AudioFlinger.cpp revision 1c594b637df26499ce1dae2db34f2b3290efd838 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <dirent.h>
24#include <math.h>
25#include <signal.h>
26#include <sys/time.h>
27#include <sys/resource.h>
28
29#include <binder/IPCThreadState.h>
30#include <binder/IServiceManager.h>
31#include <utils/Log.h>
32#include <utils/Trace.h>
33#include <binder/Parcel.h>
34#include <utils/String16.h>
35#include <utils/threads.h>
36#include <utils/Atomic.h>
37
38#include <cutils/bitops.h>
39#include <cutils/properties.h>
40
41#include <system/audio.h>
42#include <hardware/audio.h>
43
44#include "AudioMixer.h"
45#include "AudioFlinger.h"
46#include "ServiceUtilities.h"
47
48#include <media/EffectsFactoryApi.h>
49#include <audio_effects/effect_visualizer.h>
50#include <audio_effects/effect_ns.h>
51#include <audio_effects/effect_aec.h>
52
53#include <audio_utils/primitives.h>
54
55#include <powermanager/PowerManager.h>
56
57#include <common_time/cc_helper.h>
58
59#include <media/IMediaLogService.h>
60
61#include <media/nbaio/Pipe.h>
62#include <media/nbaio/PipeReader.h>
63#include <media/AudioParameter.h>
64#include <private/android_filesystem_config.h>
65
66// ----------------------------------------------------------------------------
67
68// Note: the following macro is used for extremely verbose logging message.  In
69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
70// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
71// are so verbose that we want to suppress them even when we have ALOG_ASSERT
72// turned on.  Do not uncomment the #def below unless you really know what you
73// are doing and want to see all of the extremely verbose messages.
74//#define VERY_VERY_VERBOSE_LOGGING
75#ifdef VERY_VERY_VERBOSE_LOGGING
76#define ALOGVV ALOGV
77#else
78#define ALOGVV(a...) do { } while(0)
79#endif
80
81namespace android {
82
83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
84static const char kHardwareLockedString[] = "Hardware lock is taken\n";
85
86
87nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
88
89uint32_t AudioFlinger::mScreenState;
90
91#ifdef TEE_SINK
92bool AudioFlinger::mTeeSinkInputEnabled = false;
93bool AudioFlinger::mTeeSinkOutputEnabled = false;
94bool AudioFlinger::mTeeSinkTrackEnabled = false;
95
96size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
97size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
98size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
99#endif
100
101// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
102// we define a minimum time during which a global effect is considered enabled.
103static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
104
105// ----------------------------------------------------------------------------
106
107static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
108{
109    const hw_module_t *mod;
110    int rc;
111
112    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
113    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
114                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
115    if (rc) {
116        goto out;
117    }
118    rc = audio_hw_device_open(mod, dev);
119    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
120                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
121    if (rc) {
122        goto out;
123    }
124    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
125        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
126        rc = BAD_VALUE;
127        goto out;
128    }
129    return 0;
130
131out:
132    *dev = NULL;
133    return rc;
134}
135
136// ----------------------------------------------------------------------------
137
138AudioFlinger::AudioFlinger()
139    : BnAudioFlinger(),
140      mPrimaryHardwareDev(NULL),
141      mHardwareStatus(AUDIO_HW_IDLE),
142      mMasterVolume(1.0f),
143      mMasterMute(false),
144      mNextUniqueId(1),
145      mMode(AUDIO_MODE_INVALID),
146      mBtNrecIsOff(false),
147      mIsLowRamDevice(true),
148      mIsDeviceTypeKnown(false),
149      mGlobalEffectEnableTime(0)
150{
151    getpid_cached = getpid();
152    char value[PROPERTY_VALUE_MAX];
153    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
154    if (doLog) {
155        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
156    }
157#ifdef TEE_SINK
158    (void) property_get("ro.debuggable", value, "0");
159    int debuggable = atoi(value);
160    int teeEnabled = 0;
161    if (debuggable) {
162        (void) property_get("af.tee", value, "0");
163        teeEnabled = atoi(value);
164    }
165    if (teeEnabled & 1)
166        mTeeSinkInputEnabled = true;
167    if (teeEnabled & 2)
168        mTeeSinkOutputEnabled = true;
169    if (teeEnabled & 4)
170        mTeeSinkTrackEnabled = true;
171#endif
172}
173
174void AudioFlinger::onFirstRef()
175{
176    int rc = 0;
177
178    Mutex::Autolock _l(mLock);
179
180    /* TODO: move all this work into an Init() function */
181    char val_str[PROPERTY_VALUE_MAX] = { 0 };
182    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
183        uint32_t int_val;
184        if (1 == sscanf(val_str, "%u", &int_val)) {
185            mStandbyTimeInNsecs = milliseconds(int_val);
186            ALOGI("Using %u mSec as standby time.", int_val);
187        } else {
188            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
189            ALOGI("Using default %u mSec as standby time.",
190                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
191        }
192    }
193
194    mMode = AUDIO_MODE_NORMAL;
195}
196
197AudioFlinger::~AudioFlinger()
198{
199    while (!mRecordThreads.isEmpty()) {
200        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
201        closeInput_nonvirtual(mRecordThreads.keyAt(0));
202    }
203    while (!mPlaybackThreads.isEmpty()) {
204        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
205        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
206    }
207
208    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
209        // no mHardwareLock needed, as there are no other references to this
210        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
211        delete mAudioHwDevs.valueAt(i);
212    }
213}
214
215static const char * const audio_interfaces[] = {
216    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
217    AUDIO_HARDWARE_MODULE_ID_A2DP,
218    AUDIO_HARDWARE_MODULE_ID_USB,
219};
220#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
221
222AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
223        audio_module_handle_t module,
224        audio_devices_t devices)
225{
226    // if module is 0, the request comes from an old policy manager and we should load
227    // well known modules
228    if (module == 0) {
229        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
230        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
231            loadHwModule_l(audio_interfaces[i]);
232        }
233        // then try to find a module supporting the requested device.
234        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
235            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
236            audio_hw_device_t *dev = audioHwDevice->hwDevice();
237            if ((dev->get_supported_devices != NULL) &&
238                    (dev->get_supported_devices(dev) & devices) == devices)
239                return audioHwDevice;
240        }
241    } else {
242        // check a match for the requested module handle
243        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
244        if (audioHwDevice != NULL) {
245            return audioHwDevice;
246        }
247    }
248
249    return NULL;
250}
251
252void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
253{
254    const size_t SIZE = 256;
255    char buffer[SIZE];
256    String8 result;
257
258    result.append("Clients:\n");
259    for (size_t i = 0; i < mClients.size(); ++i) {
260        sp<Client> client = mClients.valueAt(i).promote();
261        if (client != 0) {
262            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
263            result.append(buffer);
264        }
265    }
266
267    result.append("Notification Clients:\n");
268    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
269        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
270        result.append(buffer);
271    }
272
273    result.append("Global session refs:\n");
274    result.append(" session pid count\n");
275    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
276        AudioSessionRef *r = mAudioSessionRefs[i];
277        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
278        result.append(buffer);
279    }
280    write(fd, result.string(), result.size());
281}
282
283
284void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
285{
286    const size_t SIZE = 256;
287    char buffer[SIZE];
288    String8 result;
289    hardware_call_state hardwareStatus = mHardwareStatus;
290
291    snprintf(buffer, SIZE, "Hardware status: %d\n"
292                           "Standby Time mSec: %u\n",
293                            hardwareStatus,
294                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
295    result.append(buffer);
296    write(fd, result.string(), result.size());
297}
298
299void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
300{
301    const size_t SIZE = 256;
302    char buffer[SIZE];
303    String8 result;
304    snprintf(buffer, SIZE, "Permission Denial: "
305            "can't dump AudioFlinger from pid=%d, uid=%d\n",
306            IPCThreadState::self()->getCallingPid(),
307            IPCThreadState::self()->getCallingUid());
308    result.append(buffer);
309    write(fd, result.string(), result.size());
310}
311
312bool AudioFlinger::dumpTryLock(Mutex& mutex)
313{
314    bool locked = false;
315    for (int i = 0; i < kDumpLockRetries; ++i) {
316        if (mutex.tryLock() == NO_ERROR) {
317            locked = true;
318            break;
319        }
320        usleep(kDumpLockSleepUs);
321    }
322    return locked;
323}
324
325status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
326{
327    if (!dumpAllowed()) {
328        dumpPermissionDenial(fd, args);
329    } else {
330        // get state of hardware lock
331        bool hardwareLocked = dumpTryLock(mHardwareLock);
332        if (!hardwareLocked) {
333            String8 result(kHardwareLockedString);
334            write(fd, result.string(), result.size());
335        } else {
336            mHardwareLock.unlock();
337        }
338
339        bool locked = dumpTryLock(mLock);
340
341        // failed to lock - AudioFlinger is probably deadlocked
342        if (!locked) {
343            String8 result(kDeadlockedString);
344            write(fd, result.string(), result.size());
345        }
346
347        dumpClients(fd, args);
348        dumpInternals(fd, args);
349
350        // dump playback threads
351        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
352            mPlaybackThreads.valueAt(i)->dump(fd, args);
353        }
354
355        // dump record threads
356        for (size_t i = 0; i < mRecordThreads.size(); i++) {
357            mRecordThreads.valueAt(i)->dump(fd, args);
358        }
359
360        // dump all hardware devs
361        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
362            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
363            dev->dump(dev, fd);
364        }
365
366#ifdef TEE_SINK
367        // dump the serially shared record tee sink
368        if (mRecordTeeSource != 0) {
369            dumpTee(fd, mRecordTeeSource);
370        }
371#endif
372
373        if (locked) {
374            mLock.unlock();
375        }
376
377        // append a copy of media.log here by forwarding fd to it, but don't attempt
378        // to lookup the service if it's not running, as it will block for a second
379        if (mLogMemoryDealer != 0) {
380            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
381            if (binder != 0) {
382                fdprintf(fd, "\nmedia.log:\n");
383                Vector<String16> args;
384                binder->dump(fd, args);
385            }
386        }
387    }
388    return NO_ERROR;
389}
390
391sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
392{
393    // If pid is already in the mClients wp<> map, then use that entry
394    // (for which promote() is always != 0), otherwise create a new entry and Client.
395    sp<Client> client = mClients.valueFor(pid).promote();
396    if (client == 0) {
397        client = new Client(this, pid);
398        mClients.add(pid, client);
399    }
400
401    return client;
402}
403
404sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
405{
406    if (mLogMemoryDealer == 0) {
407        return new NBLog::Writer();
408    }
409    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
410    sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
411    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
412    if (binder != 0) {
413        interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
414    }
415    return writer;
416}
417
418void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
419{
420    if (writer == 0) {
421        return;
422    }
423    sp<IMemory> iMemory(writer->getIMemory());
424    if (iMemory == 0) {
425        return;
426    }
427    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
428    if (binder != 0) {
429        interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
430        // Now the media.log remote reference to IMemory is gone.
431        // When our last local reference to IMemory also drops to zero,
432        // the IMemory destructor will deallocate the region from mMemoryDealer.
433    }
434}
435
436// IAudioFlinger interface
437
438
439sp<IAudioTrack> AudioFlinger::createTrack(
440        audio_stream_type_t streamType,
441        uint32_t sampleRate,
442        audio_format_t format,
443        audio_channel_mask_t channelMask,
444        size_t frameCount,
445        IAudioFlinger::track_flags_t *flags,
446        const sp<IMemory>& sharedBuffer,
447        audio_io_handle_t output,
448        pid_t tid,
449        int *sessionId,
450        String8& name,
451        int clientUid,
452        status_t *status)
453{
454    sp<PlaybackThread::Track> track;
455    sp<TrackHandle> trackHandle;
456    sp<Client> client;
457    status_t lStatus;
458    int lSessionId;
459
460    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
461    // but if someone uses binder directly they could bypass that and cause us to crash
462    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
463        ALOGE("createTrack() invalid stream type %d", streamType);
464        lStatus = BAD_VALUE;
465        goto Exit;
466    }
467
468    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
469    // and we don't yet support 8.24 or 32-bit PCM
470    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
471        ALOGE("createTrack() invalid format %d", format);
472        lStatus = BAD_VALUE;
473        goto Exit;
474    }
475
476    {
477        Mutex::Autolock _l(mLock);
478        PlaybackThread *thread = checkPlaybackThread_l(output);
479        PlaybackThread *effectThread = NULL;
480        if (thread == NULL) {
481            ALOGE("no playback thread found for output handle %d", output);
482            lStatus = BAD_VALUE;
483            goto Exit;
484        }
485
486        pid_t pid = IPCThreadState::self()->getCallingPid();
487
488        client = registerPid_l(pid);
489
490        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
491        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
492            // check if an effect chain with the same session ID is present on another
493            // output thread and move it here.
494            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
495                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
496                if (mPlaybackThreads.keyAt(i) != output) {
497                    uint32_t sessions = t->hasAudioSession(*sessionId);
498                    if (sessions & PlaybackThread::EFFECT_SESSION) {
499                        effectThread = t.get();
500                        break;
501                    }
502                }
503            }
504            lSessionId = *sessionId;
505        } else {
506            // if no audio session id is provided, create one here
507            lSessionId = nextUniqueId();
508            if (sessionId != NULL) {
509                *sessionId = lSessionId;
510            }
511        }
512        ALOGV("createTrack() lSessionId: %d", lSessionId);
513
514        track = thread->createTrack_l(client, streamType, sampleRate, format,
515                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
516
517        // move effect chain to this output thread if an effect on same session was waiting
518        // for a track to be created
519        if (lStatus == NO_ERROR && effectThread != NULL) {
520            Mutex::Autolock _dl(thread->mLock);
521            Mutex::Autolock _sl(effectThread->mLock);
522            moveEffectChain_l(lSessionId, effectThread, thread, true);
523        }
524
525        // Look for sync events awaiting for a session to be used.
526        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
527            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
528                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
529                    if (lStatus == NO_ERROR) {
530                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
531                    } else {
532                        mPendingSyncEvents[i]->cancel();
533                    }
534                    mPendingSyncEvents.removeAt(i);
535                    i--;
536                }
537            }
538        }
539    }
540    if (lStatus == NO_ERROR) {
541        // s for server's pid, n for normal mixer name, f for fast index
542        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
543                track->fastIndex());
544        trackHandle = new TrackHandle(track);
545    } else {
546        // remove local strong reference to Client before deleting the Track so that the Client
547        // destructor is called by the TrackBase destructor with mLock held
548        client.clear();
549        track.clear();
550    }
551
552Exit:
553    if (status != NULL) {
554        *status = lStatus;
555    }
556    return trackHandle;
557}
558
559uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
560{
561    Mutex::Autolock _l(mLock);
562    PlaybackThread *thread = checkPlaybackThread_l(output);
563    if (thread == NULL) {
564        ALOGW("sampleRate() unknown thread %d", output);
565        return 0;
566    }
567    return thread->sampleRate();
568}
569
570int AudioFlinger::channelCount(audio_io_handle_t output) const
571{
572    Mutex::Autolock _l(mLock);
573    PlaybackThread *thread = checkPlaybackThread_l(output);
574    if (thread == NULL) {
575        ALOGW("channelCount() unknown thread %d", output);
576        return 0;
577    }
578    return thread->channelCount();
579}
580
581audio_format_t AudioFlinger::format(audio_io_handle_t output) const
582{
583    Mutex::Autolock _l(mLock);
584    PlaybackThread *thread = checkPlaybackThread_l(output);
585    if (thread == NULL) {
586        ALOGW("format() unknown thread %d", output);
587        return AUDIO_FORMAT_INVALID;
588    }
589    return thread->format();
590}
591
592size_t AudioFlinger::frameCount(audio_io_handle_t output) const
593{
594    Mutex::Autolock _l(mLock);
595    PlaybackThread *thread = checkPlaybackThread_l(output);
596    if (thread == NULL) {
597        ALOGW("frameCount() unknown thread %d", output);
598        return 0;
599    }
600    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
601    //       should examine all callers and fix them to handle smaller counts
602    return thread->frameCount();
603}
604
605uint32_t AudioFlinger::latency(audio_io_handle_t output) const
606{
607    Mutex::Autolock _l(mLock);
608    PlaybackThread *thread = checkPlaybackThread_l(output);
609    if (thread == NULL) {
610        ALOGW("latency(): no playback thread found for output handle %d", output);
611        return 0;
612    }
613    return thread->latency();
614}
615
616status_t AudioFlinger::setMasterVolume(float value)
617{
618    status_t ret = initCheck();
619    if (ret != NO_ERROR) {
620        return ret;
621    }
622
623    // check calling permissions
624    if (!settingsAllowed()) {
625        return PERMISSION_DENIED;
626    }
627
628    Mutex::Autolock _l(mLock);
629    mMasterVolume = value;
630
631    // Set master volume in the HALs which support it.
632    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
633        AutoMutex lock(mHardwareLock);
634        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
635
636        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
637        if (dev->canSetMasterVolume()) {
638            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
639        }
640        mHardwareStatus = AUDIO_HW_IDLE;
641    }
642
643    // Now set the master volume in each playback thread.  Playback threads
644    // assigned to HALs which do not have master volume support will apply
645    // master volume during the mix operation.  Threads with HALs which do
646    // support master volume will simply ignore the setting.
647    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
648        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
649
650    return NO_ERROR;
651}
652
653status_t AudioFlinger::setMode(audio_mode_t mode)
654{
655    status_t ret = initCheck();
656    if (ret != NO_ERROR) {
657        return ret;
658    }
659
660    // check calling permissions
661    if (!settingsAllowed()) {
662        return PERMISSION_DENIED;
663    }
664    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
665        ALOGW("Illegal value: setMode(%d)", mode);
666        return BAD_VALUE;
667    }
668
669    { // scope for the lock
670        AutoMutex lock(mHardwareLock);
671        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
672        mHardwareStatus = AUDIO_HW_SET_MODE;
673        ret = dev->set_mode(dev, mode);
674        mHardwareStatus = AUDIO_HW_IDLE;
675    }
676
677    if (NO_ERROR == ret) {
678        Mutex::Autolock _l(mLock);
679        mMode = mode;
680        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
681            mPlaybackThreads.valueAt(i)->setMode(mode);
682    }
683
684    return ret;
685}
686
687status_t AudioFlinger::setMicMute(bool state)
688{
689    status_t ret = initCheck();
690    if (ret != NO_ERROR) {
691        return ret;
692    }
693
694    // check calling permissions
695    if (!settingsAllowed()) {
696        return PERMISSION_DENIED;
697    }
698
699    AutoMutex lock(mHardwareLock);
700    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
701    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
702    ret = dev->set_mic_mute(dev, state);
703    mHardwareStatus = AUDIO_HW_IDLE;
704    return ret;
705}
706
707bool AudioFlinger::getMicMute() const
708{
709    status_t ret = initCheck();
710    if (ret != NO_ERROR) {
711        return false;
712    }
713
714    bool state = AUDIO_MODE_INVALID;
715    AutoMutex lock(mHardwareLock);
716    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
717    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
718    dev->get_mic_mute(dev, &state);
719    mHardwareStatus = AUDIO_HW_IDLE;
720    return state;
721}
722
723status_t AudioFlinger::setMasterMute(bool muted)
724{
725    status_t ret = initCheck();
726    if (ret != NO_ERROR) {
727        return ret;
728    }
729
730    // check calling permissions
731    if (!settingsAllowed()) {
732        return PERMISSION_DENIED;
733    }
734
735    Mutex::Autolock _l(mLock);
736    mMasterMute = muted;
737
738    // Set master mute in the HALs which support it.
739    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
740        AutoMutex lock(mHardwareLock);
741        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
742
743        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
744        if (dev->canSetMasterMute()) {
745            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
746        }
747        mHardwareStatus = AUDIO_HW_IDLE;
748    }
749
750    // Now set the master mute in each playback thread.  Playback threads
751    // assigned to HALs which do not have master mute support will apply master
752    // mute during the mix operation.  Threads with HALs which do support master
753    // mute will simply ignore the setting.
754    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
755        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
756
757    return NO_ERROR;
758}
759
760float AudioFlinger::masterVolume() const
761{
762    Mutex::Autolock _l(mLock);
763    return masterVolume_l();
764}
765
766bool AudioFlinger::masterMute() const
767{
768    Mutex::Autolock _l(mLock);
769    return masterMute_l();
770}
771
772float AudioFlinger::masterVolume_l() const
773{
774    return mMasterVolume;
775}
776
777bool AudioFlinger::masterMute_l() const
778{
779    return mMasterMute;
780}
781
782status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
783        audio_io_handle_t output)
784{
785    // check calling permissions
786    if (!settingsAllowed()) {
787        return PERMISSION_DENIED;
788    }
789
790    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
791        ALOGE("setStreamVolume() invalid stream %d", stream);
792        return BAD_VALUE;
793    }
794
795    AutoMutex lock(mLock);
796    PlaybackThread *thread = NULL;
797    if (output) {
798        thread = checkPlaybackThread_l(output);
799        if (thread == NULL) {
800            return BAD_VALUE;
801        }
802    }
803
804    mStreamTypes[stream].volume = value;
805
806    if (thread == NULL) {
807        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
808            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
809        }
810    } else {
811        thread->setStreamVolume(stream, value);
812    }
813
814    return NO_ERROR;
815}
816
817status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
818{
819    // check calling permissions
820    if (!settingsAllowed()) {
821        return PERMISSION_DENIED;
822    }
823
824    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
825        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
826        ALOGE("setStreamMute() invalid stream %d", stream);
827        return BAD_VALUE;
828    }
829
830    AutoMutex lock(mLock);
831    mStreamTypes[stream].mute = muted;
832    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
833        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
834
835    return NO_ERROR;
836}
837
838float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
839{
840    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
841        return 0.0f;
842    }
843
844    AutoMutex lock(mLock);
845    float volume;
846    if (output) {
847        PlaybackThread *thread = checkPlaybackThread_l(output);
848        if (thread == NULL) {
849            return 0.0f;
850        }
851        volume = thread->streamVolume(stream);
852    } else {
853        volume = streamVolume_l(stream);
854    }
855
856    return volume;
857}
858
859bool AudioFlinger::streamMute(audio_stream_type_t stream) const
860{
861    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
862        return true;
863    }
864
865    AutoMutex lock(mLock);
866    return streamMute_l(stream);
867}
868
869status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
870{
871    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
872            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
873
874    // check calling permissions
875    if (!settingsAllowed()) {
876        return PERMISSION_DENIED;
877    }
878
879    // ioHandle == 0 means the parameters are global to the audio hardware interface
880    if (ioHandle == 0) {
881        Mutex::Autolock _l(mLock);
882        status_t final_result = NO_ERROR;
883        {
884            AutoMutex lock(mHardwareLock);
885            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
886            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
887                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
888                status_t result = dev->set_parameters(dev, keyValuePairs.string());
889                final_result = result ?: final_result;
890            }
891            mHardwareStatus = AUDIO_HW_IDLE;
892        }
893        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
894        AudioParameter param = AudioParameter(keyValuePairs);
895        String8 value;
896        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
897            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
898            if (mBtNrecIsOff != btNrecIsOff) {
899                for (size_t i = 0; i < mRecordThreads.size(); i++) {
900                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
901                    audio_devices_t device = thread->inDevice();
902                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
903                    // collect all of the thread's session IDs
904                    KeyedVector<int, bool> ids = thread->sessionIds();
905                    // suspend effects associated with those session IDs
906                    for (size_t j = 0; j < ids.size(); ++j) {
907                        int sessionId = ids.keyAt(j);
908                        thread->setEffectSuspended(FX_IID_AEC,
909                                                   suspend,
910                                                   sessionId);
911                        thread->setEffectSuspended(FX_IID_NS,
912                                                   suspend,
913                                                   sessionId);
914                    }
915                }
916                mBtNrecIsOff = btNrecIsOff;
917            }
918        }
919        String8 screenState;
920        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
921            bool isOff = screenState == "off";
922            if (isOff != (AudioFlinger::mScreenState & 1)) {
923                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
924            }
925        }
926        return final_result;
927    }
928
929    // hold a strong ref on thread in case closeOutput() or closeInput() is called
930    // and the thread is exited once the lock is released
931    sp<ThreadBase> thread;
932    {
933        Mutex::Autolock _l(mLock);
934        thread = checkPlaybackThread_l(ioHandle);
935        if (thread == 0) {
936            thread = checkRecordThread_l(ioHandle);
937        } else if (thread == primaryPlaybackThread_l()) {
938            // indicate output device change to all input threads for pre processing
939            AudioParameter param = AudioParameter(keyValuePairs);
940            int value;
941            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
942                    (value != 0)) {
943                for (size_t i = 0; i < mRecordThreads.size(); i++) {
944                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
945                }
946            }
947        }
948    }
949    if (thread != 0) {
950        return thread->setParameters(keyValuePairs);
951    }
952    return BAD_VALUE;
953}
954
955String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
956{
957    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
958            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
959
960    Mutex::Autolock _l(mLock);
961
962    if (ioHandle == 0) {
963        String8 out_s8;
964
965        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
966            char *s;
967            {
968            AutoMutex lock(mHardwareLock);
969            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
970            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
971            s = dev->get_parameters(dev, keys.string());
972            mHardwareStatus = AUDIO_HW_IDLE;
973            }
974            out_s8 += String8(s ? s : "");
975            free(s);
976        }
977        return out_s8;
978    }
979
980    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
981    if (playbackThread != NULL) {
982        return playbackThread->getParameters(keys);
983    }
984    RecordThread *recordThread = checkRecordThread_l(ioHandle);
985    if (recordThread != NULL) {
986        return recordThread->getParameters(keys);
987    }
988    return String8("");
989}
990
991size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
992        audio_channel_mask_t channelMask) const
993{
994    status_t ret = initCheck();
995    if (ret != NO_ERROR) {
996        return 0;
997    }
998
999    AutoMutex lock(mHardwareLock);
1000    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1001    struct audio_config config;
1002    memset(&config, 0, sizeof(config));
1003    config.sample_rate = sampleRate;
1004    config.channel_mask = channelMask;
1005    config.format = format;
1006
1007    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1008    size_t size = dev->get_input_buffer_size(dev, &config);
1009    mHardwareStatus = AUDIO_HW_IDLE;
1010    return size;
1011}
1012
1013unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1014{
1015    Mutex::Autolock _l(mLock);
1016
1017    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1018    if (recordThread != NULL) {
1019        return recordThread->getInputFramesLost();
1020    }
1021    return 0;
1022}
1023
1024status_t AudioFlinger::setVoiceVolume(float value)
1025{
1026    status_t ret = initCheck();
1027    if (ret != NO_ERROR) {
1028        return ret;
1029    }
1030
1031    // check calling permissions
1032    if (!settingsAllowed()) {
1033        return PERMISSION_DENIED;
1034    }
1035
1036    AutoMutex lock(mHardwareLock);
1037    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1038    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1039    ret = dev->set_voice_volume(dev, value);
1040    mHardwareStatus = AUDIO_HW_IDLE;
1041
1042    return ret;
1043}
1044
1045status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1046        audio_io_handle_t output) const
1047{
1048    status_t status;
1049
1050    Mutex::Autolock _l(mLock);
1051
1052    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1053    if (playbackThread != NULL) {
1054        return playbackThread->getRenderPosition(halFrames, dspFrames);
1055    }
1056
1057    return BAD_VALUE;
1058}
1059
1060void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1061{
1062
1063    Mutex::Autolock _l(mLock);
1064
1065    pid_t pid = IPCThreadState::self()->getCallingPid();
1066    if (mNotificationClients.indexOfKey(pid) < 0) {
1067        sp<NotificationClient> notificationClient = new NotificationClient(this,
1068                                                                            client,
1069                                                                            pid);
1070        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1071
1072        mNotificationClients.add(pid, notificationClient);
1073
1074        sp<IBinder> binder = client->asBinder();
1075        binder->linkToDeath(notificationClient);
1076
1077        // the config change is always sent from playback or record threads to avoid deadlock
1078        // with AudioSystem::gLock
1079        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1080            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1081        }
1082
1083        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1084            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1085        }
1086    }
1087}
1088
1089void AudioFlinger::removeNotificationClient(pid_t pid)
1090{
1091    Mutex::Autolock _l(mLock);
1092
1093    mNotificationClients.removeItem(pid);
1094
1095    ALOGV("%d died, releasing its sessions", pid);
1096    size_t num = mAudioSessionRefs.size();
1097    bool removed = false;
1098    for (size_t i = 0; i< num; ) {
1099        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1100        ALOGV(" pid %d @ %d", ref->mPid, i);
1101        if (ref->mPid == pid) {
1102            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1103            mAudioSessionRefs.removeAt(i);
1104            delete ref;
1105            removed = true;
1106            num--;
1107        } else {
1108            i++;
1109        }
1110    }
1111    if (removed) {
1112        purgeStaleEffects_l();
1113    }
1114}
1115
1116// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1117void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1118{
1119    size_t size = mNotificationClients.size();
1120    for (size_t i = 0; i < size; i++) {
1121        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1122                                                                               param2);
1123    }
1124}
1125
1126// removeClient_l() must be called with AudioFlinger::mLock held
1127void AudioFlinger::removeClient_l(pid_t pid)
1128{
1129    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1130            IPCThreadState::self()->getCallingPid());
1131    mClients.removeItem(pid);
1132}
1133
1134// getEffectThread_l() must be called with AudioFlinger::mLock held
1135sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1136{
1137    sp<PlaybackThread> thread;
1138
1139    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1140        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1141            ALOG_ASSERT(thread == 0);
1142            thread = mPlaybackThreads.valueAt(i);
1143        }
1144    }
1145
1146    return thread;
1147}
1148
1149
1150
1151// ----------------------------------------------------------------------------
1152
1153AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1154    :   RefBase(),
1155        mAudioFlinger(audioFlinger),
1156        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1157        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1158        mPid(pid),
1159        mTimedTrackCount(0)
1160{
1161    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1162}
1163
1164// Client destructor must be called with AudioFlinger::mLock held
1165AudioFlinger::Client::~Client()
1166{
1167    mAudioFlinger->removeClient_l(mPid);
1168}
1169
1170sp<MemoryDealer> AudioFlinger::Client::heap() const
1171{
1172    return mMemoryDealer;
1173}
1174
1175// Reserve one of the limited slots for a timed audio track associated
1176// with this client
1177bool AudioFlinger::Client::reserveTimedTrack()
1178{
1179    const int kMaxTimedTracksPerClient = 4;
1180
1181    Mutex::Autolock _l(mTimedTrackLock);
1182
1183    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1184        ALOGW("can not create timed track - pid %d has exceeded the limit",
1185             mPid);
1186        return false;
1187    }
1188
1189    mTimedTrackCount++;
1190    return true;
1191}
1192
1193// Release a slot for a timed audio track
1194void AudioFlinger::Client::releaseTimedTrack()
1195{
1196    Mutex::Autolock _l(mTimedTrackLock);
1197    mTimedTrackCount--;
1198}
1199
1200// ----------------------------------------------------------------------------
1201
1202AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1203                                                     const sp<IAudioFlingerClient>& client,
1204                                                     pid_t pid)
1205    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1206{
1207}
1208
1209AudioFlinger::NotificationClient::~NotificationClient()
1210{
1211}
1212
1213void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1214{
1215    sp<NotificationClient> keep(this);
1216    mAudioFlinger->removeNotificationClient(mPid);
1217}
1218
1219
1220// ----------------------------------------------------------------------------
1221
1222static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1223    return audio_is_remote_submix_device(inDevice);
1224}
1225
1226sp<IAudioRecord> AudioFlinger::openRecord(
1227        audio_io_handle_t input,
1228        uint32_t sampleRate,
1229        audio_format_t format,
1230        audio_channel_mask_t channelMask,
1231        size_t frameCount,
1232        IAudioFlinger::track_flags_t *flags,
1233        pid_t tid,
1234        int *sessionId,
1235        status_t *status)
1236{
1237    sp<RecordThread::RecordTrack> recordTrack;
1238    sp<RecordHandle> recordHandle;
1239    sp<Client> client;
1240    status_t lStatus;
1241    RecordThread *thread;
1242    size_t inFrameCount;
1243    int lSessionId;
1244
1245    // check calling permissions
1246    if (!recordingAllowed()) {
1247        ALOGE("openRecord() permission denied: recording not allowed");
1248        lStatus = PERMISSION_DENIED;
1249        goto Exit;
1250    }
1251
1252    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1253        ALOGE("openRecord() invalid format %d", format);
1254        lStatus = BAD_VALUE;
1255        goto Exit;
1256    }
1257
1258    // add client to list
1259    { // scope for mLock
1260        Mutex::Autolock _l(mLock);
1261        thread = checkRecordThread_l(input);
1262        if (thread == NULL) {
1263            ALOGE("openRecord() checkRecordThread_l failed");
1264            lStatus = BAD_VALUE;
1265            goto Exit;
1266        }
1267
1268        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1269                && !captureAudioOutputAllowed()) {
1270            ALOGE("openRecord() permission denied: capture not allowed");
1271            lStatus = PERMISSION_DENIED;
1272            goto Exit;
1273        }
1274
1275        pid_t pid = IPCThreadState::self()->getCallingPid();
1276        client = registerPid_l(pid);
1277
1278        // If no audio session id is provided, create one here
1279        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1280            lSessionId = *sessionId;
1281        } else {
1282            lSessionId = nextUniqueId();
1283            if (sessionId != NULL) {
1284                *sessionId = lSessionId;
1285            }
1286        }
1287        // create new record track.
1288        // The record track uses one track in mHardwareMixerThread by convention.
1289        // TODO: the uid should be passed in as a parameter to openRecord
1290        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1291                                                  frameCount, lSessionId,
1292                                                  IPCThreadState::self()->getCallingUid(),
1293                                                  flags, tid, &lStatus);
1294        LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
1295    }
1296    if (lStatus != NO_ERROR) {
1297        // remove local strong reference to Client before deleting the RecordTrack so that the
1298        // Client destructor is called by the TrackBase destructor with mLock held
1299        client.clear();
1300        recordTrack.clear();
1301        goto Exit;
1302    }
1303
1304    // return to handle to client
1305    recordHandle = new RecordHandle(recordTrack);
1306    lStatus = NO_ERROR;
1307
1308Exit:
1309    if (status) {
1310        *status = lStatus;
1311    }
1312    return recordHandle;
1313}
1314
1315
1316
1317// ----------------------------------------------------------------------------
1318
1319audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1320{
1321    if (!settingsAllowed()) {
1322        return 0;
1323    }
1324    Mutex::Autolock _l(mLock);
1325    return loadHwModule_l(name);
1326}
1327
1328// loadHwModule_l() must be called with AudioFlinger::mLock held
1329audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1330{
1331    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1332        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1333            ALOGW("loadHwModule() module %s already loaded", name);
1334            return mAudioHwDevs.keyAt(i);
1335        }
1336    }
1337
1338    audio_hw_device_t *dev;
1339
1340    int rc = load_audio_interface(name, &dev);
1341    if (rc) {
1342        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1343        return 0;
1344    }
1345
1346    mHardwareStatus = AUDIO_HW_INIT;
1347    rc = dev->init_check(dev);
1348    mHardwareStatus = AUDIO_HW_IDLE;
1349    if (rc) {
1350        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1351        return 0;
1352    }
1353
1354    // Check and cache this HAL's level of support for master mute and master
1355    // volume.  If this is the first HAL opened, and it supports the get
1356    // methods, use the initial values provided by the HAL as the current
1357    // master mute and volume settings.
1358
1359    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1360    {  // scope for auto-lock pattern
1361        AutoMutex lock(mHardwareLock);
1362
1363        if (0 == mAudioHwDevs.size()) {
1364            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1365            if (NULL != dev->get_master_volume) {
1366                float mv;
1367                if (OK == dev->get_master_volume(dev, &mv)) {
1368                    mMasterVolume = mv;
1369                }
1370            }
1371
1372            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1373            if (NULL != dev->get_master_mute) {
1374                bool mm;
1375                if (OK == dev->get_master_mute(dev, &mm)) {
1376                    mMasterMute = mm;
1377                }
1378            }
1379        }
1380
1381        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1382        if ((NULL != dev->set_master_volume) &&
1383            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1384            flags = static_cast<AudioHwDevice::Flags>(flags |
1385                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1386        }
1387
1388        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1389        if ((NULL != dev->set_master_mute) &&
1390            (OK == dev->set_master_mute(dev, mMasterMute))) {
1391            flags = static_cast<AudioHwDevice::Flags>(flags |
1392                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1393        }
1394
1395        mHardwareStatus = AUDIO_HW_IDLE;
1396    }
1397
1398    audio_module_handle_t handle = nextUniqueId();
1399    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1400
1401    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1402          name, dev->common.module->name, dev->common.module->id, handle);
1403
1404    return handle;
1405
1406}
1407
1408// ----------------------------------------------------------------------------
1409
1410uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1411{
1412    Mutex::Autolock _l(mLock);
1413    PlaybackThread *thread = primaryPlaybackThread_l();
1414    return thread != NULL ? thread->sampleRate() : 0;
1415}
1416
1417size_t AudioFlinger::getPrimaryOutputFrameCount()
1418{
1419    Mutex::Autolock _l(mLock);
1420    PlaybackThread *thread = primaryPlaybackThread_l();
1421    return thread != NULL ? thread->frameCountHAL() : 0;
1422}
1423
1424// ----------------------------------------------------------------------------
1425
1426status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1427{
1428    uid_t uid = IPCThreadState::self()->getCallingUid();
1429    if (uid != AID_SYSTEM) {
1430        return PERMISSION_DENIED;
1431    }
1432    Mutex::Autolock _l(mLock);
1433    if (mIsDeviceTypeKnown) {
1434        return INVALID_OPERATION;
1435    }
1436    mIsLowRamDevice = isLowRamDevice;
1437    mIsDeviceTypeKnown = true;
1438    return NO_ERROR;
1439}
1440
1441// ----------------------------------------------------------------------------
1442
1443audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1444                                           audio_devices_t *pDevices,
1445                                           uint32_t *pSamplingRate,
1446                                           audio_format_t *pFormat,
1447                                           audio_channel_mask_t *pChannelMask,
1448                                           uint32_t *pLatencyMs,
1449                                           audio_output_flags_t flags,
1450                                           const audio_offload_info_t *offloadInfo)
1451{
1452    PlaybackThread *thread = NULL;
1453    struct audio_config config;
1454    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1455    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1456    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1457    if (offloadInfo) {
1458        config.offload_info = *offloadInfo;
1459    }
1460
1461    audio_stream_out_t *outStream = NULL;
1462    AudioHwDevice *outHwDev;
1463
1464    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1465              module,
1466              (pDevices != NULL) ? *pDevices : 0,
1467              config.sample_rate,
1468              config.format,
1469              config.channel_mask,
1470              flags);
1471    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1472          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version );
1473
1474    if (pDevices == NULL || *pDevices == 0) {
1475        return 0;
1476    }
1477
1478    Mutex::Autolock _l(mLock);
1479
1480    outHwDev = findSuitableHwDev_l(module, *pDevices);
1481    if (outHwDev == NULL)
1482        return 0;
1483
1484    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1485    audio_io_handle_t id = nextUniqueId();
1486
1487    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1488
1489    status_t status = hwDevHal->open_output_stream(hwDevHal,
1490                                          id,
1491                                          *pDevices,
1492                                          (audio_output_flags_t)flags,
1493                                          &config,
1494                                          &outStream);
1495
1496    mHardwareStatus = AUDIO_HW_IDLE;
1497    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1498            "Channels %x, status %d",
1499            outStream,
1500            config.sample_rate,
1501            config.format,
1502            config.channel_mask,
1503            status);
1504
1505    if (status == NO_ERROR && outStream != NULL) {
1506        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1507
1508        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1509            thread = new OffloadThread(this, output, id, *pDevices);
1510            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1511        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1512            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1513            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1514            thread = new DirectOutputThread(this, output, id, *pDevices);
1515            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1516        } else {
1517            thread = new MixerThread(this, output, id, *pDevices);
1518            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1519        }
1520        mPlaybackThreads.add(id, thread);
1521
1522        if (pSamplingRate != NULL) {
1523            *pSamplingRate = config.sample_rate;
1524        }
1525        if (pFormat != NULL) {
1526            *pFormat = config.format;
1527        }
1528        if (pChannelMask != NULL) {
1529            *pChannelMask = config.channel_mask;
1530        }
1531        if (pLatencyMs != NULL) {
1532            *pLatencyMs = thread->latency();
1533        }
1534
1535        // notify client processes of the new output creation
1536        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1537
1538        // the first primary output opened designates the primary hw device
1539        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1540            ALOGI("Using module %d has the primary audio interface", module);
1541            mPrimaryHardwareDev = outHwDev;
1542
1543            AutoMutex lock(mHardwareLock);
1544            mHardwareStatus = AUDIO_HW_SET_MODE;
1545            hwDevHal->set_mode(hwDevHal, mMode);
1546            mHardwareStatus = AUDIO_HW_IDLE;
1547        }
1548        return id;
1549    }
1550
1551    return 0;
1552}
1553
1554audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1555        audio_io_handle_t output2)
1556{
1557    Mutex::Autolock _l(mLock);
1558    MixerThread *thread1 = checkMixerThread_l(output1);
1559    MixerThread *thread2 = checkMixerThread_l(output2);
1560
1561    if (thread1 == NULL || thread2 == NULL) {
1562        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1563                output2);
1564        return 0;
1565    }
1566
1567    audio_io_handle_t id = nextUniqueId();
1568    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1569    thread->addOutputTrack(thread2);
1570    mPlaybackThreads.add(id, thread);
1571    // notify client processes of the new output creation
1572    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1573    return id;
1574}
1575
1576status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1577{
1578    return closeOutput_nonvirtual(output);
1579}
1580
1581status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1582{
1583    // keep strong reference on the playback thread so that
1584    // it is not destroyed while exit() is executed
1585    sp<PlaybackThread> thread;
1586    {
1587        Mutex::Autolock _l(mLock);
1588        thread = checkPlaybackThread_l(output);
1589        if (thread == NULL) {
1590            return BAD_VALUE;
1591        }
1592
1593        ALOGV("closeOutput() %d", output);
1594
1595        if (thread->type() == ThreadBase::MIXER) {
1596            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1597                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1598                    DuplicatingThread *dupThread =
1599                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1600                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1601
1602                }
1603            }
1604        }
1605
1606
1607        mPlaybackThreads.removeItem(output);
1608        // save all effects to the default thread
1609        if (mPlaybackThreads.size()) {
1610            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1611            if (dstThread != NULL) {
1612                // audioflinger lock is held here so the acquisition order of thread locks does not
1613                // matter
1614                Mutex::Autolock _dl(dstThread->mLock);
1615                Mutex::Autolock _sl(thread->mLock);
1616                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1617                for (size_t i = 0; i < effectChains.size(); i ++) {
1618                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1619                }
1620            }
1621        }
1622        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1623    }
1624    thread->exit();
1625    // The thread entity (active unit of execution) is no longer running here,
1626    // but the ThreadBase container still exists.
1627
1628    if (thread->type() != ThreadBase::DUPLICATING) {
1629        AudioStreamOut *out = thread->clearOutput();
1630        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1631        // from now on thread->mOutput is NULL
1632        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1633        delete out;
1634    }
1635    return NO_ERROR;
1636}
1637
1638status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1639{
1640    Mutex::Autolock _l(mLock);
1641    PlaybackThread *thread = checkPlaybackThread_l(output);
1642
1643    if (thread == NULL) {
1644        return BAD_VALUE;
1645    }
1646
1647    ALOGV("suspendOutput() %d", output);
1648    thread->suspend();
1649
1650    return NO_ERROR;
1651}
1652
1653status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1654{
1655    Mutex::Autolock _l(mLock);
1656    PlaybackThread *thread = checkPlaybackThread_l(output);
1657
1658    if (thread == NULL) {
1659        return BAD_VALUE;
1660    }
1661
1662    ALOGV("restoreOutput() %d", output);
1663
1664    thread->restore();
1665
1666    return NO_ERROR;
1667}
1668
1669audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1670                                          audio_devices_t *pDevices,
1671                                          uint32_t *pSamplingRate,
1672                                          audio_format_t *pFormat,
1673                                          audio_channel_mask_t *pChannelMask)
1674{
1675    status_t status;
1676    RecordThread *thread = NULL;
1677    struct audio_config config;
1678    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1679    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1680    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1681
1682    uint32_t reqSamplingRate = config.sample_rate;
1683    audio_format_t reqFormat = config.format;
1684    audio_channel_mask_t reqChannels = config.channel_mask;
1685    audio_stream_in_t *inStream = NULL;
1686    AudioHwDevice *inHwDev;
1687
1688    if (pDevices == NULL || *pDevices == 0) {
1689        return 0;
1690    }
1691
1692    Mutex::Autolock _l(mLock);
1693
1694    inHwDev = findSuitableHwDev_l(module, *pDevices);
1695    if (inHwDev == NULL)
1696        return 0;
1697
1698    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1699    audio_io_handle_t id = nextUniqueId();
1700
1701    status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1702                                        &inStream);
1703    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1704            "status %d",
1705            inStream,
1706            config.sample_rate,
1707            config.format,
1708            config.channel_mask,
1709            status);
1710
1711    // If the input could not be opened with the requested parameters and we can handle the
1712    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1713    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1714    if (status == BAD_VALUE &&
1715        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1716        (config.sample_rate <= 2 * reqSamplingRate) &&
1717        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
1718        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1719        inStream = NULL;
1720        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1721    }
1722
1723    if (status == NO_ERROR && inStream != NULL) {
1724
1725#ifdef TEE_SINK
1726        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1727        // or (re-)create if current Pipe is idle and does not match the new format
1728        sp<NBAIO_Sink> teeSink;
1729        enum {
1730            TEE_SINK_NO,    // don't copy input
1731            TEE_SINK_NEW,   // copy input using a new pipe
1732            TEE_SINK_OLD,   // copy input using an existing pipe
1733        } kind;
1734        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1735                                        popcount(inStream->common.get_channels(&inStream->common)));
1736        if (!mTeeSinkInputEnabled) {
1737            kind = TEE_SINK_NO;
1738        } else if (format == Format_Invalid) {
1739            kind = TEE_SINK_NO;
1740        } else if (mRecordTeeSink == 0) {
1741            kind = TEE_SINK_NEW;
1742        } else if (mRecordTeeSink->getStrongCount() != 1) {
1743            kind = TEE_SINK_NO;
1744        } else if (format == mRecordTeeSink->format()) {
1745            kind = TEE_SINK_OLD;
1746        } else {
1747            kind = TEE_SINK_NEW;
1748        }
1749        switch (kind) {
1750        case TEE_SINK_NEW: {
1751            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1752            size_t numCounterOffers = 0;
1753            const NBAIO_Format offers[1] = {format};
1754            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1755            ALOG_ASSERT(index == 0);
1756            PipeReader *pipeReader = new PipeReader(*pipe);
1757            numCounterOffers = 0;
1758            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1759            ALOG_ASSERT(index == 0);
1760            mRecordTeeSink = pipe;
1761            mRecordTeeSource = pipeReader;
1762            teeSink = pipe;
1763            }
1764            break;
1765        case TEE_SINK_OLD:
1766            teeSink = mRecordTeeSink;
1767            break;
1768        case TEE_SINK_NO:
1769        default:
1770            break;
1771        }
1772#endif
1773
1774        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1775
1776        // Start record thread
1777        // RecordThread requires both input and output device indication to forward to audio
1778        // pre processing modules
1779        thread = new RecordThread(this,
1780                                  input,
1781                                  reqSamplingRate,
1782                                  reqChannels,
1783                                  id,
1784                                  primaryOutputDevice_l(),
1785                                  *pDevices
1786#ifdef TEE_SINK
1787                                  , teeSink
1788#endif
1789                                  );
1790        mRecordThreads.add(id, thread);
1791        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1792        if (pSamplingRate != NULL) {
1793            *pSamplingRate = reqSamplingRate;
1794        }
1795        if (pFormat != NULL) {
1796            *pFormat = config.format;
1797        }
1798        if (pChannelMask != NULL) {
1799            *pChannelMask = reqChannels;
1800        }
1801
1802        // notify client processes of the new input creation
1803        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1804        return id;
1805    }
1806
1807    return 0;
1808}
1809
1810status_t AudioFlinger::closeInput(audio_io_handle_t input)
1811{
1812    return closeInput_nonvirtual(input);
1813}
1814
1815status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1816{
1817    // keep strong reference on the record thread so that
1818    // it is not destroyed while exit() is executed
1819    sp<RecordThread> thread;
1820    {
1821        Mutex::Autolock _l(mLock);
1822        thread = checkRecordThread_l(input);
1823        if (thread == 0) {
1824            return BAD_VALUE;
1825        }
1826
1827        ALOGV("closeInput() %d", input);
1828        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1829        mRecordThreads.removeItem(input);
1830    }
1831    thread->exit();
1832    // The thread entity (active unit of execution) is no longer running here,
1833    // but the ThreadBase container still exists.
1834
1835    AudioStreamIn *in = thread->clearInput();
1836    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1837    // from now on thread->mInput is NULL
1838    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1839    delete in;
1840
1841    return NO_ERROR;
1842}
1843
1844status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1845{
1846    Mutex::Autolock _l(mLock);
1847    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1848
1849    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1850        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1851        thread->invalidateTracks(stream);
1852    }
1853
1854    return NO_ERROR;
1855}
1856
1857
1858int AudioFlinger::newAudioSessionId()
1859{
1860    return nextUniqueId();
1861}
1862
1863void AudioFlinger::acquireAudioSessionId(int audioSession)
1864{
1865    Mutex::Autolock _l(mLock);
1866    pid_t caller = IPCThreadState::self()->getCallingPid();
1867    ALOGV("acquiring %d from %d", audioSession, caller);
1868
1869    // Ignore requests received from processes not known as notification client. The request
1870    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1871    // called from a different pid leaving a stale session reference.  Also we don't know how
1872    // to clear this reference if the client process dies.
1873    if (mNotificationClients.indexOfKey(caller) < 0) {
1874        ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1875        return;
1876    }
1877
1878    size_t num = mAudioSessionRefs.size();
1879    for (size_t i = 0; i< num; i++) {
1880        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1881        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1882            ref->mCnt++;
1883            ALOGV(" incremented refcount to %d", ref->mCnt);
1884            return;
1885        }
1886    }
1887    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1888    ALOGV(" added new entry for %d", audioSession);
1889}
1890
1891void AudioFlinger::releaseAudioSessionId(int audioSession)
1892{
1893    Mutex::Autolock _l(mLock);
1894    pid_t caller = IPCThreadState::self()->getCallingPid();
1895    ALOGV("releasing %d from %d", audioSession, caller);
1896    size_t num = mAudioSessionRefs.size();
1897    for (size_t i = 0; i< num; i++) {
1898        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1899        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1900            ref->mCnt--;
1901            ALOGV(" decremented refcount to %d", ref->mCnt);
1902            if (ref->mCnt == 0) {
1903                mAudioSessionRefs.removeAt(i);
1904                delete ref;
1905                purgeStaleEffects_l();
1906            }
1907            return;
1908        }
1909    }
1910    // If the caller is mediaserver it is likely that the session being released was acquired
1911    // on behalf of a process not in notification clients and we ignore the warning.
1912    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
1913}
1914
1915void AudioFlinger::purgeStaleEffects_l() {
1916
1917    ALOGV("purging stale effects");
1918
1919    Vector< sp<EffectChain> > chains;
1920
1921    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1922        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1923        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1924            sp<EffectChain> ec = t->mEffectChains[j];
1925            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1926                chains.push(ec);
1927            }
1928        }
1929    }
1930    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1931        sp<RecordThread> t = mRecordThreads.valueAt(i);
1932        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1933            sp<EffectChain> ec = t->mEffectChains[j];
1934            chains.push(ec);
1935        }
1936    }
1937
1938    for (size_t i = 0; i < chains.size(); i++) {
1939        sp<EffectChain> ec = chains[i];
1940        int sessionid = ec->sessionId();
1941        sp<ThreadBase> t = ec->mThread.promote();
1942        if (t == 0) {
1943            continue;
1944        }
1945        size_t numsessionrefs = mAudioSessionRefs.size();
1946        bool found = false;
1947        for (size_t k = 0; k < numsessionrefs; k++) {
1948            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1949            if (ref->mSessionid == sessionid) {
1950                ALOGV(" session %d still exists for %d with %d refs",
1951                    sessionid, ref->mPid, ref->mCnt);
1952                found = true;
1953                break;
1954            }
1955        }
1956        if (!found) {
1957            Mutex::Autolock _l (t->mLock);
1958            // remove all effects from the chain
1959            while (ec->mEffects.size()) {
1960                sp<EffectModule> effect = ec->mEffects[0];
1961                effect->unPin();
1962                t->removeEffect_l(effect);
1963                if (effect->purgeHandles()) {
1964                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
1965                }
1966                AudioSystem::unregisterEffect(effect->id());
1967            }
1968        }
1969    }
1970    return;
1971}
1972
1973// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
1974AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
1975{
1976    return mPlaybackThreads.valueFor(output).get();
1977}
1978
1979// checkMixerThread_l() must be called with AudioFlinger::mLock held
1980AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
1981{
1982    PlaybackThread *thread = checkPlaybackThread_l(output);
1983    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
1984}
1985
1986// checkRecordThread_l() must be called with AudioFlinger::mLock held
1987AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
1988{
1989    return mRecordThreads.valueFor(input).get();
1990}
1991
1992uint32_t AudioFlinger::nextUniqueId()
1993{
1994    return android_atomic_inc(&mNextUniqueId);
1995}
1996
1997AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
1998{
1999    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2000        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2001        AudioStreamOut *output = thread->getOutput();
2002        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2003            return thread;
2004        }
2005    }
2006    return NULL;
2007}
2008
2009audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2010{
2011    PlaybackThread *thread = primaryPlaybackThread_l();
2012
2013    if (thread == NULL) {
2014        return 0;
2015    }
2016
2017    return thread->outDevice();
2018}
2019
2020sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2021                                    int triggerSession,
2022                                    int listenerSession,
2023                                    sync_event_callback_t callBack,
2024                                    void *cookie)
2025{
2026    Mutex::Autolock _l(mLock);
2027
2028    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2029    status_t playStatus = NAME_NOT_FOUND;
2030    status_t recStatus = NAME_NOT_FOUND;
2031    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2032        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2033        if (playStatus == NO_ERROR) {
2034            return event;
2035        }
2036    }
2037    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2038        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2039        if (recStatus == NO_ERROR) {
2040            return event;
2041        }
2042    }
2043    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2044        mPendingSyncEvents.add(event);
2045    } else {
2046        ALOGV("createSyncEvent() invalid event %d", event->type());
2047        event.clear();
2048    }
2049    return event;
2050}
2051
2052// ----------------------------------------------------------------------------
2053//  Effect management
2054// ----------------------------------------------------------------------------
2055
2056
2057status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2058{
2059    Mutex::Autolock _l(mLock);
2060    return EffectQueryNumberEffects(numEffects);
2061}
2062
2063status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2064{
2065    Mutex::Autolock _l(mLock);
2066    return EffectQueryEffect(index, descriptor);
2067}
2068
2069status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2070        effect_descriptor_t *descriptor) const
2071{
2072    Mutex::Autolock _l(mLock);
2073    return EffectGetDescriptor(pUuid, descriptor);
2074}
2075
2076
2077sp<IEffect> AudioFlinger::createEffect(
2078        effect_descriptor_t *pDesc,
2079        const sp<IEffectClient>& effectClient,
2080        int32_t priority,
2081        audio_io_handle_t io,
2082        int sessionId,
2083        status_t *status,
2084        int *id,
2085        int *enabled)
2086{
2087    status_t lStatus = NO_ERROR;
2088    sp<EffectHandle> handle;
2089    effect_descriptor_t desc;
2090
2091    pid_t pid = IPCThreadState::self()->getCallingPid();
2092    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2093            pid, effectClient.get(), priority, sessionId, io);
2094
2095    if (pDesc == NULL) {
2096        lStatus = BAD_VALUE;
2097        goto Exit;
2098    }
2099
2100    // check audio settings permission for global effects
2101    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2102        lStatus = PERMISSION_DENIED;
2103        goto Exit;
2104    }
2105
2106    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2107    // that can only be created by audio policy manager (running in same process)
2108    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2109        lStatus = PERMISSION_DENIED;
2110        goto Exit;
2111    }
2112
2113    {
2114        if (!EffectIsNullUuid(&pDesc->uuid)) {
2115            // if uuid is specified, request effect descriptor
2116            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2117            if (lStatus < 0) {
2118                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2119                goto Exit;
2120            }
2121        } else {
2122            // if uuid is not specified, look for an available implementation
2123            // of the required type in effect factory
2124            if (EffectIsNullUuid(&pDesc->type)) {
2125                ALOGW("createEffect() no effect type");
2126                lStatus = BAD_VALUE;
2127                goto Exit;
2128            }
2129            uint32_t numEffects = 0;
2130            effect_descriptor_t d;
2131            d.flags = 0; // prevent compiler warning
2132            bool found = false;
2133
2134            lStatus = EffectQueryNumberEffects(&numEffects);
2135            if (lStatus < 0) {
2136                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2137                goto Exit;
2138            }
2139            for (uint32_t i = 0; i < numEffects; i++) {
2140                lStatus = EffectQueryEffect(i, &desc);
2141                if (lStatus < 0) {
2142                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2143                    continue;
2144                }
2145                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2146                    // If matching type found save effect descriptor. If the session is
2147                    // 0 and the effect is not auxiliary, continue enumeration in case
2148                    // an auxiliary version of this effect type is available
2149                    found = true;
2150                    d = desc;
2151                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2152                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2153                        break;
2154                    }
2155                }
2156            }
2157            if (!found) {
2158                lStatus = BAD_VALUE;
2159                ALOGW("createEffect() effect not found");
2160                goto Exit;
2161            }
2162            // For same effect type, chose auxiliary version over insert version if
2163            // connect to output mix (Compliance to OpenSL ES)
2164            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2165                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2166                desc = d;
2167            }
2168        }
2169
2170        // Do not allow auxiliary effects on a session different from 0 (output mix)
2171        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2172             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2173            lStatus = INVALID_OPERATION;
2174            goto Exit;
2175        }
2176
2177        // check recording permission for visualizer
2178        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2179            !recordingAllowed()) {
2180            lStatus = PERMISSION_DENIED;
2181            goto Exit;
2182        }
2183
2184        // return effect descriptor
2185        *pDesc = desc;
2186        if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2187            // if the output returned by getOutputForEffect() is removed before we lock the
2188            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2189            // and we will exit safely
2190            io = AudioSystem::getOutputForEffect(&desc);
2191            ALOGV("createEffect got output %d", io);
2192        }
2193
2194        Mutex::Autolock _l(mLock);
2195
2196        // If output is not specified try to find a matching audio session ID in one of the
2197        // output threads.
2198        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2199        // because of code checking output when entering the function.
2200        // Note: io is never 0 when creating an effect on an input
2201        if (io == 0) {
2202            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2203                // output must be specified by AudioPolicyManager when using session
2204                // AUDIO_SESSION_OUTPUT_STAGE
2205                lStatus = BAD_VALUE;
2206                goto Exit;
2207            }
2208            // look for the thread where the specified audio session is present
2209            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2210                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2211                    io = mPlaybackThreads.keyAt(i);
2212                    break;
2213                }
2214            }
2215            if (io == 0) {
2216                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2217                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2218                        io = mRecordThreads.keyAt(i);
2219                        break;
2220                    }
2221                }
2222            }
2223            // If no output thread contains the requested session ID, default to
2224            // first output. The effect chain will be moved to the correct output
2225            // thread when a track with the same session ID is created
2226            if (io == 0 && mPlaybackThreads.size()) {
2227                io = mPlaybackThreads.keyAt(0);
2228            }
2229            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2230        }
2231        ThreadBase *thread = checkRecordThread_l(io);
2232        if (thread == NULL) {
2233            thread = checkPlaybackThread_l(io);
2234            if (thread == NULL) {
2235                ALOGE("createEffect() unknown output thread");
2236                lStatus = BAD_VALUE;
2237                goto Exit;
2238            }
2239        }
2240
2241        sp<Client> client = registerPid_l(pid);
2242
2243        // create effect on selected output thread
2244        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2245                &desc, enabled, &lStatus);
2246        if (handle != 0 && id != NULL) {
2247            *id = handle->id();
2248        }
2249    }
2250
2251Exit:
2252    if (status != NULL) {
2253        *status = lStatus;
2254    }
2255    return handle;
2256}
2257
2258status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2259        audio_io_handle_t dstOutput)
2260{
2261    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2262            sessionId, srcOutput, dstOutput);
2263    Mutex::Autolock _l(mLock);
2264    if (srcOutput == dstOutput) {
2265        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2266        return NO_ERROR;
2267    }
2268    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2269    if (srcThread == NULL) {
2270        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2271        return BAD_VALUE;
2272    }
2273    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2274    if (dstThread == NULL) {
2275        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2276        return BAD_VALUE;
2277    }
2278
2279    Mutex::Autolock _dl(dstThread->mLock);
2280    Mutex::Autolock _sl(srcThread->mLock);
2281    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2282}
2283
2284// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2285status_t AudioFlinger::moveEffectChain_l(int sessionId,
2286                                   AudioFlinger::PlaybackThread *srcThread,
2287                                   AudioFlinger::PlaybackThread *dstThread,
2288                                   bool reRegister)
2289{
2290    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2291            sessionId, srcThread, dstThread);
2292
2293    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2294    if (chain == 0) {
2295        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2296                sessionId, srcThread);
2297        return INVALID_OPERATION;
2298    }
2299
2300    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2301    // so that a new chain is created with correct parameters when first effect is added. This is
2302    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2303    // removed.
2304    srcThread->removeEffectChain_l(chain);
2305
2306    // transfer all effects one by one so that new effect chain is created on new thread with
2307    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2308    sp<EffectChain> dstChain;
2309    uint32_t strategy = 0; // prevent compiler warning
2310    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2311    Vector< sp<EffectModule> > removed;
2312    status_t status = NO_ERROR;
2313    while (effect != 0) {
2314        srcThread->removeEffect_l(effect);
2315        removed.add(effect);
2316        status = dstThread->addEffect_l(effect);
2317        if (status != NO_ERROR) {
2318            break;
2319        }
2320        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2321        if (effect->state() == EffectModule::ACTIVE ||
2322                effect->state() == EffectModule::STOPPING) {
2323            effect->start();
2324        }
2325        // if the move request is not received from audio policy manager, the effect must be
2326        // re-registered with the new strategy and output
2327        if (dstChain == 0) {
2328            dstChain = effect->chain().promote();
2329            if (dstChain == 0) {
2330                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2331                status = NO_INIT;
2332                break;
2333            }
2334            strategy = dstChain->strategy();
2335        }
2336        if (reRegister) {
2337            AudioSystem::unregisterEffect(effect->id());
2338            AudioSystem::registerEffect(&effect->desc(),
2339                                        dstThread->id(),
2340                                        strategy,
2341                                        sessionId,
2342                                        effect->id());
2343            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2344        }
2345        effect = chain->getEffectFromId_l(0);
2346    }
2347
2348    if (status != NO_ERROR) {
2349        for (size_t i = 0; i < removed.size(); i++) {
2350            srcThread->addEffect_l(removed[i]);
2351            if (dstChain != 0 && reRegister) {
2352                AudioSystem::unregisterEffect(removed[i]->id());
2353                AudioSystem::registerEffect(&removed[i]->desc(),
2354                                            srcThread->id(),
2355                                            strategy,
2356                                            sessionId,
2357                                            removed[i]->id());
2358                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2359            }
2360        }
2361    }
2362
2363    return status;
2364}
2365
2366bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2367{
2368    if (mGlobalEffectEnableTime != 0 &&
2369            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2370        return true;
2371    }
2372
2373    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2374        sp<EffectChain> ec =
2375                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2376        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2377            return true;
2378        }
2379    }
2380    return false;
2381}
2382
2383void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2384{
2385    Mutex::Autolock _l(mLock);
2386
2387    mGlobalEffectEnableTime = systemTime();
2388
2389    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2390        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2391        if (t->mType == ThreadBase::OFFLOAD) {
2392            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2393        }
2394    }
2395
2396}
2397
2398struct Entry {
2399#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2400    char mName[MAX_NAME];
2401};
2402
2403int comparEntry(const void *p1, const void *p2)
2404{
2405    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2406}
2407
2408#ifdef TEE_SINK
2409void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2410{
2411    NBAIO_Source *teeSource = source.get();
2412    if (teeSource != NULL) {
2413        // .wav rotation
2414        // There is a benign race condition if 2 threads call this simultaneously.
2415        // They would both traverse the directory, but the result would simply be
2416        // failures at unlink() which are ignored.  It's also unlikely since
2417        // normally dumpsys is only done by bugreport or from the command line.
2418        char teePath[32+256];
2419        strcpy(teePath, "/data/misc/media");
2420        size_t teePathLen = strlen(teePath);
2421        DIR *dir = opendir(teePath);
2422        teePath[teePathLen++] = '/';
2423        if (dir != NULL) {
2424#define MAX_SORT 20 // number of entries to sort
2425#define MAX_KEEP 10 // number of entries to keep
2426            struct Entry entries[MAX_SORT];
2427            size_t entryCount = 0;
2428            while (entryCount < MAX_SORT) {
2429                struct dirent de;
2430                struct dirent *result = NULL;
2431                int rc = readdir_r(dir, &de, &result);
2432                if (rc != 0) {
2433                    ALOGW("readdir_r failed %d", rc);
2434                    break;
2435                }
2436                if (result == NULL) {
2437                    break;
2438                }
2439                if (result != &de) {
2440                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2441                    break;
2442                }
2443                // ignore non .wav file entries
2444                size_t nameLen = strlen(de.d_name);
2445                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2446                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2447                    continue;
2448                }
2449                strcpy(entries[entryCount++].mName, de.d_name);
2450            }
2451            (void) closedir(dir);
2452            if (entryCount > MAX_KEEP) {
2453                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2454                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2455                    strcpy(&teePath[teePathLen], entries[i].mName);
2456                    (void) unlink(teePath);
2457                }
2458            }
2459        } else {
2460            if (fd >= 0) {
2461                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2462            }
2463        }
2464        char teeTime[16];
2465        struct timeval tv;
2466        gettimeofday(&tv, NULL);
2467        struct tm tm;
2468        localtime_r(&tv.tv_sec, &tm);
2469        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2470        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2471        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2472        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2473        if (teeFd >= 0) {
2474            char wavHeader[44];
2475            memcpy(wavHeader,
2476                "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",
2477                sizeof(wavHeader));
2478            NBAIO_Format format = teeSource->format();
2479            unsigned channelCount = Format_channelCount(format);
2480            ALOG_ASSERT(channelCount <= FCC_2);
2481            uint32_t sampleRate = Format_sampleRate(format);
2482            wavHeader[22] = channelCount;       // number of channels
2483            wavHeader[24] = sampleRate;         // sample rate
2484            wavHeader[25] = sampleRate >> 8;
2485            wavHeader[32] = channelCount * 2;   // block alignment
2486            write(teeFd, wavHeader, sizeof(wavHeader));
2487            size_t total = 0;
2488            bool firstRead = true;
2489            for (;;) {
2490#define TEE_SINK_READ 1024
2491                short buffer[TEE_SINK_READ * FCC_2];
2492                size_t count = TEE_SINK_READ;
2493                ssize_t actual = teeSource->read(buffer, count,
2494                        AudioBufferProvider::kInvalidPTS);
2495                bool wasFirstRead = firstRead;
2496                firstRead = false;
2497                if (actual <= 0) {
2498                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2499                        continue;
2500                    }
2501                    break;
2502                }
2503                ALOG_ASSERT(actual <= (ssize_t)count);
2504                write(teeFd, buffer, actual * channelCount * sizeof(short));
2505                total += actual;
2506            }
2507            lseek(teeFd, (off_t) 4, SEEK_SET);
2508            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2509            write(teeFd, &temp, sizeof(temp));
2510            lseek(teeFd, (off_t) 40, SEEK_SET);
2511            temp =  total * channelCount * sizeof(short);
2512            write(teeFd, &temp, sizeof(temp));
2513            close(teeFd);
2514            if (fd >= 0) {
2515                fdprintf(fd, "tee copied to %s\n", teePath);
2516            }
2517        } else {
2518            if (fd >= 0) {
2519                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2520            }
2521        }
2522    }
2523}
2524#endif
2525
2526// ----------------------------------------------------------------------------
2527
2528status_t AudioFlinger::onTransact(
2529        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2530{
2531    return BnAudioFlinger::onTransact(code, data, reply, flags);
2532}
2533
2534}; // namespace android
2535