AudioFlinger.cpp revision fa319e6d918b84f93fb5457af5d1cca6421ac517 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <dirent.h>
24#include <math.h>
25#include <signal.h>
26#include <sys/time.h>
27#include <sys/resource.h>
28
29#include <binder/IPCThreadState.h>
30#include <binder/IServiceManager.h>
31#include <utils/Log.h>
32#include <utils/Trace.h>
33#include <binder/Parcel.h>
34#include <utils/String16.h>
35#include <utils/threads.h>
36#include <utils/Atomic.h>
37
38#include <cutils/bitops.h>
39#include <cutils/properties.h>
40
41#include <system/audio.h>
42#include <hardware/audio.h>
43
44#include "AudioMixer.h"
45#include "AudioFlinger.h"
46#include "ServiceUtilities.h"
47
48#include <media/EffectsFactoryApi.h>
49#include <audio_effects/effect_visualizer.h>
50#include <audio_effects/effect_ns.h>
51#include <audio_effects/effect_aec.h>
52
53#include <audio_utils/primitives.h>
54
55#include <powermanager/PowerManager.h>
56
57#include <common_time/cc_helper.h>
58
59#include <media/IMediaLogService.h>
60
61#include <media/nbaio/Pipe.h>
62#include <media/nbaio/PipeReader.h>
63#include <media/AudioParameter.h>
64#include <private/android_filesystem_config.h>
65
66// ----------------------------------------------------------------------------
67
68// Note: the following macro is used for extremely verbose logging message.  In
69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
70// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
71// are so verbose that we want to suppress them even when we have ALOG_ASSERT
72// turned on.  Do not uncomment the #def below unless you really know what you
73// are doing and want to see all of the extremely verbose messages.
74//#define VERY_VERY_VERBOSE_LOGGING
75#ifdef VERY_VERY_VERBOSE_LOGGING
76#define ALOGVV ALOGV
77#else
78#define ALOGVV(a...) do { } while(0)
79#endif
80
81namespace android {
82
83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
84static const char kHardwareLockedString[] = "Hardware lock is taken\n";
85
86
87nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
88
89uint32_t AudioFlinger::mScreenState;
90
91#ifdef TEE_SINK
92bool AudioFlinger::mTeeSinkInputEnabled = false;
93bool AudioFlinger::mTeeSinkOutputEnabled = false;
94bool AudioFlinger::mTeeSinkTrackEnabled = false;
95
96size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
97size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
98size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
99#endif
100
101// ----------------------------------------------------------------------------
102
103static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
104{
105    const hw_module_t *mod;
106    int rc;
107
108    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
109    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
110                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
111    if (rc) {
112        goto out;
113    }
114    rc = audio_hw_device_open(mod, dev);
115    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
116                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
117    if (rc) {
118        goto out;
119    }
120    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
121        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
122        rc = BAD_VALUE;
123        goto out;
124    }
125    return 0;
126
127out:
128    *dev = NULL;
129    return rc;
130}
131
132// ----------------------------------------------------------------------------
133
134AudioFlinger::AudioFlinger()
135    : BnAudioFlinger(),
136      mPrimaryHardwareDev(NULL),
137      mHardwareStatus(AUDIO_HW_IDLE),
138      mMasterVolume(1.0f),
139      mMasterMute(false),
140      mNextUniqueId(1),
141      mMode(AUDIO_MODE_INVALID),
142      mBtNrecIsOff(false),
143      mIsLowRamDevice(true),
144      mIsDeviceTypeKnown(false)
145{
146    getpid_cached = getpid();
147    char value[PROPERTY_VALUE_MAX];
148    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
149    if (doLog) {
150        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
151    }
152#ifdef TEE_SINK
153    (void) property_get("ro.debuggable", value, "0");
154    int debuggable = atoi(value);
155    int teeEnabled = 0;
156    if (debuggable) {
157        (void) property_get("af.tee", value, "0");
158        teeEnabled = atoi(value);
159    }
160    if (teeEnabled & 1)
161        mTeeSinkInputEnabled = true;
162    if (teeEnabled & 2)
163        mTeeSinkOutputEnabled = true;
164    if (teeEnabled & 4)
165        mTeeSinkTrackEnabled = true;
166#endif
167}
168
169void AudioFlinger::onFirstRef()
170{
171    int rc = 0;
172
173    Mutex::Autolock _l(mLock);
174
175    /* TODO: move all this work into an Init() function */
176    char val_str[PROPERTY_VALUE_MAX] = { 0 };
177    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
178        uint32_t int_val;
179        if (1 == sscanf(val_str, "%u", &int_val)) {
180            mStandbyTimeInNsecs = milliseconds(int_val);
181            ALOGI("Using %u mSec as standby time.", int_val);
182        } else {
183            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
184            ALOGI("Using default %u mSec as standby time.",
185                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
186        }
187    }
188
189    mMode = AUDIO_MODE_NORMAL;
190}
191
192AudioFlinger::~AudioFlinger()
193{
194    while (!mRecordThreads.isEmpty()) {
195        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
196        closeInput_nonvirtual(mRecordThreads.keyAt(0));
197    }
198    while (!mPlaybackThreads.isEmpty()) {
199        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
200        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
201    }
202
203    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
204        // no mHardwareLock needed, as there are no other references to this
205        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
206        delete mAudioHwDevs.valueAt(i);
207    }
208}
209
210static const char * const audio_interfaces[] = {
211    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
212    AUDIO_HARDWARE_MODULE_ID_A2DP,
213    AUDIO_HARDWARE_MODULE_ID_USB,
214};
215#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
216
217AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
218        audio_module_handle_t module,
219        audio_devices_t devices)
220{
221    // if module is 0, the request comes from an old policy manager and we should load
222    // well known modules
223    if (module == 0) {
224        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
225        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
226            loadHwModule_l(audio_interfaces[i]);
227        }
228        // then try to find a module supporting the requested device.
229        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
230            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
231            audio_hw_device_t *dev = audioHwDevice->hwDevice();
232            if ((dev->get_supported_devices != NULL) &&
233                    (dev->get_supported_devices(dev) & devices) == devices)
234                return audioHwDevice;
235        }
236    } else {
237        // check a match for the requested module handle
238        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
239        if (audioHwDevice != NULL) {
240            return audioHwDevice;
241        }
242    }
243
244    return NULL;
245}
246
247void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
248{
249    const size_t SIZE = 256;
250    char buffer[SIZE];
251    String8 result;
252
253    result.append("Clients:\n");
254    for (size_t i = 0; i < mClients.size(); ++i) {
255        sp<Client> client = mClients.valueAt(i).promote();
256        if (client != 0) {
257            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
258            result.append(buffer);
259        }
260    }
261
262    result.append("Global session refs:\n");
263    result.append(" session pid count\n");
264    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
265        AudioSessionRef *r = mAudioSessionRefs[i];
266        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
267        result.append(buffer);
268    }
269    write(fd, result.string(), result.size());
270}
271
272
273void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
274{
275    const size_t SIZE = 256;
276    char buffer[SIZE];
277    String8 result;
278    hardware_call_state hardwareStatus = mHardwareStatus;
279
280    snprintf(buffer, SIZE, "Hardware status: %d\n"
281                           "Standby Time mSec: %u\n",
282                            hardwareStatus,
283                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
284    result.append(buffer);
285    write(fd, result.string(), result.size());
286}
287
288void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
289{
290    const size_t SIZE = 256;
291    char buffer[SIZE];
292    String8 result;
293    snprintf(buffer, SIZE, "Permission Denial: "
294            "can't dump AudioFlinger from pid=%d, uid=%d\n",
295            IPCThreadState::self()->getCallingPid(),
296            IPCThreadState::self()->getCallingUid());
297    result.append(buffer);
298    write(fd, result.string(), result.size());
299}
300
301bool AudioFlinger::dumpTryLock(Mutex& mutex)
302{
303    bool locked = false;
304    for (int i = 0; i < kDumpLockRetries; ++i) {
305        if (mutex.tryLock() == NO_ERROR) {
306            locked = true;
307            break;
308        }
309        usleep(kDumpLockSleepUs);
310    }
311    return locked;
312}
313
314status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
315{
316    if (!dumpAllowed()) {
317        dumpPermissionDenial(fd, args);
318    } else {
319        // get state of hardware lock
320        bool hardwareLocked = dumpTryLock(mHardwareLock);
321        if (!hardwareLocked) {
322            String8 result(kHardwareLockedString);
323            write(fd, result.string(), result.size());
324        } else {
325            mHardwareLock.unlock();
326        }
327
328        bool locked = dumpTryLock(mLock);
329
330        // failed to lock - AudioFlinger is probably deadlocked
331        if (!locked) {
332            String8 result(kDeadlockedString);
333            write(fd, result.string(), result.size());
334        }
335
336        dumpClients(fd, args);
337        dumpInternals(fd, args);
338
339        // dump playback threads
340        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
341            mPlaybackThreads.valueAt(i)->dump(fd, args);
342        }
343
344        // dump record threads
345        for (size_t i = 0; i < mRecordThreads.size(); i++) {
346            mRecordThreads.valueAt(i)->dump(fd, args);
347        }
348
349        // dump all hardware devs
350        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
351            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
352            dev->dump(dev, fd);
353        }
354
355#ifdef TEE_SINK
356        // dump the serially shared record tee sink
357        if (mRecordTeeSource != 0) {
358            dumpTee(fd, mRecordTeeSource);
359        }
360#endif
361
362        if (locked) {
363            mLock.unlock();
364        }
365
366        // append a copy of media.log here by forwarding fd to it, but don't attempt
367        // to lookup the service if it's not running, as it will block for a second
368        if (mLogMemoryDealer != 0) {
369            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
370            if (binder != 0) {
371                fdprintf(fd, "\nmedia.log:\n");
372                Vector<String16> args;
373                binder->dump(fd, args);
374            }
375        }
376    }
377    return NO_ERROR;
378}
379
380sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
381{
382    // If pid is already in the mClients wp<> map, then use that entry
383    // (for which promote() is always != 0), otherwise create a new entry and Client.
384    sp<Client> client = mClients.valueFor(pid).promote();
385    if (client == 0) {
386        client = new Client(this, pid);
387        mClients.add(pid, client);
388    }
389
390    return client;
391}
392
393sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
394{
395    if (mLogMemoryDealer == 0) {
396        return new NBLog::Writer();
397    }
398    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
399    sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
400    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
401    if (binder != 0) {
402        interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
403    }
404    return writer;
405}
406
407void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
408{
409    if (writer == 0) {
410        return;
411    }
412    sp<IMemory> iMemory(writer->getIMemory());
413    if (iMemory == 0) {
414        return;
415    }
416    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
417    if (binder != 0) {
418        interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
419        // Now the media.log remote reference to IMemory is gone.
420        // When our last local reference to IMemory also drops to zero,
421        // the IMemory destructor will deallocate the region from mMemoryDealer.
422    }
423}
424
425// IAudioFlinger interface
426
427
428sp<IAudioTrack> AudioFlinger::createTrack(
429        audio_stream_type_t streamType,
430        uint32_t sampleRate,
431        audio_format_t format,
432        audio_channel_mask_t channelMask,
433        size_t frameCount,
434        IAudioFlinger::track_flags_t *flags,
435        const sp<IMemory>& sharedBuffer,
436        audio_io_handle_t output,
437        pid_t tid,
438        int *sessionId,
439        status_t *status)
440{
441    sp<PlaybackThread::Track> track;
442    sp<TrackHandle> trackHandle;
443    sp<Client> client;
444    status_t lStatus;
445    int lSessionId;
446
447    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
448    // but if someone uses binder directly they could bypass that and cause us to crash
449    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
450        ALOGE("createTrack() invalid stream type %d", streamType);
451        lStatus = BAD_VALUE;
452        goto Exit;
453    }
454
455    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
456    // and we don't yet support 8.24 or 32-bit PCM
457    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
458        ALOGE("createTrack() invalid format %d", format);
459        lStatus = BAD_VALUE;
460        goto Exit;
461    }
462
463    {
464        Mutex::Autolock _l(mLock);
465        PlaybackThread *thread = checkPlaybackThread_l(output);
466        PlaybackThread *effectThread = NULL;
467        if (thread == NULL) {
468            ALOGE("no playback thread found for output handle %d", output);
469            lStatus = BAD_VALUE;
470            goto Exit;
471        }
472
473        pid_t pid = IPCThreadState::self()->getCallingPid();
474        client = registerPid_l(pid);
475
476        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
477        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
478            // check if an effect chain with the same session ID is present on another
479            // output thread and move it here.
480            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
481                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
482                if (mPlaybackThreads.keyAt(i) != output) {
483                    uint32_t sessions = t->hasAudioSession(*sessionId);
484                    if (sessions & PlaybackThread::EFFECT_SESSION) {
485                        effectThread = t.get();
486                        break;
487                    }
488                }
489            }
490            lSessionId = *sessionId;
491        } else {
492            // if no audio session id is provided, create one here
493            lSessionId = nextUniqueId();
494            if (sessionId != NULL) {
495                *sessionId = lSessionId;
496            }
497        }
498        ALOGV("createTrack() lSessionId: %d", lSessionId);
499
500        track = thread->createTrack_l(client, streamType, sampleRate, format,
501                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus);
502
503        // move effect chain to this output thread if an effect on same session was waiting
504        // for a track to be created
505        if (lStatus == NO_ERROR && effectThread != NULL) {
506            Mutex::Autolock _dl(thread->mLock);
507            Mutex::Autolock _sl(effectThread->mLock);
508            moveEffectChain_l(lSessionId, effectThread, thread, true);
509        }
510
511        // Look for sync events awaiting for a session to be used.
512        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
513            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
514                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
515                    if (lStatus == NO_ERROR) {
516                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
517                    } else {
518                        mPendingSyncEvents[i]->cancel();
519                    }
520                    mPendingSyncEvents.removeAt(i);
521                    i--;
522                }
523            }
524        }
525    }
526    if (lStatus == NO_ERROR) {
527        trackHandle = new TrackHandle(track);
528    } else {
529        // remove local strong reference to Client before deleting the Track so that the Client
530        // destructor is called by the TrackBase destructor with mLock held
531        client.clear();
532        track.clear();
533    }
534
535Exit:
536    if (status != NULL) {
537        *status = lStatus;
538    }
539    return trackHandle;
540}
541
542uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
543{
544    Mutex::Autolock _l(mLock);
545    PlaybackThread *thread = checkPlaybackThread_l(output);
546    if (thread == NULL) {
547        ALOGW("sampleRate() unknown thread %d", output);
548        return 0;
549    }
550    return thread->sampleRate();
551}
552
553int AudioFlinger::channelCount(audio_io_handle_t output) const
554{
555    Mutex::Autolock _l(mLock);
556    PlaybackThread *thread = checkPlaybackThread_l(output);
557    if (thread == NULL) {
558        ALOGW("channelCount() unknown thread %d", output);
559        return 0;
560    }
561    return thread->channelCount();
562}
563
564audio_format_t AudioFlinger::format(audio_io_handle_t output) const
565{
566    Mutex::Autolock _l(mLock);
567    PlaybackThread *thread = checkPlaybackThread_l(output);
568    if (thread == NULL) {
569        ALOGW("format() unknown thread %d", output);
570        return AUDIO_FORMAT_INVALID;
571    }
572    return thread->format();
573}
574
575size_t AudioFlinger::frameCount(audio_io_handle_t output) const
576{
577    Mutex::Autolock _l(mLock);
578    PlaybackThread *thread = checkPlaybackThread_l(output);
579    if (thread == NULL) {
580        ALOGW("frameCount() unknown thread %d", output);
581        return 0;
582    }
583    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
584    //       should examine all callers and fix them to handle smaller counts
585    return thread->frameCount();
586}
587
588uint32_t AudioFlinger::latency(audio_io_handle_t output) const
589{
590    Mutex::Autolock _l(mLock);
591    PlaybackThread *thread = checkPlaybackThread_l(output);
592    if (thread == NULL) {
593        ALOGW("latency(): no playback thread found for output handle %d", output);
594        return 0;
595    }
596    return thread->latency();
597}
598
599status_t AudioFlinger::setMasterVolume(float value)
600{
601    status_t ret = initCheck();
602    if (ret != NO_ERROR) {
603        return ret;
604    }
605
606    // check calling permissions
607    if (!settingsAllowed()) {
608        return PERMISSION_DENIED;
609    }
610
611    Mutex::Autolock _l(mLock);
612    mMasterVolume = value;
613
614    // Set master volume in the HALs which support it.
615    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
616        AutoMutex lock(mHardwareLock);
617        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
618
619        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
620        if (dev->canSetMasterVolume()) {
621            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
622        }
623        mHardwareStatus = AUDIO_HW_IDLE;
624    }
625
626    // Now set the master volume in each playback thread.  Playback threads
627    // assigned to HALs which do not have master volume support will apply
628    // master volume during the mix operation.  Threads with HALs which do
629    // support master volume will simply ignore the setting.
630    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
631        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
632
633    return NO_ERROR;
634}
635
636status_t AudioFlinger::setMode(audio_mode_t mode)
637{
638    status_t ret = initCheck();
639    if (ret != NO_ERROR) {
640        return ret;
641    }
642
643    // check calling permissions
644    if (!settingsAllowed()) {
645        return PERMISSION_DENIED;
646    }
647    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
648        ALOGW("Illegal value: setMode(%d)", mode);
649        return BAD_VALUE;
650    }
651
652    { // scope for the lock
653        AutoMutex lock(mHardwareLock);
654        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
655        mHardwareStatus = AUDIO_HW_SET_MODE;
656        ret = dev->set_mode(dev, mode);
657        mHardwareStatus = AUDIO_HW_IDLE;
658    }
659
660    if (NO_ERROR == ret) {
661        Mutex::Autolock _l(mLock);
662        mMode = mode;
663        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
664            mPlaybackThreads.valueAt(i)->setMode(mode);
665    }
666
667    return ret;
668}
669
670status_t AudioFlinger::setMicMute(bool state)
671{
672    status_t ret = initCheck();
673    if (ret != NO_ERROR) {
674        return ret;
675    }
676
677    // check calling permissions
678    if (!settingsAllowed()) {
679        return PERMISSION_DENIED;
680    }
681
682    AutoMutex lock(mHardwareLock);
683    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
684    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
685    ret = dev->set_mic_mute(dev, state);
686    mHardwareStatus = AUDIO_HW_IDLE;
687    return ret;
688}
689
690bool AudioFlinger::getMicMute() const
691{
692    status_t ret = initCheck();
693    if (ret != NO_ERROR) {
694        return false;
695    }
696
697    bool state = AUDIO_MODE_INVALID;
698    AutoMutex lock(mHardwareLock);
699    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
700    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
701    dev->get_mic_mute(dev, &state);
702    mHardwareStatus = AUDIO_HW_IDLE;
703    return state;
704}
705
706status_t AudioFlinger::setMasterMute(bool muted)
707{
708    status_t ret = initCheck();
709    if (ret != NO_ERROR) {
710        return ret;
711    }
712
713    // check calling permissions
714    if (!settingsAllowed()) {
715        return PERMISSION_DENIED;
716    }
717
718    Mutex::Autolock _l(mLock);
719    mMasterMute = muted;
720
721    // Set master mute in the HALs which support it.
722    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
723        AutoMutex lock(mHardwareLock);
724        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
725
726        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
727        if (dev->canSetMasterMute()) {
728            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
729        }
730        mHardwareStatus = AUDIO_HW_IDLE;
731    }
732
733    // Now set the master mute in each playback thread.  Playback threads
734    // assigned to HALs which do not have master mute support will apply master
735    // mute during the mix operation.  Threads with HALs which do support master
736    // mute will simply ignore the setting.
737    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
738        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
739
740    return NO_ERROR;
741}
742
743float AudioFlinger::masterVolume() const
744{
745    Mutex::Autolock _l(mLock);
746    return masterVolume_l();
747}
748
749bool AudioFlinger::masterMute() const
750{
751    Mutex::Autolock _l(mLock);
752    return masterMute_l();
753}
754
755float AudioFlinger::masterVolume_l() const
756{
757    return mMasterVolume;
758}
759
760bool AudioFlinger::masterMute_l() const
761{
762    return mMasterMute;
763}
764
765status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
766        audio_io_handle_t output)
767{
768    // check calling permissions
769    if (!settingsAllowed()) {
770        return PERMISSION_DENIED;
771    }
772
773    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
774        ALOGE("setStreamVolume() invalid stream %d", stream);
775        return BAD_VALUE;
776    }
777
778    AutoMutex lock(mLock);
779    PlaybackThread *thread = NULL;
780    if (output) {
781        thread = checkPlaybackThread_l(output);
782        if (thread == NULL) {
783            return BAD_VALUE;
784        }
785    }
786
787    mStreamTypes[stream].volume = value;
788
789    if (thread == NULL) {
790        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
791            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
792        }
793    } else {
794        thread->setStreamVolume(stream, value);
795    }
796
797    return NO_ERROR;
798}
799
800status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
801{
802    // check calling permissions
803    if (!settingsAllowed()) {
804        return PERMISSION_DENIED;
805    }
806
807    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
808        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
809        ALOGE("setStreamMute() invalid stream %d", stream);
810        return BAD_VALUE;
811    }
812
813    AutoMutex lock(mLock);
814    mStreamTypes[stream].mute = muted;
815    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
816        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
817
818    return NO_ERROR;
819}
820
821float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
822{
823    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
824        return 0.0f;
825    }
826
827    AutoMutex lock(mLock);
828    float volume;
829    if (output) {
830        PlaybackThread *thread = checkPlaybackThread_l(output);
831        if (thread == NULL) {
832            return 0.0f;
833        }
834        volume = thread->streamVolume(stream);
835    } else {
836        volume = streamVolume_l(stream);
837    }
838
839    return volume;
840}
841
842bool AudioFlinger::streamMute(audio_stream_type_t stream) const
843{
844    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
845        return true;
846    }
847
848    AutoMutex lock(mLock);
849    return streamMute_l(stream);
850}
851
852status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
853{
854    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
855            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
856
857    // check calling permissions
858    if (!settingsAllowed()) {
859        return PERMISSION_DENIED;
860    }
861
862    // ioHandle == 0 means the parameters are global to the audio hardware interface
863    if (ioHandle == 0) {
864        Mutex::Autolock _l(mLock);
865        status_t final_result = NO_ERROR;
866        {
867            AutoMutex lock(mHardwareLock);
868            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
869            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
870                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
871                status_t result = dev->set_parameters(dev, keyValuePairs.string());
872                final_result = result ?: final_result;
873            }
874            mHardwareStatus = AUDIO_HW_IDLE;
875        }
876        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
877        AudioParameter param = AudioParameter(keyValuePairs);
878        String8 value;
879        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
880            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
881            if (mBtNrecIsOff != btNrecIsOff) {
882                for (size_t i = 0; i < mRecordThreads.size(); i++) {
883                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
884                    audio_devices_t device = thread->inDevice();
885                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
886                    // collect all of the thread's session IDs
887                    KeyedVector<int, bool> ids = thread->sessionIds();
888                    // suspend effects associated with those session IDs
889                    for (size_t j = 0; j < ids.size(); ++j) {
890                        int sessionId = ids.keyAt(j);
891                        thread->setEffectSuspended(FX_IID_AEC,
892                                                   suspend,
893                                                   sessionId);
894                        thread->setEffectSuspended(FX_IID_NS,
895                                                   suspend,
896                                                   sessionId);
897                    }
898                }
899                mBtNrecIsOff = btNrecIsOff;
900            }
901        }
902        String8 screenState;
903        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
904            bool isOff = screenState == "off";
905            if (isOff != (AudioFlinger::mScreenState & 1)) {
906                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
907            }
908        }
909        return final_result;
910    }
911
912    // hold a strong ref on thread in case closeOutput() or closeInput() is called
913    // and the thread is exited once the lock is released
914    sp<ThreadBase> thread;
915    {
916        Mutex::Autolock _l(mLock);
917        thread = checkPlaybackThread_l(ioHandle);
918        if (thread == 0) {
919            thread = checkRecordThread_l(ioHandle);
920        } else if (thread == primaryPlaybackThread_l()) {
921            // indicate output device change to all input threads for pre processing
922            AudioParameter param = AudioParameter(keyValuePairs);
923            int value;
924            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
925                    (value != 0)) {
926                for (size_t i = 0; i < mRecordThreads.size(); i++) {
927                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
928                }
929            }
930        }
931    }
932    if (thread != 0) {
933        return thread->setParameters(keyValuePairs);
934    }
935    return BAD_VALUE;
936}
937
938String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
939{
940    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
941            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
942
943    Mutex::Autolock _l(mLock);
944
945    if (ioHandle == 0) {
946        String8 out_s8;
947
948        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
949            char *s;
950            {
951            AutoMutex lock(mHardwareLock);
952            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
953            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
954            s = dev->get_parameters(dev, keys.string());
955            mHardwareStatus = AUDIO_HW_IDLE;
956            }
957            out_s8 += String8(s ? s : "");
958            free(s);
959        }
960        return out_s8;
961    }
962
963    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
964    if (playbackThread != NULL) {
965        return playbackThread->getParameters(keys);
966    }
967    RecordThread *recordThread = checkRecordThread_l(ioHandle);
968    if (recordThread != NULL) {
969        return recordThread->getParameters(keys);
970    }
971    return String8("");
972}
973
974size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
975        audio_channel_mask_t channelMask) const
976{
977    status_t ret = initCheck();
978    if (ret != NO_ERROR) {
979        return 0;
980    }
981
982    AutoMutex lock(mHardwareLock);
983    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
984    struct audio_config config;
985    memset(&config, 0, sizeof(config));
986    config.sample_rate = sampleRate;
987    config.channel_mask = channelMask;
988    config.format = format;
989
990    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
991    size_t size = dev->get_input_buffer_size(dev, &config);
992    mHardwareStatus = AUDIO_HW_IDLE;
993    return size;
994}
995
996unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
997{
998    Mutex::Autolock _l(mLock);
999
1000    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1001    if (recordThread != NULL) {
1002        return recordThread->getInputFramesLost();
1003    }
1004    return 0;
1005}
1006
1007status_t AudioFlinger::setVoiceVolume(float value)
1008{
1009    status_t ret = initCheck();
1010    if (ret != NO_ERROR) {
1011        return ret;
1012    }
1013
1014    // check calling permissions
1015    if (!settingsAllowed()) {
1016        return PERMISSION_DENIED;
1017    }
1018
1019    AutoMutex lock(mHardwareLock);
1020    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1021    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1022    ret = dev->set_voice_volume(dev, value);
1023    mHardwareStatus = AUDIO_HW_IDLE;
1024
1025    return ret;
1026}
1027
1028status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1029        audio_io_handle_t output) const
1030{
1031    status_t status;
1032
1033    Mutex::Autolock _l(mLock);
1034
1035    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1036    if (playbackThread != NULL) {
1037        return playbackThread->getRenderPosition(halFrames, dspFrames);
1038    }
1039
1040    return BAD_VALUE;
1041}
1042
1043void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1044{
1045
1046    Mutex::Autolock _l(mLock);
1047
1048    pid_t pid = IPCThreadState::self()->getCallingPid();
1049    if (mNotificationClients.indexOfKey(pid) < 0) {
1050        sp<NotificationClient> notificationClient = new NotificationClient(this,
1051                                                                            client,
1052                                                                            pid);
1053        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1054
1055        mNotificationClients.add(pid, notificationClient);
1056
1057        sp<IBinder> binder = client->asBinder();
1058        binder->linkToDeath(notificationClient);
1059
1060        // the config change is always sent from playback or record threads to avoid deadlock
1061        // with AudioSystem::gLock
1062        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1063            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1064        }
1065
1066        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1067            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1068        }
1069    }
1070}
1071
1072void AudioFlinger::removeNotificationClient(pid_t pid)
1073{
1074    Mutex::Autolock _l(mLock);
1075
1076    mNotificationClients.removeItem(pid);
1077
1078    ALOGV("%d died, releasing its sessions", pid);
1079    size_t num = mAudioSessionRefs.size();
1080    bool removed = false;
1081    for (size_t i = 0; i< num; ) {
1082        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1083        ALOGV(" pid %d @ %d", ref->mPid, i);
1084        if (ref->mPid == pid) {
1085            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1086            mAudioSessionRefs.removeAt(i);
1087            delete ref;
1088            removed = true;
1089            num--;
1090        } else {
1091            i++;
1092        }
1093    }
1094    if (removed) {
1095        purgeStaleEffects_l();
1096    }
1097}
1098
1099// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1100void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1101{
1102    size_t size = mNotificationClients.size();
1103    for (size_t i = 0; i < size; i++) {
1104        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1105                                                                               param2);
1106    }
1107}
1108
1109// removeClient_l() must be called with AudioFlinger::mLock held
1110void AudioFlinger::removeClient_l(pid_t pid)
1111{
1112    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1113            IPCThreadState::self()->getCallingPid());
1114    mClients.removeItem(pid);
1115}
1116
1117// getEffectThread_l() must be called with AudioFlinger::mLock held
1118sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1119{
1120    sp<PlaybackThread> thread;
1121
1122    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1123        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1124            ALOG_ASSERT(thread == 0);
1125            thread = mPlaybackThreads.valueAt(i);
1126        }
1127    }
1128
1129    return thread;
1130}
1131
1132
1133
1134// ----------------------------------------------------------------------------
1135
1136AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1137    :   RefBase(),
1138        mAudioFlinger(audioFlinger),
1139        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1140        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1141        mPid(pid),
1142        mTimedTrackCount(0)
1143{
1144    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1145}
1146
1147// Client destructor must be called with AudioFlinger::mLock held
1148AudioFlinger::Client::~Client()
1149{
1150    mAudioFlinger->removeClient_l(mPid);
1151}
1152
1153sp<MemoryDealer> AudioFlinger::Client::heap() const
1154{
1155    return mMemoryDealer;
1156}
1157
1158// Reserve one of the limited slots for a timed audio track associated
1159// with this client
1160bool AudioFlinger::Client::reserveTimedTrack()
1161{
1162    const int kMaxTimedTracksPerClient = 4;
1163
1164    Mutex::Autolock _l(mTimedTrackLock);
1165
1166    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1167        ALOGW("can not create timed track - pid %d has exceeded the limit",
1168             mPid);
1169        return false;
1170    }
1171
1172    mTimedTrackCount++;
1173    return true;
1174}
1175
1176// Release a slot for a timed audio track
1177void AudioFlinger::Client::releaseTimedTrack()
1178{
1179    Mutex::Autolock _l(mTimedTrackLock);
1180    mTimedTrackCount--;
1181}
1182
1183// ----------------------------------------------------------------------------
1184
1185AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1186                                                     const sp<IAudioFlingerClient>& client,
1187                                                     pid_t pid)
1188    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1189{
1190}
1191
1192AudioFlinger::NotificationClient::~NotificationClient()
1193{
1194}
1195
1196void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1197{
1198    sp<NotificationClient> keep(this);
1199    mAudioFlinger->removeNotificationClient(mPid);
1200}
1201
1202
1203// ----------------------------------------------------------------------------
1204
1205sp<IAudioRecord> AudioFlinger::openRecord(
1206        audio_io_handle_t input,
1207        uint32_t sampleRate,
1208        audio_format_t format,
1209        audio_channel_mask_t channelMask,
1210        size_t frameCount,
1211        IAudioFlinger::track_flags_t flags,
1212        pid_t tid,
1213        int *sessionId,
1214        status_t *status)
1215{
1216    sp<RecordThread::RecordTrack> recordTrack;
1217    sp<RecordHandle> recordHandle;
1218    sp<Client> client;
1219    status_t lStatus;
1220    RecordThread *thread;
1221    size_t inFrameCount;
1222    int lSessionId;
1223
1224    // check calling permissions
1225    if (!recordingAllowed()) {
1226        lStatus = PERMISSION_DENIED;
1227        goto Exit;
1228    }
1229
1230    // add client to list
1231    { // scope for mLock
1232        Mutex::Autolock _l(mLock);
1233        thread = checkRecordThread_l(input);
1234        if (thread == NULL) {
1235            lStatus = BAD_VALUE;
1236            goto Exit;
1237        }
1238
1239        pid_t pid = IPCThreadState::self()->getCallingPid();
1240        client = registerPid_l(pid);
1241
1242        // If no audio session id is provided, create one here
1243        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1244            lSessionId = *sessionId;
1245        } else {
1246            lSessionId = nextUniqueId();
1247            if (sessionId != NULL) {
1248                *sessionId = lSessionId;
1249            }
1250        }
1251        // create new record track.
1252        // The record track uses one track in mHardwareMixerThread by convention.
1253        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1254                                                  frameCount, lSessionId, flags, tid, &lStatus);
1255    }
1256    if (lStatus != NO_ERROR) {
1257        // remove local strong reference to Client before deleting the RecordTrack so that the
1258        // Client destructor is called by the TrackBase destructor with mLock held
1259        client.clear();
1260        recordTrack.clear();
1261        goto Exit;
1262    }
1263
1264    // return to handle to client
1265    recordHandle = new RecordHandle(recordTrack);
1266    lStatus = NO_ERROR;
1267
1268Exit:
1269    if (status) {
1270        *status = lStatus;
1271    }
1272    return recordHandle;
1273}
1274
1275
1276
1277// ----------------------------------------------------------------------------
1278
1279audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1280{
1281    if (!settingsAllowed()) {
1282        return 0;
1283    }
1284    Mutex::Autolock _l(mLock);
1285    return loadHwModule_l(name);
1286}
1287
1288// loadHwModule_l() must be called with AudioFlinger::mLock held
1289audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1290{
1291    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1292        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1293            ALOGW("loadHwModule() module %s already loaded", name);
1294            return mAudioHwDevs.keyAt(i);
1295        }
1296    }
1297
1298    audio_hw_device_t *dev;
1299
1300    int rc = load_audio_interface(name, &dev);
1301    if (rc) {
1302        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1303        return 0;
1304    }
1305
1306    mHardwareStatus = AUDIO_HW_INIT;
1307    rc = dev->init_check(dev);
1308    mHardwareStatus = AUDIO_HW_IDLE;
1309    if (rc) {
1310        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1311        return 0;
1312    }
1313
1314    // Check and cache this HAL's level of support for master mute and master
1315    // volume.  If this is the first HAL opened, and it supports the get
1316    // methods, use the initial values provided by the HAL as the current
1317    // master mute and volume settings.
1318
1319    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1320    {  // scope for auto-lock pattern
1321        AutoMutex lock(mHardwareLock);
1322
1323        if (0 == mAudioHwDevs.size()) {
1324            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1325            if (NULL != dev->get_master_volume) {
1326                float mv;
1327                if (OK == dev->get_master_volume(dev, &mv)) {
1328                    mMasterVolume = mv;
1329                }
1330            }
1331
1332            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1333            if (NULL != dev->get_master_mute) {
1334                bool mm;
1335                if (OK == dev->get_master_mute(dev, &mm)) {
1336                    mMasterMute = mm;
1337                }
1338            }
1339        }
1340
1341        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1342        if ((NULL != dev->set_master_volume) &&
1343            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1344            flags = static_cast<AudioHwDevice::Flags>(flags |
1345                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1346        }
1347
1348        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1349        if ((NULL != dev->set_master_mute) &&
1350            (OK == dev->set_master_mute(dev, mMasterMute))) {
1351            flags = static_cast<AudioHwDevice::Flags>(flags |
1352                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1353        }
1354
1355        mHardwareStatus = AUDIO_HW_IDLE;
1356    }
1357
1358    audio_module_handle_t handle = nextUniqueId();
1359    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1360
1361    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1362          name, dev->common.module->name, dev->common.module->id, handle);
1363
1364    return handle;
1365
1366}
1367
1368// ----------------------------------------------------------------------------
1369
1370uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1371{
1372    Mutex::Autolock _l(mLock);
1373    PlaybackThread *thread = primaryPlaybackThread_l();
1374    return thread != NULL ? thread->sampleRate() : 0;
1375}
1376
1377size_t AudioFlinger::getPrimaryOutputFrameCount()
1378{
1379    Mutex::Autolock _l(mLock);
1380    PlaybackThread *thread = primaryPlaybackThread_l();
1381    return thread != NULL ? thread->frameCountHAL() : 0;
1382}
1383
1384// ----------------------------------------------------------------------------
1385
1386status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1387{
1388    uid_t uid = IPCThreadState::self()->getCallingUid();
1389    if (uid != AID_SYSTEM) {
1390        return PERMISSION_DENIED;
1391    }
1392    Mutex::Autolock _l(mLock);
1393    if (mIsDeviceTypeKnown) {
1394        return INVALID_OPERATION;
1395    }
1396    mIsLowRamDevice = isLowRamDevice;
1397    mIsDeviceTypeKnown = true;
1398    return NO_ERROR;
1399}
1400
1401// ----------------------------------------------------------------------------
1402
1403audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1404                                           audio_devices_t *pDevices,
1405                                           uint32_t *pSamplingRate,
1406                                           audio_format_t *pFormat,
1407                                           audio_channel_mask_t *pChannelMask,
1408                                           uint32_t *pLatencyMs,
1409                                           audio_output_flags_t flags,
1410                                           const audio_offload_info_t *offloadInfo)
1411{
1412    PlaybackThread *thread = NULL;
1413    struct audio_config config;
1414    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1415    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1416    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1417    if (offloadInfo) {
1418        config.offload_info = *offloadInfo;
1419    }
1420
1421    audio_stream_out_t *outStream = NULL;
1422    AudioHwDevice *outHwDev;
1423
1424    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1425              module,
1426              (pDevices != NULL) ? *pDevices : 0,
1427              config.sample_rate,
1428              config.format,
1429              config.channel_mask,
1430              flags);
1431    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1432          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version );
1433
1434    if (pDevices == NULL || *pDevices == 0) {
1435        return 0;
1436    }
1437
1438    Mutex::Autolock _l(mLock);
1439
1440    outHwDev = findSuitableHwDev_l(module, *pDevices);
1441    if (outHwDev == NULL)
1442        return 0;
1443
1444    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1445    audio_io_handle_t id = nextUniqueId();
1446
1447    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1448
1449    status_t status = hwDevHal->open_output_stream(hwDevHal,
1450                                          id,
1451                                          *pDevices,
1452                                          (audio_output_flags_t)flags,
1453                                          &config,
1454                                          &outStream);
1455
1456    mHardwareStatus = AUDIO_HW_IDLE;
1457    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1458            "Channels %x, status %d",
1459            outStream,
1460            config.sample_rate,
1461            config.format,
1462            config.channel_mask,
1463            status);
1464
1465    if (status == NO_ERROR && outStream != NULL) {
1466        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1467
1468        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1469            thread = new OffloadThread(this, output, id, *pDevices);
1470            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1471        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1472            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1473            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1474            thread = new DirectOutputThread(this, output, id, *pDevices);
1475            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1476        } else {
1477            thread = new MixerThread(this, output, id, *pDevices);
1478            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1479        }
1480        mPlaybackThreads.add(id, thread);
1481
1482        if (pSamplingRate != NULL) {
1483            *pSamplingRate = config.sample_rate;
1484        }
1485        if (pFormat != NULL) {
1486            *pFormat = config.format;
1487        }
1488        if (pChannelMask != NULL) {
1489            *pChannelMask = config.channel_mask;
1490        }
1491        if (pLatencyMs != NULL) {
1492            *pLatencyMs = thread->latency();
1493        }
1494
1495        // notify client processes of the new output creation
1496        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1497
1498        // the first primary output opened designates the primary hw device
1499        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1500            ALOGI("Using module %d has the primary audio interface", module);
1501            mPrimaryHardwareDev = outHwDev;
1502
1503            AutoMutex lock(mHardwareLock);
1504            mHardwareStatus = AUDIO_HW_SET_MODE;
1505            hwDevHal->set_mode(hwDevHal, mMode);
1506            mHardwareStatus = AUDIO_HW_IDLE;
1507        }
1508        return id;
1509    }
1510
1511    return 0;
1512}
1513
1514audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1515        audio_io_handle_t output2)
1516{
1517    Mutex::Autolock _l(mLock);
1518    MixerThread *thread1 = checkMixerThread_l(output1);
1519    MixerThread *thread2 = checkMixerThread_l(output2);
1520
1521    if (thread1 == NULL || thread2 == NULL) {
1522        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1523                output2);
1524        return 0;
1525    }
1526
1527    audio_io_handle_t id = nextUniqueId();
1528    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1529    thread->addOutputTrack(thread2);
1530    mPlaybackThreads.add(id, thread);
1531    // notify client processes of the new output creation
1532    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1533    return id;
1534}
1535
1536status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1537{
1538    return closeOutput_nonvirtual(output);
1539}
1540
1541status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1542{
1543    // keep strong reference on the playback thread so that
1544    // it is not destroyed while exit() is executed
1545    sp<PlaybackThread> thread;
1546    {
1547        Mutex::Autolock _l(mLock);
1548        thread = checkPlaybackThread_l(output);
1549        if (thread == NULL) {
1550            return BAD_VALUE;
1551        }
1552
1553        ALOGV("closeOutput() %d", output);
1554
1555        if (thread->type() == ThreadBase::MIXER) {
1556            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1557                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1558                    DuplicatingThread *dupThread =
1559                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1560                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1561
1562                }
1563            }
1564        }
1565
1566
1567        mPlaybackThreads.removeItem(output);
1568        // save all effects to the default thread
1569        if (mPlaybackThreads.size()) {
1570            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1571            if (dstThread != NULL) {
1572                // audioflinger lock is held here so the acquisition order of thread locks does not
1573                // matter
1574                Mutex::Autolock _dl(dstThread->mLock);
1575                Mutex::Autolock _sl(thread->mLock);
1576                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1577                for (size_t i = 0; i < effectChains.size(); i ++) {
1578                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1579                }
1580            }
1581        }
1582        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1583    }
1584    thread->exit();
1585    // The thread entity (active unit of execution) is no longer running here,
1586    // but the ThreadBase container still exists.
1587
1588    if (thread->type() != ThreadBase::DUPLICATING) {
1589        AudioStreamOut *out = thread->clearOutput();
1590        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1591        // from now on thread->mOutput is NULL
1592        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1593        delete out;
1594    }
1595    return NO_ERROR;
1596}
1597
1598status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1599{
1600    Mutex::Autolock _l(mLock);
1601    PlaybackThread *thread = checkPlaybackThread_l(output);
1602
1603    if (thread == NULL) {
1604        return BAD_VALUE;
1605    }
1606
1607    ALOGV("suspendOutput() %d", output);
1608    thread->suspend();
1609
1610    return NO_ERROR;
1611}
1612
1613status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1614{
1615    Mutex::Autolock _l(mLock);
1616    PlaybackThread *thread = checkPlaybackThread_l(output);
1617
1618    if (thread == NULL) {
1619        return BAD_VALUE;
1620    }
1621
1622    ALOGV("restoreOutput() %d", output);
1623
1624    thread->restore();
1625
1626    return NO_ERROR;
1627}
1628
1629audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1630                                          audio_devices_t *pDevices,
1631                                          uint32_t *pSamplingRate,
1632                                          audio_format_t *pFormat,
1633                                          audio_channel_mask_t *pChannelMask)
1634{
1635    status_t status;
1636    RecordThread *thread = NULL;
1637    struct audio_config config;
1638    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1639    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1640    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1641
1642    uint32_t reqSamplingRate = config.sample_rate;
1643    audio_format_t reqFormat = config.format;
1644    audio_channel_mask_t reqChannels = config.channel_mask;
1645    audio_stream_in_t *inStream = NULL;
1646    AudioHwDevice *inHwDev;
1647
1648    if (pDevices == NULL || *pDevices == 0) {
1649        return 0;
1650    }
1651
1652    Mutex::Autolock _l(mLock);
1653
1654    inHwDev = findSuitableHwDev_l(module, *pDevices);
1655    if (inHwDev == NULL)
1656        return 0;
1657
1658    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1659    audio_io_handle_t id = nextUniqueId();
1660
1661    status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1662                                        &inStream);
1663    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1664            "status %d",
1665            inStream,
1666            config.sample_rate,
1667            config.format,
1668            config.channel_mask,
1669            status);
1670
1671    // If the input could not be opened with the requested parameters and we can handle the
1672    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1673    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1674    if (status == BAD_VALUE &&
1675        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1676        (config.sample_rate <= 2 * reqSamplingRate) &&
1677        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
1678        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1679        inStream = NULL;
1680        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1681    }
1682
1683    if (status == NO_ERROR && inStream != NULL) {
1684
1685#ifdef TEE_SINK
1686        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1687        // or (re-)create if current Pipe is idle and does not match the new format
1688        sp<NBAIO_Sink> teeSink;
1689        enum {
1690            TEE_SINK_NO,    // don't copy input
1691            TEE_SINK_NEW,   // copy input using a new pipe
1692            TEE_SINK_OLD,   // copy input using an existing pipe
1693        } kind;
1694        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1695                                        popcount(inStream->common.get_channels(&inStream->common)));
1696        if (!mTeeSinkInputEnabled) {
1697            kind = TEE_SINK_NO;
1698        } else if (format == Format_Invalid) {
1699            kind = TEE_SINK_NO;
1700        } else if (mRecordTeeSink == 0) {
1701            kind = TEE_SINK_NEW;
1702        } else if (mRecordTeeSink->getStrongCount() != 1) {
1703            kind = TEE_SINK_NO;
1704        } else if (format == mRecordTeeSink->format()) {
1705            kind = TEE_SINK_OLD;
1706        } else {
1707            kind = TEE_SINK_NEW;
1708        }
1709        switch (kind) {
1710        case TEE_SINK_NEW: {
1711            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1712            size_t numCounterOffers = 0;
1713            const NBAIO_Format offers[1] = {format};
1714            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1715            ALOG_ASSERT(index == 0);
1716            PipeReader *pipeReader = new PipeReader(*pipe);
1717            numCounterOffers = 0;
1718            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1719            ALOG_ASSERT(index == 0);
1720            mRecordTeeSink = pipe;
1721            mRecordTeeSource = pipeReader;
1722            teeSink = pipe;
1723            }
1724            break;
1725        case TEE_SINK_OLD:
1726            teeSink = mRecordTeeSink;
1727            break;
1728        case TEE_SINK_NO:
1729        default:
1730            break;
1731        }
1732#endif
1733
1734        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1735
1736        // Start record thread
1737        // RecorThread require both input and output device indication to forward to audio
1738        // pre processing modules
1739        thread = new RecordThread(this,
1740                                  input,
1741                                  reqSamplingRate,
1742                                  reqChannels,
1743                                  id,
1744                                  primaryOutputDevice_l(),
1745                                  *pDevices
1746#ifdef TEE_SINK
1747                                  , teeSink
1748#endif
1749                                  );
1750        mRecordThreads.add(id, thread);
1751        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1752        if (pSamplingRate != NULL) {
1753            *pSamplingRate = reqSamplingRate;
1754        }
1755        if (pFormat != NULL) {
1756            *pFormat = config.format;
1757        }
1758        if (pChannelMask != NULL) {
1759            *pChannelMask = reqChannels;
1760        }
1761
1762        // notify client processes of the new input creation
1763        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1764        return id;
1765    }
1766
1767    return 0;
1768}
1769
1770status_t AudioFlinger::closeInput(audio_io_handle_t input)
1771{
1772    return closeInput_nonvirtual(input);
1773}
1774
1775status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1776{
1777    // keep strong reference on the record thread so that
1778    // it is not destroyed while exit() is executed
1779    sp<RecordThread> thread;
1780    {
1781        Mutex::Autolock _l(mLock);
1782        thread = checkRecordThread_l(input);
1783        if (thread == 0) {
1784            return BAD_VALUE;
1785        }
1786
1787        ALOGV("closeInput() %d", input);
1788        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1789        mRecordThreads.removeItem(input);
1790    }
1791    thread->exit();
1792    // The thread entity (active unit of execution) is no longer running here,
1793    // but the ThreadBase container still exists.
1794
1795    AudioStreamIn *in = thread->clearInput();
1796    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1797    // from now on thread->mInput is NULL
1798    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1799    delete in;
1800
1801    return NO_ERROR;
1802}
1803
1804status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1805{
1806    Mutex::Autolock _l(mLock);
1807    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1808
1809    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1810        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1811        thread->invalidateTracks(stream);
1812    }
1813
1814    return NO_ERROR;
1815}
1816
1817
1818int AudioFlinger::newAudioSessionId()
1819{
1820    return nextUniqueId();
1821}
1822
1823void AudioFlinger::acquireAudioSessionId(int audioSession)
1824{
1825    Mutex::Autolock _l(mLock);
1826    pid_t caller = IPCThreadState::self()->getCallingPid();
1827    ALOGV("acquiring %d from %d", audioSession, caller);
1828    size_t num = mAudioSessionRefs.size();
1829    for (size_t i = 0; i< num; i++) {
1830        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1831        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1832            ref->mCnt++;
1833            ALOGV(" incremented refcount to %d", ref->mCnt);
1834            return;
1835        }
1836    }
1837    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1838    ALOGV(" added new entry for %d", audioSession);
1839}
1840
1841void AudioFlinger::releaseAudioSessionId(int audioSession)
1842{
1843    Mutex::Autolock _l(mLock);
1844    pid_t caller = IPCThreadState::self()->getCallingPid();
1845    ALOGV("releasing %d from %d", audioSession, caller);
1846    size_t num = mAudioSessionRefs.size();
1847    for (size_t i = 0; i< num; i++) {
1848        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1849        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1850            ref->mCnt--;
1851            ALOGV(" decremented refcount to %d", ref->mCnt);
1852            if (ref->mCnt == 0) {
1853                mAudioSessionRefs.removeAt(i);
1854                delete ref;
1855                purgeStaleEffects_l();
1856            }
1857            return;
1858        }
1859    }
1860    ALOGW("session id %d not found for pid %d", audioSession, caller);
1861}
1862
1863void AudioFlinger::purgeStaleEffects_l() {
1864
1865    ALOGV("purging stale effects");
1866
1867    Vector< sp<EffectChain> > chains;
1868
1869    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1870        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1871        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1872            sp<EffectChain> ec = t->mEffectChains[j];
1873            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1874                chains.push(ec);
1875            }
1876        }
1877    }
1878    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1879        sp<RecordThread> t = mRecordThreads.valueAt(i);
1880        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1881            sp<EffectChain> ec = t->mEffectChains[j];
1882            chains.push(ec);
1883        }
1884    }
1885
1886    for (size_t i = 0; i < chains.size(); i++) {
1887        sp<EffectChain> ec = chains[i];
1888        int sessionid = ec->sessionId();
1889        sp<ThreadBase> t = ec->mThread.promote();
1890        if (t == 0) {
1891            continue;
1892        }
1893        size_t numsessionrefs = mAudioSessionRefs.size();
1894        bool found = false;
1895        for (size_t k = 0; k < numsessionrefs; k++) {
1896            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1897            if (ref->mSessionid == sessionid) {
1898                ALOGV(" session %d still exists for %d with %d refs",
1899                    sessionid, ref->mPid, ref->mCnt);
1900                found = true;
1901                break;
1902            }
1903        }
1904        if (!found) {
1905            Mutex::Autolock _l (t->mLock);
1906            // remove all effects from the chain
1907            while (ec->mEffects.size()) {
1908                sp<EffectModule> effect = ec->mEffects[0];
1909                effect->unPin();
1910                t->removeEffect_l(effect);
1911                if (effect->purgeHandles()) {
1912                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
1913                }
1914                AudioSystem::unregisterEffect(effect->id());
1915            }
1916        }
1917    }
1918    return;
1919}
1920
1921// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
1922AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
1923{
1924    return mPlaybackThreads.valueFor(output).get();
1925}
1926
1927// checkMixerThread_l() must be called with AudioFlinger::mLock held
1928AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
1929{
1930    PlaybackThread *thread = checkPlaybackThread_l(output);
1931    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
1932}
1933
1934// checkRecordThread_l() must be called with AudioFlinger::mLock held
1935AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
1936{
1937    return mRecordThreads.valueFor(input).get();
1938}
1939
1940uint32_t AudioFlinger::nextUniqueId()
1941{
1942    return android_atomic_inc(&mNextUniqueId);
1943}
1944
1945AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
1946{
1947    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1948        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1949        AudioStreamOut *output = thread->getOutput();
1950        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
1951            return thread;
1952        }
1953    }
1954    return NULL;
1955}
1956
1957audio_devices_t AudioFlinger::primaryOutputDevice_l() const
1958{
1959    PlaybackThread *thread = primaryPlaybackThread_l();
1960
1961    if (thread == NULL) {
1962        return 0;
1963    }
1964
1965    return thread->outDevice();
1966}
1967
1968sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
1969                                    int triggerSession,
1970                                    int listenerSession,
1971                                    sync_event_callback_t callBack,
1972                                    void *cookie)
1973{
1974    Mutex::Autolock _l(mLock);
1975
1976    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
1977    status_t playStatus = NAME_NOT_FOUND;
1978    status_t recStatus = NAME_NOT_FOUND;
1979    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1980        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
1981        if (playStatus == NO_ERROR) {
1982            return event;
1983        }
1984    }
1985    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1986        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
1987        if (recStatus == NO_ERROR) {
1988            return event;
1989        }
1990    }
1991    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
1992        mPendingSyncEvents.add(event);
1993    } else {
1994        ALOGV("createSyncEvent() invalid event %d", event->type());
1995        event.clear();
1996    }
1997    return event;
1998}
1999
2000// ----------------------------------------------------------------------------
2001//  Effect management
2002// ----------------------------------------------------------------------------
2003
2004
2005status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2006{
2007    Mutex::Autolock _l(mLock);
2008    return EffectQueryNumberEffects(numEffects);
2009}
2010
2011status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2012{
2013    Mutex::Autolock _l(mLock);
2014    return EffectQueryEffect(index, descriptor);
2015}
2016
2017status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2018        effect_descriptor_t *descriptor) const
2019{
2020    Mutex::Autolock _l(mLock);
2021    return EffectGetDescriptor(pUuid, descriptor);
2022}
2023
2024
2025sp<IEffect> AudioFlinger::createEffect(
2026        effect_descriptor_t *pDesc,
2027        const sp<IEffectClient>& effectClient,
2028        int32_t priority,
2029        audio_io_handle_t io,
2030        int sessionId,
2031        status_t *status,
2032        int *id,
2033        int *enabled)
2034{
2035    status_t lStatus = NO_ERROR;
2036    sp<EffectHandle> handle;
2037    effect_descriptor_t desc;
2038
2039    pid_t pid = IPCThreadState::self()->getCallingPid();
2040    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2041            pid, effectClient.get(), priority, sessionId, io);
2042
2043    if (pDesc == NULL) {
2044        lStatus = BAD_VALUE;
2045        goto Exit;
2046    }
2047
2048    // check audio settings permission for global effects
2049    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2050        lStatus = PERMISSION_DENIED;
2051        goto Exit;
2052    }
2053
2054    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2055    // that can only be created by audio policy manager (running in same process)
2056    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2057        lStatus = PERMISSION_DENIED;
2058        goto Exit;
2059    }
2060
2061    if (io == 0) {
2062        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2063            // output must be specified by AudioPolicyManager when using session
2064            // AUDIO_SESSION_OUTPUT_STAGE
2065            lStatus = BAD_VALUE;
2066            goto Exit;
2067        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2068            // if the output returned by getOutputForEffect() is removed before we lock the
2069            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2070            // and we will exit safely
2071            io = AudioSystem::getOutputForEffect(&desc);
2072        }
2073    }
2074
2075    {
2076        Mutex::Autolock _l(mLock);
2077
2078
2079        if (!EffectIsNullUuid(&pDesc->uuid)) {
2080            // if uuid is specified, request effect descriptor
2081            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2082            if (lStatus < 0) {
2083                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2084                goto Exit;
2085            }
2086        } else {
2087            // if uuid is not specified, look for an available implementation
2088            // of the required type in effect factory
2089            if (EffectIsNullUuid(&pDesc->type)) {
2090                ALOGW("createEffect() no effect type");
2091                lStatus = BAD_VALUE;
2092                goto Exit;
2093            }
2094            uint32_t numEffects = 0;
2095            effect_descriptor_t d;
2096            d.flags = 0; // prevent compiler warning
2097            bool found = false;
2098
2099            lStatus = EffectQueryNumberEffects(&numEffects);
2100            if (lStatus < 0) {
2101                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2102                goto Exit;
2103            }
2104            for (uint32_t i = 0; i < numEffects; i++) {
2105                lStatus = EffectQueryEffect(i, &desc);
2106                if (lStatus < 0) {
2107                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2108                    continue;
2109                }
2110                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2111                    // If matching type found save effect descriptor. If the session is
2112                    // 0 and the effect is not auxiliary, continue enumeration in case
2113                    // an auxiliary version of this effect type is available
2114                    found = true;
2115                    d = desc;
2116                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2117                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2118                        break;
2119                    }
2120                }
2121            }
2122            if (!found) {
2123                lStatus = BAD_VALUE;
2124                ALOGW("createEffect() effect not found");
2125                goto Exit;
2126            }
2127            // For same effect type, chose auxiliary version over insert version if
2128            // connect to output mix (Compliance to OpenSL ES)
2129            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2130                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2131                desc = d;
2132            }
2133        }
2134
2135        // Do not allow auxiliary effects on a session different from 0 (output mix)
2136        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2137             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2138            lStatus = INVALID_OPERATION;
2139            goto Exit;
2140        }
2141
2142        // check recording permission for visualizer
2143        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2144            !recordingAllowed()) {
2145            lStatus = PERMISSION_DENIED;
2146            goto Exit;
2147        }
2148
2149        // return effect descriptor
2150        *pDesc = desc;
2151
2152        // If output is not specified try to find a matching audio session ID in one of the
2153        // output threads.
2154        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2155        // because of code checking output when entering the function.
2156        // Note: io is never 0 when creating an effect on an input
2157        if (io == 0) {
2158            // look for the thread where the specified audio session is present
2159            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2160                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2161                    io = mPlaybackThreads.keyAt(i);
2162                    break;
2163                }
2164            }
2165            if (io == 0) {
2166                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2167                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2168                        io = mRecordThreads.keyAt(i);
2169                        break;
2170                    }
2171                }
2172            }
2173            // If no output thread contains the requested session ID, default to
2174            // first output. The effect chain will be moved to the correct output
2175            // thread when a track with the same session ID is created
2176            if (io == 0 && mPlaybackThreads.size()) {
2177                io = mPlaybackThreads.keyAt(0);
2178            }
2179            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2180        }
2181        ThreadBase *thread = checkRecordThread_l(io);
2182        if (thread == NULL) {
2183            thread = checkPlaybackThread_l(io);
2184            if (thread == NULL) {
2185                ALOGE("createEffect() unknown output thread");
2186                lStatus = BAD_VALUE;
2187                goto Exit;
2188            }
2189        }
2190
2191        sp<Client> client = registerPid_l(pid);
2192
2193        // create effect on selected output thread
2194        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2195                &desc, enabled, &lStatus);
2196        if (handle != 0 && id != NULL) {
2197            *id = handle->id();
2198        }
2199    }
2200
2201Exit:
2202    if (status != NULL) {
2203        *status = lStatus;
2204    }
2205    return handle;
2206}
2207
2208status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2209        audio_io_handle_t dstOutput)
2210{
2211    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2212            sessionId, srcOutput, dstOutput);
2213    Mutex::Autolock _l(mLock);
2214    if (srcOutput == dstOutput) {
2215        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2216        return NO_ERROR;
2217    }
2218    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2219    if (srcThread == NULL) {
2220        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2221        return BAD_VALUE;
2222    }
2223    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2224    if (dstThread == NULL) {
2225        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2226        return BAD_VALUE;
2227    }
2228
2229    Mutex::Autolock _dl(dstThread->mLock);
2230    Mutex::Autolock _sl(srcThread->mLock);
2231    moveEffectChain_l(sessionId, srcThread, dstThread, false);
2232
2233    return NO_ERROR;
2234}
2235
2236// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2237status_t AudioFlinger::moveEffectChain_l(int sessionId,
2238                                   AudioFlinger::PlaybackThread *srcThread,
2239                                   AudioFlinger::PlaybackThread *dstThread,
2240                                   bool reRegister)
2241{
2242    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2243            sessionId, srcThread, dstThread);
2244
2245    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2246    if (chain == 0) {
2247        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2248                sessionId, srcThread);
2249        return INVALID_OPERATION;
2250    }
2251
2252    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2253    // so that a new chain is created with correct parameters when first effect is added. This is
2254    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2255    // removed.
2256    srcThread->removeEffectChain_l(chain);
2257
2258    // transfer all effects one by one so that new effect chain is created on new thread with
2259    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2260    audio_io_handle_t dstOutput = dstThread->id();
2261    sp<EffectChain> dstChain;
2262    uint32_t strategy = 0; // prevent compiler warning
2263    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2264    while (effect != 0) {
2265        srcThread->removeEffect_l(effect);
2266        dstThread->addEffect_l(effect);
2267        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2268        if (effect->state() == EffectModule::ACTIVE ||
2269                effect->state() == EffectModule::STOPPING) {
2270            effect->start();
2271        }
2272        // if the move request is not received from audio policy manager, the effect must be
2273        // re-registered with the new strategy and output
2274        if (dstChain == 0) {
2275            dstChain = effect->chain().promote();
2276            if (dstChain == 0) {
2277                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2278                srcThread->addEffect_l(effect);
2279                return NO_INIT;
2280            }
2281            strategy = dstChain->strategy();
2282        }
2283        if (reRegister) {
2284            AudioSystem::unregisterEffect(effect->id());
2285            AudioSystem::registerEffect(&effect->desc(),
2286                                        dstOutput,
2287                                        strategy,
2288                                        sessionId,
2289                                        effect->id());
2290        }
2291        effect = chain->getEffectFromId_l(0);
2292    }
2293
2294    return NO_ERROR;
2295}
2296
2297struct Entry {
2298#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2299    char mName[MAX_NAME];
2300};
2301
2302int comparEntry(const void *p1, const void *p2)
2303{
2304    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2305}
2306
2307#ifdef TEE_SINK
2308void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2309{
2310    NBAIO_Source *teeSource = source.get();
2311    if (teeSource != NULL) {
2312        // .wav rotation
2313        // There is a benign race condition if 2 threads call this simultaneously.
2314        // They would both traverse the directory, but the result would simply be
2315        // failures at unlink() which are ignored.  It's also unlikely since
2316        // normally dumpsys is only done by bugreport or from the command line.
2317        char teePath[32+256];
2318        strcpy(teePath, "/data/misc/media");
2319        size_t teePathLen = strlen(teePath);
2320        DIR *dir = opendir(teePath);
2321        teePath[teePathLen++] = '/';
2322        if (dir != NULL) {
2323#define MAX_SORT 20 // number of entries to sort
2324#define MAX_KEEP 10 // number of entries to keep
2325            struct Entry entries[MAX_SORT];
2326            size_t entryCount = 0;
2327            while (entryCount < MAX_SORT) {
2328                struct dirent de;
2329                struct dirent *result = NULL;
2330                int rc = readdir_r(dir, &de, &result);
2331                if (rc != 0) {
2332                    ALOGW("readdir_r failed %d", rc);
2333                    break;
2334                }
2335                if (result == NULL) {
2336                    break;
2337                }
2338                if (result != &de) {
2339                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2340                    break;
2341                }
2342                // ignore non .wav file entries
2343                size_t nameLen = strlen(de.d_name);
2344                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2345                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2346                    continue;
2347                }
2348                strcpy(entries[entryCount++].mName, de.d_name);
2349            }
2350            (void) closedir(dir);
2351            if (entryCount > MAX_KEEP) {
2352                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2353                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2354                    strcpy(&teePath[teePathLen], entries[i].mName);
2355                    (void) unlink(teePath);
2356                }
2357            }
2358        } else {
2359            if (fd >= 0) {
2360                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2361            }
2362        }
2363        char teeTime[16];
2364        struct timeval tv;
2365        gettimeofday(&tv, NULL);
2366        struct tm tm;
2367        localtime_r(&tv.tv_sec, &tm);
2368        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2369        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2370        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2371        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2372        if (teeFd >= 0) {
2373            char wavHeader[44];
2374            memcpy(wavHeader,
2375                "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",
2376                sizeof(wavHeader));
2377            NBAIO_Format format = teeSource->format();
2378            unsigned channelCount = Format_channelCount(format);
2379            ALOG_ASSERT(channelCount <= FCC_2);
2380            uint32_t sampleRate = Format_sampleRate(format);
2381            wavHeader[22] = channelCount;       // number of channels
2382            wavHeader[24] = sampleRate;         // sample rate
2383            wavHeader[25] = sampleRate >> 8;
2384            wavHeader[32] = channelCount * 2;   // block alignment
2385            write(teeFd, wavHeader, sizeof(wavHeader));
2386            size_t total = 0;
2387            bool firstRead = true;
2388            for (;;) {
2389#define TEE_SINK_READ 1024
2390                short buffer[TEE_SINK_READ * FCC_2];
2391                size_t count = TEE_SINK_READ;
2392                ssize_t actual = teeSource->read(buffer, count,
2393                        AudioBufferProvider::kInvalidPTS);
2394                bool wasFirstRead = firstRead;
2395                firstRead = false;
2396                if (actual <= 0) {
2397                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2398                        continue;
2399                    }
2400                    break;
2401                }
2402                ALOG_ASSERT(actual <= (ssize_t)count);
2403                write(teeFd, buffer, actual * channelCount * sizeof(short));
2404                total += actual;
2405            }
2406            lseek(teeFd, (off_t) 4, SEEK_SET);
2407            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2408            write(teeFd, &temp, sizeof(temp));
2409            lseek(teeFd, (off_t) 40, SEEK_SET);
2410            temp =  total * channelCount * sizeof(short);
2411            write(teeFd, &temp, sizeof(temp));
2412            close(teeFd);
2413            if (fd >= 0) {
2414                fdprintf(fd, "tee copied to %s\n", teePath);
2415            }
2416        } else {
2417            if (fd >= 0) {
2418                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2419            }
2420        }
2421    }
2422}
2423#endif
2424
2425// ----------------------------------------------------------------------------
2426
2427status_t AudioFlinger::onTransact(
2428        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2429{
2430    return BnAudioFlinger::onTransact(code, data, reply, flags);
2431}
2432
2433}; // namespace android
2434