AudioFlinger.cpp revision e93cf2ca27ae6f4a81d4ef548bbf10a34db6d98f 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <dirent.h>
24#include <math.h>
25#include <signal.h>
26#include <sys/time.h>
27#include <sys/resource.h>
28
29#include <binder/IPCThreadState.h>
30#include <binder/IServiceManager.h>
31#include <utils/Log.h>
32#include <utils/Trace.h>
33#include <binder/Parcel.h>
34#include <utils/String16.h>
35#include <utils/threads.h>
36#include <utils/Atomic.h>
37
38#include <cutils/bitops.h>
39#include <cutils/properties.h>
40
41#include <system/audio.h>
42#include <hardware/audio.h>
43
44#include "AudioMixer.h"
45#include "AudioFlinger.h"
46#include "ServiceUtilities.h"
47
48#include <media/EffectsFactoryApi.h>
49#include <audio_effects/effect_visualizer.h>
50#include <audio_effects/effect_ns.h>
51#include <audio_effects/effect_aec.h>
52
53#include <audio_utils/primitives.h>
54
55#include <powermanager/PowerManager.h>
56
57#include <common_time/cc_helper.h>
58
59#include <media/IMediaLogService.h>
60
61#include <media/nbaio/Pipe.h>
62#include <media/nbaio/PipeReader.h>
63#include <media/AudioParameter.h>
64#include <private/android_filesystem_config.h>
65
66// ----------------------------------------------------------------------------
67
68// Note: the following macro is used for extremely verbose logging message.  In
69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
70// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
71// are so verbose that we want to suppress them even when we have ALOG_ASSERT
72// turned on.  Do not uncomment the #def below unless you really know what you
73// are doing and want to see all of the extremely verbose messages.
74//#define VERY_VERY_VERBOSE_LOGGING
75#ifdef VERY_VERY_VERBOSE_LOGGING
76#define ALOGVV ALOGV
77#else
78#define ALOGVV(a...) do { } while(0)
79#endif
80
81namespace android {
82
83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
84static const char kHardwareLockedString[] = "Hardware lock is taken\n";
85
86
87nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
88
89uint32_t AudioFlinger::mScreenState;
90
91#ifdef TEE_SINK
92bool AudioFlinger::mTeeSinkInputEnabled = false;
93bool AudioFlinger::mTeeSinkOutputEnabled = false;
94bool AudioFlinger::mTeeSinkTrackEnabled = false;
95
96size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
97size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
98size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
99#endif
100
101// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
102// we define a minimum time during which a global effect is considered enabled.
103static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
104
105// ----------------------------------------------------------------------------
106
107static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
108{
109    const hw_module_t *mod;
110    int rc;
111
112    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
113    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
114                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
115    if (rc) {
116        goto out;
117    }
118    rc = audio_hw_device_open(mod, dev);
119    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
120                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
121    if (rc) {
122        goto out;
123    }
124    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
125        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
126        rc = BAD_VALUE;
127        goto out;
128    }
129    return 0;
130
131out:
132    *dev = NULL;
133    return rc;
134}
135
136// ----------------------------------------------------------------------------
137
138AudioFlinger::AudioFlinger()
139    : BnAudioFlinger(),
140      mPrimaryHardwareDev(NULL),
141      mHardwareStatus(AUDIO_HW_IDLE),
142      mMasterVolume(1.0f),
143      mMasterMute(false),
144      mNextUniqueId(1),
145      mMode(AUDIO_MODE_INVALID),
146      mBtNrecIsOff(false),
147      mIsLowRamDevice(true),
148      mIsDeviceTypeKnown(false),
149      mGlobalEffectEnableTime(0)
150{
151    getpid_cached = getpid();
152    char value[PROPERTY_VALUE_MAX];
153    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
154    if (doLog) {
155        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
156    }
157#ifdef TEE_SINK
158    (void) property_get("ro.debuggable", value, "0");
159    int debuggable = atoi(value);
160    int teeEnabled = 0;
161    if (debuggable) {
162        (void) property_get("af.tee", value, "0");
163        teeEnabled = atoi(value);
164    }
165    if (teeEnabled & 1)
166        mTeeSinkInputEnabled = true;
167    if (teeEnabled & 2)
168        mTeeSinkOutputEnabled = true;
169    if (teeEnabled & 4)
170        mTeeSinkTrackEnabled = true;
171#endif
172}
173
174void AudioFlinger::onFirstRef()
175{
176    int rc = 0;
177
178    Mutex::Autolock _l(mLock);
179
180    /* TODO: move all this work into an Init() function */
181    char val_str[PROPERTY_VALUE_MAX] = { 0 };
182    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
183        uint32_t int_val;
184        if (1 == sscanf(val_str, "%u", &int_val)) {
185            mStandbyTimeInNsecs = milliseconds(int_val);
186            ALOGI("Using %u mSec as standby time.", int_val);
187        } else {
188            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
189            ALOGI("Using default %u mSec as standby time.",
190                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
191        }
192    }
193
194    mMode = AUDIO_MODE_NORMAL;
195}
196
197AudioFlinger::~AudioFlinger()
198{
199    while (!mRecordThreads.isEmpty()) {
200        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
201        closeInput_nonvirtual(mRecordThreads.keyAt(0));
202    }
203    while (!mPlaybackThreads.isEmpty()) {
204        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
205        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
206    }
207
208    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
209        // no mHardwareLock needed, as there are no other references to this
210        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
211        delete mAudioHwDevs.valueAt(i);
212    }
213}
214
215static const char * const audio_interfaces[] = {
216    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
217    AUDIO_HARDWARE_MODULE_ID_A2DP,
218    AUDIO_HARDWARE_MODULE_ID_USB,
219};
220#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
221
222AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
223        audio_module_handle_t module,
224        audio_devices_t devices)
225{
226    // if module is 0, the request comes from an old policy manager and we should load
227    // well known modules
228    if (module == 0) {
229        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
230        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
231            loadHwModule_l(audio_interfaces[i]);
232        }
233        // then try to find a module supporting the requested device.
234        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
235            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
236            audio_hw_device_t *dev = audioHwDevice->hwDevice();
237            if ((dev->get_supported_devices != NULL) &&
238                    (dev->get_supported_devices(dev) & devices) == devices)
239                return audioHwDevice;
240        }
241    } else {
242        // check a match for the requested module handle
243        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
244        if (audioHwDevice != NULL) {
245            return audioHwDevice;
246        }
247    }
248
249    return NULL;
250}
251
252void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
253{
254    const size_t SIZE = 256;
255    char buffer[SIZE];
256    String8 result;
257
258    result.append("Clients:\n");
259    for (size_t i = 0; i < mClients.size(); ++i) {
260        sp<Client> client = mClients.valueAt(i).promote();
261        if (client != 0) {
262            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
263            result.append(buffer);
264        }
265    }
266
267    result.append("Notification Clients:\n");
268    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
269        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
270        result.append(buffer);
271    }
272
273    result.append("Global session refs:\n");
274    result.append(" session pid count\n");
275    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
276        AudioSessionRef *r = mAudioSessionRefs[i];
277        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
278        result.append(buffer);
279    }
280    write(fd, result.string(), result.size());
281}
282
283
284void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
285{
286    const size_t SIZE = 256;
287    char buffer[SIZE];
288    String8 result;
289    hardware_call_state hardwareStatus = mHardwareStatus;
290
291    snprintf(buffer, SIZE, "Hardware status: %d\n"
292                           "Standby Time mSec: %u\n",
293                            hardwareStatus,
294                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
295    result.append(buffer);
296    write(fd, result.string(), result.size());
297}
298
299void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
300{
301    const size_t SIZE = 256;
302    char buffer[SIZE];
303    String8 result;
304    snprintf(buffer, SIZE, "Permission Denial: "
305            "can't dump AudioFlinger from pid=%d, uid=%d\n",
306            IPCThreadState::self()->getCallingPid(),
307            IPCThreadState::self()->getCallingUid());
308    result.append(buffer);
309    write(fd, result.string(), result.size());
310}
311
312bool AudioFlinger::dumpTryLock(Mutex& mutex)
313{
314    bool locked = false;
315    for (int i = 0; i < kDumpLockRetries; ++i) {
316        if (mutex.tryLock() == NO_ERROR) {
317            locked = true;
318            break;
319        }
320        usleep(kDumpLockSleepUs);
321    }
322    return locked;
323}
324
325status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
326{
327    if (!dumpAllowed()) {
328        dumpPermissionDenial(fd, args);
329    } else {
330        // get state of hardware lock
331        bool hardwareLocked = dumpTryLock(mHardwareLock);
332        if (!hardwareLocked) {
333            String8 result(kHardwareLockedString);
334            write(fd, result.string(), result.size());
335        } else {
336            mHardwareLock.unlock();
337        }
338
339        bool locked = dumpTryLock(mLock);
340
341        // failed to lock - AudioFlinger is probably deadlocked
342        if (!locked) {
343            String8 result(kDeadlockedString);
344            write(fd, result.string(), result.size());
345        }
346
347        dumpClients(fd, args);
348        dumpInternals(fd, args);
349
350        // dump playback threads
351        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
352            mPlaybackThreads.valueAt(i)->dump(fd, args);
353        }
354
355        // dump record threads
356        for (size_t i = 0; i < mRecordThreads.size(); i++) {
357            mRecordThreads.valueAt(i)->dump(fd, args);
358        }
359
360        // dump all hardware devs
361        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
362            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
363            dev->dump(dev, fd);
364        }
365
366#ifdef TEE_SINK
367        // dump the serially shared record tee sink
368        if (mRecordTeeSource != 0) {
369            dumpTee(fd, mRecordTeeSource);
370        }
371#endif
372
373        if (locked) {
374            mLock.unlock();
375        }
376
377        // append a copy of media.log here by forwarding fd to it, but don't attempt
378        // to lookup the service if it's not running, as it will block for a second
379        if (mLogMemoryDealer != 0) {
380            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
381            if (binder != 0) {
382                fdprintf(fd, "\nmedia.log:\n");
383                Vector<String16> args;
384                binder->dump(fd, args);
385            }
386        }
387    }
388    return NO_ERROR;
389}
390
391sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
392{
393    // If pid is already in the mClients wp<> map, then use that entry
394    // (for which promote() is always != 0), otherwise create a new entry and Client.
395    sp<Client> client = mClients.valueFor(pid).promote();
396    if (client == 0) {
397        client = new Client(this, pid);
398        mClients.add(pid, client);
399    }
400
401    return client;
402}
403
404sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
405{
406    if (mLogMemoryDealer == 0) {
407        return new NBLog::Writer();
408    }
409    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
410    sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
411    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
412    if (binder != 0) {
413        interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
414    }
415    return writer;
416}
417
418void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
419{
420    if (writer == 0) {
421        return;
422    }
423    sp<IMemory> iMemory(writer->getIMemory());
424    if (iMemory == 0) {
425        return;
426    }
427    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
428    if (binder != 0) {
429        interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
430        // Now the media.log remote reference to IMemory is gone.
431        // When our last local reference to IMemory also drops to zero,
432        // the IMemory destructor will deallocate the region from mMemoryDealer.
433    }
434}
435
436// IAudioFlinger interface
437
438
439sp<IAudioTrack> AudioFlinger::createTrack(
440        audio_stream_type_t streamType,
441        uint32_t sampleRate,
442        audio_format_t format,
443        audio_channel_mask_t channelMask,
444        size_t frameCount,
445        IAudioFlinger::track_flags_t *flags,
446        const sp<IMemory>& sharedBuffer,
447        audio_io_handle_t output,
448        pid_t tid,
449        int *sessionId,
450        String8& name,
451        status_t *status)
452{
453    sp<PlaybackThread::Track> track;
454    sp<TrackHandle> trackHandle;
455    sp<Client> client;
456    status_t lStatus;
457    int lSessionId;
458
459    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
460    // but if someone uses binder directly they could bypass that and cause us to crash
461    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
462        ALOGE("createTrack() invalid stream type %d", streamType);
463        lStatus = BAD_VALUE;
464        goto Exit;
465    }
466
467    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
468    // and we don't yet support 8.24 or 32-bit PCM
469    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
470        ALOGE("createTrack() invalid format %d", format);
471        lStatus = BAD_VALUE;
472        goto Exit;
473    }
474
475    {
476        Mutex::Autolock _l(mLock);
477        PlaybackThread *thread = checkPlaybackThread_l(output);
478        PlaybackThread *effectThread = NULL;
479        if (thread == NULL) {
480            ALOGE("no playback thread found for output handle %d", output);
481            lStatus = BAD_VALUE;
482            goto Exit;
483        }
484
485        pid_t pid = IPCThreadState::self()->getCallingPid();
486        client = registerPid_l(pid);
487
488        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
489        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
490            // check if an effect chain with the same session ID is present on another
491            // output thread and move it here.
492            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
493                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
494                if (mPlaybackThreads.keyAt(i) != output) {
495                    uint32_t sessions = t->hasAudioSession(*sessionId);
496                    if (sessions & PlaybackThread::EFFECT_SESSION) {
497                        effectThread = t.get();
498                        break;
499                    }
500                }
501            }
502            lSessionId = *sessionId;
503        } else {
504            // if no audio session id is provided, create one here
505            lSessionId = nextUniqueId();
506            if (sessionId != NULL) {
507                *sessionId = lSessionId;
508            }
509        }
510        ALOGV("createTrack() lSessionId: %d", lSessionId);
511
512        track = thread->createTrack_l(client, streamType, sampleRate, format,
513                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus);
514
515        // move effect chain to this output thread if an effect on same session was waiting
516        // for a track to be created
517        if (lStatus == NO_ERROR && effectThread != NULL) {
518            Mutex::Autolock _dl(thread->mLock);
519            Mutex::Autolock _sl(effectThread->mLock);
520            moveEffectChain_l(lSessionId, effectThread, thread, true);
521        }
522
523        // Look for sync events awaiting for a session to be used.
524        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
525            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
526                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
527                    if (lStatus == NO_ERROR) {
528                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
529                    } else {
530                        mPendingSyncEvents[i]->cancel();
531                    }
532                    mPendingSyncEvents.removeAt(i);
533                    i--;
534                }
535            }
536        }
537    }
538    if (lStatus == NO_ERROR) {
539        // s for server's pid, n for normal mixer name, f for fast index
540        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
541                track->fastIndex());
542        trackHandle = new TrackHandle(track);
543    } else {
544        // remove local strong reference to Client before deleting the Track so that the Client
545        // destructor is called by the TrackBase destructor with mLock held
546        client.clear();
547        track.clear();
548    }
549
550Exit:
551    if (status != NULL) {
552        *status = lStatus;
553    }
554    return trackHandle;
555}
556
557uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
558{
559    Mutex::Autolock _l(mLock);
560    PlaybackThread *thread = checkPlaybackThread_l(output);
561    if (thread == NULL) {
562        ALOGW("sampleRate() unknown thread %d", output);
563        return 0;
564    }
565    return thread->sampleRate();
566}
567
568int AudioFlinger::channelCount(audio_io_handle_t output) const
569{
570    Mutex::Autolock _l(mLock);
571    PlaybackThread *thread = checkPlaybackThread_l(output);
572    if (thread == NULL) {
573        ALOGW("channelCount() unknown thread %d", output);
574        return 0;
575    }
576    return thread->channelCount();
577}
578
579audio_format_t AudioFlinger::format(audio_io_handle_t output) const
580{
581    Mutex::Autolock _l(mLock);
582    PlaybackThread *thread = checkPlaybackThread_l(output);
583    if (thread == NULL) {
584        ALOGW("format() unknown thread %d", output);
585        return AUDIO_FORMAT_INVALID;
586    }
587    return thread->format();
588}
589
590size_t AudioFlinger::frameCount(audio_io_handle_t output) const
591{
592    Mutex::Autolock _l(mLock);
593    PlaybackThread *thread = checkPlaybackThread_l(output);
594    if (thread == NULL) {
595        ALOGW("frameCount() unknown thread %d", output);
596        return 0;
597    }
598    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
599    //       should examine all callers and fix them to handle smaller counts
600    return thread->frameCount();
601}
602
603uint32_t AudioFlinger::latency(audio_io_handle_t output) const
604{
605    Mutex::Autolock _l(mLock);
606    PlaybackThread *thread = checkPlaybackThread_l(output);
607    if (thread == NULL) {
608        ALOGW("latency(): no playback thread found for output handle %d", output);
609        return 0;
610    }
611    return thread->latency();
612}
613
614status_t AudioFlinger::setMasterVolume(float value)
615{
616    status_t ret = initCheck();
617    if (ret != NO_ERROR) {
618        return ret;
619    }
620
621    // check calling permissions
622    if (!settingsAllowed()) {
623        return PERMISSION_DENIED;
624    }
625
626    Mutex::Autolock _l(mLock);
627    mMasterVolume = value;
628
629    // Set master volume in the HALs which support it.
630    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
631        AutoMutex lock(mHardwareLock);
632        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
633
634        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
635        if (dev->canSetMasterVolume()) {
636            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
637        }
638        mHardwareStatus = AUDIO_HW_IDLE;
639    }
640
641    // Now set the master volume in each playback thread.  Playback threads
642    // assigned to HALs which do not have master volume support will apply
643    // master volume during the mix operation.  Threads with HALs which do
644    // support master volume will simply ignore the setting.
645    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
646        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
647
648    return NO_ERROR;
649}
650
651status_t AudioFlinger::setMode(audio_mode_t mode)
652{
653    status_t ret = initCheck();
654    if (ret != NO_ERROR) {
655        return ret;
656    }
657
658    // check calling permissions
659    if (!settingsAllowed()) {
660        return PERMISSION_DENIED;
661    }
662    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
663        ALOGW("Illegal value: setMode(%d)", mode);
664        return BAD_VALUE;
665    }
666
667    { // scope for the lock
668        AutoMutex lock(mHardwareLock);
669        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
670        mHardwareStatus = AUDIO_HW_SET_MODE;
671        ret = dev->set_mode(dev, mode);
672        mHardwareStatus = AUDIO_HW_IDLE;
673    }
674
675    if (NO_ERROR == ret) {
676        Mutex::Autolock _l(mLock);
677        mMode = mode;
678        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
679            mPlaybackThreads.valueAt(i)->setMode(mode);
680    }
681
682    return ret;
683}
684
685status_t AudioFlinger::setMicMute(bool state)
686{
687    status_t ret = initCheck();
688    if (ret != NO_ERROR) {
689        return ret;
690    }
691
692    // check calling permissions
693    if (!settingsAllowed()) {
694        return PERMISSION_DENIED;
695    }
696
697    AutoMutex lock(mHardwareLock);
698    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
699    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
700    ret = dev->set_mic_mute(dev, state);
701    mHardwareStatus = AUDIO_HW_IDLE;
702    return ret;
703}
704
705bool AudioFlinger::getMicMute() const
706{
707    status_t ret = initCheck();
708    if (ret != NO_ERROR) {
709        return false;
710    }
711
712    bool state = AUDIO_MODE_INVALID;
713    AutoMutex lock(mHardwareLock);
714    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
715    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
716    dev->get_mic_mute(dev, &state);
717    mHardwareStatus = AUDIO_HW_IDLE;
718    return state;
719}
720
721status_t AudioFlinger::setMasterMute(bool muted)
722{
723    status_t ret = initCheck();
724    if (ret != NO_ERROR) {
725        return ret;
726    }
727
728    // check calling permissions
729    if (!settingsAllowed()) {
730        return PERMISSION_DENIED;
731    }
732
733    Mutex::Autolock _l(mLock);
734    mMasterMute = muted;
735
736    // Set master mute in the HALs which support it.
737    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
738        AutoMutex lock(mHardwareLock);
739        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
740
741        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
742        if (dev->canSetMasterMute()) {
743            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
744        }
745        mHardwareStatus = AUDIO_HW_IDLE;
746    }
747
748    // Now set the master mute in each playback thread.  Playback threads
749    // assigned to HALs which do not have master mute support will apply master
750    // mute during the mix operation.  Threads with HALs which do support master
751    // mute will simply ignore the setting.
752    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
753        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
754
755    return NO_ERROR;
756}
757
758float AudioFlinger::masterVolume() const
759{
760    Mutex::Autolock _l(mLock);
761    return masterVolume_l();
762}
763
764bool AudioFlinger::masterMute() const
765{
766    Mutex::Autolock _l(mLock);
767    return masterMute_l();
768}
769
770float AudioFlinger::masterVolume_l() const
771{
772    return mMasterVolume;
773}
774
775bool AudioFlinger::masterMute_l() const
776{
777    return mMasterMute;
778}
779
780status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
781        audio_io_handle_t output)
782{
783    // check calling permissions
784    if (!settingsAllowed()) {
785        return PERMISSION_DENIED;
786    }
787
788    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
789        ALOGE("setStreamVolume() invalid stream %d", stream);
790        return BAD_VALUE;
791    }
792
793    AutoMutex lock(mLock);
794    PlaybackThread *thread = NULL;
795    if (output) {
796        thread = checkPlaybackThread_l(output);
797        if (thread == NULL) {
798            return BAD_VALUE;
799        }
800    }
801
802    mStreamTypes[stream].volume = value;
803
804    if (thread == NULL) {
805        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
806            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
807        }
808    } else {
809        thread->setStreamVolume(stream, value);
810    }
811
812    return NO_ERROR;
813}
814
815status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
816{
817    // check calling permissions
818    if (!settingsAllowed()) {
819        return PERMISSION_DENIED;
820    }
821
822    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
823        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
824        ALOGE("setStreamMute() invalid stream %d", stream);
825        return BAD_VALUE;
826    }
827
828    AutoMutex lock(mLock);
829    mStreamTypes[stream].mute = muted;
830    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
831        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
832
833    return NO_ERROR;
834}
835
836float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
837{
838    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
839        return 0.0f;
840    }
841
842    AutoMutex lock(mLock);
843    float volume;
844    if (output) {
845        PlaybackThread *thread = checkPlaybackThread_l(output);
846        if (thread == NULL) {
847            return 0.0f;
848        }
849        volume = thread->streamVolume(stream);
850    } else {
851        volume = streamVolume_l(stream);
852    }
853
854    return volume;
855}
856
857bool AudioFlinger::streamMute(audio_stream_type_t stream) const
858{
859    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
860        return true;
861    }
862
863    AutoMutex lock(mLock);
864    return streamMute_l(stream);
865}
866
867status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
868{
869    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
870            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
871
872    // check calling permissions
873    if (!settingsAllowed()) {
874        return PERMISSION_DENIED;
875    }
876
877    // ioHandle == 0 means the parameters are global to the audio hardware interface
878    if (ioHandle == 0) {
879        Mutex::Autolock _l(mLock);
880        status_t final_result = NO_ERROR;
881        {
882            AutoMutex lock(mHardwareLock);
883            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
884            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
885                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
886                status_t result = dev->set_parameters(dev, keyValuePairs.string());
887                final_result = result ?: final_result;
888            }
889            mHardwareStatus = AUDIO_HW_IDLE;
890        }
891        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
892        AudioParameter param = AudioParameter(keyValuePairs);
893        String8 value;
894        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
895            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
896            if (mBtNrecIsOff != btNrecIsOff) {
897                for (size_t i = 0; i < mRecordThreads.size(); i++) {
898                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
899                    audio_devices_t device = thread->inDevice();
900                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
901                    // collect all of the thread's session IDs
902                    KeyedVector<int, bool> ids = thread->sessionIds();
903                    // suspend effects associated with those session IDs
904                    for (size_t j = 0; j < ids.size(); ++j) {
905                        int sessionId = ids.keyAt(j);
906                        thread->setEffectSuspended(FX_IID_AEC,
907                                                   suspend,
908                                                   sessionId);
909                        thread->setEffectSuspended(FX_IID_NS,
910                                                   suspend,
911                                                   sessionId);
912                    }
913                }
914                mBtNrecIsOff = btNrecIsOff;
915            }
916        }
917        String8 screenState;
918        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
919            bool isOff = screenState == "off";
920            if (isOff != (AudioFlinger::mScreenState & 1)) {
921                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
922            }
923        }
924        return final_result;
925    }
926
927    // hold a strong ref on thread in case closeOutput() or closeInput() is called
928    // and the thread is exited once the lock is released
929    sp<ThreadBase> thread;
930    {
931        Mutex::Autolock _l(mLock);
932        thread = checkPlaybackThread_l(ioHandle);
933        if (thread == 0) {
934            thread = checkRecordThread_l(ioHandle);
935        } else if (thread == primaryPlaybackThread_l()) {
936            // indicate output device change to all input threads for pre processing
937            AudioParameter param = AudioParameter(keyValuePairs);
938            int value;
939            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
940                    (value != 0)) {
941                for (size_t i = 0; i < mRecordThreads.size(); i++) {
942                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
943                }
944            }
945        }
946    }
947    if (thread != 0) {
948        return thread->setParameters(keyValuePairs);
949    }
950    return BAD_VALUE;
951}
952
953String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
954{
955    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
956            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
957
958    Mutex::Autolock _l(mLock);
959
960    if (ioHandle == 0) {
961        String8 out_s8;
962
963        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
964            char *s;
965            {
966            AutoMutex lock(mHardwareLock);
967            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
968            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
969            s = dev->get_parameters(dev, keys.string());
970            mHardwareStatus = AUDIO_HW_IDLE;
971            }
972            out_s8 += String8(s ? s : "");
973            free(s);
974        }
975        return out_s8;
976    }
977
978    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
979    if (playbackThread != NULL) {
980        return playbackThread->getParameters(keys);
981    }
982    RecordThread *recordThread = checkRecordThread_l(ioHandle);
983    if (recordThread != NULL) {
984        return recordThread->getParameters(keys);
985    }
986    return String8("");
987}
988
989size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
990        audio_channel_mask_t channelMask) const
991{
992    status_t ret = initCheck();
993    if (ret != NO_ERROR) {
994        return 0;
995    }
996
997    AutoMutex lock(mHardwareLock);
998    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
999    struct audio_config config;
1000    memset(&config, 0, sizeof(config));
1001    config.sample_rate = sampleRate;
1002    config.channel_mask = channelMask;
1003    config.format = format;
1004
1005    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1006    size_t size = dev->get_input_buffer_size(dev, &config);
1007    mHardwareStatus = AUDIO_HW_IDLE;
1008    return size;
1009}
1010
1011unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1012{
1013    Mutex::Autolock _l(mLock);
1014
1015    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1016    if (recordThread != NULL) {
1017        return recordThread->getInputFramesLost();
1018    }
1019    return 0;
1020}
1021
1022status_t AudioFlinger::setVoiceVolume(float value)
1023{
1024    status_t ret = initCheck();
1025    if (ret != NO_ERROR) {
1026        return ret;
1027    }
1028
1029    // check calling permissions
1030    if (!settingsAllowed()) {
1031        return PERMISSION_DENIED;
1032    }
1033
1034    AutoMutex lock(mHardwareLock);
1035    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1036    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1037    ret = dev->set_voice_volume(dev, value);
1038    mHardwareStatus = AUDIO_HW_IDLE;
1039
1040    return ret;
1041}
1042
1043status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1044        audio_io_handle_t output) const
1045{
1046    status_t status;
1047
1048    Mutex::Autolock _l(mLock);
1049
1050    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1051    if (playbackThread != NULL) {
1052        return playbackThread->getRenderPosition(halFrames, dspFrames);
1053    }
1054
1055    return BAD_VALUE;
1056}
1057
1058void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1059{
1060
1061    Mutex::Autolock _l(mLock);
1062
1063    pid_t pid = IPCThreadState::self()->getCallingPid();
1064    if (mNotificationClients.indexOfKey(pid) < 0) {
1065        sp<NotificationClient> notificationClient = new NotificationClient(this,
1066                                                                            client,
1067                                                                            pid);
1068        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1069
1070        mNotificationClients.add(pid, notificationClient);
1071
1072        sp<IBinder> binder = client->asBinder();
1073        binder->linkToDeath(notificationClient);
1074
1075        // the config change is always sent from playback or record threads to avoid deadlock
1076        // with AudioSystem::gLock
1077        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1078            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1079        }
1080
1081        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1082            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1083        }
1084    }
1085}
1086
1087void AudioFlinger::removeNotificationClient(pid_t pid)
1088{
1089    Mutex::Autolock _l(mLock);
1090
1091    mNotificationClients.removeItem(pid);
1092
1093    ALOGV("%d died, releasing its sessions", pid);
1094    size_t num = mAudioSessionRefs.size();
1095    bool removed = false;
1096    for (size_t i = 0; i< num; ) {
1097        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1098        ALOGV(" pid %d @ %d", ref->mPid, i);
1099        if (ref->mPid == pid) {
1100            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1101            mAudioSessionRefs.removeAt(i);
1102            delete ref;
1103            removed = true;
1104            num--;
1105        } else {
1106            i++;
1107        }
1108    }
1109    if (removed) {
1110        purgeStaleEffects_l();
1111    }
1112}
1113
1114// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1115void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1116{
1117    size_t size = mNotificationClients.size();
1118    for (size_t i = 0; i < size; i++) {
1119        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1120                                                                               param2);
1121    }
1122}
1123
1124// removeClient_l() must be called with AudioFlinger::mLock held
1125void AudioFlinger::removeClient_l(pid_t pid)
1126{
1127    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1128            IPCThreadState::self()->getCallingPid());
1129    mClients.removeItem(pid);
1130}
1131
1132// getEffectThread_l() must be called with AudioFlinger::mLock held
1133sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1134{
1135    sp<PlaybackThread> thread;
1136
1137    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1138        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1139            ALOG_ASSERT(thread == 0);
1140            thread = mPlaybackThreads.valueAt(i);
1141        }
1142    }
1143
1144    return thread;
1145}
1146
1147
1148
1149// ----------------------------------------------------------------------------
1150
1151AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1152    :   RefBase(),
1153        mAudioFlinger(audioFlinger),
1154        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1155        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1156        mPid(pid),
1157        mTimedTrackCount(0)
1158{
1159    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1160}
1161
1162// Client destructor must be called with AudioFlinger::mLock held
1163AudioFlinger::Client::~Client()
1164{
1165    mAudioFlinger->removeClient_l(mPid);
1166}
1167
1168sp<MemoryDealer> AudioFlinger::Client::heap() const
1169{
1170    return mMemoryDealer;
1171}
1172
1173// Reserve one of the limited slots for a timed audio track associated
1174// with this client
1175bool AudioFlinger::Client::reserveTimedTrack()
1176{
1177    const int kMaxTimedTracksPerClient = 4;
1178
1179    Mutex::Autolock _l(mTimedTrackLock);
1180
1181    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1182        ALOGW("can not create timed track - pid %d has exceeded the limit",
1183             mPid);
1184        return false;
1185    }
1186
1187    mTimedTrackCount++;
1188    return true;
1189}
1190
1191// Release a slot for a timed audio track
1192void AudioFlinger::Client::releaseTimedTrack()
1193{
1194    Mutex::Autolock _l(mTimedTrackLock);
1195    mTimedTrackCount--;
1196}
1197
1198// ----------------------------------------------------------------------------
1199
1200AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1201                                                     const sp<IAudioFlingerClient>& client,
1202                                                     pid_t pid)
1203    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1204{
1205}
1206
1207AudioFlinger::NotificationClient::~NotificationClient()
1208{
1209}
1210
1211void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1212{
1213    sp<NotificationClient> keep(this);
1214    mAudioFlinger->removeNotificationClient(mPid);
1215}
1216
1217
1218// ----------------------------------------------------------------------------
1219
1220static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1221    return audio_is_remote_submix_device(inDevice);
1222}
1223
1224sp<IAudioRecord> AudioFlinger::openRecord(
1225        audio_io_handle_t input,
1226        uint32_t sampleRate,
1227        audio_format_t format,
1228        audio_channel_mask_t channelMask,
1229        size_t frameCount,
1230        IAudioFlinger::track_flags_t *flags,
1231        pid_t tid,
1232        int *sessionId,
1233        status_t *status)
1234{
1235    sp<RecordThread::RecordTrack> recordTrack;
1236    sp<RecordHandle> recordHandle;
1237    sp<Client> client;
1238    status_t lStatus;
1239    RecordThread *thread;
1240    size_t inFrameCount;
1241    int lSessionId;
1242
1243    // check calling permissions
1244    if (!recordingAllowed()) {
1245        ALOGE("openRecord() permission denied: recording not allowed");
1246        lStatus = PERMISSION_DENIED;
1247        goto Exit;
1248    }
1249
1250    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1251        ALOGE("openRecord() invalid format %d", format);
1252        lStatus = BAD_VALUE;
1253        goto Exit;
1254    }
1255
1256    // add client to list
1257    { // scope for mLock
1258        Mutex::Autolock _l(mLock);
1259        thread = checkRecordThread_l(input);
1260        if (thread == NULL) {
1261            ALOGE("openRecord() checkRecordThread_l failed");
1262            lStatus = BAD_VALUE;
1263            goto Exit;
1264        }
1265
1266        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1267                && !captureAudioOutputAllowed()) {
1268            ALOGE("openRecord() permission denied: capture not allowed");
1269            lStatus = PERMISSION_DENIED;
1270            goto Exit;
1271        }
1272
1273        pid_t pid = IPCThreadState::self()->getCallingPid();
1274        client = registerPid_l(pid);
1275
1276        // If no audio session id is provided, create one here
1277        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1278            lSessionId = *sessionId;
1279        } else {
1280            lSessionId = nextUniqueId();
1281            if (sessionId != NULL) {
1282                *sessionId = lSessionId;
1283            }
1284        }
1285        // create new record track.
1286        // The record track uses one track in mHardwareMixerThread by convention.
1287        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1288                                                  frameCount, lSessionId, flags, tid, &lStatus);
1289        LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
1290    }
1291    if (lStatus != NO_ERROR) {
1292        // remove local strong reference to Client before deleting the RecordTrack so that the
1293        // Client destructor is called by the TrackBase destructor with mLock held
1294        client.clear();
1295        recordTrack.clear();
1296        goto Exit;
1297    }
1298
1299    // return to handle to client
1300    recordHandle = new RecordHandle(recordTrack);
1301    lStatus = NO_ERROR;
1302
1303Exit:
1304    if (status) {
1305        *status = lStatus;
1306    }
1307    return recordHandle;
1308}
1309
1310
1311
1312// ----------------------------------------------------------------------------
1313
1314audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1315{
1316    if (!settingsAllowed()) {
1317        return 0;
1318    }
1319    Mutex::Autolock _l(mLock);
1320    return loadHwModule_l(name);
1321}
1322
1323// loadHwModule_l() must be called with AudioFlinger::mLock held
1324audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1325{
1326    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1327        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1328            ALOGW("loadHwModule() module %s already loaded", name);
1329            return mAudioHwDevs.keyAt(i);
1330        }
1331    }
1332
1333    audio_hw_device_t *dev;
1334
1335    int rc = load_audio_interface(name, &dev);
1336    if (rc) {
1337        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1338        return 0;
1339    }
1340
1341    mHardwareStatus = AUDIO_HW_INIT;
1342    rc = dev->init_check(dev);
1343    mHardwareStatus = AUDIO_HW_IDLE;
1344    if (rc) {
1345        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1346        return 0;
1347    }
1348
1349    // Check and cache this HAL's level of support for master mute and master
1350    // volume.  If this is the first HAL opened, and it supports the get
1351    // methods, use the initial values provided by the HAL as the current
1352    // master mute and volume settings.
1353
1354    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1355    {  // scope for auto-lock pattern
1356        AutoMutex lock(mHardwareLock);
1357
1358        if (0 == mAudioHwDevs.size()) {
1359            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1360            if (NULL != dev->get_master_volume) {
1361                float mv;
1362                if (OK == dev->get_master_volume(dev, &mv)) {
1363                    mMasterVolume = mv;
1364                }
1365            }
1366
1367            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1368            if (NULL != dev->get_master_mute) {
1369                bool mm;
1370                if (OK == dev->get_master_mute(dev, &mm)) {
1371                    mMasterMute = mm;
1372                }
1373            }
1374        }
1375
1376        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1377        if ((NULL != dev->set_master_volume) &&
1378            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1379            flags = static_cast<AudioHwDevice::Flags>(flags |
1380                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1381        }
1382
1383        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1384        if ((NULL != dev->set_master_mute) &&
1385            (OK == dev->set_master_mute(dev, mMasterMute))) {
1386            flags = static_cast<AudioHwDevice::Flags>(flags |
1387                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1388        }
1389
1390        mHardwareStatus = AUDIO_HW_IDLE;
1391    }
1392
1393    audio_module_handle_t handle = nextUniqueId();
1394    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1395
1396    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1397          name, dev->common.module->name, dev->common.module->id, handle);
1398
1399    return handle;
1400
1401}
1402
1403// ----------------------------------------------------------------------------
1404
1405uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1406{
1407    Mutex::Autolock _l(mLock);
1408    PlaybackThread *thread = primaryPlaybackThread_l();
1409    return thread != NULL ? thread->sampleRate() : 0;
1410}
1411
1412size_t AudioFlinger::getPrimaryOutputFrameCount()
1413{
1414    Mutex::Autolock _l(mLock);
1415    PlaybackThread *thread = primaryPlaybackThread_l();
1416    return thread != NULL ? thread->frameCountHAL() : 0;
1417}
1418
1419// ----------------------------------------------------------------------------
1420
1421status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1422{
1423    uid_t uid = IPCThreadState::self()->getCallingUid();
1424    if (uid != AID_SYSTEM) {
1425        return PERMISSION_DENIED;
1426    }
1427    Mutex::Autolock _l(mLock);
1428    if (mIsDeviceTypeKnown) {
1429        return INVALID_OPERATION;
1430    }
1431    mIsLowRamDevice = isLowRamDevice;
1432    mIsDeviceTypeKnown = true;
1433    return NO_ERROR;
1434}
1435
1436// ----------------------------------------------------------------------------
1437
1438audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1439                                           audio_devices_t *pDevices,
1440                                           uint32_t *pSamplingRate,
1441                                           audio_format_t *pFormat,
1442                                           audio_channel_mask_t *pChannelMask,
1443                                           uint32_t *pLatencyMs,
1444                                           audio_output_flags_t flags,
1445                                           const audio_offload_info_t *offloadInfo)
1446{
1447    PlaybackThread *thread = NULL;
1448    struct audio_config config;
1449    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1450    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1451    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1452    if (offloadInfo) {
1453        config.offload_info = *offloadInfo;
1454    }
1455
1456    audio_stream_out_t *outStream = NULL;
1457    AudioHwDevice *outHwDev;
1458
1459    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1460              module,
1461              (pDevices != NULL) ? *pDevices : 0,
1462              config.sample_rate,
1463              config.format,
1464              config.channel_mask,
1465              flags);
1466    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1467          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version );
1468
1469    if (pDevices == NULL || *pDevices == 0) {
1470        return 0;
1471    }
1472
1473    Mutex::Autolock _l(mLock);
1474
1475    outHwDev = findSuitableHwDev_l(module, *pDevices);
1476    if (outHwDev == NULL)
1477        return 0;
1478
1479    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1480    audio_io_handle_t id = nextUniqueId();
1481
1482    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1483
1484    status_t status = hwDevHal->open_output_stream(hwDevHal,
1485                                          id,
1486                                          *pDevices,
1487                                          (audio_output_flags_t)flags,
1488                                          &config,
1489                                          &outStream);
1490
1491    mHardwareStatus = AUDIO_HW_IDLE;
1492    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1493            "Channels %x, status %d",
1494            outStream,
1495            config.sample_rate,
1496            config.format,
1497            config.channel_mask,
1498            status);
1499
1500    if (status == NO_ERROR && outStream != NULL) {
1501        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1502
1503        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1504            thread = new OffloadThread(this, output, id, *pDevices);
1505            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1506        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1507            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1508            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1509            thread = new DirectOutputThread(this, output, id, *pDevices);
1510            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1511        } else {
1512            thread = new MixerThread(this, output, id, *pDevices);
1513            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1514        }
1515        mPlaybackThreads.add(id, thread);
1516
1517        if (pSamplingRate != NULL) {
1518            *pSamplingRate = config.sample_rate;
1519        }
1520        if (pFormat != NULL) {
1521            *pFormat = config.format;
1522        }
1523        if (pChannelMask != NULL) {
1524            *pChannelMask = config.channel_mask;
1525        }
1526        if (pLatencyMs != NULL) {
1527            *pLatencyMs = thread->latency();
1528        }
1529
1530        // notify client processes of the new output creation
1531        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1532
1533        // the first primary output opened designates the primary hw device
1534        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1535            ALOGI("Using module %d has the primary audio interface", module);
1536            mPrimaryHardwareDev = outHwDev;
1537
1538            AutoMutex lock(mHardwareLock);
1539            mHardwareStatus = AUDIO_HW_SET_MODE;
1540            hwDevHal->set_mode(hwDevHal, mMode);
1541            mHardwareStatus = AUDIO_HW_IDLE;
1542        }
1543        return id;
1544    }
1545
1546    return 0;
1547}
1548
1549audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1550        audio_io_handle_t output2)
1551{
1552    Mutex::Autolock _l(mLock);
1553    MixerThread *thread1 = checkMixerThread_l(output1);
1554    MixerThread *thread2 = checkMixerThread_l(output2);
1555
1556    if (thread1 == NULL || thread2 == NULL) {
1557        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1558                output2);
1559        return 0;
1560    }
1561
1562    audio_io_handle_t id = nextUniqueId();
1563    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1564    thread->addOutputTrack(thread2);
1565    mPlaybackThreads.add(id, thread);
1566    // notify client processes of the new output creation
1567    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1568    return id;
1569}
1570
1571status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1572{
1573    return closeOutput_nonvirtual(output);
1574}
1575
1576status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1577{
1578    // keep strong reference on the playback thread so that
1579    // it is not destroyed while exit() is executed
1580    sp<PlaybackThread> thread;
1581    {
1582        Mutex::Autolock _l(mLock);
1583        thread = checkPlaybackThread_l(output);
1584        if (thread == NULL) {
1585            return BAD_VALUE;
1586        }
1587
1588        ALOGV("closeOutput() %d", output);
1589
1590        if (thread->type() == ThreadBase::MIXER) {
1591            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1592                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1593                    DuplicatingThread *dupThread =
1594                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1595                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1596
1597                }
1598            }
1599        }
1600
1601
1602        mPlaybackThreads.removeItem(output);
1603        // save all effects to the default thread
1604        if (mPlaybackThreads.size()) {
1605            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1606            if (dstThread != NULL) {
1607                // audioflinger lock is held here so the acquisition order of thread locks does not
1608                // matter
1609                Mutex::Autolock _dl(dstThread->mLock);
1610                Mutex::Autolock _sl(thread->mLock);
1611                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1612                for (size_t i = 0; i < effectChains.size(); i ++) {
1613                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1614                }
1615            }
1616        }
1617        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1618    }
1619    thread->exit();
1620    // The thread entity (active unit of execution) is no longer running here,
1621    // but the ThreadBase container still exists.
1622
1623    if (thread->type() != ThreadBase::DUPLICATING) {
1624        AudioStreamOut *out = thread->clearOutput();
1625        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1626        // from now on thread->mOutput is NULL
1627        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1628        delete out;
1629    }
1630    return NO_ERROR;
1631}
1632
1633status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1634{
1635    Mutex::Autolock _l(mLock);
1636    PlaybackThread *thread = checkPlaybackThread_l(output);
1637
1638    if (thread == NULL) {
1639        return BAD_VALUE;
1640    }
1641
1642    ALOGV("suspendOutput() %d", output);
1643    thread->suspend();
1644
1645    return NO_ERROR;
1646}
1647
1648status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1649{
1650    Mutex::Autolock _l(mLock);
1651    PlaybackThread *thread = checkPlaybackThread_l(output);
1652
1653    if (thread == NULL) {
1654        return BAD_VALUE;
1655    }
1656
1657    ALOGV("restoreOutput() %d", output);
1658
1659    thread->restore();
1660
1661    return NO_ERROR;
1662}
1663
1664audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1665                                          audio_devices_t *pDevices,
1666                                          uint32_t *pSamplingRate,
1667                                          audio_format_t *pFormat,
1668                                          audio_channel_mask_t *pChannelMask)
1669{
1670    status_t status;
1671    RecordThread *thread = NULL;
1672    struct audio_config config;
1673    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1674    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1675    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1676
1677    uint32_t reqSamplingRate = config.sample_rate;
1678    audio_format_t reqFormat = config.format;
1679    audio_channel_mask_t reqChannels = config.channel_mask;
1680    audio_stream_in_t *inStream = NULL;
1681    AudioHwDevice *inHwDev;
1682
1683    if (pDevices == NULL || *pDevices == 0) {
1684        return 0;
1685    }
1686
1687    Mutex::Autolock _l(mLock);
1688
1689    inHwDev = findSuitableHwDev_l(module, *pDevices);
1690    if (inHwDev == NULL)
1691        return 0;
1692
1693    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1694    audio_io_handle_t id = nextUniqueId();
1695
1696    status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1697                                        &inStream);
1698    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1699            "status %d",
1700            inStream,
1701            config.sample_rate,
1702            config.format,
1703            config.channel_mask,
1704            status);
1705
1706    // If the input could not be opened with the requested parameters and we can handle the
1707    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1708    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1709    if (status == BAD_VALUE &&
1710        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1711        (config.sample_rate <= 2 * reqSamplingRate) &&
1712        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
1713        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1714        inStream = NULL;
1715        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1716    }
1717
1718    if (status == NO_ERROR && inStream != NULL) {
1719
1720#ifdef TEE_SINK
1721        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1722        // or (re-)create if current Pipe is idle and does not match the new format
1723        sp<NBAIO_Sink> teeSink;
1724        enum {
1725            TEE_SINK_NO,    // don't copy input
1726            TEE_SINK_NEW,   // copy input using a new pipe
1727            TEE_SINK_OLD,   // copy input using an existing pipe
1728        } kind;
1729        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1730                                        popcount(inStream->common.get_channels(&inStream->common)));
1731        if (!mTeeSinkInputEnabled) {
1732            kind = TEE_SINK_NO;
1733        } else if (format == Format_Invalid) {
1734            kind = TEE_SINK_NO;
1735        } else if (mRecordTeeSink == 0) {
1736            kind = TEE_SINK_NEW;
1737        } else if (mRecordTeeSink->getStrongCount() != 1) {
1738            kind = TEE_SINK_NO;
1739        } else if (format == mRecordTeeSink->format()) {
1740            kind = TEE_SINK_OLD;
1741        } else {
1742            kind = TEE_SINK_NEW;
1743        }
1744        switch (kind) {
1745        case TEE_SINK_NEW: {
1746            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1747            size_t numCounterOffers = 0;
1748            const NBAIO_Format offers[1] = {format};
1749            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1750            ALOG_ASSERT(index == 0);
1751            PipeReader *pipeReader = new PipeReader(*pipe);
1752            numCounterOffers = 0;
1753            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1754            ALOG_ASSERT(index == 0);
1755            mRecordTeeSink = pipe;
1756            mRecordTeeSource = pipeReader;
1757            teeSink = pipe;
1758            }
1759            break;
1760        case TEE_SINK_OLD:
1761            teeSink = mRecordTeeSink;
1762            break;
1763        case TEE_SINK_NO:
1764        default:
1765            break;
1766        }
1767#endif
1768
1769        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1770
1771        // Start record thread
1772        // RecordThread requires both input and output device indication to forward to audio
1773        // pre processing modules
1774        thread = new RecordThread(this,
1775                                  input,
1776                                  reqSamplingRate,
1777                                  reqChannels,
1778                                  id,
1779                                  primaryOutputDevice_l(),
1780                                  *pDevices
1781#ifdef TEE_SINK
1782                                  , teeSink
1783#endif
1784                                  );
1785        mRecordThreads.add(id, thread);
1786        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1787        if (pSamplingRate != NULL) {
1788            *pSamplingRate = reqSamplingRate;
1789        }
1790        if (pFormat != NULL) {
1791            *pFormat = config.format;
1792        }
1793        if (pChannelMask != NULL) {
1794            *pChannelMask = reqChannels;
1795        }
1796
1797        // notify client processes of the new input creation
1798        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1799        return id;
1800    }
1801
1802    return 0;
1803}
1804
1805status_t AudioFlinger::closeInput(audio_io_handle_t input)
1806{
1807    return closeInput_nonvirtual(input);
1808}
1809
1810status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1811{
1812    // keep strong reference on the record thread so that
1813    // it is not destroyed while exit() is executed
1814    sp<RecordThread> thread;
1815    {
1816        Mutex::Autolock _l(mLock);
1817        thread = checkRecordThread_l(input);
1818        if (thread == 0) {
1819            return BAD_VALUE;
1820        }
1821
1822        ALOGV("closeInput() %d", input);
1823        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1824        mRecordThreads.removeItem(input);
1825    }
1826    thread->exit();
1827    // The thread entity (active unit of execution) is no longer running here,
1828    // but the ThreadBase container still exists.
1829
1830    AudioStreamIn *in = thread->clearInput();
1831    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1832    // from now on thread->mInput is NULL
1833    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1834    delete in;
1835
1836    return NO_ERROR;
1837}
1838
1839status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1840{
1841    Mutex::Autolock _l(mLock);
1842    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1843
1844    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1845        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1846        thread->invalidateTracks(stream);
1847    }
1848
1849    return NO_ERROR;
1850}
1851
1852
1853int AudioFlinger::newAudioSessionId()
1854{
1855    return nextUniqueId();
1856}
1857
1858void AudioFlinger::acquireAudioSessionId(int audioSession)
1859{
1860    Mutex::Autolock _l(mLock);
1861    pid_t caller = IPCThreadState::self()->getCallingPid();
1862    ALOGV("acquiring %d from %d", audioSession, caller);
1863
1864    // Ignore requests received from processes not known as notification client. The request
1865    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1866    // called from a different pid leaving a stale session reference.  Also we don't know how
1867    // to clear this reference if the client process dies.
1868    if (mNotificationClients.indexOfKey(caller) < 0) {
1869        ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1870        return;
1871    }
1872
1873    size_t num = mAudioSessionRefs.size();
1874    for (size_t i = 0; i< num; i++) {
1875        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1876        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1877            ref->mCnt++;
1878            ALOGV(" incremented refcount to %d", ref->mCnt);
1879            return;
1880        }
1881    }
1882    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1883    ALOGV(" added new entry for %d", audioSession);
1884}
1885
1886void AudioFlinger::releaseAudioSessionId(int audioSession)
1887{
1888    Mutex::Autolock _l(mLock);
1889    pid_t caller = IPCThreadState::self()->getCallingPid();
1890    ALOGV("releasing %d from %d", audioSession, caller);
1891    size_t num = mAudioSessionRefs.size();
1892    for (size_t i = 0; i< num; i++) {
1893        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1894        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1895            ref->mCnt--;
1896            ALOGV(" decremented refcount to %d", ref->mCnt);
1897            if (ref->mCnt == 0) {
1898                mAudioSessionRefs.removeAt(i);
1899                delete ref;
1900                purgeStaleEffects_l();
1901            }
1902            return;
1903        }
1904    }
1905    // If the caller is mediaserver it is likely that the session being released was acquired
1906    // on behalf of a process not in notification clients and we ignore the warning.
1907    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
1908}
1909
1910void AudioFlinger::purgeStaleEffects_l() {
1911
1912    ALOGV("purging stale effects");
1913
1914    Vector< sp<EffectChain> > chains;
1915
1916    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1917        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1918        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1919            sp<EffectChain> ec = t->mEffectChains[j];
1920            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1921                chains.push(ec);
1922            }
1923        }
1924    }
1925    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1926        sp<RecordThread> t = mRecordThreads.valueAt(i);
1927        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1928            sp<EffectChain> ec = t->mEffectChains[j];
1929            chains.push(ec);
1930        }
1931    }
1932
1933    for (size_t i = 0; i < chains.size(); i++) {
1934        sp<EffectChain> ec = chains[i];
1935        int sessionid = ec->sessionId();
1936        sp<ThreadBase> t = ec->mThread.promote();
1937        if (t == 0) {
1938            continue;
1939        }
1940        size_t numsessionrefs = mAudioSessionRefs.size();
1941        bool found = false;
1942        for (size_t k = 0; k < numsessionrefs; k++) {
1943            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1944            if (ref->mSessionid == sessionid) {
1945                ALOGV(" session %d still exists for %d with %d refs",
1946                    sessionid, ref->mPid, ref->mCnt);
1947                found = true;
1948                break;
1949            }
1950        }
1951        if (!found) {
1952            Mutex::Autolock _l (t->mLock);
1953            // remove all effects from the chain
1954            while (ec->mEffects.size()) {
1955                sp<EffectModule> effect = ec->mEffects[0];
1956                effect->unPin();
1957                t->removeEffect_l(effect);
1958                if (effect->purgeHandles()) {
1959                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
1960                }
1961                AudioSystem::unregisterEffect(effect->id());
1962            }
1963        }
1964    }
1965    return;
1966}
1967
1968// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
1969AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
1970{
1971    return mPlaybackThreads.valueFor(output).get();
1972}
1973
1974// checkMixerThread_l() must be called with AudioFlinger::mLock held
1975AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
1976{
1977    PlaybackThread *thread = checkPlaybackThread_l(output);
1978    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
1979}
1980
1981// checkRecordThread_l() must be called with AudioFlinger::mLock held
1982AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
1983{
1984    return mRecordThreads.valueFor(input).get();
1985}
1986
1987uint32_t AudioFlinger::nextUniqueId()
1988{
1989    return android_atomic_inc(&mNextUniqueId);
1990}
1991
1992AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
1993{
1994    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1995        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1996        AudioStreamOut *output = thread->getOutput();
1997        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
1998            return thread;
1999        }
2000    }
2001    return NULL;
2002}
2003
2004audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2005{
2006    PlaybackThread *thread = primaryPlaybackThread_l();
2007
2008    if (thread == NULL) {
2009        return 0;
2010    }
2011
2012    return thread->outDevice();
2013}
2014
2015sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2016                                    int triggerSession,
2017                                    int listenerSession,
2018                                    sync_event_callback_t callBack,
2019                                    void *cookie)
2020{
2021    Mutex::Autolock _l(mLock);
2022
2023    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2024    status_t playStatus = NAME_NOT_FOUND;
2025    status_t recStatus = NAME_NOT_FOUND;
2026    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2027        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2028        if (playStatus == NO_ERROR) {
2029            return event;
2030        }
2031    }
2032    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2033        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2034        if (recStatus == NO_ERROR) {
2035            return event;
2036        }
2037    }
2038    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2039        mPendingSyncEvents.add(event);
2040    } else {
2041        ALOGV("createSyncEvent() invalid event %d", event->type());
2042        event.clear();
2043    }
2044    return event;
2045}
2046
2047// ----------------------------------------------------------------------------
2048//  Effect management
2049// ----------------------------------------------------------------------------
2050
2051
2052status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2053{
2054    Mutex::Autolock _l(mLock);
2055    return EffectQueryNumberEffects(numEffects);
2056}
2057
2058status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2059{
2060    Mutex::Autolock _l(mLock);
2061    return EffectQueryEffect(index, descriptor);
2062}
2063
2064status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2065        effect_descriptor_t *descriptor) const
2066{
2067    Mutex::Autolock _l(mLock);
2068    return EffectGetDescriptor(pUuid, descriptor);
2069}
2070
2071
2072sp<IEffect> AudioFlinger::createEffect(
2073        effect_descriptor_t *pDesc,
2074        const sp<IEffectClient>& effectClient,
2075        int32_t priority,
2076        audio_io_handle_t io,
2077        int sessionId,
2078        status_t *status,
2079        int *id,
2080        int *enabled)
2081{
2082    status_t lStatus = NO_ERROR;
2083    sp<EffectHandle> handle;
2084    effect_descriptor_t desc;
2085
2086    pid_t pid = IPCThreadState::self()->getCallingPid();
2087    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2088            pid, effectClient.get(), priority, sessionId, io);
2089
2090    if (pDesc == NULL) {
2091        lStatus = BAD_VALUE;
2092        goto Exit;
2093    }
2094
2095    // check audio settings permission for global effects
2096    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2097        lStatus = PERMISSION_DENIED;
2098        goto Exit;
2099    }
2100
2101    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2102    // that can only be created by audio policy manager (running in same process)
2103    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2104        lStatus = PERMISSION_DENIED;
2105        goto Exit;
2106    }
2107
2108    {
2109        Mutex::Autolock _l(mLock);
2110
2111
2112        if (!EffectIsNullUuid(&pDesc->uuid)) {
2113            // if uuid is specified, request effect descriptor
2114            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2115            if (lStatus < 0) {
2116                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2117                goto Exit;
2118            }
2119        } else {
2120            // if uuid is not specified, look for an available implementation
2121            // of the required type in effect factory
2122            if (EffectIsNullUuid(&pDesc->type)) {
2123                ALOGW("createEffect() no effect type");
2124                lStatus = BAD_VALUE;
2125                goto Exit;
2126            }
2127            uint32_t numEffects = 0;
2128            effect_descriptor_t d;
2129            d.flags = 0; // prevent compiler warning
2130            bool found = false;
2131
2132            lStatus = EffectQueryNumberEffects(&numEffects);
2133            if (lStatus < 0) {
2134                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2135                goto Exit;
2136            }
2137            for (uint32_t i = 0; i < numEffects; i++) {
2138                lStatus = EffectQueryEffect(i, &desc);
2139                if (lStatus < 0) {
2140                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2141                    continue;
2142                }
2143                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2144                    // If matching type found save effect descriptor. If the session is
2145                    // 0 and the effect is not auxiliary, continue enumeration in case
2146                    // an auxiliary version of this effect type is available
2147                    found = true;
2148                    d = desc;
2149                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2150                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2151                        break;
2152                    }
2153                }
2154            }
2155            if (!found) {
2156                lStatus = BAD_VALUE;
2157                ALOGW("createEffect() effect not found");
2158                goto Exit;
2159            }
2160            // For same effect type, chose auxiliary version over insert version if
2161            // connect to output mix (Compliance to OpenSL ES)
2162            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2163                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2164                desc = d;
2165            }
2166        }
2167
2168        // Do not allow auxiliary effects on a session different from 0 (output mix)
2169        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2170             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2171            lStatus = INVALID_OPERATION;
2172            goto Exit;
2173        }
2174
2175        // check recording permission for visualizer
2176        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2177            !recordingAllowed()) {
2178            lStatus = PERMISSION_DENIED;
2179            goto Exit;
2180        }
2181
2182        // return effect descriptor
2183        *pDesc = desc;
2184
2185        // If output is not specified try to find a matching audio session ID in one of the
2186        // output threads.
2187        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2188        // because of code checking output when entering the function.
2189        // Note: io is never 0 when creating an effect on an input
2190        if (io == 0) {
2191            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2192                // output must be specified by AudioPolicyManager when using session
2193                // AUDIO_SESSION_OUTPUT_STAGE
2194                lStatus = BAD_VALUE;
2195                goto Exit;
2196            }
2197            if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2198                // if the output returned by getOutputForEffect() is removed before we lock the
2199                // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2200                // and we will exit safely
2201                io = AudioSystem::getOutputForEffect(&desc);
2202                ALOGV("createEffect got output %d", io);
2203            }
2204            if (io == 0) {
2205                // look for the thread where the specified audio session is present
2206                for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2207                    if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2208                        io = mPlaybackThreads.keyAt(i);
2209                        break;
2210                    }
2211                }
2212                if (io == 0) {
2213                    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2214                        if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2215                            io = mRecordThreads.keyAt(i);
2216                            break;
2217                        }
2218                    }
2219                }
2220            }
2221            // If no output thread contains the requested session ID, default to
2222            // first output. The effect chain will be moved to the correct output
2223            // thread when a track with the same session ID is created
2224            if (io == 0 && mPlaybackThreads.size()) {
2225                io = mPlaybackThreads.keyAt(0);
2226            }
2227            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2228        }
2229        ThreadBase *thread = checkRecordThread_l(io);
2230        if (thread == NULL) {
2231            thread = checkPlaybackThread_l(io);
2232            if (thread == NULL) {
2233                ALOGE("createEffect() unknown output thread");
2234                lStatus = BAD_VALUE;
2235                goto Exit;
2236            }
2237        }
2238
2239        sp<Client> client = registerPid_l(pid);
2240
2241        // create effect on selected output thread
2242        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2243                &desc, enabled, &lStatus);
2244        if (handle != 0 && id != NULL) {
2245            *id = handle->id();
2246        }
2247    }
2248
2249Exit:
2250    if (status != NULL) {
2251        *status = lStatus;
2252    }
2253    return handle;
2254}
2255
2256status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2257        audio_io_handle_t dstOutput)
2258{
2259    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2260            sessionId, srcOutput, dstOutput);
2261    Mutex::Autolock _l(mLock);
2262    if (srcOutput == dstOutput) {
2263        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2264        return NO_ERROR;
2265    }
2266    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2267    if (srcThread == NULL) {
2268        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2269        return BAD_VALUE;
2270    }
2271    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2272    if (dstThread == NULL) {
2273        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2274        return BAD_VALUE;
2275    }
2276
2277    Mutex::Autolock _dl(dstThread->mLock);
2278    Mutex::Autolock _sl(srcThread->mLock);
2279    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2280}
2281
2282// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2283status_t AudioFlinger::moveEffectChain_l(int sessionId,
2284                                   AudioFlinger::PlaybackThread *srcThread,
2285                                   AudioFlinger::PlaybackThread *dstThread,
2286                                   bool reRegister)
2287{
2288    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2289            sessionId, srcThread, dstThread);
2290
2291    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2292    if (chain == 0) {
2293        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2294                sessionId, srcThread);
2295        return INVALID_OPERATION;
2296    }
2297
2298    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2299    // so that a new chain is created with correct parameters when first effect is added. This is
2300    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2301    // removed.
2302    srcThread->removeEffectChain_l(chain);
2303
2304    // transfer all effects one by one so that new effect chain is created on new thread with
2305    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2306    sp<EffectChain> dstChain;
2307    uint32_t strategy = 0; // prevent compiler warning
2308    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2309    Vector< sp<EffectModule> > removed;
2310    status_t status = NO_ERROR;
2311    while (effect != 0) {
2312        srcThread->removeEffect_l(effect);
2313        removed.add(effect);
2314        status = dstThread->addEffect_l(effect);
2315        if (status != NO_ERROR) {
2316            break;
2317        }
2318        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2319        if (effect->state() == EffectModule::ACTIVE ||
2320                effect->state() == EffectModule::STOPPING) {
2321            effect->start();
2322        }
2323        // if the move request is not received from audio policy manager, the effect must be
2324        // re-registered with the new strategy and output
2325        if (dstChain == 0) {
2326            dstChain = effect->chain().promote();
2327            if (dstChain == 0) {
2328                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2329                status = NO_INIT;
2330                break;
2331            }
2332            strategy = dstChain->strategy();
2333        }
2334        if (reRegister) {
2335            AudioSystem::unregisterEffect(effect->id());
2336            AudioSystem::registerEffect(&effect->desc(),
2337                                        dstThread->id(),
2338                                        strategy,
2339                                        sessionId,
2340                                        effect->id());
2341        }
2342        effect = chain->getEffectFromId_l(0);
2343    }
2344
2345    if (status != NO_ERROR) {
2346        for (size_t i = 0; i < removed.size(); i++) {
2347            srcThread->addEffect_l(removed[i]);
2348            if (dstChain != 0 && reRegister) {
2349                AudioSystem::unregisterEffect(removed[i]->id());
2350                AudioSystem::registerEffect(&removed[i]->desc(),
2351                                            srcThread->id(),
2352                                            strategy,
2353                                            sessionId,
2354                                            removed[i]->id());
2355            }
2356        }
2357    }
2358
2359    return status;
2360}
2361
2362bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2363{
2364    if (mGlobalEffectEnableTime != 0 &&
2365            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2366        return true;
2367    }
2368
2369    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2370        sp<EffectChain> ec =
2371                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2372        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2373            return true;
2374        }
2375    }
2376    return false;
2377}
2378
2379void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2380{
2381    Mutex::Autolock _l(mLock);
2382
2383    mGlobalEffectEnableTime = systemTime();
2384
2385    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2386        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2387        if (t->mType == ThreadBase::OFFLOAD) {
2388            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2389        }
2390    }
2391
2392}
2393
2394struct Entry {
2395#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2396    char mName[MAX_NAME];
2397};
2398
2399int comparEntry(const void *p1, const void *p2)
2400{
2401    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2402}
2403
2404#ifdef TEE_SINK
2405void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2406{
2407    NBAIO_Source *teeSource = source.get();
2408    if (teeSource != NULL) {
2409        // .wav rotation
2410        // There is a benign race condition if 2 threads call this simultaneously.
2411        // They would both traverse the directory, but the result would simply be
2412        // failures at unlink() which are ignored.  It's also unlikely since
2413        // normally dumpsys is only done by bugreport or from the command line.
2414        char teePath[32+256];
2415        strcpy(teePath, "/data/misc/media");
2416        size_t teePathLen = strlen(teePath);
2417        DIR *dir = opendir(teePath);
2418        teePath[teePathLen++] = '/';
2419        if (dir != NULL) {
2420#define MAX_SORT 20 // number of entries to sort
2421#define MAX_KEEP 10 // number of entries to keep
2422            struct Entry entries[MAX_SORT];
2423            size_t entryCount = 0;
2424            while (entryCount < MAX_SORT) {
2425                struct dirent de;
2426                struct dirent *result = NULL;
2427                int rc = readdir_r(dir, &de, &result);
2428                if (rc != 0) {
2429                    ALOGW("readdir_r failed %d", rc);
2430                    break;
2431                }
2432                if (result == NULL) {
2433                    break;
2434                }
2435                if (result != &de) {
2436                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2437                    break;
2438                }
2439                // ignore non .wav file entries
2440                size_t nameLen = strlen(de.d_name);
2441                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2442                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2443                    continue;
2444                }
2445                strcpy(entries[entryCount++].mName, de.d_name);
2446            }
2447            (void) closedir(dir);
2448            if (entryCount > MAX_KEEP) {
2449                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2450                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2451                    strcpy(&teePath[teePathLen], entries[i].mName);
2452                    (void) unlink(teePath);
2453                }
2454            }
2455        } else {
2456            if (fd >= 0) {
2457                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2458            }
2459        }
2460        char teeTime[16];
2461        struct timeval tv;
2462        gettimeofday(&tv, NULL);
2463        struct tm tm;
2464        localtime_r(&tv.tv_sec, &tm);
2465        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2466        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2467        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2468        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2469        if (teeFd >= 0) {
2470            char wavHeader[44];
2471            memcpy(wavHeader,
2472                "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",
2473                sizeof(wavHeader));
2474            NBAIO_Format format = teeSource->format();
2475            unsigned channelCount = Format_channelCount(format);
2476            ALOG_ASSERT(channelCount <= FCC_2);
2477            uint32_t sampleRate = Format_sampleRate(format);
2478            wavHeader[22] = channelCount;       // number of channels
2479            wavHeader[24] = sampleRate;         // sample rate
2480            wavHeader[25] = sampleRate >> 8;
2481            wavHeader[32] = channelCount * 2;   // block alignment
2482            write(teeFd, wavHeader, sizeof(wavHeader));
2483            size_t total = 0;
2484            bool firstRead = true;
2485            for (;;) {
2486#define TEE_SINK_READ 1024
2487                short buffer[TEE_SINK_READ * FCC_2];
2488                size_t count = TEE_SINK_READ;
2489                ssize_t actual = teeSource->read(buffer, count,
2490                        AudioBufferProvider::kInvalidPTS);
2491                bool wasFirstRead = firstRead;
2492                firstRead = false;
2493                if (actual <= 0) {
2494                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2495                        continue;
2496                    }
2497                    break;
2498                }
2499                ALOG_ASSERT(actual <= (ssize_t)count);
2500                write(teeFd, buffer, actual * channelCount * sizeof(short));
2501                total += actual;
2502            }
2503            lseek(teeFd, (off_t) 4, SEEK_SET);
2504            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2505            write(teeFd, &temp, sizeof(temp));
2506            lseek(teeFd, (off_t) 40, SEEK_SET);
2507            temp =  total * channelCount * sizeof(short);
2508            write(teeFd, &temp, sizeof(temp));
2509            close(teeFd);
2510            if (fd >= 0) {
2511                fdprintf(fd, "tee copied to %s\n", teePath);
2512            }
2513        } else {
2514            if (fd >= 0) {
2515                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2516            }
2517        }
2518    }
2519}
2520#endif
2521
2522// ----------------------------------------------------------------------------
2523
2524status_t AudioFlinger::onTransact(
2525        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2526{
2527    return BnAudioFlinger::onTransact(code, data, reply, flags);
2528}
2529
2530}; // namespace android
2531