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