AudioFlinger.cpp revision 663c2247b71086e30bfd3192979d1dd7f15c539e 
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    }
168    if (teeEnabled & 2) {
169        mTeeSinkOutputEnabled = true;
170    }
171    if (teeEnabled & 4) {
172        mTeeSinkTrackEnabled = true;
173    }
174#endif
175}
176
177void AudioFlinger::onFirstRef()
178{
179    int rc = 0;
180
181    Mutex::Autolock _l(mLock);
182
183    /* TODO: move all this work into an Init() function */
184    char val_str[PROPERTY_VALUE_MAX] = { 0 };
185    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
186        uint32_t int_val;
187        if (1 == sscanf(val_str, "%u", &int_val)) {
188            mStandbyTimeInNsecs = milliseconds(int_val);
189            ALOGI("Using %u mSec as standby time.", int_val);
190        } else {
191            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
192            ALOGI("Using default %u mSec as standby time.",
193                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
194        }
195    }
196
197    mMode = AUDIO_MODE_NORMAL;
198}
199
200AudioFlinger::~AudioFlinger()
201{
202    while (!mRecordThreads.isEmpty()) {
203        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
204        closeInput_nonvirtual(mRecordThreads.keyAt(0));
205    }
206    while (!mPlaybackThreads.isEmpty()) {
207        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
208        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
209    }
210
211    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
212        // no mHardwareLock needed, as there are no other references to this
213        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
214        delete mAudioHwDevs.valueAt(i);
215    }
216}
217
218static const char * const audio_interfaces[] = {
219    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
220    AUDIO_HARDWARE_MODULE_ID_A2DP,
221    AUDIO_HARDWARE_MODULE_ID_USB,
222};
223#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
224
225AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
226        audio_module_handle_t module,
227        audio_devices_t devices)
228{
229    // if module is 0, the request comes from an old policy manager and we should load
230    // well known modules
231    if (module == 0) {
232        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
233        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
234            loadHwModule_l(audio_interfaces[i]);
235        }
236        // then try to find a module supporting the requested device.
237        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
238            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
239            audio_hw_device_t *dev = audioHwDevice->hwDevice();
240            if ((dev->get_supported_devices != NULL) &&
241                    (dev->get_supported_devices(dev) & devices) == devices)
242                return audioHwDevice;
243        }
244    } else {
245        // check a match for the requested module handle
246        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
247        if (audioHwDevice != NULL) {
248            return audioHwDevice;
249        }
250    }
251
252    return NULL;
253}
254
255void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
256{
257    const size_t SIZE = 256;
258    char buffer[SIZE];
259    String8 result;
260
261    result.append("Clients:\n");
262    for (size_t i = 0; i < mClients.size(); ++i) {
263        sp<Client> client = mClients.valueAt(i).promote();
264        if (client != 0) {
265            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
266            result.append(buffer);
267        }
268    }
269
270    result.append("Notification Clients:\n");
271    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
272        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
273        result.append(buffer);
274    }
275
276    result.append("Global session refs:\n");
277    result.append(" session pid count\n");
278    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
279        AudioSessionRef *r = mAudioSessionRefs[i];
280        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
281        result.append(buffer);
282    }
283    write(fd, result.string(), result.size());
284}
285
286
287void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
288{
289    const size_t SIZE = 256;
290    char buffer[SIZE];
291    String8 result;
292    hardware_call_state hardwareStatus = mHardwareStatus;
293
294    snprintf(buffer, SIZE, "Hardware status: %d\n"
295                           "Standby Time mSec: %u\n",
296                            hardwareStatus,
297                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
298    result.append(buffer);
299    write(fd, result.string(), result.size());
300}
301
302void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
303{
304    const size_t SIZE = 256;
305    char buffer[SIZE];
306    String8 result;
307    snprintf(buffer, SIZE, "Permission Denial: "
308            "can't dump AudioFlinger from pid=%d, uid=%d\n",
309            IPCThreadState::self()->getCallingPid(),
310            IPCThreadState::self()->getCallingUid());
311    result.append(buffer);
312    write(fd, result.string(), result.size());
313}
314
315bool AudioFlinger::dumpTryLock(Mutex& mutex)
316{
317    bool locked = false;
318    for (int i = 0; i < kDumpLockRetries; ++i) {
319        if (mutex.tryLock() == NO_ERROR) {
320            locked = true;
321            break;
322        }
323        usleep(kDumpLockSleepUs);
324    }
325    return locked;
326}
327
328status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
329{
330    if (!dumpAllowed()) {
331        dumpPermissionDenial(fd, args);
332    } else {
333        // get state of hardware lock
334        bool hardwareLocked = dumpTryLock(mHardwareLock);
335        if (!hardwareLocked) {
336            String8 result(kHardwareLockedString);
337            write(fd, result.string(), result.size());
338        } else {
339            mHardwareLock.unlock();
340        }
341
342        bool locked = dumpTryLock(mLock);
343
344        // failed to lock - AudioFlinger is probably deadlocked
345        if (!locked) {
346            String8 result(kDeadlockedString);
347            write(fd, result.string(), result.size());
348        }
349
350        dumpClients(fd, args);
351        dumpInternals(fd, args);
352
353        // dump playback threads
354        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
355            mPlaybackThreads.valueAt(i)->dump(fd, args);
356        }
357
358        // dump record threads
359        for (size_t i = 0; i < mRecordThreads.size(); i++) {
360            mRecordThreads.valueAt(i)->dump(fd, args);
361        }
362
363        // dump all hardware devs
364        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
365            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
366            dev->dump(dev, fd);
367        }
368
369#ifdef TEE_SINK
370        // dump the serially shared record tee sink
371        if (mRecordTeeSource != 0) {
372            dumpTee(fd, mRecordTeeSource);
373        }
374#endif
375
376        if (locked) {
377            mLock.unlock();
378        }
379
380        // append a copy of media.log here by forwarding fd to it, but don't attempt
381        // to lookup the service if it's not running, as it will block for a second
382        if (mLogMemoryDealer != 0) {
383            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
384            if (binder != 0) {
385                fdprintf(fd, "\nmedia.log:\n");
386                Vector<String16> args;
387                binder->dump(fd, args);
388            }
389        }
390    }
391    return NO_ERROR;
392}
393
394sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
395{
396    // If pid is already in the mClients wp<> map, then use that entry
397    // (for which promote() is always != 0), otherwise create a new entry and Client.
398    sp<Client> client = mClients.valueFor(pid).promote();
399    if (client == 0) {
400        client = new Client(this, pid);
401        mClients.add(pid, client);
402    }
403
404    return client;
405}
406
407sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
408{
409    if (mLogMemoryDealer == 0) {
410        return new NBLog::Writer();
411    }
412    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
413    sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
414    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
415    if (binder != 0) {
416        interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
417    }
418    return writer;
419}
420
421void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
422{
423    if (writer == 0) {
424        return;
425    }
426    sp<IMemory> iMemory(writer->getIMemory());
427    if (iMemory == 0) {
428        return;
429    }
430    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
431    if (binder != 0) {
432        interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
433        // Now the media.log remote reference to IMemory is gone.
434        // When our last local reference to IMemory also drops to zero,
435        // the IMemory destructor will deallocate the region from mMemoryDealer.
436    }
437}
438
439// IAudioFlinger interface
440
441
442sp<IAudioTrack> AudioFlinger::createTrack(
443        audio_stream_type_t streamType,
444        uint32_t sampleRate,
445        audio_format_t format,
446        audio_channel_mask_t channelMask,
447        size_t frameCount,
448        IAudioFlinger::track_flags_t *flags,
449        const sp<IMemory>& sharedBuffer,
450        audio_io_handle_t output,
451        pid_t tid,
452        int *sessionId,
453        String8& name,
454        int clientUid,
455        status_t *status)
456{
457    sp<PlaybackThread::Track> track;
458    sp<TrackHandle> trackHandle;
459    sp<Client> client;
460    status_t lStatus;
461    int lSessionId;
462
463    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
464    // but if someone uses binder directly they could bypass that and cause us to crash
465    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
466        ALOGE("createTrack() invalid stream type %d", streamType);
467        lStatus = BAD_VALUE;
468        goto Exit;
469    }
470
471    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
472    // and we don't yet support 8.24 or 32-bit PCM
473    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
474        ALOGE("createTrack() invalid format %d", format);
475        lStatus = BAD_VALUE;
476        goto Exit;
477    }
478
479    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
480        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
481        lStatus = BAD_VALUE;
482        goto Exit;
483    }
484
485    {
486        Mutex::Autolock _l(mLock);
487        PlaybackThread *thread = checkPlaybackThread_l(output);
488        PlaybackThread *effectThread = NULL;
489        if (thread == NULL) {
490            ALOGE("no playback thread found for output handle %d", output);
491            lStatus = BAD_VALUE;
492            goto Exit;
493        }
494
495        pid_t pid = IPCThreadState::self()->getCallingPid();
496
497        client = registerPid_l(pid);
498
499        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
500        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
501            // check if an effect chain with the same session ID is present on another
502            // output thread and move it here.
503            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
504                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
505                if (mPlaybackThreads.keyAt(i) != output) {
506                    uint32_t sessions = t->hasAudioSession(*sessionId);
507                    if (sessions & PlaybackThread::EFFECT_SESSION) {
508                        effectThread = t.get();
509                        break;
510                    }
511                }
512            }
513            lSessionId = *sessionId;
514        } else {
515            // if no audio session id is provided, create one here
516            lSessionId = nextUniqueId();
517            if (sessionId != NULL) {
518                *sessionId = lSessionId;
519            }
520        }
521        ALOGV("createTrack() lSessionId: %d", lSessionId);
522
523        track = thread->createTrack_l(client, streamType, sampleRate, format,
524                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
525        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
526
527        // move effect chain to this output thread if an effect on same session was waiting
528        // for a track to be created
529        if (lStatus == NO_ERROR && effectThread != NULL) {
530            // no risk of deadlock because AudioFlinger::mLock is held
531            Mutex::Autolock _dl(thread->mLock);
532            Mutex::Autolock _sl(effectThread->mLock);
533            moveEffectChain_l(lSessionId, effectThread, thread, true);
534        }
535
536        // Look for sync events awaiting for a session to be used.
537        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
538            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
539                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
540                    if (lStatus == NO_ERROR) {
541                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
542                    } else {
543                        mPendingSyncEvents[i]->cancel();
544                    }
545                    mPendingSyncEvents.removeAt(i);
546                    i--;
547                }
548            }
549        }
550
551    }
552
553    if (lStatus == NO_ERROR) {
554        // s for server's pid, n for normal mixer name, f for fast index
555        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
556                track->fastIndex());
557        trackHandle = new TrackHandle(track);
558    } else {
559        // remove local strong reference to Client before deleting the Track so that the Client
560        // destructor is called by the TrackBase destructor with mLock held
561        client.clear();
562        track.clear();
563    }
564
565Exit:
566    *status = lStatus;
567    return trackHandle;
568}
569
570uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
571{
572    Mutex::Autolock _l(mLock);
573    PlaybackThread *thread = checkPlaybackThread_l(output);
574    if (thread == NULL) {
575        ALOGW("sampleRate() unknown thread %d", output);
576        return 0;
577    }
578    return thread->sampleRate();
579}
580
581int AudioFlinger::channelCount(audio_io_handle_t output) const
582{
583    Mutex::Autolock _l(mLock);
584    PlaybackThread *thread = checkPlaybackThread_l(output);
585    if (thread == NULL) {
586        ALOGW("channelCount() unknown thread %d", output);
587        return 0;
588    }
589    return thread->channelCount();
590}
591
592audio_format_t AudioFlinger::format(audio_io_handle_t output) const
593{
594    Mutex::Autolock _l(mLock);
595    PlaybackThread *thread = checkPlaybackThread_l(output);
596    if (thread == NULL) {
597        ALOGW("format() unknown thread %d", output);
598        return AUDIO_FORMAT_INVALID;
599    }
600    return thread->format();
601}
602
603size_t AudioFlinger::frameCount(audio_io_handle_t output) const
604{
605    Mutex::Autolock _l(mLock);
606    PlaybackThread *thread = checkPlaybackThread_l(output);
607    if (thread == NULL) {
608        ALOGW("frameCount() unknown thread %d", output);
609        return 0;
610    }
611    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
612    //       should examine all callers and fix them to handle smaller counts
613    return thread->frameCount();
614}
615
616uint32_t AudioFlinger::latency(audio_io_handle_t output) const
617{
618    Mutex::Autolock _l(mLock);
619    PlaybackThread *thread = checkPlaybackThread_l(output);
620    if (thread == NULL) {
621        ALOGW("latency(): no playback thread found for output handle %d", output);
622        return 0;
623    }
624    return thread->latency();
625}
626
627status_t AudioFlinger::setMasterVolume(float value)
628{
629    status_t ret = initCheck();
630    if (ret != NO_ERROR) {
631        return ret;
632    }
633
634    // check calling permissions
635    if (!settingsAllowed()) {
636        return PERMISSION_DENIED;
637    }
638
639    Mutex::Autolock _l(mLock);
640    mMasterVolume = value;
641
642    // Set master volume in the HALs which support it.
643    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
644        AutoMutex lock(mHardwareLock);
645        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
646
647        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
648        if (dev->canSetMasterVolume()) {
649            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
650        }
651        mHardwareStatus = AUDIO_HW_IDLE;
652    }
653
654    // Now set the master volume in each playback thread.  Playback threads
655    // assigned to HALs which do not have master volume support will apply
656    // master volume during the mix operation.  Threads with HALs which do
657    // support master volume will simply ignore the setting.
658    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
659        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
660
661    return NO_ERROR;
662}
663
664status_t AudioFlinger::setMode(audio_mode_t mode)
665{
666    status_t ret = initCheck();
667    if (ret != NO_ERROR) {
668        return ret;
669    }
670
671    // check calling permissions
672    if (!settingsAllowed()) {
673        return PERMISSION_DENIED;
674    }
675    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
676        ALOGW("Illegal value: setMode(%d)", mode);
677        return BAD_VALUE;
678    }
679
680    { // scope for the lock
681        AutoMutex lock(mHardwareLock);
682        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
683        mHardwareStatus = AUDIO_HW_SET_MODE;
684        ret = dev->set_mode(dev, mode);
685        mHardwareStatus = AUDIO_HW_IDLE;
686    }
687
688    if (NO_ERROR == ret) {
689        Mutex::Autolock _l(mLock);
690        mMode = mode;
691        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
692            mPlaybackThreads.valueAt(i)->setMode(mode);
693    }
694
695    return ret;
696}
697
698status_t AudioFlinger::setMicMute(bool state)
699{
700    status_t ret = initCheck();
701    if (ret != NO_ERROR) {
702        return ret;
703    }
704
705    // check calling permissions
706    if (!settingsAllowed()) {
707        return PERMISSION_DENIED;
708    }
709
710    AutoMutex lock(mHardwareLock);
711    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
712    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
713    ret = dev->set_mic_mute(dev, state);
714    mHardwareStatus = AUDIO_HW_IDLE;
715    return ret;
716}
717
718bool AudioFlinger::getMicMute() const
719{
720    status_t ret = initCheck();
721    if (ret != NO_ERROR) {
722        return false;
723    }
724
725    bool state = AUDIO_MODE_INVALID;
726    AutoMutex lock(mHardwareLock);
727    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
728    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
729    dev->get_mic_mute(dev, &state);
730    mHardwareStatus = AUDIO_HW_IDLE;
731    return state;
732}
733
734status_t AudioFlinger::setMasterMute(bool muted)
735{
736    status_t ret = initCheck();
737    if (ret != NO_ERROR) {
738        return ret;
739    }
740
741    // check calling permissions
742    if (!settingsAllowed()) {
743        return PERMISSION_DENIED;
744    }
745
746    Mutex::Autolock _l(mLock);
747    mMasterMute = muted;
748
749    // Set master mute in the HALs which support it.
750    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
751        AutoMutex lock(mHardwareLock);
752        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
753
754        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
755        if (dev->canSetMasterMute()) {
756            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
757        }
758        mHardwareStatus = AUDIO_HW_IDLE;
759    }
760
761    // Now set the master mute in each playback thread.  Playback threads
762    // assigned to HALs which do not have master mute support will apply master
763    // mute during the mix operation.  Threads with HALs which do support master
764    // mute will simply ignore the setting.
765    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
766        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
767
768    return NO_ERROR;
769}
770
771float AudioFlinger::masterVolume() const
772{
773    Mutex::Autolock _l(mLock);
774    return masterVolume_l();
775}
776
777bool AudioFlinger::masterMute() const
778{
779    Mutex::Autolock _l(mLock);
780    return masterMute_l();
781}
782
783float AudioFlinger::masterVolume_l() const
784{
785    return mMasterVolume;
786}
787
788bool AudioFlinger::masterMute_l() const
789{
790    return mMasterMute;
791}
792
793status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
794        audio_io_handle_t output)
795{
796    // check calling permissions
797    if (!settingsAllowed()) {
798        return PERMISSION_DENIED;
799    }
800
801    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
802        ALOGE("setStreamVolume() invalid stream %d", stream);
803        return BAD_VALUE;
804    }
805
806    AutoMutex lock(mLock);
807    PlaybackThread *thread = NULL;
808    if (output) {
809        thread = checkPlaybackThread_l(output);
810        if (thread == NULL) {
811            return BAD_VALUE;
812        }
813    }
814
815    mStreamTypes[stream].volume = value;
816
817    if (thread == NULL) {
818        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
819            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
820        }
821    } else {
822        thread->setStreamVolume(stream, value);
823    }
824
825    return NO_ERROR;
826}
827
828status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
829{
830    // check calling permissions
831    if (!settingsAllowed()) {
832        return PERMISSION_DENIED;
833    }
834
835    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
836        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
837        ALOGE("setStreamMute() invalid stream %d", stream);
838        return BAD_VALUE;
839    }
840
841    AutoMutex lock(mLock);
842    mStreamTypes[stream].mute = muted;
843    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
844        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
845
846    return NO_ERROR;
847}
848
849float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
850{
851    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
852        return 0.0f;
853    }
854
855    AutoMutex lock(mLock);
856    float volume;
857    if (output) {
858        PlaybackThread *thread = checkPlaybackThread_l(output);
859        if (thread == NULL) {
860            return 0.0f;
861        }
862        volume = thread->streamVolume(stream);
863    } else {
864        volume = streamVolume_l(stream);
865    }
866
867    return volume;
868}
869
870bool AudioFlinger::streamMute(audio_stream_type_t stream) const
871{
872    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
873        return true;
874    }
875
876    AutoMutex lock(mLock);
877    return streamMute_l(stream);
878}
879
880status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
881{
882    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
883            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
884
885    // check calling permissions
886    if (!settingsAllowed()) {
887        return PERMISSION_DENIED;
888    }
889
890    // ioHandle == 0 means the parameters are global to the audio hardware interface
891    if (ioHandle == 0) {
892        Mutex::Autolock _l(mLock);
893        status_t final_result = NO_ERROR;
894        {
895            AutoMutex lock(mHardwareLock);
896            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
897            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
898                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
899                status_t result = dev->set_parameters(dev, keyValuePairs.string());
900                final_result = result ?: final_result;
901            }
902            mHardwareStatus = AUDIO_HW_IDLE;
903        }
904        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
905        AudioParameter param = AudioParameter(keyValuePairs);
906        String8 value;
907        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
908            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
909            if (mBtNrecIsOff != btNrecIsOff) {
910                for (size_t i = 0; i < mRecordThreads.size(); i++) {
911                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
912                    audio_devices_t device = thread->inDevice();
913                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
914                    // collect all of the thread's session IDs
915                    KeyedVector<int, bool> ids = thread->sessionIds();
916                    // suspend effects associated with those session IDs
917                    for (size_t j = 0; j < ids.size(); ++j) {
918                        int sessionId = ids.keyAt(j);
919                        thread->setEffectSuspended(FX_IID_AEC,
920                                                   suspend,
921                                                   sessionId);
922                        thread->setEffectSuspended(FX_IID_NS,
923                                                   suspend,
924                                                   sessionId);
925                    }
926                }
927                mBtNrecIsOff = btNrecIsOff;
928            }
929        }
930        String8 screenState;
931        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
932            bool isOff = screenState == "off";
933            if (isOff != (AudioFlinger::mScreenState & 1)) {
934                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
935            }
936        }
937        return final_result;
938    }
939
940    // hold a strong ref on thread in case closeOutput() or closeInput() is called
941    // and the thread is exited once the lock is released
942    sp<ThreadBase> thread;
943    {
944        Mutex::Autolock _l(mLock);
945        thread = checkPlaybackThread_l(ioHandle);
946        if (thread == 0) {
947            thread = checkRecordThread_l(ioHandle);
948        } else if (thread == primaryPlaybackThread_l()) {
949            // indicate output device change to all input threads for pre processing
950            AudioParameter param = AudioParameter(keyValuePairs);
951            int value;
952            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
953                    (value != 0)) {
954                for (size_t i = 0; i < mRecordThreads.size(); i++) {
955                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
956                }
957            }
958        }
959    }
960    if (thread != 0) {
961        return thread->setParameters(keyValuePairs);
962    }
963    return BAD_VALUE;
964}
965
966String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
967{
968    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
969            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
970
971    Mutex::Autolock _l(mLock);
972
973    if (ioHandle == 0) {
974        String8 out_s8;
975
976        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
977            char *s;
978            {
979            AutoMutex lock(mHardwareLock);
980            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
981            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
982            s = dev->get_parameters(dev, keys.string());
983            mHardwareStatus = AUDIO_HW_IDLE;
984            }
985            out_s8 += String8(s ? s : "");
986            free(s);
987        }
988        return out_s8;
989    }
990
991    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
992    if (playbackThread != NULL) {
993        return playbackThread->getParameters(keys);
994    }
995    RecordThread *recordThread = checkRecordThread_l(ioHandle);
996    if (recordThread != NULL) {
997        return recordThread->getParameters(keys);
998    }
999    return String8("");
1000}
1001
1002size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1003        audio_channel_mask_t channelMask) const
1004{
1005    status_t ret = initCheck();
1006    if (ret != NO_ERROR) {
1007        return 0;
1008    }
1009
1010    AutoMutex lock(mHardwareLock);
1011    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1012    struct audio_config config;
1013    memset(&config, 0, sizeof(config));
1014    config.sample_rate = sampleRate;
1015    config.channel_mask = channelMask;
1016    config.format = format;
1017
1018    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1019    size_t size = dev->get_input_buffer_size(dev, &config);
1020    mHardwareStatus = AUDIO_HW_IDLE;
1021    return size;
1022}
1023
1024unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1025{
1026    Mutex::Autolock _l(mLock);
1027
1028    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1029    if (recordThread != NULL) {
1030        return recordThread->getInputFramesLost();
1031    }
1032    return 0;
1033}
1034
1035status_t AudioFlinger::setVoiceVolume(float value)
1036{
1037    status_t ret = initCheck();
1038    if (ret != NO_ERROR) {
1039        return ret;
1040    }
1041
1042    // check calling permissions
1043    if (!settingsAllowed()) {
1044        return PERMISSION_DENIED;
1045    }
1046
1047    AutoMutex lock(mHardwareLock);
1048    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1049    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1050    ret = dev->set_voice_volume(dev, value);
1051    mHardwareStatus = AUDIO_HW_IDLE;
1052
1053    return ret;
1054}
1055
1056status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1057        audio_io_handle_t output) const
1058{
1059    status_t status;
1060
1061    Mutex::Autolock _l(mLock);
1062
1063    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1064    if (playbackThread != NULL) {
1065        return playbackThread->getRenderPosition(halFrames, dspFrames);
1066    }
1067
1068    return BAD_VALUE;
1069}
1070
1071void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1072{
1073
1074    Mutex::Autolock _l(mLock);
1075
1076    pid_t pid = IPCThreadState::self()->getCallingPid();
1077    if (mNotificationClients.indexOfKey(pid) < 0) {
1078        sp<NotificationClient> notificationClient = new NotificationClient(this,
1079                                                                            client,
1080                                                                            pid);
1081        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1082
1083        mNotificationClients.add(pid, notificationClient);
1084
1085        sp<IBinder> binder = client->asBinder();
1086        binder->linkToDeath(notificationClient);
1087
1088        // the config change is always sent from playback or record threads to avoid deadlock
1089        // with AudioSystem::gLock
1090        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1091            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1092        }
1093
1094        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1095            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1096        }
1097    }
1098}
1099
1100void AudioFlinger::removeNotificationClient(pid_t pid)
1101{
1102    Mutex::Autolock _l(mLock);
1103
1104    mNotificationClients.removeItem(pid);
1105
1106    ALOGV("%d died, releasing its sessions", pid);
1107    size_t num = mAudioSessionRefs.size();
1108    bool removed = false;
1109    for (size_t i = 0; i< num; ) {
1110        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1111        ALOGV(" pid %d @ %d", ref->mPid, i);
1112        if (ref->mPid == pid) {
1113            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1114            mAudioSessionRefs.removeAt(i);
1115            delete ref;
1116            removed = true;
1117            num--;
1118        } else {
1119            i++;
1120        }
1121    }
1122    if (removed) {
1123        purgeStaleEffects_l();
1124    }
1125}
1126
1127// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1128void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1129{
1130    size_t size = mNotificationClients.size();
1131    for (size_t i = 0; i < size; i++) {
1132        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1133                                                                               param2);
1134    }
1135}
1136
1137// removeClient_l() must be called with AudioFlinger::mLock held
1138void AudioFlinger::removeClient_l(pid_t pid)
1139{
1140    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1141            IPCThreadState::self()->getCallingPid());
1142    mClients.removeItem(pid);
1143}
1144
1145// getEffectThread_l() must be called with AudioFlinger::mLock held
1146sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1147{
1148    sp<PlaybackThread> thread;
1149
1150    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1151        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1152            ALOG_ASSERT(thread == 0);
1153            thread = mPlaybackThreads.valueAt(i);
1154        }
1155    }
1156
1157    return thread;
1158}
1159
1160
1161
1162// ----------------------------------------------------------------------------
1163
1164AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1165    :   RefBase(),
1166        mAudioFlinger(audioFlinger),
1167        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1168        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1169        mPid(pid),
1170        mTimedTrackCount(0)
1171{
1172    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1173}
1174
1175// Client destructor must be called with AudioFlinger::mLock held
1176AudioFlinger::Client::~Client()
1177{
1178    mAudioFlinger->removeClient_l(mPid);
1179}
1180
1181sp<MemoryDealer> AudioFlinger::Client::heap() const
1182{
1183    return mMemoryDealer;
1184}
1185
1186// Reserve one of the limited slots for a timed audio track associated
1187// with this client
1188bool AudioFlinger::Client::reserveTimedTrack()
1189{
1190    const int kMaxTimedTracksPerClient = 4;
1191
1192    Mutex::Autolock _l(mTimedTrackLock);
1193
1194    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1195        ALOGW("can not create timed track - pid %d has exceeded the limit",
1196             mPid);
1197        return false;
1198    }
1199
1200    mTimedTrackCount++;
1201    return true;
1202}
1203
1204// Release a slot for a timed audio track
1205void AudioFlinger::Client::releaseTimedTrack()
1206{
1207    Mutex::Autolock _l(mTimedTrackLock);
1208    mTimedTrackCount--;
1209}
1210
1211// ----------------------------------------------------------------------------
1212
1213AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1214                                                     const sp<IAudioFlingerClient>& client,
1215                                                     pid_t pid)
1216    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1217{
1218}
1219
1220AudioFlinger::NotificationClient::~NotificationClient()
1221{
1222}
1223
1224void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1225{
1226    sp<NotificationClient> keep(this);
1227    mAudioFlinger->removeNotificationClient(mPid);
1228}
1229
1230
1231// ----------------------------------------------------------------------------
1232
1233static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1234    return audio_is_remote_submix_device(inDevice);
1235}
1236
1237sp<IAudioRecord> AudioFlinger::openRecord(
1238        audio_io_handle_t input,
1239        uint32_t sampleRate,
1240        audio_format_t format,
1241        audio_channel_mask_t channelMask,
1242        size_t frameCount,
1243        IAudioFlinger::track_flags_t *flags,
1244        pid_t tid,
1245        int *sessionId,
1246        status_t *status)
1247{
1248    sp<RecordThread::RecordTrack> recordTrack;
1249    sp<RecordHandle> recordHandle;
1250    sp<Client> client;
1251    status_t lStatus;
1252    RecordThread *thread;
1253    size_t inFrameCount;
1254    int lSessionId;
1255
1256    // check calling permissions
1257    if (!recordingAllowed()) {
1258        ALOGE("openRecord() permission denied: recording not allowed");
1259        lStatus = PERMISSION_DENIED;
1260        goto Exit;
1261    }
1262
1263    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1264        ALOGE("openRecord() invalid format %d", format);
1265        lStatus = BAD_VALUE;
1266        goto Exit;
1267    }
1268
1269    // add client to list
1270    { // scope for mLock
1271        Mutex::Autolock _l(mLock);
1272        thread = checkRecordThread_l(input);
1273        if (thread == NULL) {
1274            ALOGE("openRecord() checkRecordThread_l failed");
1275            lStatus = BAD_VALUE;
1276            goto Exit;
1277        }
1278
1279        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1280                && !captureAudioOutputAllowed()) {
1281            ALOGE("openRecord() permission denied: capture not allowed");
1282            lStatus = PERMISSION_DENIED;
1283            goto Exit;
1284        }
1285
1286        pid_t pid = IPCThreadState::self()->getCallingPid();
1287        client = registerPid_l(pid);
1288
1289        // If no audio session id is provided, create one here
1290        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1291            lSessionId = *sessionId;
1292        } else {
1293            lSessionId = nextUniqueId();
1294            if (sessionId != NULL) {
1295                *sessionId = lSessionId;
1296            }
1297        }
1298        // create new record track.
1299        // The record track uses one track in mHardwareMixerThread by convention.
1300        // TODO: the uid should be passed in as a parameter to openRecord
1301        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1302                                                  frameCount, lSessionId,
1303                                                  IPCThreadState::self()->getCallingUid(),
1304                                                  flags, tid, &lStatus);
1305        LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
1306    }
1307
1308    if (lStatus != NO_ERROR) {
1309        // remove local strong reference to Client before deleting the RecordTrack so that the
1310        // Client destructor is called by the TrackBase destructor with mLock held
1311        client.clear();
1312        recordTrack.clear();
1313        goto Exit;
1314    }
1315
1316    // return handle to client
1317    recordHandle = new RecordHandle(recordTrack);
1318
1319Exit:
1320    *status = lStatus;
1321    return recordHandle;
1322}
1323
1324
1325
1326// ----------------------------------------------------------------------------
1327
1328audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1329{
1330    if (!settingsAllowed()) {
1331        return 0;
1332    }
1333    Mutex::Autolock _l(mLock);
1334    return loadHwModule_l(name);
1335}
1336
1337// loadHwModule_l() must be called with AudioFlinger::mLock held
1338audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1339{
1340    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1341        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1342            ALOGW("loadHwModule() module %s already loaded", name);
1343            return mAudioHwDevs.keyAt(i);
1344        }
1345    }
1346
1347    audio_hw_device_t *dev;
1348
1349    int rc = load_audio_interface(name, &dev);
1350    if (rc) {
1351        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1352        return 0;
1353    }
1354
1355    mHardwareStatus = AUDIO_HW_INIT;
1356    rc = dev->init_check(dev);
1357    mHardwareStatus = AUDIO_HW_IDLE;
1358    if (rc) {
1359        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1360        return 0;
1361    }
1362
1363    // Check and cache this HAL's level of support for master mute and master
1364    // volume.  If this is the first HAL opened, and it supports the get
1365    // methods, use the initial values provided by the HAL as the current
1366    // master mute and volume settings.
1367
1368    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1369    {  // scope for auto-lock pattern
1370        AutoMutex lock(mHardwareLock);
1371
1372        if (0 == mAudioHwDevs.size()) {
1373            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1374            if (NULL != dev->get_master_volume) {
1375                float mv;
1376                if (OK == dev->get_master_volume(dev, &mv)) {
1377                    mMasterVolume = mv;
1378                }
1379            }
1380
1381            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1382            if (NULL != dev->get_master_mute) {
1383                bool mm;
1384                if (OK == dev->get_master_mute(dev, &mm)) {
1385                    mMasterMute = mm;
1386                }
1387            }
1388        }
1389
1390        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1391        if ((NULL != dev->set_master_volume) &&
1392            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1393            flags = static_cast<AudioHwDevice::Flags>(flags |
1394                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1395        }
1396
1397        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1398        if ((NULL != dev->set_master_mute) &&
1399            (OK == dev->set_master_mute(dev, mMasterMute))) {
1400            flags = static_cast<AudioHwDevice::Flags>(flags |
1401                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1402        }
1403
1404        mHardwareStatus = AUDIO_HW_IDLE;
1405    }
1406
1407    audio_module_handle_t handle = nextUniqueId();
1408    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1409
1410    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1411          name, dev->common.module->name, dev->common.module->id, handle);
1412
1413    return handle;
1414
1415}
1416
1417// ----------------------------------------------------------------------------
1418
1419uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1420{
1421    Mutex::Autolock _l(mLock);
1422    PlaybackThread *thread = primaryPlaybackThread_l();
1423    return thread != NULL ? thread->sampleRate() : 0;
1424}
1425
1426size_t AudioFlinger::getPrimaryOutputFrameCount()
1427{
1428    Mutex::Autolock _l(mLock);
1429    PlaybackThread *thread = primaryPlaybackThread_l();
1430    return thread != NULL ? thread->frameCountHAL() : 0;
1431}
1432
1433// ----------------------------------------------------------------------------
1434
1435status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1436{
1437    uid_t uid = IPCThreadState::self()->getCallingUid();
1438    if (uid != AID_SYSTEM) {
1439        return PERMISSION_DENIED;
1440    }
1441    Mutex::Autolock _l(mLock);
1442    if (mIsDeviceTypeKnown) {
1443        return INVALID_OPERATION;
1444    }
1445    mIsLowRamDevice = isLowRamDevice;
1446    mIsDeviceTypeKnown = true;
1447    return NO_ERROR;
1448}
1449
1450// ----------------------------------------------------------------------------
1451
1452audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1453                                           audio_devices_t *pDevices,
1454                                           uint32_t *pSamplingRate,
1455                                           audio_format_t *pFormat,
1456                                           audio_channel_mask_t *pChannelMask,
1457                                           uint32_t *pLatencyMs,
1458                                           audio_output_flags_t flags,
1459                                           const audio_offload_info_t *offloadInfo)
1460{
1461    struct audio_config config;
1462    memset(&config, 0, sizeof(config));
1463    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1464    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1465    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1466    if (offloadInfo != NULL) {
1467        config.offload_info = *offloadInfo;
1468    }
1469
1470    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1471              module,
1472              (pDevices != NULL) ? *pDevices : 0,
1473              config.sample_rate,
1474              config.format,
1475              config.channel_mask,
1476              flags);
1477    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1478          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1479
1480    if (pDevices == NULL || *pDevices == 0) {
1481        return 0;
1482    }
1483
1484    Mutex::Autolock _l(mLock);
1485
1486    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1487    if (outHwDev == NULL) {
1488        return 0;
1489    }
1490
1491    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1492    audio_io_handle_t id = nextUniqueId();
1493
1494    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1495
1496    audio_stream_out_t *outStream = NULL;
1497    status_t status = hwDevHal->open_output_stream(hwDevHal,
1498                                          id,
1499                                          *pDevices,
1500                                          (audio_output_flags_t)flags,
1501                                          &config,
1502                                          &outStream);
1503
1504    mHardwareStatus = AUDIO_HW_IDLE;
1505    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1506            "Channels %x, status %d",
1507            outStream,
1508            config.sample_rate,
1509            config.format,
1510            config.channel_mask,
1511            status);
1512
1513    if (status == NO_ERROR && outStream != NULL) {
1514        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1515
1516        PlaybackThread *thread;
1517        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1518            thread = new OffloadThread(this, output, id, *pDevices);
1519            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1520        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1521            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1522            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1523            thread = new DirectOutputThread(this, output, id, *pDevices);
1524            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1525        } else {
1526            thread = new MixerThread(this, output, id, *pDevices);
1527            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1528        }
1529        mPlaybackThreads.add(id, thread);
1530
1531        if (pSamplingRate != NULL) {
1532            *pSamplingRate = config.sample_rate;
1533        }
1534        if (pFormat != NULL) {
1535            *pFormat = config.format;
1536        }
1537        if (pChannelMask != NULL) {
1538            *pChannelMask = config.channel_mask;
1539        }
1540        if (pLatencyMs != NULL) {
1541            *pLatencyMs = thread->latency();
1542        }
1543
1544        // notify client processes of the new output creation
1545        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1546
1547        // the first primary output opened designates the primary hw device
1548        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1549            ALOGI("Using module %d has the primary audio interface", module);
1550            mPrimaryHardwareDev = outHwDev;
1551
1552            AutoMutex lock(mHardwareLock);
1553            mHardwareStatus = AUDIO_HW_SET_MODE;
1554            hwDevHal->set_mode(hwDevHal, mMode);
1555            mHardwareStatus = AUDIO_HW_IDLE;
1556        }
1557        return id;
1558    }
1559
1560    return 0;
1561}
1562
1563audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1564        audio_io_handle_t output2)
1565{
1566    Mutex::Autolock _l(mLock);
1567    MixerThread *thread1 = checkMixerThread_l(output1);
1568    MixerThread *thread2 = checkMixerThread_l(output2);
1569
1570    if (thread1 == NULL || thread2 == NULL) {
1571        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1572                output2);
1573        return 0;
1574    }
1575
1576    audio_io_handle_t id = nextUniqueId();
1577    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1578    thread->addOutputTrack(thread2);
1579    mPlaybackThreads.add(id, thread);
1580    // notify client processes of the new output creation
1581    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1582    return id;
1583}
1584
1585status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1586{
1587    return closeOutput_nonvirtual(output);
1588}
1589
1590status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1591{
1592    // keep strong reference on the playback thread so that
1593    // it is not destroyed while exit() is executed
1594    sp<PlaybackThread> thread;
1595    {
1596        Mutex::Autolock _l(mLock);
1597        thread = checkPlaybackThread_l(output);
1598        if (thread == NULL) {
1599            return BAD_VALUE;
1600        }
1601
1602        ALOGV("closeOutput() %d", output);
1603
1604        if (thread->type() == ThreadBase::MIXER) {
1605            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1606                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1607                    DuplicatingThread *dupThread =
1608                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1609                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1610
1611                }
1612            }
1613        }
1614
1615
1616        mPlaybackThreads.removeItem(output);
1617        // save all effects to the default thread
1618        if (mPlaybackThreads.size()) {
1619            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1620            if (dstThread != NULL) {
1621                // audioflinger lock is held here so the acquisition order of thread locks does not
1622                // matter
1623                Mutex::Autolock _dl(dstThread->mLock);
1624                Mutex::Autolock _sl(thread->mLock);
1625                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1626                for (size_t i = 0; i < effectChains.size(); i ++) {
1627                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1628                }
1629            }
1630        }
1631        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1632    }
1633    thread->exit();
1634    // The thread entity (active unit of execution) is no longer running here,
1635    // but the ThreadBase container still exists.
1636
1637    if (thread->type() != ThreadBase::DUPLICATING) {
1638        AudioStreamOut *out = thread->clearOutput();
1639        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1640        // from now on thread->mOutput is NULL
1641        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1642        delete out;
1643    }
1644    return NO_ERROR;
1645}
1646
1647status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1648{
1649    Mutex::Autolock _l(mLock);
1650    PlaybackThread *thread = checkPlaybackThread_l(output);
1651
1652    if (thread == NULL) {
1653        return BAD_VALUE;
1654    }
1655
1656    ALOGV("suspendOutput() %d", output);
1657    thread->suspend();
1658
1659    return NO_ERROR;
1660}
1661
1662status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1663{
1664    Mutex::Autolock _l(mLock);
1665    PlaybackThread *thread = checkPlaybackThread_l(output);
1666
1667    if (thread == NULL) {
1668        return BAD_VALUE;
1669    }
1670
1671    ALOGV("restoreOutput() %d", output);
1672
1673    thread->restore();
1674
1675    return NO_ERROR;
1676}
1677
1678audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1679                                          audio_devices_t *pDevices,
1680                                          uint32_t *pSamplingRate,
1681                                          audio_format_t *pFormat,
1682                                          audio_channel_mask_t *pChannelMask)
1683{
1684    struct audio_config config;
1685    memset(&config, 0, sizeof(config));
1686    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1687    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1688    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1689
1690    uint32_t reqSamplingRate = config.sample_rate;
1691    audio_format_t reqFormat = config.format;
1692    audio_channel_mask_t reqChannelMask = config.channel_mask;
1693
1694    if (pDevices == NULL || *pDevices == 0) {
1695        return 0;
1696    }
1697
1698    Mutex::Autolock _l(mLock);
1699
1700    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1701    if (inHwDev == NULL) {
1702        return 0;
1703    }
1704
1705    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1706    audio_io_handle_t id = nextUniqueId();
1707
1708    audio_stream_in_t *inStream = NULL;
1709    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1710                                        &inStream);
1711    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1712            "status %d",
1713            inStream,
1714            config.sample_rate,
1715            config.format,
1716            config.channel_mask,
1717            status);
1718
1719    // If the input could not be opened with the requested parameters and we can handle the
1720    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1721    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1722    if (status == BAD_VALUE &&
1723        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1724        (config.sample_rate <= 2 * reqSamplingRate) &&
1725        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1726        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1727        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1728        inStream = NULL;
1729        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1730        // FIXME log this new status; HAL should not propose any further changes
1731    }
1732
1733    if (status == NO_ERROR && inStream != NULL) {
1734
1735#ifdef TEE_SINK
1736        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1737        // or (re-)create if current Pipe is idle and does not match the new format
1738        sp<NBAIO_Sink> teeSink;
1739        enum {
1740            TEE_SINK_NO,    // don't copy input
1741            TEE_SINK_NEW,   // copy input using a new pipe
1742            TEE_SINK_OLD,   // copy input using an existing pipe
1743        } kind;
1744        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1745                                        popcount(inStream->common.get_channels(&inStream->common)));
1746        if (!mTeeSinkInputEnabled) {
1747            kind = TEE_SINK_NO;
1748        } else if (format == Format_Invalid) {
1749            kind = TEE_SINK_NO;
1750        } else if (mRecordTeeSink == 0) {
1751            kind = TEE_SINK_NEW;
1752        } else if (mRecordTeeSink->getStrongCount() != 1) {
1753            kind = TEE_SINK_NO;
1754        } else if (format == mRecordTeeSink->format()) {
1755            kind = TEE_SINK_OLD;
1756        } else {
1757            kind = TEE_SINK_NEW;
1758        }
1759        switch (kind) {
1760        case TEE_SINK_NEW: {
1761            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1762            size_t numCounterOffers = 0;
1763            const NBAIO_Format offers[1] = {format};
1764            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1765            ALOG_ASSERT(index == 0);
1766            PipeReader *pipeReader = new PipeReader(*pipe);
1767            numCounterOffers = 0;
1768            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1769            ALOG_ASSERT(index == 0);
1770            mRecordTeeSink = pipe;
1771            mRecordTeeSource = pipeReader;
1772            teeSink = pipe;
1773            }
1774            break;
1775        case TEE_SINK_OLD:
1776            teeSink = mRecordTeeSink;
1777            break;
1778        case TEE_SINK_NO:
1779        default:
1780            break;
1781        }
1782#endif
1783
1784        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1785
1786        // Start record thread
1787        // RecordThread requires both input and output device indication to forward to audio
1788        // pre processing modules
1789        RecordThread *thread = new RecordThread(this,
1790                                  input,
1791                                  reqSamplingRate,
1792                                  reqChannelMask,
1793                                  id,
1794                                  primaryOutputDevice_l(),
1795                                  *pDevices
1796#ifdef TEE_SINK
1797                                  , teeSink
1798#endif
1799                                  );
1800        mRecordThreads.add(id, thread);
1801        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1802        if (pSamplingRate != NULL) {
1803            *pSamplingRate = reqSamplingRate;
1804        }
1805        if (pFormat != NULL) {
1806            *pFormat = config.format;
1807        }
1808        if (pChannelMask != NULL) {
1809            *pChannelMask = reqChannelMask;
1810        }
1811
1812        // notify client processes of the new input creation
1813        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1814        return id;
1815    }
1816
1817    return 0;
1818}
1819
1820status_t AudioFlinger::closeInput(audio_io_handle_t input)
1821{
1822    return closeInput_nonvirtual(input);
1823}
1824
1825status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1826{
1827    // keep strong reference on the record thread so that
1828    // it is not destroyed while exit() is executed
1829    sp<RecordThread> thread;
1830    {
1831        Mutex::Autolock _l(mLock);
1832        thread = checkRecordThread_l(input);
1833        if (thread == 0) {
1834            return BAD_VALUE;
1835        }
1836
1837        ALOGV("closeInput() %d", input);
1838        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1839        mRecordThreads.removeItem(input);
1840    }
1841    thread->exit();
1842    // The thread entity (active unit of execution) is no longer running here,
1843    // but the ThreadBase container still exists.
1844
1845    AudioStreamIn *in = thread->clearInput();
1846    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1847    // from now on thread->mInput is NULL
1848    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1849    delete in;
1850
1851    return NO_ERROR;
1852}
1853
1854status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1855{
1856    Mutex::Autolock _l(mLock);
1857    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1858
1859    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1860        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1861        thread->invalidateTracks(stream);
1862    }
1863
1864    return NO_ERROR;
1865}
1866
1867
1868int AudioFlinger::newAudioSessionId()
1869{
1870    return nextUniqueId();
1871}
1872
1873void AudioFlinger::acquireAudioSessionId(int audioSession)
1874{
1875    Mutex::Autolock _l(mLock);
1876    pid_t caller = IPCThreadState::self()->getCallingPid();
1877    ALOGV("acquiring %d from %d", audioSession, caller);
1878
1879    // Ignore requests received from processes not known as notification client. The request
1880    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1881    // called from a different pid leaving a stale session reference.  Also we don't know how
1882    // to clear this reference if the client process dies.
1883    if (mNotificationClients.indexOfKey(caller) < 0) {
1884        ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1885        return;
1886    }
1887
1888    size_t num = mAudioSessionRefs.size();
1889    for (size_t i = 0; i< num; i++) {
1890        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1891        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1892            ref->mCnt++;
1893            ALOGV(" incremented refcount to %d", ref->mCnt);
1894            return;
1895        }
1896    }
1897    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1898    ALOGV(" added new entry for %d", audioSession);
1899}
1900
1901void AudioFlinger::releaseAudioSessionId(int audioSession)
1902{
1903    Mutex::Autolock _l(mLock);
1904    pid_t caller = IPCThreadState::self()->getCallingPid();
1905    ALOGV("releasing %d from %d", audioSession, caller);
1906    size_t num = mAudioSessionRefs.size();
1907    for (size_t i = 0; i< num; i++) {
1908        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1909        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1910            ref->mCnt--;
1911            ALOGV(" decremented refcount to %d", ref->mCnt);
1912            if (ref->mCnt == 0) {
1913                mAudioSessionRefs.removeAt(i);
1914                delete ref;
1915                purgeStaleEffects_l();
1916            }
1917            return;
1918        }
1919    }
1920    // If the caller is mediaserver it is likely that the session being released was acquired
1921    // on behalf of a process not in notification clients and we ignore the warning.
1922    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
1923}
1924
1925void AudioFlinger::purgeStaleEffects_l() {
1926
1927    ALOGV("purging stale effects");
1928
1929    Vector< sp<EffectChain> > chains;
1930
1931    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1932        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1933        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1934            sp<EffectChain> ec = t->mEffectChains[j];
1935            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1936                chains.push(ec);
1937            }
1938        }
1939    }
1940    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1941        sp<RecordThread> t = mRecordThreads.valueAt(i);
1942        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1943            sp<EffectChain> ec = t->mEffectChains[j];
1944            chains.push(ec);
1945        }
1946    }
1947
1948    for (size_t i = 0; i < chains.size(); i++) {
1949        sp<EffectChain> ec = chains[i];
1950        int sessionid = ec->sessionId();
1951        sp<ThreadBase> t = ec->mThread.promote();
1952        if (t == 0) {
1953            continue;
1954        }
1955        size_t numsessionrefs = mAudioSessionRefs.size();
1956        bool found = false;
1957        for (size_t k = 0; k < numsessionrefs; k++) {
1958            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1959            if (ref->mSessionid == sessionid) {
1960                ALOGV(" session %d still exists for %d with %d refs",
1961                    sessionid, ref->mPid, ref->mCnt);
1962                found = true;
1963                break;
1964            }
1965        }
1966        if (!found) {
1967            Mutex::Autolock _l(t->mLock);
1968            // remove all effects from the chain
1969            while (ec->mEffects.size()) {
1970                sp<EffectModule> effect = ec->mEffects[0];
1971                effect->unPin();
1972                t->removeEffect_l(effect);
1973                if (effect->purgeHandles()) {
1974                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
1975                }
1976                AudioSystem::unregisterEffect(effect->id());
1977            }
1978        }
1979    }
1980    return;
1981}
1982
1983// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
1984AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
1985{
1986    return mPlaybackThreads.valueFor(output).get();
1987}
1988
1989// checkMixerThread_l() must be called with AudioFlinger::mLock held
1990AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
1991{
1992    PlaybackThread *thread = checkPlaybackThread_l(output);
1993    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
1994}
1995
1996// checkRecordThread_l() must be called with AudioFlinger::mLock held
1997AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
1998{
1999    return mRecordThreads.valueFor(input).get();
2000}
2001
2002uint32_t AudioFlinger::nextUniqueId()
2003{
2004    return android_atomic_inc(&mNextUniqueId);
2005}
2006
2007AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2008{
2009    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2010        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2011        AudioStreamOut *output = thread->getOutput();
2012        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2013            return thread;
2014        }
2015    }
2016    return NULL;
2017}
2018
2019audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2020{
2021    PlaybackThread *thread = primaryPlaybackThread_l();
2022
2023    if (thread == NULL) {
2024        return 0;
2025    }
2026
2027    return thread->outDevice();
2028}
2029
2030sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2031                                    int triggerSession,
2032                                    int listenerSession,
2033                                    sync_event_callback_t callBack,
2034                                    void *cookie)
2035{
2036    Mutex::Autolock _l(mLock);
2037
2038    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2039    status_t playStatus = NAME_NOT_FOUND;
2040    status_t recStatus = NAME_NOT_FOUND;
2041    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2042        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2043        if (playStatus == NO_ERROR) {
2044            return event;
2045        }
2046    }
2047    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2048        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2049        if (recStatus == NO_ERROR) {
2050            return event;
2051        }
2052    }
2053    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2054        mPendingSyncEvents.add(event);
2055    } else {
2056        ALOGV("createSyncEvent() invalid event %d", event->type());
2057        event.clear();
2058    }
2059    return event;
2060}
2061
2062// ----------------------------------------------------------------------------
2063//  Effect management
2064// ----------------------------------------------------------------------------
2065
2066
2067status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2068{
2069    Mutex::Autolock _l(mLock);
2070    return EffectQueryNumberEffects(numEffects);
2071}
2072
2073status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2074{
2075    Mutex::Autolock _l(mLock);
2076    return EffectQueryEffect(index, descriptor);
2077}
2078
2079status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2080        effect_descriptor_t *descriptor) const
2081{
2082    Mutex::Autolock _l(mLock);
2083    return EffectGetDescriptor(pUuid, descriptor);
2084}
2085
2086
2087sp<IEffect> AudioFlinger::createEffect(
2088        effect_descriptor_t *pDesc,
2089        const sp<IEffectClient>& effectClient,
2090        int32_t priority,
2091        audio_io_handle_t io,
2092        int sessionId,
2093        status_t *status,
2094        int *id,
2095        int *enabled)
2096{
2097    status_t lStatus = NO_ERROR;
2098    sp<EffectHandle> handle;
2099    effect_descriptor_t desc;
2100
2101    pid_t pid = IPCThreadState::self()->getCallingPid();
2102    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2103            pid, effectClient.get(), priority, sessionId, io);
2104
2105    if (pDesc == NULL) {
2106        lStatus = BAD_VALUE;
2107        goto Exit;
2108    }
2109
2110    // check audio settings permission for global effects
2111    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2112        lStatus = PERMISSION_DENIED;
2113        goto Exit;
2114    }
2115
2116    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2117    // that can only be created by audio policy manager (running in same process)
2118    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2119        lStatus = PERMISSION_DENIED;
2120        goto Exit;
2121    }
2122
2123    {
2124        if (!EffectIsNullUuid(&pDesc->uuid)) {
2125            // if uuid is specified, request effect descriptor
2126            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2127            if (lStatus < 0) {
2128                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2129                goto Exit;
2130            }
2131        } else {
2132            // if uuid is not specified, look for an available implementation
2133            // of the required type in effect factory
2134            if (EffectIsNullUuid(&pDesc->type)) {
2135                ALOGW("createEffect() no effect type");
2136                lStatus = BAD_VALUE;
2137                goto Exit;
2138            }
2139            uint32_t numEffects = 0;
2140            effect_descriptor_t d;
2141            d.flags = 0; // prevent compiler warning
2142            bool found = false;
2143
2144            lStatus = EffectQueryNumberEffects(&numEffects);
2145            if (lStatus < 0) {
2146                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2147                goto Exit;
2148            }
2149            for (uint32_t i = 0; i < numEffects; i++) {
2150                lStatus = EffectQueryEffect(i, &desc);
2151                if (lStatus < 0) {
2152                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2153                    continue;
2154                }
2155                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2156                    // If matching type found save effect descriptor. If the session is
2157                    // 0 and the effect is not auxiliary, continue enumeration in case
2158                    // an auxiliary version of this effect type is available
2159                    found = true;
2160                    d = desc;
2161                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2162                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2163                        break;
2164                    }
2165                }
2166            }
2167            if (!found) {
2168                lStatus = BAD_VALUE;
2169                ALOGW("createEffect() effect not found");
2170                goto Exit;
2171            }
2172            // For same effect type, chose auxiliary version over insert version if
2173            // connect to output mix (Compliance to OpenSL ES)
2174            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2175                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2176                desc = d;
2177            }
2178        }
2179
2180        // Do not allow auxiliary effects on a session different from 0 (output mix)
2181        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2182             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2183            lStatus = INVALID_OPERATION;
2184            goto Exit;
2185        }
2186
2187        // check recording permission for visualizer
2188        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2189            !recordingAllowed()) {
2190            lStatus = PERMISSION_DENIED;
2191            goto Exit;
2192        }
2193
2194        // return effect descriptor
2195        *pDesc = desc;
2196        if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2197            // if the output returned by getOutputForEffect() is removed before we lock the
2198            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2199            // and we will exit safely
2200            io = AudioSystem::getOutputForEffect(&desc);
2201            ALOGV("createEffect got output %d", io);
2202        }
2203
2204        Mutex::Autolock _l(mLock);
2205
2206        // If output is not specified try to find a matching audio session ID in one of the
2207        // output threads.
2208        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2209        // because of code checking output when entering the function.
2210        // Note: io is never 0 when creating an effect on an input
2211        if (io == 0) {
2212            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2213                // output must be specified by AudioPolicyManager when using session
2214                // AUDIO_SESSION_OUTPUT_STAGE
2215                lStatus = BAD_VALUE;
2216                goto Exit;
2217            }
2218            // look for the thread where the specified audio session is present
2219            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2220                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2221                    io = mPlaybackThreads.keyAt(i);
2222                    break;
2223                }
2224            }
2225            if (io == 0) {
2226                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2227                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2228                        io = mRecordThreads.keyAt(i);
2229                        break;
2230                    }
2231                }
2232            }
2233            // If no output thread contains the requested session ID, default to
2234            // first output. The effect chain will be moved to the correct output
2235            // thread when a track with the same session ID is created
2236            if (io == 0 && mPlaybackThreads.size()) {
2237                io = mPlaybackThreads.keyAt(0);
2238            }
2239            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2240        }
2241        ThreadBase *thread = checkRecordThread_l(io);
2242        if (thread == NULL) {
2243            thread = checkPlaybackThread_l(io);
2244            if (thread == NULL) {
2245                ALOGE("createEffect() unknown output thread");
2246                lStatus = BAD_VALUE;
2247                goto Exit;
2248            }
2249        }
2250
2251        sp<Client> client = registerPid_l(pid);
2252
2253        // create effect on selected output thread
2254        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2255                &desc, enabled, &lStatus);
2256        if (handle != 0 && id != NULL) {
2257            *id = handle->id();
2258        }
2259    }
2260
2261Exit:
2262    *status = lStatus;
2263    return handle;
2264}
2265
2266status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2267        audio_io_handle_t dstOutput)
2268{
2269    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2270            sessionId, srcOutput, dstOutput);
2271    Mutex::Autolock _l(mLock);
2272    if (srcOutput == dstOutput) {
2273        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2274        return NO_ERROR;
2275    }
2276    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2277    if (srcThread == NULL) {
2278        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2279        return BAD_VALUE;
2280    }
2281    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2282    if (dstThread == NULL) {
2283        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2284        return BAD_VALUE;
2285    }
2286
2287    Mutex::Autolock _dl(dstThread->mLock);
2288    Mutex::Autolock _sl(srcThread->mLock);
2289    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2290}
2291
2292// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2293status_t AudioFlinger::moveEffectChain_l(int sessionId,
2294                                   AudioFlinger::PlaybackThread *srcThread,
2295                                   AudioFlinger::PlaybackThread *dstThread,
2296                                   bool reRegister)
2297{
2298    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2299            sessionId, srcThread, dstThread);
2300
2301    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2302    if (chain == 0) {
2303        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2304                sessionId, srcThread);
2305        return INVALID_OPERATION;
2306    }
2307
2308    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2309    // so that a new chain is created with correct parameters when first effect is added. This is
2310    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2311    // removed.
2312    srcThread->removeEffectChain_l(chain);
2313
2314    // transfer all effects one by one so that new effect chain is created on new thread with
2315    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2316    sp<EffectChain> dstChain;
2317    uint32_t strategy = 0; // prevent compiler warning
2318    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2319    Vector< sp<EffectModule> > removed;
2320    status_t status = NO_ERROR;
2321    while (effect != 0) {
2322        srcThread->removeEffect_l(effect);
2323        removed.add(effect);
2324        status = dstThread->addEffect_l(effect);
2325        if (status != NO_ERROR) {
2326            break;
2327        }
2328        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2329        if (effect->state() == EffectModule::ACTIVE ||
2330                effect->state() == EffectModule::STOPPING) {
2331            effect->start();
2332        }
2333        // if the move request is not received from audio policy manager, the effect must be
2334        // re-registered with the new strategy and output
2335        if (dstChain == 0) {
2336            dstChain = effect->chain().promote();
2337            if (dstChain == 0) {
2338                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2339                status = NO_INIT;
2340                break;
2341            }
2342            strategy = dstChain->strategy();
2343        }
2344        if (reRegister) {
2345            AudioSystem::unregisterEffect(effect->id());
2346            AudioSystem::registerEffect(&effect->desc(),
2347                                        dstThread->id(),
2348                                        strategy,
2349                                        sessionId,
2350                                        effect->id());
2351            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2352        }
2353        effect = chain->getEffectFromId_l(0);
2354    }
2355
2356    if (status != NO_ERROR) {
2357        for (size_t i = 0; i < removed.size(); i++) {
2358            srcThread->addEffect_l(removed[i]);
2359            if (dstChain != 0 && reRegister) {
2360                AudioSystem::unregisterEffect(removed[i]->id());
2361                AudioSystem::registerEffect(&removed[i]->desc(),
2362                                            srcThread->id(),
2363                                            strategy,
2364                                            sessionId,
2365                                            removed[i]->id());
2366                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2367            }
2368        }
2369    }
2370
2371    return status;
2372}
2373
2374bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2375{
2376    if (mGlobalEffectEnableTime != 0 &&
2377            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2378        return true;
2379    }
2380
2381    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2382        sp<EffectChain> ec =
2383                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2384        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2385            return true;
2386        }
2387    }
2388    return false;
2389}
2390
2391void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2392{
2393    Mutex::Autolock _l(mLock);
2394
2395    mGlobalEffectEnableTime = systemTime();
2396
2397    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2398        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2399        if (t->mType == ThreadBase::OFFLOAD) {
2400            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2401        }
2402    }
2403
2404}
2405
2406struct Entry {
2407#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2408    char mName[MAX_NAME];
2409};
2410
2411int comparEntry(const void *p1, const void *p2)
2412{
2413    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2414}
2415
2416#ifdef TEE_SINK
2417void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2418{
2419    NBAIO_Source *teeSource = source.get();
2420    if (teeSource != NULL) {
2421        // .wav rotation
2422        // There is a benign race condition if 2 threads call this simultaneously.
2423        // They would both traverse the directory, but the result would simply be
2424        // failures at unlink() which are ignored.  It's also unlikely since
2425        // normally dumpsys is only done by bugreport or from the command line.
2426        char teePath[32+256];
2427        strcpy(teePath, "/data/misc/media");
2428        size_t teePathLen = strlen(teePath);
2429        DIR *dir = opendir(teePath);
2430        teePath[teePathLen++] = '/';
2431        if (dir != NULL) {
2432#define MAX_SORT 20 // number of entries to sort
2433#define MAX_KEEP 10 // number of entries to keep
2434            struct Entry entries[MAX_SORT];
2435            size_t entryCount = 0;
2436            while (entryCount < MAX_SORT) {
2437                struct dirent de;
2438                struct dirent *result = NULL;
2439                int rc = readdir_r(dir, &de, &result);
2440                if (rc != 0) {
2441                    ALOGW("readdir_r failed %d", rc);
2442                    break;
2443                }
2444                if (result == NULL) {
2445                    break;
2446                }
2447                if (result != &de) {
2448                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2449                    break;
2450                }
2451                // ignore non .wav file entries
2452                size_t nameLen = strlen(de.d_name);
2453                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2454                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2455                    continue;
2456                }
2457                strcpy(entries[entryCount++].mName, de.d_name);
2458            }
2459            (void) closedir(dir);
2460            if (entryCount > MAX_KEEP) {
2461                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2462                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2463                    strcpy(&teePath[teePathLen], entries[i].mName);
2464                    (void) unlink(teePath);
2465                }
2466            }
2467        } else {
2468            if (fd >= 0) {
2469                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2470            }
2471        }
2472        char teeTime[16];
2473        struct timeval tv;
2474        gettimeofday(&tv, NULL);
2475        struct tm tm;
2476        localtime_r(&tv.tv_sec, &tm);
2477        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2478        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2479        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2480        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2481        if (teeFd >= 0) {
2482            char wavHeader[44];
2483            memcpy(wavHeader,
2484                "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",
2485                sizeof(wavHeader));
2486            NBAIO_Format format = teeSource->format();
2487            unsigned channelCount = Format_channelCount(format);
2488            ALOG_ASSERT(channelCount <= FCC_2);
2489            uint32_t sampleRate = Format_sampleRate(format);
2490            wavHeader[22] = channelCount;       // number of channels
2491            wavHeader[24] = sampleRate;         // sample rate
2492            wavHeader[25] = sampleRate >> 8;
2493            wavHeader[32] = channelCount * 2;   // block alignment
2494            write(teeFd, wavHeader, sizeof(wavHeader));
2495            size_t total = 0;
2496            bool firstRead = true;
2497            for (;;) {
2498#define TEE_SINK_READ 1024
2499                short buffer[TEE_SINK_READ * FCC_2];
2500                size_t count = TEE_SINK_READ;
2501                ssize_t actual = teeSource->read(buffer, count,
2502                        AudioBufferProvider::kInvalidPTS);
2503                bool wasFirstRead = firstRead;
2504                firstRead = false;
2505                if (actual <= 0) {
2506                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2507                        continue;
2508                    }
2509                    break;
2510                }
2511                ALOG_ASSERT(actual <= (ssize_t)count);
2512                write(teeFd, buffer, actual * channelCount * sizeof(short));
2513                total += actual;
2514            }
2515            lseek(teeFd, (off_t) 4, SEEK_SET);
2516            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2517            write(teeFd, &temp, sizeof(temp));
2518            lseek(teeFd, (off_t) 40, SEEK_SET);
2519            temp =  total * channelCount * sizeof(short);
2520            write(teeFd, &temp, sizeof(temp));
2521            close(teeFd);
2522            if (fd >= 0) {
2523                fdprintf(fd, "tee copied to %s\n", teePath);
2524            }
2525        } else {
2526            if (fd >= 0) {
2527                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2528            }
2529        }
2530    }
2531}
2532#endif
2533
2534// ----------------------------------------------------------------------------
2535
2536status_t AudioFlinger::onTransact(
2537        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2538{
2539    return BnAudioFlinger::onTransact(code, data, reply, flags);
2540}
2541
2542}; // namespace android
2543