i++) {
408            mRecordThreads.valueAt(i)->dump(fd, args);
409        }
410
411        // dump all hardware devs
412        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
413            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
414            dev->dump(dev, fd);
415        }
416
417#ifdef TEE_SINK
418        // dump the serially shared record tee sink
419        if (mRecordTeeSource != 0) {
420            dumpTee(fd, mRecordTeeSource);
421        }
422#endif
423
424        if (locked) {
425            mLock.unlock();
426        }
427
428        // append a copy of media.log here by forwarding fd to it, but don't attempt
429        // to lookup the service if it's not running, as it will block for a second
430        if (mLogMemoryDealer != 0) {
431            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
432            if (binder != 0) {
433                fdprintf(fd, "\nmedia.log:\n");
434                Vector<String16> args;
435                binder->dump(fd, args);
436            }
437        }
438    }
439    return NO_ERROR;
440}
441
442sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
443{
444    Mutex::Autolock _cl(mClientLock);
445    // If pid is already in the mClients wp<> map, then use that entry
446    // (for which promote() is always != 0), otherwise create a new entry and Client.
447    sp<Client> client = mClients.valueFor(pid).promote();
448    if (client == 0) {
449        client = new Client(this, pid);
450        mClients.add(pid, client);
451    }
452
453    return client;
454}
455
456sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
457{
458    // If there is no memory allocated for logs, return a dummy writer that does nothing
459    if (mLogMemoryDealer == 0) {
460        return new NBLog::Writer();
461    }
462    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
463    // Similarly if we can't contact the media.log service, also return a dummy writer
464    if (binder == 0) {
465        return new NBLog::Writer();
466    }
467    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
468    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
469    // If allocation fails, consult the vector of previously unregistered writers
470    // and garbage-collect one or more them until an allocation succeeds
471    if (shared == 0) {
472        Mutex::Autolock _l(mUnregisteredWritersLock);
473        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
474            {
475                // Pick the oldest stale writer to garbage-collect
476                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
477                mUnregisteredWriters.removeAt(0);
478                mediaLogService->unregisterWriter(iMemory);
479                // Now the media.log remote reference to IMemory is gone.  When our last local
480                // reference to IMemory also drops to zero at end of this block,
481                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
482            }
483            // Re-attempt the allocation
484            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
485            if (shared != 0) {
486                goto success;
487            }
488        }
489        // Even after garbage-collecting all old writers, there is still not enough memory,
490        // so return a dummy writer
491        return new NBLog::Writer();
492    }
493success:
494    mediaLogService->registerWriter(shared, size, name);
495    return new NBLog::Writer(size, shared);
496}
497
498void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
499{
500    if (writer == 0) {
501        return;
502    }
503    sp<IMemory> iMemory(writer->getIMemory());
504    if (iMemory == 0) {
505        return;
506    }
507    // Rather than removing the writer immediately, append it to a queue of old writers to
508    // be garbage-collected later.  This allows us to continue to view old logs for a while.
509    Mutex::Autolock _l(mUnregisteredWritersLock);
510    mUnregisteredWriters.push(writer);
511}
512
513// IAudioFlinger interface
514
515
516sp<IAudioTrack> AudioFlinger::createTrack(
517        audio_stream_type_t streamType,
518        uint32_t sampleRate,
519        audio_format_t format,
520        audio_channel_mask_t channelMask,
521        size_t *frameCount,
522        IAudioFlinger::track_flags_t *flags,
523        const sp<IMemory>& sharedBuffer,
524        audio_io_handle_t output,
525        pid_t tid,
526        int *sessionId,
527        int clientUid,
528        status_t *status)
529{
530    sp<PlaybackThread::Track> track;
531    sp<TrackHandle> trackHandle;
532    sp<Client> client;
533    status_t lStatus;
534    int lSessionId;
535
536    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
537    // but if someone uses binder directly they could bypass that and cause us to crash
538    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
539        ALOGE("createTrack() invalid stream type %d", streamType);
540        lStatus = BAD_VALUE;
541        goto Exit;
542    }
543
544    // further sample rate checks are performed by createTrack_l() depending on the thread type
545    if (sampleRate == 0) {
546        ALOGE("createTrack() invalid sample rate %u", sampleRate);
547        lStatus = BAD_VALUE;
548        goto Exit;
549    }
550
551    // further channel mask checks are performed by createTrack_l() depending on the thread type
552    if (!audio_is_output_channel(channelMask)) {
553        ALOGE("createTrack() invalid channel mask %#x", channelMask);
554        lStatus = BAD_VALUE;
555        goto Exit;
556    }
557
558    // further format checks are performed by createTrack_l() depending on the thread type
559    if (!audio_is_valid_format(format)) {
560        ALOGE("createTrack() invalid format %#x", format);
561        lStatus = BAD_VALUE;
562        goto Exit;
563    }
564
565    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
566        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
567        lStatus = BAD_VALUE;
568        goto Exit;
569    }
570
571    {
572        Mutex::Autolock _l(mLock);
573        PlaybackThread *thread = checkPlaybackThread_l(output);
574        if (thread == NULL) {
575            ALOGE("no playback thread found for output handle %d", output);
576            lStatus = BAD_VALUE;
577            goto Exit;
578        }
579
580        pid_t pid = IPCThreadState::self()->getCallingPid();
581        client = registerPid(pid);
582
583        PlaybackThread *effectThread = NULL;
584        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
585            lSessionId = *sessionId;
586            // check if an effect chain with the same session ID is present on another
587            // output thread and move it here.
588            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
589                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
590                if (mPlaybackThreads.keyAt(i) != output) {
591                    uint32_t sessions = t->hasAudioSession(lSessionId);
592                    if (sessions & PlaybackThread::EFFECT_SESSION) {
593                        effectThread = t.get();
594                        break;
595                    }
596                }
597            }
598        } else {
599            // if no audio session id is provided, create one here
600            lSessionId = nextUniqueId();
601            if (sessionId != NULL) {
602                *sessionId = lSessionId;
603            }
604        }
605        ALOGV("createTrack() lSessionId: %d", lSessionId);
606
607        track = thread->createTrack_l(client, streamType, sampleRate, format,
608                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
609        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
610        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
611
612        // move effect chain to this output thread if an effect on same session was waiting
613        // for a track to be created
614        if (lStatus == NO_ERROR && effectThread != NULL) {
615            // no risk of deadlock because AudioFlinger::mLock is held
616            Mutex::Autolock _dl(thread->mLock);
617            Mutex::Autolock _sl(effectThread->mLock);
618            moveEffectChain_l(lSessionId, effectThread, thread, true);
619        }
620
621        // Look for sync events awaiting for a session to be used.
622        for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
623            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
624                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
625                    if (lStatus == NO_ERROR) {
626                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
627                    } else {
628                        mPendingSyncEvents[i]->cancel();
629                    }
630                    mPendingSyncEvents.removeAt(i);
631                    i--;
632                }
633            }
634        }
635
636    }
637
638    if (lStatus != NO_ERROR) {
639        // remove local strong reference to Client before deleting the Track so that the
640        // Client destructor is called by the TrackBase destructor with mClientLock held
641        // Don't hold mClientLock when releasing the reference on the track as the
642        // destructor will acquire it.
643        {
644            Mutex::Autolock _cl(mClientLock);
645            client.clear();
646        }
647        track.clear();
648        goto Exit;
649    }
650
651    // return handle to client
652    trackHandle = new TrackHandle(track);
653
654Exit:
655    *status = lStatus;
656    return trackHandle;
657}
658
659uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
660{
661    Mutex::Autolock _l(mLock);
662    PlaybackThread *thread = checkPlaybackThread_l(output);
663    if (thread == NULL) {
664        ALOGW("sampleRate() unknown thread %d", output);
665        return 0;
666    }
667    return thread->sampleRate();
668}
669
670int AudioFlinger::channelCount(audio_io_handle_t output) const
671{
672    Mutex::Autolock _l(mLock);
673    PlaybackThread *thread = checkPlaybackThread_l(output);
674    if (thread == NULL) {
675        ALOGW("channelCount() unknown thread %d", output);
676        return 0;
677    }
678    return thread->channelCount();
679}
680
681audio_format_t AudioFlinger::format(audio_io_handle_t output) const
682{
683    Mutex::Autolock _l(mLock);
684    PlaybackThread *thread = checkPlaybackThread_l(output);
685    if (thread == NULL) {
686        ALOGW("format() unknown thread %d", output);
687        return AUDIO_FORMAT_INVALID;
688    }
689    return thread->format();
690}
691
692size_t AudioFlinger::frameCount(audio_io_handle_t output) const
693{
694    Mutex::Autolock _l(mLock);
695    PlaybackThread *thread = checkPlaybackThread_l(output);
696    if (thread == NULL) {
697        ALOGW("frameCount() unknown thread %d", output);
698        return 0;
699    }
700    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
701    //       should examine all callers and fix them to handle smaller counts
702    return thread->frameCount();
703}
704
705uint32_t AudioFlinger::latency(audio_io_handle_t output) const
706{
707    Mutex::Autolock _l(mLock);
708    PlaybackThread *thread = checkPlaybackThread_l(output);
709    if (thread == NULL) {
710        ALOGW("latency(): no playback thread found for output handle %d", output);
711        return 0;
712    }
713    return thread->latency();
714}
715
716status_t AudioFlinger::setMasterVolume(float value)
717{
718    status_t ret = initCheck();
719    if (ret != NO_ERROR) {
720        return ret;
721    }
722
723    // check calling permissions
724    if (!settingsAllowed()) {
725        return PERMISSION_DENIED;
726    }
727
728    Mutex::Autolock _l(mLock);
729    mMasterVolume = value;
730
731    // Set master volume in the HALs which support it.
732    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
733        AutoMutex lock(mHardwareLock);
734        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
735
736        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
737        if (dev->canSetMasterVolume()) {
738            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
739        }
740        mHardwareStatus = AUDIO_HW_IDLE;
741    }
742
743    // Now set the master volume in each playback thread.  Playback threads
744    // assigned to HALs which do not have master volume support will apply
745    // master volume during the mix operation.  Threads with HALs which do
746    // support master volume will simply ignore the setting.
747    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
748        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
749
750    return NO_ERROR;
751}
752
753status_t AudioFlinger::setMode(audio_mode_t mode)
754{
755    status_t ret = initCheck();
756    if (ret != NO_ERROR) {
757        return ret;
758    }
759
760    // check calling permissions
761    if (!settingsAllowed()) {
762        return PERMISSION_DENIED;
763    }
764    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
765        ALOGW("Illegal value: setMode(%d)", mode);
766        return BAD_VALUE;
767    }
768
769    { // scope for the lock
770        AutoMutex lock(mHardwareLock);
771        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
772        mHardwareStatus = AUDIO_HW_SET_MODE;
773        ret = dev->set_mode(dev, mode);
774        mHardwareStatus = AUDIO_HW_IDLE;
775    }
776
777    if (NO_ERROR == ret) {
778        Mutex::Autolock _l(mLock);
779        mMode = mode;
780        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
781            mPlaybackThreads.valueAt(i)->setMode(mode);
782    }
783
784    return ret;
785}
786
787status_t AudioFlinger::setMicMute(bool state)
788{
789    status_t ret = initCheck();
790    if (ret != NO_ERROR) {
791        return ret;
792    }
793
794    // check calling permissions
795    if (!settingsAllowed()) {
796        return PERMISSION_DENIED;
797    }
798
799    AutoMutex lock(mHardwareLock);
800    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
801    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
802    ret = dev->set_mic_mute(dev, state);
803    mHardwareStatus = AUDIO_HW_IDLE;
804    return ret;
805}
806
807bool AudioFlinger::getMicMute() const
808{
809    status_t ret = initCheck();
810    if (ret != NO_ERROR) {
811        return false;
812    }
813
814    bool state = AUDIO_MODE_INVALID;
815    AutoMutex lock(mHardwareLock);
816    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
817    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
818    dev->get_mic_mute(dev, &state);
819    mHardwareStatus = AUDIO_HW_IDLE;
820    return state;
821}
822
823status_t AudioFlinger::setMasterMute(bool muted)
824{
825    status_t ret = initCheck();
826    if (ret != NO_ERROR) {
827        return ret;
828    }
829
830    // check calling permissions
831    if (!settingsAllowed()) {
832        return PERMISSION_DENIED;
833    }
834
835    Mutex::Autolock _l(mLock);
836    mMasterMute = muted;
837
838    // Set master mute in the HALs which support it.
839    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
840        AutoMutex lock(mHardwareLock);
841        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
842
843        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
844        if (dev->canSetMasterMute()) {
845            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
846        }
847        mHardwareStatus = AUDIO_HW_IDLE;
848    }
849
850    // Now set the master mute in each playback thread.  Playback threads
851    // assigned to HALs which do not have master mute support will apply master
852    // mute during the mix operation.  Threads with HALs which do support master
853    // mute will simply ignore the setting.
854    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
855        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
856
857    return NO_ERROR;
858}
859
860float AudioFlinger::masterVolume() const
861{
862    Mutex::Autolock _l(mLock);
863    return masterVolume_l();
864}
865
866bool AudioFlinger::masterMute() const
867{
868    Mutex::Autolock _l(mLock);
869    return masterMute_l();
870}
871
872float AudioFlinger::masterVolume_l() const
873{
874    return mMasterVolume;
875}
876
877bool AudioFlinger::masterMute_l() const
878{
879    return mMasterMute;
880}
881
882status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
883        audio_io_handle_t output)
884{
885    // check calling permissions
886    if (!settingsAllowed()) {
887        return PERMISSION_DENIED;
888    }
889
890    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
891        ALOGE("setStreamVolume() invalid stream %d", stream);
892        return BAD_VALUE;
893    }
894
895    AutoMutex lock(mLock);
896    PlaybackThread *thread = NULL;
897    if (output != AUDIO_IO_HANDLE_NONE) {
898        thread = checkPlaybackThread_l(output);
899        if (thread == NULL) {
900            return BAD_VALUE;
901        }
902    }
903
904    mStreamTypes[stream].volume = value;
905
906    if (thread == NULL) {
907        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
908            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
909        }
910    } else {
911        thread->setStreamVolume(stream, value);
912    }
913
914    return NO_ERROR;
915}
916
917status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
918{
919    // check calling permissions
920    if (!settingsAllowed()) {
921        return PERMISSION_DENIED;
922    }
923
924    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
925        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
926        ALOGE("setStreamMute() invalid stream %d", stream);
927        return BAD_VALUE;
928    }
929
930    AutoMutex lock(mLock);
931    mStreamTypes[stream].mute = muted;
932    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
933        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
934
935    return NO_ERROR;
936}
937
938float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
939{
940    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
941        return 0.0f;
942    }
943
944    AutoMutex lock(mLock);
945    float volume;
946    if (output != AUDIO_IO_HANDLE_NONE) {
947        PlaybackThread *thread = checkPlaybackThread_l(output);
948        if (thread == NULL) {
949            return 0.0f;
950        }
951        volume = thread->streamVolume(stream);
952    } else {
953        volume = streamVolume_l(stream);
954    }
955
956    return volume;
957}
958
959bool AudioFlinger::streamMute(audio_stream_type_t stream) const
960{
961    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
962        return true;
963    }
964
965    AutoMutex lock(mLock);
966    return streamMute_l(stream);
967}
968
969status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
970{
971    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
972            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
973
974    // check calling permissions
975    if (!settingsAllowed()) {
976        return PERMISSION_DENIED;
977    }
978
979    // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
980    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
981        Mutex::Autolock _l(mLock);
982        status_t final_result = NO_ERROR;
983        {
984            AutoMutex lock(mHardwareLock);
985            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
986            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
987                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
988                status_t result = dev->set_parameters(dev, keyValuePairs.string());
989                final_result = result ?: final_result;
990            }
991            mHardwareStatus = AUDIO_HW_IDLE;
992        }
993        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
994        AudioParameter param = AudioParameter(keyValuePairs);
995        String8 value;
996        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
997            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
998            if (mBtNrecIsOff != btNrecIsOff) {
999                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1000                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
1001                    audio_devices_t device = thread->inDevice();
1002                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
1003                    // collect all of the thread's session IDs
1004                    KeyedVector<int, bool> ids = thread->sessionIds();
1005                    // suspend effects associated with those session IDs
1006                    for (size_t j = 0; j < ids.size(); ++j) {
1007                        int sessionId = ids.keyAt(j);
1008                        thread->setEffectSuspended(FX_IID_AEC,
1009                                                   suspend,
1010                                                   sessionId);
1011                        thread->setEffectSuspended(FX_IID_NS,
1012                                                   suspend,
1013                                                   sessionId);
1014                    }
1015                }
1016                mBtNrecIsOff = btNrecIsOff;
1017            }
1018        }
1019        String8 screenState;
1020        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1021            bool isOff = screenState == "off";
1022            if (isOff != (AudioFlinger::mScreenState & 1)) {
1023                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1024            }
1025        }
1026        return final_result;
1027    }
1028
1029    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1030    // and the thread is exited once the lock is released
1031    sp<ThreadBase> thread;
1032    {
1033        Mutex::Autolock _l(mLock);
1034        thread = checkPlaybackThread_l(ioHandle);
1035        if (thread == 0) {
1036            thread = checkRecordThread_l(ioHandle);
1037        } else if (thread == primaryPlaybackThread_l()) {
1038            // indicate output device change to all input threads for pre processing
1039            AudioParameter param = AudioParameter(keyValuePairs);
1040            int value;
1041            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1042                    (value != 0)) {
1043                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1044                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1045                }
1046            }
1047        }
1048    }
1049    if (thread != 0) {
1050        return thread->setParameters(keyValuePairs);
1051    }
1052    return BAD_VALUE;
1053}
1054
1055String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1056{
1057    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1058            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1059
1060    Mutex::Autolock _l(mLock);
1061
1062    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1063        String8 out_s8;
1064
1065        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1066            char *s;
1067            {
1068            AutoMutex lock(mHardwareLock);
1069            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1070            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1071            s = dev->get_parameters(dev, keys.string());
1072            mHardwareStatus = AUDIO_HW_IDLE;
1073            }
1074            out_s8 += String8(s ? s : "");
1075            free(s);
1076        }
1077        return out_s8;
1078    }
1079
1080    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1081    if (playbackThread != NULL) {
1082        return playbackThread->getParameters(keys);
1083    }
1084    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1085    if (recordThread != NULL) {
1086        return recordThread->getParameters(keys);
1087    }
1088    return String8("");
1089}
1090
1091size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1092        audio_channel_mask_t channelMask) const
1093{
1094    status_t ret = initCheck();
1095    if (ret != NO_ERROR) {
1096        return 0;
1097    }
1098
1099    AutoMutex lock(mHardwareLock);
1100    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1101    struct audio_config config;
1102    memset(&config, 0, sizeof(config));
1103    config.sample_rate = sampleRate;
1104    config.channel_mask = channelMask;
1105    config.format = format;
1106
1107    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1108    size_t size = dev->get_input_buffer_size(dev, &config);
1109    mHardwareStatus = AUDIO_HW_IDLE;
1110    return size;
1111}
1112
1113uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1114{
1115    Mutex::Autolock _l(mLock);
1116
1117    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1118    if (recordThread != NULL) {
1119        return recordThread->getInputFramesLost();
1120    }
1121    return 0;
1122}
1123
1124status_t AudioFlinger::setVoiceVolume(float value)
1125{
1126    status_t ret = initCheck();
1127    if (ret != NO_ERROR) {
1128        return ret;
1129    }
1130
1131    // check calling permissions
1132    if (!settingsAllowed()) {
1133        return PERMISSION_DENIED;
1134    }
1135
1136    AutoMutex lock(mHardwareLock);
1137    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1138    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1139    ret = dev->set_voice_volume(dev, value);
1140    mHardwareStatus = AUDIO_HW_IDLE;
1141
1142    return ret;
1143}
1144
1145status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1146        audio_io_handle_t output) const
1147{
1148    status_t status;
1149
1150    Mutex::Autolock _l(mLock);
1151
1152    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1153    if (playbackThread != NULL) {
1154        return playbackThread->getRenderPosition(halFrames, dspFrames);
1155    }
1156
1157    return BAD_VALUE;
1158}
1159
1160void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1161{
1162    Mutex::Autolock _l(mLock);
1163    bool clientAdded = false;
1164    {
1165        Mutex::Autolock _cl(mClientLock);
1166
1167        pid_t pid = IPCThreadState::self()->getCallingPid();
1168        if (mNotificationClients.indexOfKey(pid) < 0) {
1169            sp<NotificationClient> notificationClient = new NotificationClient(this,
1170                                                                                client,
1171                                                                                pid);
1172            ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1173
1174            mNotificationClients.add(pid, notificationClient);
1175
1176            sp<IBinder> binder = client->asBinder();
1177            binder->linkToDeath(notificationClient);
1178            clientAdded = true;
1179        }
1180    }
1181
1182    // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
1183    // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock.
1184    if (clientAdded) {
1185        // the config change is always sent from playback or record threads to avoid deadlock
1186        // with AudioSystem::gLock
1187        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1188            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1189        }
1190
1191        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1192            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1193        }
1194    }
1195}
1196
1197void AudioFlinger::removeNotificationClient(pid_t pid)
1198{
1199    Mutex::Autolock _l(mLock);
1200    {
1201        Mutex::Autolock _cl(mClientLock);
1202        mNotificationClients.removeItem(pid);
1203    }
1204
1205    ALOGV("%d died, releasing its sessions", pid);
1206    size_t num = mAudioSessionRefs.size();
1207    bool removed = false;
1208    for (size_t i = 0; i< num; ) {
1209        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1210        ALOGV(" pid %d @ %d", ref->mPid, i);
1211        if (ref->mPid == pid) {
1212            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1213            mAudioSessionRefs.removeAt(i);
1214            delete ref;
1215            removed = true;
1216            num--;
1217        } else {
1218            i++;
1219        }
1220    }
1221    if (removed) {
1222        purgeStaleEffects_l();
1223    }
1224}
1225
1226void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2)
1227{
1228    Mutex::Autolock _l(mClientLock);
1229    size_t size = mNotificationClients.size();
1230    for (size_t i = 0; i < size; i++) {
1231        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event,
1232                                                                              ioHandle,
1233                                                                              param2);
1234    }
1235}
1236
1237// removeClient_l() must be called with AudioFlinger::mClientLock held
1238void AudioFlinger::removeClient_l(pid_t pid)
1239{
1240    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1241            IPCThreadState::self()->getCallingPid());
1242    mClients.removeItem(pid);
1243}
1244
1245// getEffectThread_l() must be called with AudioFlinger::mLock held
1246sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1247{
1248    sp<PlaybackThread> thread;
1249
1250    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1251        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1252            ALOG_ASSERT(thread == 0);
1253            thread = mPlaybackThreads.valueAt(i);
1254        }
1255    }
1256
1257    return thread;
1258}
1259
1260
1261
1262// ----------------------------------------------------------------------------
1263
1264AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1265    :   RefBase(),
1266        mAudioFlinger(audioFlinger),
1267        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1268        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1269        mPid(pid),
1270        mTimedTrackCount(0)
1271{
1272    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1273}
1274
1275// Client destructor must be called with AudioFlinger::mClientLock held
1276AudioFlinger::Client::~Client()
1277{
1278    mAudioFlinger->removeClient_l(mPid);
1279}
1280
1281sp<MemoryDealer> AudioFlinger::Client::heap() const
1282{
1283    return mMemoryDealer;
1284}
1285
1286// Reserve one of the limited slots for a timed audio track associated
1287// with this client
1288bool AudioFlinger::Client::reserveTimedTrack()
1289{
1290    const int kMaxTimedTracksPerClient = 4;
1291
1292    Mutex::Autolock _l(mTimedTrackLock);
1293
1294    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1295        ALOGW("can not create timed track - pid %d has exceeded the limit",
1296             mPid);
1297        return false;
1298    }
1299
1300    mTimedTrackCount++;
1301    return true;
1302}
1303
1304// Release a slot for a timed audio track
1305void AudioFlinger::Client::releaseTimedTrack()
1306{
1307    Mutex::Autolock _l(mTimedTrackLock);
1308    mTimedTrackCount--;
1309}
1310
1311// ----------------------------------------------------------------------------
1312
1313AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1314                                                     const sp<IAudioFlingerClient>& client,
1315                                                     pid_t pid)
1316    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1317{
1318}
1319
1320AudioFlinger::NotificationClient::~NotificationClient()
1321{
1322}
1323
1324void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1325{
1326    sp<NotificationClient> keep(this);
1327    mAudioFlinger->removeNotificationClient(mPid);
1328}
1329
1330
1331// ----------------------------------------------------------------------------
1332
1333static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1334    return audio_is_remote_submix_device(inDevice);
1335}
1336
1337sp<IAudioRecord> AudioFlinger::openRecord(
1338        audio_io_handle_t input,
1339        uint32_t sampleRate,
1340        audio_format_t format,
1341        audio_channel_mask_t channelMask,
1342        size_t *frameCount,
1343        IAudioFlinger::track_flags_t *flags,
1344        pid_t tid,
1345        int *sessionId,
1346        sp<IMemory>& cblk,
1347        sp<IMemory>& buffers,
1348        status_t *status)
1349{
1350    sp<RecordThread::RecordTrack> recordTrack;
1351    sp<RecordHandle> recordHandle;
1352    sp<Client> client;
1353    status_t lStatus;
1354    int lSessionId;
1355
1356    cblk.clear();
1357    buffers.clear();
1358
1359    // check calling permissions
1360    if (!recordingAllowed()) {
1361        ALOGE("openRecord() permission denied: recording not allowed");
1362        lStatus = PERMISSION_DENIED;
1363        goto Exit;
1364    }
1365
1366    // further sample rate checks are performed by createRecordTrack_l()
1367    if (sampleRate == 0) {
1368        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1369        lStatus = BAD_VALUE;
1370        goto Exit;
1371    }
1372
1373    // we don't yet support anything other than 16-bit PCM
1374    if (!(audio_is_valid_format(format) &&
1375            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
1376        ALOGE("openRecord() invalid format %#x", format);
1377        lStatus = BAD_VALUE;
1378        goto Exit;
1379    }
1380
1381    // further channel mask checks are performed by createRecordTrack_l()
1382    if (!audio_is_input_channel(channelMask)) {
1383        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1384        lStatus = BAD_VALUE;
1385        goto Exit;
1386    }
1387
1388    {
1389        Mutex::Autolock _l(mLock);
1390        RecordThread *thread = checkRecordThread_l(input);
1391        if (thread == NULL) {
1392            ALOGE("openRecord() checkRecordThread_l failed");
1393            lStatus = BAD_VALUE;
1394            goto Exit;
1395        }
1396
1397        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1398                && !captureAudioOutputAllowed()) {
1399            ALOGE("openRecord() permission denied: capture not allowed");
1400            lStatus = PERMISSION_DENIED;
1401            goto Exit;
1402        }
1403
1404        pid_t pid = IPCThreadState::self()->getCallingPid();
1405        client = registerPid(pid);
1406
1407        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1408            lSessionId = *sessionId;
1409        } else {
1410            // if no audio session id is provided, create one here
1411            lSessionId = nextUniqueId();
1412            if (sessionId != NULL) {
1413                *sessionId = lSessionId;
1414            }
1415        }
1416        ALOGV("openRecord() lSessionId: %d", lSessionId);
1417
1418        // TODO: the uid should be passed in as a parameter to openRecord
1419        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1420                                                  frameCount, lSessionId,
1421                                                  IPCThreadState::self()->getCallingUid(),
1422                                                  flags, tid, &lStatus);
1423        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1424    }
1425
1426    if (lStatus != NO_ERROR) {
1427        // remove local strong reference to Client before deleting the RecordTrack so that the
1428        // Client destructor is called by the TrackBase destructor with mClientLock held
1429        // Don't hold mClientLock when releasing the reference on the track as the
1430        // destructor will acquire it.
1431        {
1432            Mutex::Autolock _cl(mClientLock);
1433            client.clear();
1434        }
1435        recordTrack.clear();
1436        goto Exit;
1437    }
1438
1439    cblk = recordTrack->getCblk();
1440    buffers = recordTrack->getBuffers();
1441
1442    // return handle to client
1443    recordHandle = new RecordHandle(recordTrack);
1444
1445Exit:
1446    *status = lStatus;
1447    return recordHandle;
1448}
1449
1450
1451
1452// ----------------------------------------------------------------------------
1453
1454audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1455{
1456    if (!settingsAllowed()) {
1457        return 0;
1458    }
1459    Mutex::Autolock _l(mLock);
1460    return loadHwModule_l(name);
1461}
1462
1463// loadHwModule_l() must be called with AudioFlinger::mLock held
1464audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1465{
1466    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1467        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1468            ALOGW("loadHwModule() module %s already loaded", name);
1469            return mAudioHwDevs.keyAt(i);
1470        }
1471    }
1472
1473    audio_hw_device_t *dev;
1474
1475    int rc = load_audio_interface(name, &dev);
1476    if (rc) {
1477        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1478        return 0;
1479    }
1480
1481    mHardwareStatus = AUDIO_HW_INIT;
1482    rc = dev->init_check(dev);
1483    mHardwareStatus = AUDIO_HW_IDLE;
1484    if (rc) {
1485        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1486        return 0;
1487    }
1488
1489    // Check and cache this HAL's level of support for master mute and master
1490    // volume.  If this is the first HAL opened, and it supports the get
1491    // methods, use the initial values provided by the HAL as the current
1492    // master mute and volume settings.
1493
1494    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1495    {  // scope for auto-lock pattern
1496        AutoMutex lock(mHardwareLock);
1497
1498        if (0 == mAudioHwDevs.size()) {
1499            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1500            if (NULL != dev->get_master_volume) {
1501                float mv;
1502                if (OK == dev->get_master_volume(dev, &mv)) {
1503                    mMasterVolume = mv;
1504                }
1505            }
1506
1507            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1508            if (NULL != dev->get_master_mute) {
1509                bool mm;
1510                if (OK == dev->get_master_mute(dev, &mm)) {
1511                    mMasterMute = mm;
1512                }
1513            }
1514        }
1515
1516        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1517        if ((NULL != dev->set_master_volume) &&
1518            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1519            flags = static_cast<AudioHwDevice::Flags>(flags |
1520                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1521        }
1522
1523        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1524        if ((NULL != dev->set_master_mute) &&
1525            (OK == dev->set_master_mute(dev, mMasterMute))) {
1526            flags = static_cast<AudioHwDevice::Flags>(flags |
1527                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1528        }
1529
1530        mHardwareStatus = AUDIO_HW_IDLE;
1531    }
1532
1533    audio_module_handle_t handle = nextUniqueId();
1534    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1535
1536    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1537          name, dev->common.module->name, dev->common.module->id, handle);
1538
1539    return handle;
1540
1541}
1542
1543// ----------------------------------------------------------------------------
1544
1545uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1546{
1547    Mutex::Autolock _l(mLock);
1548    PlaybackThread *thread = primaryPlaybackThread_l();
1549    return thread != NULL ? thread->sampleRate() : 0;
1550}
1551
1552size_t AudioFlinger::getPrimaryOutputFrameCount()
1553{
1554    Mutex::Autolock _l(mLock);
1555    PlaybackThread *thread = primaryPlaybackThread_l();
1556    return thread != NULL ? thread->frameCountHAL() : 0;
1557}
1558
1559// ----------------------------------------------------------------------------
1560
1561status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1562{
1563    uid_t uid = IPCThreadState::self()->getCallingUid();
1564    if (uid != AID_SYSTEM) {
1565        return PERMISSION_DENIED;
1566    }
1567    Mutex::Autolock _l(mLock);
1568    if (mIsDeviceTypeKnown) {
1569        return INVALID_OPERATION;
1570    }
1571    mIsLowRamDevice = isLowRamDevice;
1572    mIsDeviceTypeKnown = true;
1573    return NO_ERROR;
1574}
1575
1576// ----------------------------------------------------------------------------
1577
1578audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1579                                           audio_devices_t *pDevices,
1580                                           uint32_t *pSamplingRate,
1581                                           audio_format_t *pFormat,
1582                                           audio_channel_mask_t *pChannelMask,
1583                                           uint32_t *pLatencyMs,
1584                                           audio_output_flags_t flags,
1585                                           const audio_offload_info_t *offloadInfo)
1586{
1587    struct audio_config config;
1588    memset(&config, 0, sizeof(config));
1589    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1590    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1591    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1592    if (offloadInfo != NULL) {
1593        config.offload_info = *offloadInfo;
1594    }
1595
1596    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1597              module,
1598              (pDevices != NULL) ? *pDevices : 0,
1599              config.sample_rate,
1600              config.format,
1601              config.channel_mask,
1602              flags);
1603    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1604          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1605
1606    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1607        return AUDIO_IO_HANDLE_NONE;
1608    }
1609
1610    Mutex::Autolock _l(mLock);
1611
1612    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1613    if (outHwDev == NULL) {
1614        return AUDIO_IO_HANDLE_NONE;
1615    }
1616
1617    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1618    audio_io_handle_t id = nextUniqueId();
1619
1620    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1621
1622    audio_stream_out_t *outStream = NULL;
1623    status_t status = hwDevHal->open_output_stream(hwDevHal,
1624                                          id,
1625                                          *pDevices,
1626                                          (audio_output_flags_t)flags,
1627                                          &config,
1628                                          &outStream);
1629
1630    mHardwareStatus = AUDIO_HW_IDLE;
1631    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1632            "Channels %x, status %d",
1633            outStream,
1634            config.sample_rate,
1635            config.format,
1636            config.channel_mask,
1637            status);
1638
1639    if (status == NO_ERROR && outStream != NULL) {
1640        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1641
1642        PlaybackThread *thread;
1643        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1644            thread = new OffloadThread(this, output, id, *pDevices);
1645            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1646        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1647            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1648            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1649            thread = new DirectOutputThread(this, output, id, *pDevices);
1650            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1651        } else {
1652            thread = new MixerThread(this, output, id, *pDevices);
1653            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1654        }
1655        mPlaybackThreads.add(id, thread);
1656
1657        if (pSamplingRate != NULL) {
1658            *pSamplingRate = config.sample_rate;
1659        }
1660        if (pFormat != NULL) {
1661            *pFormat = config.format;
1662        }
1663        if (pChannelMask != NULL) {
1664            *pChannelMask = config.channel_mask;
1665        }
1666        if (pLatencyMs != NULL) {
1667            *pLatencyMs = thread->latency();
1668        }
1669
1670        // notify client processes of the new output creation
1671        thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED);
1672
1673        // the first primary output opened designates the primary hw device
1674        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1675            ALOGI("Using module %d has the primary audio interface", module);
1676            mPrimaryHardwareDev = outHwDev;
1677
1678            AutoMutex lock(mHardwareLock);
1679            mHardwareStatus = AUDIO_HW_SET_MODE;
1680            hwDevHal->set_mode(hwDevHal, mMode);
1681            mHardwareStatus = AUDIO_HW_IDLE;
1682        }
1683        return id;
1684    }
1685
1686    return AUDIO_IO_HANDLE_NONE;
1687}
1688
1689audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1690        audio_io_handle_t output2)
1691{
1692    Mutex::Autolock _l(mLock);
1693    MixerThread *thread1 = checkMixerThread_l(output1);
1694    MixerThread *thread2 = checkMixerThread_l(output2);
1695
1696    if (thread1 == NULL || thread2 == NULL) {
1697        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1698                output2);
1699        return AUDIO_IO_HANDLE_NONE;
1700    }
1701
1702    audio_io_handle_t id = nextUniqueId();
1703    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1704    thread->addOutputTrack(thread2);
1705    mPlaybackThreads.add(id, thread);
1706    // notify client processes of the new output creation
1707    thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED);
1708    return id;
1709}
1710
1711status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1712{
1713    return closeOutput_nonvirtual(output);
1714}
1715
1716status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1717{
1718    // keep strong reference on the playback thread so that
1719    // it is not destroyed while exit() is executed
1720    sp<PlaybackThread> thread;
1721    {
1722        Mutex::Autolock _l(mLock);
1723        thread = checkPlaybackThread_l(output);
1724        if (thread == NULL) {
1725            return BAD_VALUE;
1726        }
1727
1728        ALOGV("closeOutput() %d", output);
1729
1730        if (thread->type() == ThreadBase::MIXER) {
1731            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1732                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1733                    DuplicatingThread *dupThread =
1734                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1735                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1736
1737                }
1738            }
1739        }
1740
1741
1742        mPlaybackThreads.removeItem(output);
1743        // save all effects to the default thread
1744        if (mPlaybackThreads.size()) {
1745            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1746            if (dstThread != NULL) {
1747                // audioflinger lock is held here so the acquisition order of thread locks does not
1748                // matter
1749                Mutex::Autolock _dl(dstThread->mLock);
1750                Mutex::Autolock _sl(thread->mLock);
1751                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1752                for (size_t i = 0; i < effectChains.size(); i ++) {
1753                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1754                }
1755            }
1756        }
1757        audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL);
1758    }
1759    thread->exit();
1760    // The thread entity (active unit of execution) is no longer running here,
1761    // but the ThreadBase container still exists.
1762
1763    if (thread->type() != ThreadBase::DUPLICATING) {
1764        AudioStreamOut *out = thread->clearOutput();
1765        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1766        // from now on thread->mOutput is NULL
1767        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1768        delete out;
1769    }
1770    return NO_ERROR;
1771}
1772
1773status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1774{
1775    Mutex::Autolock _l(mLock);
1776    PlaybackThread *thread = checkPlaybackThread_l(output);
1777
1778    if (thread == NULL) {
1779        return BAD_VALUE;
1780    }
1781
1782    ALOGV("suspendOutput() %d", output);
1783    thread->suspend();
1784
1785    return NO_ERROR;
1786}
1787
1788status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1789{
1790    Mutex::Autolock _l(mLock);
1791    PlaybackThread *thread = checkPlaybackThread_l(output);
1792
1793    if (thread == NULL) {
1794        return BAD_VALUE;
1795    }
1796
1797    ALOGV("restoreOutput() %d", output);
1798
1799    thread->restore();
1800
1801    return NO_ERROR;
1802}
1803
1804audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1805                                          audio_devices_t *pDevices,
1806                                          uint32_t *pSamplingRate,
1807                                          audio_format_t *pFormat,
1808                                          audio_channel_mask_t *pChannelMask)
1809{
1810    struct audio_config config;
1811    memset(&config, 0, sizeof(config));
1812    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1813    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1814    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1815
1816    uint32_t reqSamplingRate = config.sample_rate;
1817    audio_format_t reqFormat = config.format;
1818    audio_channel_mask_t reqChannelMask = config.channel_mask;
1819
1820    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1821        return 0;
1822    }
1823
1824    Mutex::Autolock _l(mLock);
1825
1826    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1827    if (inHwDev == NULL) {
1828        return 0;
1829    }
1830
1831    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1832    audio_io_handle_t id = nextUniqueId();
1833
1834    audio_stream_in_t *inStream = NULL;
1835    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1836                                        &inStream);
1837    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
1838            "status %d",
1839            inStream,
1840            config.sample_rate,
1841            config.format,
1842            config.channel_mask,
1843            status);
1844
1845    // If the input could not be opened with the requested parameters and we can handle the
1846    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1847    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1848    if (status == BAD_VALUE &&
1849        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1850        (config.sample_rate <= 2 * reqSamplingRate) &&
1851        (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) &&
1852        (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) {
1853        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1854        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1855        inStream = NULL;
1856        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1857        // FIXME log this new status; HAL should not propose any further changes
1858    }
1859
1860    if (status == NO_ERROR && inStream != NULL) {
1861
1862#ifdef TEE_SINK
1863        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1864        // or (re-)create if current Pipe is idle and does not match the new format
1865        sp<NBAIO_Sink> teeSink;
1866        enum {
1867            TEE_SINK_NO,    // don't copy input
1868            TEE_SINK_NEW,   // copy input using a new pipe
1869            TEE_SINK_OLD,   // copy input using an existing pipe
1870        } kind;
1871        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1872                audio_channel_count_from_in_mask(
1873                        inStream->common.get_channels(&inStream->common)));
1874        if (!mTeeSinkInputEnabled) {
1875            kind = TEE_SINK_NO;
1876        } else if (!Format_isValid(format)) {
1877            kind = TEE_SINK_NO;
1878        } else if (mRecordTeeSink == 0) {
1879            kind = TEE_SINK_NEW;
1880        } else if (mRecordTeeSink->getStrongCount() != 1) {
1881            kind = TEE_SINK_NO;
1882        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
1883            kind = TEE_SINK_OLD;
1884        } else {
1885            kind = TEE_SINK_NEW;
1886        }
1887        switch (kind) {
1888        case TEE_SINK_NEW: {
1889            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1890            size_t numCounterOffers = 0;
1891            const NBAIO_Format offers[1] = {format};
1892            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1893            ALOG_ASSERT(index == 0);
1894            PipeReader *pipeReader = new PipeReader(*pipe);
1895            numCounterOffers = 0;
1896            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1897            ALOG_ASSERT(index == 0);
1898            mRecordTeeSink = pipe;
1899            mRecordTeeSource = pipeReader;
1900            teeSink = pipe;
1901            }
1902            break;
1903        case TEE_SINK_OLD:
1904            teeSink = mRecordTeeSink;
1905            break;
1906        case TEE_SINK_NO:
1907        default:
1908            break;
1909        }
1910#endif
1911
1912        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1913
1914        // Start record thread
1915        // RecordThread requires both input and output device indication to forward to audio
1916        // pre processing modules
1917        RecordThread *thread = new RecordThread(this,
1918                                  input,
1919                                  id,
1920                                  primaryOutputDevice_l(),
1921                                  *pDevices
1922#ifdef TEE_SINK
1923                                  , teeSink
1924#endif
1925                                  );
1926        mRecordThreads.add(id, thread);
1927        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1928        if (pSamplingRate != NULL) {
1929            *pSamplingRate = reqSamplingRate;
1930        }
1931        if (pFormat != NULL) {
1932            *pFormat = config.format;
1933        }
1934        if (pChannelMask != NULL) {
1935            *pChannelMask = reqChannelMask;
1936        }
1937
1938        // notify client processes of the new input creation
1939        thread->audioConfigChanged(AudioSystem::INPUT_OPENED);
1940        return id;
1941    }
1942
1943    return 0;
1944}
1945
1946status_t AudioFlinger::closeInput(audio_io_handle_t input)
1947{
1948    return closeInput_nonvirtual(input);
1949}
1950
1951status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1952{
1953    // keep strong reference on the record thread so that
1954    // it is not destroyed while exit() is executed
1955    sp<RecordThread> thread;
1956    {
1957        Mutex::Autolock _l(mLock);
1958        thread = checkRecordThread_l(input);
1959        if (thread == 0) {
1960            return BAD_VALUE;
1961        }
1962
1963        ALOGV("closeInput() %d", input);
1964        audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL);
1965        mRecordThreads.removeItem(input);
1966    }
1967    thread->exit();
1968    // The thread entity (active unit of execution) is no longer running here,
1969    // but the ThreadBase container still exists.
1970
1971    AudioStreamIn *in = thread->clearInput();
1972    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1973    // from now on thread->mInput is NULL
1974    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1975    delete in;
1976
1977    return NO_ERROR;
1978}
1979
1980status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
1981{
1982    Mutex::Autolock _l(mLock);
1983    ALOGV("invalidateStream() stream %d", stream);
1984
1985    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1986        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1987        thread->invalidateTracks(stream);
1988    }
1989
1990    return NO_ERROR;
1991}
1992
1993
1994int AudioFlinger::newAudioSessionId()
1995{
1996    return nextUniqueId();
1997}
1998
1999void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
2000{
2001    Mutex::Autolock _l(mLock);
2002    pid_t caller = IPCThreadState::self()->getCallingPid();
2003    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
2004    if (pid != -1 && (caller == getpid_cached)) {
2005        caller = pid;
2006    }
2007
2008    {
2009        Mutex::Autolock _cl(mClientLock);
2010        // Ignore requests received from processes not known as notification client. The request
2011        // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
2012        // called from a different pid leaving a stale session reference.  Also we don't know how
2013        // to clear this reference if the client process dies.
2014        if (mNotificationClients.indexOfKey(caller) < 0) {
2015            ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
2016            return;
2017        }
2018    }
2019
2020    size_t num = mAudioSessionRefs.size();
2021    for (size_t i = 0; i< num; i++) {
2022        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
2023        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2024            ref->mCnt++;
2025            ALOGV(" incremented refcount to %d", ref->mCnt);
2026            return;
2027        }
2028    }
2029    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
2030    ALOGV(" added new entry for %d", audioSession);
2031}
2032
2033void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
2034{
2035    Mutex::Autolock _l(mLock);
2036    pid_t caller = IPCThreadState::self()->getCallingPid();
2037    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
2038    if (pid != -1 && (caller == getpid_cached)) {
2039        caller = pid;
2040    }
2041    size_t num = mAudioSessionRefs.size();
2042    for (size_t i = 0; i< num; i++) {
2043        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2044        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2045            ref->mCnt--;
2046            ALOGV(" decremented refcount to %d", ref->mCnt);
2047            if (ref->mCnt == 0) {
2048                mAudioSessionRefs.removeAt(i);
2049                delete ref;
2050                purgeStaleEffects_l();
2051            }
2052            return;
2053        }
2054    }
2055    // If the caller is mediaserver it is likely that the session being released was acquired
2056    // on behalf of a process not in notification clients and we ignore the warning.
2057    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2058}
2059
2060void AudioFlinger::purgeStaleEffects_l() {
2061
2062    ALOGV("purging stale effects");
2063
2064    Vector< sp<EffectChain> > chains;
2065
2066    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2067        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2068        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2069            sp<EffectChain> ec = t->mEffectChains[j];
2070            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2071                chains.push(ec);
2072            }
2073        }
2074    }
2075    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2076        sp<RecordThread> t = mRecordThreads.valueAt(i);
2077        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2078            sp<EffectChain> ec = t->mEffectChains[j];
2079            chains.push(ec);
2080        }
2081    }
2082
2083    for (size_t i = 0; i < chains.size(); i++) {
2084        sp<EffectChain> ec = chains[i];
2085        int sessionid = ec->sessionId();
2086        sp<ThreadBase> t = ec->mThread.promote();
2087        if (t == 0) {
2088            continue;
2089        }
2090        size_t numsessionrefs = mAudioSessionRefs.size();
2091        bool found = false;
2092        for (size_t k = 0; k < numsessionrefs; k++) {
2093            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2094            if (ref->mSessionid == sessionid) {
2095                ALOGV(" session %d still exists for %d with %d refs",
2096                    sessionid, ref->mPid, ref->mCnt);
2097                found = true;
2098                break;
2099            }
2100        }
2101        if (!found) {
2102            Mutex::Autolock _l(t->mLock);
2103            // remove all effects from the chain
2104            while (ec->mEffects.size()) {
2105                sp<EffectModule> effect = ec->mEffects[0];
2106                effect->unPin();
2107                t->removeEffect_l(effect);
2108                if (effect->purgeHandles()) {
2109                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2110                }
2111                AudioSystem::unregisterEffect(effect->id());
2112            }
2113        }
2114    }
2115    return;
2116}
2117
2118// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2119AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2120{
2121    return mPlaybackThreads.valueFor(output).get();
2122}
2123
2124// checkMixerThread_l() must be called with AudioFlinger::mLock held
2125AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2126{
2127    PlaybackThread *thread = checkPlaybackThread_l(output);
2128    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2129}
2130
2131// checkRecordThread_l() must be called with AudioFlinger::mLock held
2132AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2133{
2134    return mRecordThreads.valueFor(input).get();
2135}
2136
2137uint32_t AudioFlinger::nextUniqueId()
2138{
2139    return (uint32_t) android_atomic_inc(&mNextUniqueId);
2140}
2141
2142AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2143{
2144    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2145        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2146        AudioStreamOut *output = thread->getOutput();
2147        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2148            return thread;
2149        }
2150    }
2151    return NULL;
2152}
2153
2154audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2155{
2156    PlaybackThread *thread = primaryPlaybackThread_l();
2157
2158    if (thread == NULL) {
2159        return 0;
2160    }
2161
2162    return thread->outDevice();
2163}
2164
2165sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2166                                    int triggerSession,
2167                                    int listenerSession,
2168                                    sync_event_callback_t callBack,
2169                                    wp<RefBase> cookie)
2170{
2171    Mutex::Autolock _l(mLock);
2172
2173    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2174    status_t playStatus = NAME_NOT_FOUND;
2175    status_t recStatus = NAME_NOT_FOUND;
2176    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2177        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2178        if (playStatus == NO_ERROR) {
2179            return event;
2180        }
2181    }
2182    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2183        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2184        if (recStatus == NO_ERROR) {
2185            return event;
2186        }
2187    }
2188    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2189        mPendingSyncEvents.add(event);
2190    } else {
2191        ALOGV("createSyncEvent() invalid event %d", event->type());
2192        event.clear();
2193    }
2194    return event;
2195}
2196
2197// ----------------------------------------------------------------------------
2198//  Effect management
2199// ----------------------------------------------------------------------------
2200
2201
2202status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2203{
2204    Mutex::Autolock _l(mLock);
2205    return EffectQueryNumberEffects(numEffects);
2206}
2207
2208status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2209{
2210    Mutex::Autolock _l(mLock);
2211    return EffectQueryEffect(index, descriptor);
2212}
2213
2214status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2215        effect_descriptor_t *descriptor) const
2216{
2217    Mutex::Autolock _l(mLock);
2218    return EffectGetDescriptor(pUuid, descriptor);
2219}
2220
2221
2222sp<IEffect> AudioFlinger::createEffect(
2223        effect_descriptor_t *pDesc,
2224        const sp<IEffectClient>& effectClient,
2225        int32_t priority,
2226        audio_io_handle_t io,
2227        int sessionId,
2228        status_t *status,
2229        int *id,
2230        int *enabled)
2231{
2232    status_t lStatus = NO_ERROR;
2233    sp<EffectHandle> handle;
2234    effect_descriptor_t desc;
2235
2236    pid_t pid = IPCThreadState::self()->getCallingPid();
2237    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2238            pid, effectClient.get(), priority, sessionId, io);
2239
2240    if (pDesc == NULL) {
2241        lStatus = BAD_VALUE;
2242        goto Exit;
2243    }
2244
2245    // check audio settings permission for global effects
2246    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2247        lStatus = PERMISSION_DENIED;
2248        goto Exit;
2249    }
2250
2251    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2252    // that can only be created by audio policy manager (running in same process)
2253    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2254        lStatus = PERMISSION_DENIED;
2255        goto Exit;
2256    }
2257
2258    {
2259        if (!EffectIsNullUuid(&pDesc->uuid)) {
2260            // if uuid is specified, request effect descriptor
2261            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2262            if (lStatus < 0) {
2263                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2264                goto Exit;
2265            }
2266        } else {
2267            // if uuid is not specified, look for an available implementation
2268            // of the required type in effect factory
2269            if (EffectIsNullUuid(&pDesc->type)) {
2270                ALOGW("createEffect() no effect type");
2271                lStatus = BAD_VALUE;
2272                goto Exit;
2273            }
2274            uint32_t numEffects = 0;
2275            effect_descriptor_t d;
2276            d.flags = 0; // prevent compiler warning
2277            bool found = false;
2278
2279            lStatus = EffectQueryNumberEffects(&numEffects);
2280            if (lStatus < 0) {
2281                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2282                goto Exit;
2283            }
2284            for (uint32_t i = 0; i < numEffects; i++) {
2285                lStatus = EffectQueryEffect(i, &desc);
2286                if (lStatus < 0) {
2287                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2288                    continue;
2289                }
2290                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2291                    // If matching type found save effect descriptor. If the session is
2292                    // 0 and the effect is not auxiliary, continue enumeration in case
2293                    // an auxiliary version of this effect type is available
2294                    found = true;
2295                    d = desc;
2296                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2297                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2298                        break;
2299                    }
2300                }
2301            }
2302            if (!found) {
2303                lStatus = BAD_VALUE;
2304                ALOGW("createEffect() effect not found");
2305                goto Exit;
2306            }
2307            // For same effect type, chose auxiliary version over insert version if
2308            // connect to output mix (Compliance to OpenSL ES)
2309            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2310                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2311                desc = d;
2312            }
2313        }
2314
2315        // Do not allow auxiliary effects on a session different from 0 (output mix)
2316        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2317             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2318            lStatus = INVALID_OPERATION;
2319            goto Exit;
2320        }
2321
2322        // check recording permission for visualizer
2323        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2324            !recordingAllowed()) {
2325            lStatus = PERMISSION_DENIED;
2326            goto Exit;
2327        }
2328
2329        // return effect descriptor
2330        *pDesc = desc;
2331        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2332            // if the output returned by getOutputForEffect() is removed before we lock the
2333            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2334            // and we will exit safely
2335            io = AudioSystem::getOutputForEffect(&desc);
2336            ALOGV("createEffect got output %d", io);
2337        }
2338
2339        Mutex::Autolock _l(mLock);
2340
2341        // If output is not specified try to find a matching audio session ID in one of the
2342        // output threads.
2343        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2344        // because of code checking output when entering the function.
2345        // Note: io is never 0 when creating an effect on an input
2346        if (io == AUDIO_IO_HANDLE_NONE) {
2347            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2348                // output must be specified by AudioPolicyManager when using session
2349                // AUDIO_SESSION_OUTPUT_STAGE
2350                lStatus = BAD_VALUE;
2351                goto Exit;
2352            }
2353            // look for the thread where the specified audio session is present
2354            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2355                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2356                    io = mPlaybackThreads.keyAt(i);
2357                    break;
2358                }
2359            }
2360            if (io == 0) {
2361                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2362                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2363                        io = mRecordThreads.keyAt(i);
2364                        break;
2365                    }
2366                }
2367            }
2368            // If no output thread contains the requested session ID, default to
2369            // first output. The effect chain will be moved to the correct output
2370            // thread when a track with the same session ID is created
2371            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2372                io = mPlaybackThreads.keyAt(0);
2373            }
2374            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2375        }
2376        ThreadBase *thread = checkRecordThread_l(io);
2377        if (thread == NULL) {
2378            thread = checkPlaybackThread_l(io);
2379            if (thread == NULL) {
2380                ALOGE("createEffect() unknown output thread");
2381                lStatus = BAD_VALUE;
2382                goto Exit;
2383            }
2384        }
2385
2386        sp<Client> client = registerPid(pid);
2387
2388        // create effect on selected output thread
2389        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2390                &desc, enabled, &lStatus);
2391        if (handle != 0 && id != NULL) {
2392            *id = handle->id();
2393        }
2394        if (handle == 0) {
2395            // remove local strong reference to Client with mClientLock held
2396            Mutex::Autolock _cl(mClientLock);
2397            client.clear();
2398        }
2399    }
2400
2401Exit:
2402    *status = lStatus;
2403    return handle;
2404}
2405
2406status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2407        audio_io_handle_t dstOutput)
2408{
2409    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2410            sessionId, srcOutput, dstOutput);
2411    Mutex::Autolock _l(mLock);
2412    if (srcOutput == dstOutput) {
2413        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2414        return NO_ERROR;
2415    }
2416    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2417    if (srcThread == NULL) {
2418        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2419        return BAD_VALUE;
2420    }
2421    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2422    if (dstThread == NULL) {
2423        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2424        return BAD_VALUE;
2425    }
2426
2427    Mutex::Autolock _dl(dstThread->mLock);
2428    Mutex::Autolock _sl(srcThread->mLock);
2429    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2430}
2431
2432// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2433status_t AudioFlinger::moveEffectChain_l(int sessionId,
2434                                   AudioFlinger::PlaybackThread *srcThread,
2435                                   AudioFlinger::PlaybackThread *dstThread,
2436                                   bool reRegister)
2437{
2438    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2439            sessionId, srcThread, dstThread);
2440
2441    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2442    if (chain == 0) {
2443        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2444                sessionId, srcThread);
2445        return INVALID_OPERATION;
2446    }
2447
2448    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2449    // so that a new chain is created with correct parameters when first effect is added. This is
2450    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2451    // removed.
2452    srcThread->removeEffectChain_l(chain);
2453
2454    // transfer all effects one by one so that new effect chain is created on new thread with
2455    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2456    sp<EffectChain> dstChain;
2457    uint32_t strategy = 0; // prevent compiler warning
2458    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2459    Vector< sp<EffectModule> > removed;
2460    status_t status = NO_ERROR;
2461    while (effect != 0) {
2462        srcThread->removeEffect_l(effect);
2463        removed.add(effect);
2464        status = dstThread->addEffect_l(effect);
2465        if (status != NO_ERROR) {
2466            break;
2467        }
2468        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2469        if (effect->state() == EffectModule::ACTIVE ||
2470                effect->state() == EffectModule::STOPPING) {
2471            effect->start();
2472        }
2473        // if the move request is not received from audio policy manager, the effect must be
2474        // re-registered with the new strategy and output
2475        if (dstChain == 0) {
2476            dstChain = effect->chain().promote();
2477            if (dstChain == 0) {
2478                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2479                status = NO_INIT;
2480                break;
2481            }
2482            strategy = dstChain->strategy();
2483        }
2484        if (reRegister) {
2485            AudioSystem::unregisterEffect(effect->id());
2486            AudioSystem::registerEffect(&effect->desc(),
2487                                        dstThread->id(),
2488                                        strategy,
2489                                        sessionId,
2490                                        effect->id());
2491            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2492        }
2493        effect = chain->getEffectFromId_l(0);
2494    }
2495
2496    if (status != NO_ERROR) {
2497        for (size_t i = 0; i < removed.size(); i++) {
2498            srcThread->addEffect_l(removed[i]);
2499            if (dstChain != 0 && reRegister) {
2500                AudioSystem::unregisterEffect(removed[i]->id());
2501                AudioSystem::registerEffect(&removed[i]->desc(),
2502                                            srcThread->id(),
2503                                            strategy,
2504                                            sessionId,
2505                                            removed[i]->id());
2506                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2507            }
2508        }
2509    }
2510
2511    return status;
2512}
2513
2514bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2515{
2516    if (mGlobalEffectEnableTime != 0 &&
2517            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2518        return true;
2519    }
2520
2521    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2522        sp<EffectChain> ec =
2523                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2524        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2525            return true;
2526        }
2527    }
2528    return false;
2529}
2530
2531void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2532{
2533    Mutex::Autolock _l(mLock);
2534
2535    mGlobalEffectEnableTime = systemTime();
2536
2537    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2538        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2539        if (t->mType == ThreadBase::OFFLOAD) {
2540            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2541        }
2542    }
2543
2544}
2545
2546struct Entry {
2547#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2548    char mName[MAX_NAME];
2549};
2550
2551int comparEntry(const void *p1, const void *p2)
2552{
2553    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2554}
2555
2556#ifdef TEE_SINK
2557void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2558{
2559    NBAIO_Source *teeSource = source.get();
2560    if (teeSource != NULL) {
2561        // .wav rotation
2562        // There is a benign race condition if 2 threads call this simultaneously.
2563        // They would both traverse the directory, but the result would simply be
2564        // failures at unlink() which are ignored.  It's also unlikely since
2565        // normally dumpsys is only done by bugreport or from the command line.
2566        char teePath[32+256];
2567        strcpy(teePath, "/data/misc/media");
2568        size_t teePathLen = strlen(teePath);
2569        DIR *dir = opendir(teePath);
2570        teePath[teePathLen++] = '/';
2571        if (dir != NULL) {
2572#define MAX_SORT 20 // number of entries to sort
2573#define MAX_KEEP 10 // number of entries to keep
2574            struct Entry entries[MAX_SORT];
2575            size_t entryCount = 0;
2576            while (entryCount < MAX_SORT) {
2577                struct dirent de;
2578                struct dirent *result = NULL;
2579                int rc = readdir_r(dir, &de, &result);
2580                if (rc != 0) {
2581                    ALOGW("readdir_r failed %d", rc);
2582                    break;
2583                }
2584                if (result == NULL) {
2585                    break;
2586                }
2587                if (result != &de) {
2588                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2589                    break;
2590                }
2591                // ignore non .wav file entries
2592                size_t nameLen = strlen(de.d_name);
2593                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2594                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2595                    continue;
2596                }
2597                strcpy(entries[entryCount++].mName, de.d_name);
2598            }
2599            (void) closedir(dir);
2600            if (entryCount > MAX_KEEP) {
2601                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2602                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2603                    strcpy(&teePath[teePathLen], entries[i].mName);
2604                    (void) unlink(teePath);
2605                }
2606            }
2607        } else {
2608            if (fd >= 0) {
2609                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2610            }
2611        }
2612        char teeTime[16];
2613        struct timeval tv;
2614        gettimeofday(&tv, NULL);
2615        struct tm tm;
2616        localtime_r(&tv.tv_sec, &tm);
2617        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2618        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2619        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2620        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2621        if (teeFd >= 0) {
2622            char wavHeader[44];
2623            memcpy(wavHeader,
2624                "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",
2625                sizeof(wavHeader));
2626            NBAIO_Format format = teeSource->format();
2627            unsigned channelCount = Format_channelCount(format);
2628            ALOG_ASSERT(channelCount <= FCC_2);
2629            uint32_t sampleRate = Format_sampleRate(format);
2630            wavHeader[22] = channelCount;       // number of channels
2631            wavHeader[24] = sampleRate;         // sample rate
2632            wavHeader[25] = sampleRate >> 8;
2633            wavHeader[32] = channelCount * 2;   // block alignment
2634            write(teeFd, wavHeader, sizeof(wavHeader));
2635            size_t total = 0;
2636            bool firstRead = true;
2637            for (;;) {
2638#define TEE_SINK_READ 1024
2639                short buffer[TEE_SINK_READ * FCC_2];
2640                size_t count = TEE_SINK_READ;
2641                ssize_t actual = teeSource->read(buffer, count,
2642                        AudioBufferProvider::kInvalidPTS);
2643                bool wasFirstRead = firstRead;
2644                firstRead = false;
2645                if (actual <= 0) {
2646                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2647                        continue;
2648                    }
2649                    break;
2650                }
2651                ALOG_ASSERT(actual <= (ssize_t)count);
2652                write(teeFd, buffer, actual * channelCount * sizeof(short));
2653                total += actual;
2654            }
2655            lseek(teeFd, (off_t) 4, SEEK_SET);
2656            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2657            write(teeFd, &temp, sizeof(temp));
2658            lseek(teeFd, (off_t) 40, SEEK_SET);
2659            temp =  total * channelCount * sizeof(short);
2660            write(teeFd, &temp, sizeof(temp));
2661            close(teeFd);
2662            if (fd >= 0) {
2663                fdprintf(fd, "tee copied to %s\n", teePath);
2664            }
2665        } else {
2666            if (fd >= 0) {
2667                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2668            }
2669        }
2670    }
2671}
2672#endif
2673
2674// ----------------------------------------------------------------------------
2675
2676status_t AudioFlinger::onTransact(
2677        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2678{
2679    return BnAudioFlinger::onTransact(code, data, reply, flags);
2680}
2681
2682}; // namespace android
2683