AudioFlinger.cpp revision e198c360d5e75a9b2097844c495c10902e7e8500 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <dirent.h>
24#include <math.h>
25#include <signal.h>
26#include <sys/time.h>
27#include <sys/resource.h>
28
29#include <binder/IPCThreadState.h>
30#include <binder/IServiceManager.h>
31#include <utils/Log.h>
32#include <utils/Trace.h>
33#include <binder/Parcel.h>
34#include <utils/String16.h>
35#include <utils/threads.h>
36#include <utils/Atomic.h>
37
38#include <cutils/bitops.h>
39#include <cutils/properties.h>
40
41#include <system/audio.h>
42#include <hardware/audio.h>
43
44#include "AudioMixer.h"
45#include "AudioFlinger.h"
46#include "ServiceUtilities.h"
47
48#include <media/EffectsFactoryApi.h>
49#include <audio_effects/effect_visualizer.h>
50#include <audio_effects/effect_ns.h>
51#include <audio_effects/effect_aec.h>
52
53#include <audio_utils/primitives.h>
54
55#include <powermanager/PowerManager.h>
56
57#include <common_time/cc_helper.h>
58
59#include <media/IMediaLogService.h>
60
61#include <media/nbaio/Pipe.h>
62#include <media/nbaio/PipeReader.h>
63#include <media/AudioParameter.h>
64#include <private/android_filesystem_config.h>
65
66// ----------------------------------------------------------------------------
67
68// Note: the following macro is used for extremely verbose logging message.  In
69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
70// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
71// are so verbose that we want to suppress them even when we have ALOG_ASSERT
72// turned on.  Do not uncomment the #def below unless you really know what you
73// are doing and want to see all of the extremely verbose messages.
74//#define VERY_VERY_VERBOSE_LOGGING
75#ifdef VERY_VERY_VERBOSE_LOGGING
76#define ALOGVV ALOGV
77#else
78#define ALOGVV(a...) do { } while(0)
79#endif
80
81namespace android {
82
83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
84static const char kHardwareLockedString[] = "Hardware lock is taken\n";
85
86
87nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
88
89uint32_t AudioFlinger::mScreenState;
90
91#ifdef TEE_SINK
92bool AudioFlinger::mTeeSinkInputEnabled = false;
93bool AudioFlinger::mTeeSinkOutputEnabled = false;
94bool AudioFlinger::mTeeSinkTrackEnabled = false;
95
96size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
97size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
98size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
99#endif
100
101// ----------------------------------------------------------------------------
102
103static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
104{
105    const hw_module_t *mod;
106    int rc;
107
108    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
109    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
110                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
111    if (rc) {
112        goto out;
113    }
114    rc = audio_hw_device_open(mod, dev);
115    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
116                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
117    if (rc) {
118        goto out;
119    }
120    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
121        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
122        rc = BAD_VALUE;
123        goto out;
124    }
125    return 0;
126
127out:
128    *dev = NULL;
129    return rc;
130}
131
132// ----------------------------------------------------------------------------
133
134AudioFlinger::AudioFlinger()
135    : BnAudioFlinger(),
136      mPrimaryHardwareDev(NULL),
137      mHardwareStatus(AUDIO_HW_IDLE),
138      mMasterVolume(1.0f),
139      mMasterMute(false),
140      mNextUniqueId(1),
141      mMode(AUDIO_MODE_INVALID),
142      mBtNrecIsOff(false),
143      mIsLowRamDevice(true),
144      mIsDeviceTypeKnown(false)
145{
146    getpid_cached = getpid();
147    char value[PROPERTY_VALUE_MAX];
148    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
149    if (doLog) {
150        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
151    }
152#ifdef TEE_SINK
153    (void) property_get("ro.debuggable", value, "0");
154    int debuggable = atoi(value);
155    int teeEnabled = 0;
156    if (debuggable) {
157        (void) property_get("af.tee", value, "0");
158        teeEnabled = atoi(value);
159    }
160    if (teeEnabled & 1)
161        mTeeSinkInputEnabled = true;
162    if (teeEnabled & 2)
163        mTeeSinkOutputEnabled = true;
164    if (teeEnabled & 4)
165        mTeeSinkTrackEnabled = true;
166#endif
167}
168
169void AudioFlinger::onFirstRef()
170{
171    int rc = 0;
172
173    Mutex::Autolock _l(mLock);
174
175    /* TODO: move all this work into an Init() function */
176    char val_str[PROPERTY_VALUE_MAX] = { 0 };
177    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
178        uint32_t int_val;
179        if (1 == sscanf(val_str, "%u", &int_val)) {
180            mStandbyTimeInNsecs = milliseconds(int_val);
181            ALOGI("Using %u mSec as standby time.", int_val);
182        } else {
183            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
184            ALOGI("Using default %u mSec as standby time.",
185                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
186        }
187    }
188
189    mMode = AUDIO_MODE_NORMAL;
190}
191
192AudioFlinger::~AudioFlinger()
193{
194    while (!mRecordThreads.isEmpty()) {
195        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
196        closeInput_nonvirtual(mRecordThreads.keyAt(0));
197    }
198    while (!mPlaybackThreads.isEmpty()) {
199        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
200        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
201    }
202
203    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
204        // no mHardwareLock needed, as there are no other references to this
205        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
206        delete mAudioHwDevs.valueAt(i);
207    }
208}
209
210static const char * const audio_interfaces[] = {
211    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
212    AUDIO_HARDWARE_MODULE_ID_A2DP,
213    AUDIO_HARDWARE_MODULE_ID_USB,
214};
215#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
216
217AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
218        audio_module_handle_t module,
219        audio_devices_t devices)
220{
221    // if module is 0, the request comes from an old policy manager and we should load
222    // well known modules
223    if (module == 0) {
224        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
225        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
226            loadHwModule_l(audio_interfaces[i]);
227        }
228        // then try to find a module supporting the requested device.
229        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
230            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
231            audio_hw_device_t *dev = audioHwDevice->hwDevice();
232            if ((dev->get_supported_devices != NULL) &&
233                    (dev->get_supported_devices(dev) & devices) == devices)
234                return audioHwDevice;
235        }
236    } else {
237        // check a match for the requested module handle
238        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
239        if (audioHwDevice != NULL) {
240            return audioHwDevice;
241        }
242    }
243
244    return NULL;
245}
246
247void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
248{
249    const size_t SIZE = 256;
250    char buffer[SIZE];
251    String8 result;
252
253    result.append("Clients:\n");
254    for (size_t i = 0; i < mClients.size(); ++i) {
255        sp<Client> client = mClients.valueAt(i).promote();
256        if (client != 0) {
257            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
258            result.append(buffer);
259        }
260    }
261
262    result.append("Global session refs:\n");
263    result.append(" session pid count\n");
264    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
265        AudioSessionRef *r = mAudioSessionRefs[i];
266        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
267        result.append(buffer);
268    }
269    write(fd, result.string(), result.size());
270}
271
272
273void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
274{
275    const size_t SIZE = 256;
276    char buffer[SIZE];
277    String8 result;
278    hardware_call_state hardwareStatus = mHardwareStatus;
279
280    snprintf(buffer, SIZE, "Hardware status: %d\n"
281                           "Standby Time mSec: %u\n",
282                            hardwareStatus,
283                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
284    result.append(buffer);
285    write(fd, result.string(), result.size());
286}
287
288void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
289{
290    const size_t SIZE = 256;
291    char buffer[SIZE];
292    String8 result;
293    snprintf(buffer, SIZE, "Permission Denial: "
294            "can't dump AudioFlinger from pid=%d, uid=%d\n",
295            IPCThreadState::self()->getCallingPid(),
296            IPCThreadState::self()->getCallingUid());
297    result.append(buffer);
298    write(fd, result.string(), result.size());
299}
300
301bool AudioFlinger::dumpTryLock(Mutex& mutex)
302{
303    bool locked = false;
304    for (int i = 0; i < kDumpLockRetries; ++i) {
305        if (mutex.tryLock() == NO_ERROR) {
306            locked = true;
307            break;
308        }
309        usleep(kDumpLockSleepUs);
310    }
311    return locked;
312}
313
314status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
315{
316    if (!dumpAllowed()) {
317        dumpPermissionDenial(fd, args);
318    } else {
319        // get state of hardware lock
320        bool hardwareLocked = dumpTryLock(mHardwareLock);
321        if (!hardwareLocked) {
322            String8 result(kHardwareLockedString);
323            write(fd, result.string(), result.size());
324        } else {
325            mHardwareLock.unlock();
326        }
327
328        bool locked = dumpTryLock(mLock);
329
330        // failed to lock - AudioFlinger is probably deadlocked
331        if (!locked) {
332            String8 result(kDeadlockedString);
333            write(fd, result.string(), result.size());
334        }
335
336        dumpClients(fd, args);
337        dumpInternals(fd, args);
338
339        // dump playback threads
340        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
341            mPlaybackThreads.valueAt(i)->dump(fd, args);
342        }
343
344        // dump record threads
345        for (size_t i = 0; i < mRecordThreads.size(); i++) {
346            mRecordThreads.valueAt(i)->dump(fd, args);
347        }
348
349        // dump all hardware devs
350        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
351            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
352            dev->dump(dev, fd);
353        }
354
355#ifdef TEE_SINK
356        // dump the serially shared record tee sink
357        if (mRecordTeeSource != 0) {
358            dumpTee(fd, mRecordTeeSource);
359        }
360#endif
361
362        if (locked) {
363            mLock.unlock();
364        }
365
366        // append a copy of media.log here by forwarding fd to it, but don't attempt
367        // to lookup the service if it's not running, as it will block for a second
368        if (mLogMemoryDealer != 0) {
369            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
370            if (binder != 0) {
371                fdprintf(fd, "\nmedia.log:\n");
372                Vector<String16> args;
373                binder->dump(fd, args);
374            }
375        }
376    }
377    return NO_ERROR;
378}
379
380sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
381{
382    // If pid is already in the mClients wp<> map, then use that entry
383    // (for which promote() is always != 0), otherwise create a new entry and Client.
384    sp<Client> client = mClients.valueFor(pid).promote();
385    if (client == 0) {
386        client = new Client(this, pid);
387        mClients.add(pid, client);
388    }
389
390    return client;
391}
392
393sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
394{
395    if (mLogMemoryDealer == 0) {
396        return new NBLog::Writer();
397    }
398    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
399    sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
400    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
401    if (binder != 0) {
402        interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
403    }
404    return writer;
405}
406
407void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
408{
409    if (writer == 0) {
410        return;
411    }
412    sp<IMemory> iMemory(writer->getIMemory());
413    if (iMemory == 0) {
414        return;
415    }
416    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
417    if (binder != 0) {
418        interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
419        // Now the media.log remote reference to IMemory is gone.
420        // When our last local reference to IMemory also drops to zero,
421        // the IMemory destructor will deallocate the region from mMemoryDealer.
422    }
423}
424
425// IAudioFlinger interface
426
427
428sp<IAudioTrack> AudioFlinger::createTrack(
429        audio_stream_type_t streamType,
430        uint32_t sampleRate,
431        audio_format_t format,
432        audio_channel_mask_t channelMask,
433        size_t frameCount,
434        IAudioFlinger::track_flags_t *flags,
435        const sp<IMemory>& sharedBuffer,
436        audio_io_handle_t output,
437        pid_t tid,
438        int *sessionId,
439        String8& name,
440        status_t *status)
441{
442    sp<PlaybackThread::Track> track;
443    sp<TrackHandle> trackHandle;
444    sp<Client> client;
445    status_t lStatus;
446    int lSessionId;
447
448    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
449    // but if someone uses binder directly they could bypass that and cause us to crash
450    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
451        ALOGE("createTrack() invalid stream type %d", streamType);
452        lStatus = BAD_VALUE;
453        goto Exit;
454    }
455
456    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
457    // and we don't yet support 8.24 or 32-bit PCM
458    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
459        ALOGE("createTrack() invalid format %d", format);
460        lStatus = BAD_VALUE;
461        goto Exit;
462    }
463
464    {
465        Mutex::Autolock _l(mLock);
466        PlaybackThread *thread = checkPlaybackThread_l(output);
467        PlaybackThread *effectThread = NULL;
468        if (thread == NULL) {
469            ALOGE("no playback thread found for output handle %d", output);
470            lStatus = BAD_VALUE;
471            goto Exit;
472        }
473
474        pid_t pid = IPCThreadState::self()->getCallingPid();
475        client = registerPid_l(pid);
476
477        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
478        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
479            // check if an effect chain with the same session ID is present on another
480            // output thread and move it here.
481            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
482                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
483                if (mPlaybackThreads.keyAt(i) != output) {
484                    uint32_t sessions = t->hasAudioSession(*sessionId);
485                    if (sessions & PlaybackThread::EFFECT_SESSION) {
486                        effectThread = t.get();
487                        break;
488                    }
489                }
490            }
491            lSessionId = *sessionId;
492        } else {
493            // if no audio session id is provided, create one here
494            lSessionId = nextUniqueId();
495            if (sessionId != NULL) {
496                *sessionId = lSessionId;
497            }
498        }
499        ALOGV("createTrack() lSessionId: %d", lSessionId);
500
501        track = thread->createTrack_l(client, streamType, sampleRate, format,
502                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus);
503        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
504
505        // move effect chain to this output thread if an effect on same session was waiting
506        // for a track to be created
507        if (lStatus == NO_ERROR && effectThread != NULL) {
508            // no risk of deadlock because AudioFlinger::mLock is held
509            Mutex::Autolock _dl(thread->mLock);
510            Mutex::Autolock _sl(effectThread->mLock);
511            moveEffectChain_l(lSessionId, effectThread, thread, true);
512        }
513
514        // Look for sync events awaiting for a session to be used.
515        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
516            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
517                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
518                    if (lStatus == NO_ERROR) {
519                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
520                    } else {
521                        mPendingSyncEvents[i]->cancel();
522                    }
523                    mPendingSyncEvents.removeAt(i);
524                    i--;
525                }
526            }
527        }
528
529    }
530
531    if (lStatus == NO_ERROR) {
532        // s for server's pid, n for normal mixer name, f for fast index
533        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
534                track->fastIndex());
535        trackHandle = new TrackHandle(track);
536    } else {
537        // remove local strong reference to Client before deleting the Track so that the Client
538        // destructor is called by the TrackBase destructor with mLock held
539        client.clear();
540        track.clear();
541    }
542
543Exit:
544    *status = lStatus;
545    return trackHandle;
546}
547
548uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
549{
550    Mutex::Autolock _l(mLock);
551    PlaybackThread *thread = checkPlaybackThread_l(output);
552    if (thread == NULL) {
553        ALOGW("sampleRate() unknown thread %d", output);
554        return 0;
555    }
556    return thread->sampleRate();
557}
558
559int AudioFlinger::channelCount(audio_io_handle_t output) const
560{
561    Mutex::Autolock _l(mLock);
562    PlaybackThread *thread = checkPlaybackThread_l(output);
563    if (thread == NULL) {
564        ALOGW("channelCount() unknown thread %d", output);
565        return 0;
566    }
567    return thread->channelCount();
568}
569
570audio_format_t AudioFlinger::format(audio_io_handle_t output) const
571{
572    Mutex::Autolock _l(mLock);
573    PlaybackThread *thread = checkPlaybackThread_l(output);
574    if (thread == NULL) {
575        ALOGW("format() unknown thread %d", output);
576        return AUDIO_FORMAT_INVALID;
577    }
578    return thread->format();
579}
580
581size_t AudioFlinger::frameCount(audio_io_handle_t output) const
582{
583    Mutex::Autolock _l(mLock);
584    PlaybackThread *thread = checkPlaybackThread_l(output);
585    if (thread == NULL) {
586        ALOGW("frameCount() unknown thread %d", output);
587        return 0;
588    }
589    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
590    //       should examine all callers and fix them to handle smaller counts
591    return thread->frameCount();
592}
593
594uint32_t AudioFlinger::latency(audio_io_handle_t output) const
595{
596    Mutex::Autolock _l(mLock);
597    PlaybackThread *thread = checkPlaybackThread_l(output);
598    if (thread == NULL) {
599        ALOGW("latency(): no playback thread found for output handle %d", output);
600        return 0;
601    }
602    return thread->latency();
603}
604
605status_t AudioFlinger::setMasterVolume(float value)
606{
607    status_t ret = initCheck();
608    if (ret != NO_ERROR) {
609        return ret;
610    }
611
612    // check calling permissions
613    if (!settingsAllowed()) {
614        return PERMISSION_DENIED;
615    }
616
617    Mutex::Autolock _l(mLock);
618    mMasterVolume = value;
619
620    // Set master volume in the HALs which support it.
621    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
622        AutoMutex lock(mHardwareLock);
623        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
624
625        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
626        if (dev->canSetMasterVolume()) {
627            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
628        }
629        mHardwareStatus = AUDIO_HW_IDLE;
630    }
631
632    // Now set the master volume in each playback thread.  Playback threads
633    // assigned to HALs which do not have master volume support will apply
634    // master volume during the mix operation.  Threads with HALs which do
635    // support master volume will simply ignore the setting.
636    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
637        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
638
639    return NO_ERROR;
640}
641
642status_t AudioFlinger::setMode(audio_mode_t mode)
643{
644    status_t ret = initCheck();
645    if (ret != NO_ERROR) {
646        return ret;
647    }
648
649    // check calling permissions
650    if (!settingsAllowed()) {
651        return PERMISSION_DENIED;
652    }
653    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
654        ALOGW("Illegal value: setMode(%d)", mode);
655        return BAD_VALUE;
656    }
657
658    { // scope for the lock
659        AutoMutex lock(mHardwareLock);
660        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
661        mHardwareStatus = AUDIO_HW_SET_MODE;
662        ret = dev->set_mode(dev, mode);
663        mHardwareStatus = AUDIO_HW_IDLE;
664    }
665
666    if (NO_ERROR == ret) {
667        Mutex::Autolock _l(mLock);
668        mMode = mode;
669        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
670            mPlaybackThreads.valueAt(i)->setMode(mode);
671    }
672
673    return ret;
674}
675
676status_t AudioFlinger::setMicMute(bool state)
677{
678    status_t ret = initCheck();
679    if (ret != NO_ERROR) {
680        return ret;
681    }
682
683    // check calling permissions
684    if (!settingsAllowed()) {
685        return PERMISSION_DENIED;
686    }
687
688    AutoMutex lock(mHardwareLock);
689    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
690    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
691    ret = dev->set_mic_mute(dev, state);
692    mHardwareStatus = AUDIO_HW_IDLE;
693    return ret;
694}
695
696bool AudioFlinger::getMicMute() const
697{
698    status_t ret = initCheck();
699    if (ret != NO_ERROR) {
700        return false;
701    }
702
703    bool state = AUDIO_MODE_INVALID;
704    AutoMutex lock(mHardwareLock);
705    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
706    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
707    dev->get_mic_mute(dev, &state);
708    mHardwareStatus = AUDIO_HW_IDLE;
709    return state;
710}
711
712status_t AudioFlinger::setMasterMute(bool muted)
713{
714    status_t ret = initCheck();
715    if (ret != NO_ERROR) {
716        return ret;
717    }
718
719    // check calling permissions
720    if (!settingsAllowed()) {
721        return PERMISSION_DENIED;
722    }
723
724    Mutex::Autolock _l(mLock);
725    mMasterMute = muted;
726
727    // Set master mute in the HALs which support it.
728    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
729        AutoMutex lock(mHardwareLock);
730        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
731
732        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
733        if (dev->canSetMasterMute()) {
734            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
735        }
736        mHardwareStatus = AUDIO_HW_IDLE;
737    }
738
739    // Now set the master mute in each playback thread.  Playback threads
740    // assigned to HALs which do not have master mute support will apply master
741    // mute during the mix operation.  Threads with HALs which do support master
742    // mute will simply ignore the setting.
743    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
744        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
745
746    return NO_ERROR;
747}
748
749float AudioFlinger::masterVolume() const
750{
751    Mutex::Autolock _l(mLock);
752    return masterVolume_l();
753}
754
755bool AudioFlinger::masterMute() const
756{
757    Mutex::Autolock _l(mLock);
758    return masterMute_l();
759}
760
761float AudioFlinger::masterVolume_l() const
762{
763    return mMasterVolume;
764}
765
766bool AudioFlinger::masterMute_l() const
767{
768    return mMasterMute;
769}
770
771status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
772        audio_io_handle_t output)
773{
774    // check calling permissions
775    if (!settingsAllowed()) {
776        return PERMISSION_DENIED;
777    }
778
779    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
780        ALOGE("setStreamVolume() invalid stream %d", stream);
781        return BAD_VALUE;
782    }
783
784    AutoMutex lock(mLock);
785    PlaybackThread *thread = NULL;
786    if (output) {
787        thread = checkPlaybackThread_l(output);
788        if (thread == NULL) {
789            return BAD_VALUE;
790        }
791    }
792
793    mStreamTypes[stream].volume = value;
794
795    if (thread == NULL) {
796        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
797            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
798        }
799    } else {
800        thread->setStreamVolume(stream, value);
801    }
802
803    return NO_ERROR;
804}
805
806status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
807{
808    // check calling permissions
809    if (!settingsAllowed()) {
810        return PERMISSION_DENIED;
811    }
812
813    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
814        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
815        ALOGE("setStreamMute() invalid stream %d", stream);
816        return BAD_VALUE;
817    }
818
819    AutoMutex lock(mLock);
820    mStreamTypes[stream].mute = muted;
821    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
822        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
823
824    return NO_ERROR;
825}
826
827float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
828{
829    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
830        return 0.0f;
831    }
832
833    AutoMutex lock(mLock);
834    float volume;
835    if (output) {
836        PlaybackThread *thread = checkPlaybackThread_l(output);
837        if (thread == NULL) {
838            return 0.0f;
839        }
840        volume = thread->streamVolume(stream);
841    } else {
842        volume = streamVolume_l(stream);
843    }
844
845    return volume;
846}
847
848bool AudioFlinger::streamMute(audio_stream_type_t stream) const
849{
850    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
851        return true;
852    }
853
854    AutoMutex lock(mLock);
855    return streamMute_l(stream);
856}
857
858status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
859{
860    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
861            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
862
863    // check calling permissions
864    if (!settingsAllowed()) {
865        return PERMISSION_DENIED;
866    }
867
868    // ioHandle == 0 means the parameters are global to the audio hardware interface
869    if (ioHandle == 0) {
870        Mutex::Autolock _l(mLock);
871        status_t final_result = NO_ERROR;
872        {
873            AutoMutex lock(mHardwareLock);
874            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
875            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
876                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
877                status_t result = dev->set_parameters(dev, keyValuePairs.string());
878                final_result = result ?: final_result;
879            }
880            mHardwareStatus = AUDIO_HW_IDLE;
881        }
882        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
883        AudioParameter param = AudioParameter(keyValuePairs);
884        String8 value;
885        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
886            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
887            if (mBtNrecIsOff != btNrecIsOff) {
888                for (size_t i = 0; i < mRecordThreads.size(); i++) {
889                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
890                    audio_devices_t device = thread->inDevice();
891                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
892                    // collect all of the thread's session IDs
893                    KeyedVector<int, bool> ids = thread->sessionIds();
894                    // suspend effects associated with those session IDs
895                    for (size_t j = 0; j < ids.size(); ++j) {
896                        int sessionId = ids.keyAt(j);
897                        thread->setEffectSuspended(FX_IID_AEC,
898                                                   suspend,
899                                                   sessionId);
900                        thread->setEffectSuspended(FX_IID_NS,
901                                                   suspend,
902                                                   sessionId);
903                    }
904                }
905                mBtNrecIsOff = btNrecIsOff;
906            }
907        }
908        String8 screenState;
909        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
910            bool isOff = screenState == "off";
911            if (isOff != (AudioFlinger::mScreenState & 1)) {
912                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
913            }
914        }
915        return final_result;
916    }
917
918    // hold a strong ref on thread in case closeOutput() or closeInput() is called
919    // and the thread is exited once the lock is released
920    sp<ThreadBase> thread;
921    {
922        Mutex::Autolock _l(mLock);
923        thread = checkPlaybackThread_l(ioHandle);
924        if (thread == 0) {
925            thread = checkRecordThread_l(ioHandle);
926        } else if (thread == primaryPlaybackThread_l()) {
927            // indicate output device change to all input threads for pre processing
928            AudioParameter param = AudioParameter(keyValuePairs);
929            int value;
930            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
931                    (value != 0)) {
932                for (size_t i = 0; i < mRecordThreads.size(); i++) {
933                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
934                }
935            }
936        }
937    }
938    if (thread != 0) {
939        return thread->setParameters(keyValuePairs);
940    }
941    return BAD_VALUE;
942}
943
944String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
945{
946    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
947            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
948
949    Mutex::Autolock _l(mLock);
950
951    if (ioHandle == 0) {
952        String8 out_s8;
953
954        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
955            char *s;
956            {
957            AutoMutex lock(mHardwareLock);
958            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
959            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
960            s = dev->get_parameters(dev, keys.string());
961            mHardwareStatus = AUDIO_HW_IDLE;
962            }
963            out_s8 += String8(s ? s : "");
964            free(s);
965        }
966        return out_s8;
967    }
968
969    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
970    if (playbackThread != NULL) {
971        return playbackThread->getParameters(keys);
972    }
973    RecordThread *recordThread = checkRecordThread_l(ioHandle);
974    if (recordThread != NULL) {
975        return recordThread->getParameters(keys);
976    }
977    return String8("");
978}
979
980size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
981        audio_channel_mask_t channelMask) const
982{
983    status_t ret = initCheck();
984    if (ret != NO_ERROR) {
985        return 0;
986    }
987
988    AutoMutex lock(mHardwareLock);
989    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
990    struct audio_config config;
991    memset(&config, 0, sizeof(config));
992    config.sample_rate = sampleRate;
993    config.channel_mask = channelMask;
994    config.format = format;
995
996    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
997    size_t size = dev->get_input_buffer_size(dev, &config);
998    mHardwareStatus = AUDIO_HW_IDLE;
999    return size;
1000}
1001
1002unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1003{
1004    Mutex::Autolock _l(mLock);
1005
1006    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1007    if (recordThread != NULL) {
1008        return recordThread->getInputFramesLost();
1009    }
1010    return 0;
1011}
1012
1013status_t AudioFlinger::setVoiceVolume(float value)
1014{
1015    status_t ret = initCheck();
1016    if (ret != NO_ERROR) {
1017        return ret;
1018    }
1019
1020    // check calling permissions
1021    if (!settingsAllowed()) {
1022        return PERMISSION_DENIED;
1023    }
1024
1025    AutoMutex lock(mHardwareLock);
1026    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1027    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1028    ret = dev->set_voice_volume(dev, value);
1029    mHardwareStatus = AUDIO_HW_IDLE;
1030
1031    return ret;
1032}
1033
1034status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1035        audio_io_handle_t output) const
1036{
1037    status_t status;
1038
1039    Mutex::Autolock _l(mLock);
1040
1041    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1042    if (playbackThread != NULL) {
1043        return playbackThread->getRenderPosition(halFrames, dspFrames);
1044    }
1045
1046    return BAD_VALUE;
1047}
1048
1049void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1050{
1051
1052    Mutex::Autolock _l(mLock);
1053
1054    pid_t pid = IPCThreadState::self()->getCallingPid();
1055    if (mNotificationClients.indexOfKey(pid) < 0) {
1056        sp<NotificationClient> notificationClient = new NotificationClient(this,
1057                                                                            client,
1058                                                                            pid);
1059        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1060
1061        mNotificationClients.add(pid, notificationClient);
1062
1063        sp<IBinder> binder = client->asBinder();
1064        binder->linkToDeath(notificationClient);
1065
1066        // the config change is always sent from playback or record threads to avoid deadlock
1067        // with AudioSystem::gLock
1068        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1069            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1070        }
1071
1072        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1073            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1074        }
1075    }
1076}
1077
1078void AudioFlinger::removeNotificationClient(pid_t pid)
1079{
1080    Mutex::Autolock _l(mLock);
1081
1082    mNotificationClients.removeItem(pid);
1083
1084    ALOGV("%d died, releasing its sessions", pid);
1085    size_t num = mAudioSessionRefs.size();
1086    bool removed = false;
1087    for (size_t i = 0; i< num; ) {
1088        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1089        ALOGV(" pid %d @ %d", ref->mPid, i);
1090        if (ref->mPid == pid) {
1091            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1092            mAudioSessionRefs.removeAt(i);
1093            delete ref;
1094            removed = true;
1095            num--;
1096        } else {
1097            i++;
1098        }
1099    }
1100    if (removed) {
1101        purgeStaleEffects_l();
1102    }
1103}
1104
1105// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1106void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1107{
1108    size_t size = mNotificationClients.size();
1109    for (size_t i = 0; i < size; i++) {
1110        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1111                                                                               param2);
1112    }
1113}
1114
1115// removeClient_l() must be called with AudioFlinger::mLock held
1116void AudioFlinger::removeClient_l(pid_t pid)
1117{
1118    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1119            IPCThreadState::self()->getCallingPid());
1120    mClients.removeItem(pid);
1121}
1122
1123// getEffectThread_l() must be called with AudioFlinger::mLock held
1124sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1125{
1126    sp<PlaybackThread> thread;
1127
1128    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1129        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1130            ALOG_ASSERT(thread == 0);
1131            thread = mPlaybackThreads.valueAt(i);
1132        }
1133    }
1134
1135    return thread;
1136}
1137
1138
1139
1140// ----------------------------------------------------------------------------
1141
1142AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1143    :   RefBase(),
1144        mAudioFlinger(audioFlinger),
1145        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1146        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1147        mPid(pid),
1148        mTimedTrackCount(0)
1149{
1150    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1151}
1152
1153// Client destructor must be called with AudioFlinger::mLock held
1154AudioFlinger::Client::~Client()
1155{
1156    mAudioFlinger->removeClient_l(mPid);
1157}
1158
1159sp<MemoryDealer> AudioFlinger::Client::heap() const
1160{
1161    return mMemoryDealer;
1162}
1163
1164// Reserve one of the limited slots for a timed audio track associated
1165// with this client
1166bool AudioFlinger::Client::reserveTimedTrack()
1167{
1168    const int kMaxTimedTracksPerClient = 4;
1169
1170    Mutex::Autolock _l(mTimedTrackLock);
1171
1172    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1173        ALOGW("can not create timed track - pid %d has exceeded the limit",
1174             mPid);
1175        return false;
1176    }
1177
1178    mTimedTrackCount++;
1179    return true;
1180}
1181
1182// Release a slot for a timed audio track
1183void AudioFlinger::Client::releaseTimedTrack()
1184{
1185    Mutex::Autolock _l(mTimedTrackLock);
1186    mTimedTrackCount--;
1187}
1188
1189// ----------------------------------------------------------------------------
1190
1191AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1192                                                     const sp<IAudioFlingerClient>& client,
1193                                                     pid_t pid)
1194    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1195{
1196}
1197
1198AudioFlinger::NotificationClient::~NotificationClient()
1199{
1200}
1201
1202void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1203{
1204    sp<NotificationClient> keep(this);
1205    mAudioFlinger->removeNotificationClient(mPid);
1206}
1207
1208
1209// ----------------------------------------------------------------------------
1210
1211sp<IAudioRecord> AudioFlinger::openRecord(
1212        audio_io_handle_t input,
1213        uint32_t sampleRate,
1214        audio_format_t format,
1215        audio_channel_mask_t channelMask,
1216        size_t frameCount,
1217        IAudioFlinger::track_flags_t *flags,
1218        pid_t tid,
1219        int *sessionId,
1220        status_t *status)
1221{
1222    sp<RecordThread::RecordTrack> recordTrack;
1223    sp<RecordHandle> recordHandle;
1224    sp<Client> client;
1225    status_t lStatus;
1226    RecordThread *thread;
1227    size_t inFrameCount;
1228    int lSessionId;
1229
1230    // check calling permissions
1231    if (!recordingAllowed()) {
1232        lStatus = PERMISSION_DENIED;
1233        goto Exit;
1234    }
1235
1236    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1237        ALOGE("openRecord() invalid format %d", format);
1238        lStatus = BAD_VALUE;
1239        goto Exit;
1240    }
1241
1242    // add client to list
1243    { // scope for mLock
1244        Mutex::Autolock _l(mLock);
1245        thread = checkRecordThread_l(input);
1246        if (thread == NULL) {
1247            lStatus = BAD_VALUE;
1248            goto Exit;
1249        }
1250
1251        pid_t pid = IPCThreadState::self()->getCallingPid();
1252        client = registerPid_l(pid);
1253
1254        // If no audio session id is provided, create one here
1255        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1256            lSessionId = *sessionId;
1257        } else {
1258            lSessionId = nextUniqueId();
1259            if (sessionId != NULL) {
1260                *sessionId = lSessionId;
1261            }
1262        }
1263        // create new record track.
1264        // The record track uses one track in mHardwareMixerThread by convention.
1265        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1266                                                  frameCount, lSessionId, flags, tid, &lStatus);
1267    }
1268
1269    if (lStatus != NO_ERROR) {
1270        // remove local strong reference to Client before deleting the RecordTrack so that the
1271        // Client destructor is called by the TrackBase destructor with mLock held
1272        client.clear();
1273        recordTrack.clear();
1274        goto Exit;
1275    }
1276
1277    // return handle to client
1278    recordHandle = new RecordHandle(recordTrack);
1279
1280Exit:
1281    *status = lStatus;
1282    return recordHandle;
1283}
1284
1285
1286
1287// ----------------------------------------------------------------------------
1288
1289audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1290{
1291    if (!settingsAllowed()) {
1292        return 0;
1293    }
1294    Mutex::Autolock _l(mLock);
1295    return loadHwModule_l(name);
1296}
1297
1298// loadHwModule_l() must be called with AudioFlinger::mLock held
1299audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1300{
1301    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1302        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1303            ALOGW("loadHwModule() module %s already loaded", name);
1304            return mAudioHwDevs.keyAt(i);
1305        }
1306    }
1307
1308    audio_hw_device_t *dev;
1309
1310    int rc = load_audio_interface(name, &dev);
1311    if (rc) {
1312        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1313        return 0;
1314    }
1315
1316    mHardwareStatus = AUDIO_HW_INIT;
1317    rc = dev->init_check(dev);
1318    mHardwareStatus = AUDIO_HW_IDLE;
1319    if (rc) {
1320        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1321        return 0;
1322    }
1323
1324    // Check and cache this HAL's level of support for master mute and master
1325    // volume.  If this is the first HAL opened, and it supports the get
1326    // methods, use the initial values provided by the HAL as the current
1327    // master mute and volume settings.
1328
1329    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1330    {  // scope for auto-lock pattern
1331        AutoMutex lock(mHardwareLock);
1332
1333        if (0 == mAudioHwDevs.size()) {
1334            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1335            if (NULL != dev->get_master_volume) {
1336                float mv;
1337                if (OK == dev->get_master_volume(dev, &mv)) {
1338                    mMasterVolume = mv;
1339                }
1340            }
1341
1342            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1343            if (NULL != dev->get_master_mute) {
1344                bool mm;
1345                if (OK == dev->get_master_mute(dev, &mm)) {
1346                    mMasterMute = mm;
1347                }
1348            }
1349        }
1350
1351        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1352        if ((NULL != dev->set_master_volume) &&
1353            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1354            flags = static_cast<AudioHwDevice::Flags>(flags |
1355                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1356        }
1357
1358        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1359        if ((NULL != dev->set_master_mute) &&
1360            (OK == dev->set_master_mute(dev, mMasterMute))) {
1361            flags = static_cast<AudioHwDevice::Flags>(flags |
1362                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1363        }
1364
1365        mHardwareStatus = AUDIO_HW_IDLE;
1366    }
1367
1368    audio_module_handle_t handle = nextUniqueId();
1369    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1370
1371    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1372          name, dev->common.module->name, dev->common.module->id, handle);
1373
1374    return handle;
1375
1376}
1377
1378// ----------------------------------------------------------------------------
1379
1380uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1381{
1382    Mutex::Autolock _l(mLock);
1383    PlaybackThread *thread = primaryPlaybackThread_l();
1384    return thread != NULL ? thread->sampleRate() : 0;
1385}
1386
1387size_t AudioFlinger::getPrimaryOutputFrameCount()
1388{
1389    Mutex::Autolock _l(mLock);
1390    PlaybackThread *thread = primaryPlaybackThread_l();
1391    return thread != NULL ? thread->frameCountHAL() : 0;
1392}
1393
1394// ----------------------------------------------------------------------------
1395
1396status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1397{
1398    uid_t uid = IPCThreadState::self()->getCallingUid();
1399    if (uid != AID_SYSTEM) {
1400        return PERMISSION_DENIED;
1401    }
1402    Mutex::Autolock _l(mLock);
1403    if (mIsDeviceTypeKnown) {
1404        return INVALID_OPERATION;
1405    }
1406    mIsLowRamDevice = isLowRamDevice;
1407    mIsDeviceTypeKnown = true;
1408    return NO_ERROR;
1409}
1410
1411// ----------------------------------------------------------------------------
1412
1413audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1414                                           audio_devices_t *pDevices,
1415                                           uint32_t *pSamplingRate,
1416                                           audio_format_t *pFormat,
1417                                           audio_channel_mask_t *pChannelMask,
1418                                           uint32_t *pLatencyMs,
1419                                           audio_output_flags_t flags,
1420                                           const audio_offload_info_t *offloadInfo)
1421{
1422    PlaybackThread *thread = NULL;
1423    struct audio_config config;
1424    memset(&config, 0, sizeof(config));
1425    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1426    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1427    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1428    if (offloadInfo != NULL) {
1429        config.offload_info = *offloadInfo;
1430    }
1431
1432    audio_stream_out_t *outStream = NULL;
1433    AudioHwDevice *outHwDev;
1434
1435    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1436              module,
1437              (pDevices != NULL) ? *pDevices : 0,
1438              config.sample_rate,
1439              config.format,
1440              config.channel_mask,
1441              flags);
1442    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1443          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1444
1445    if (pDevices == NULL || *pDevices == 0) {
1446        return 0;
1447    }
1448
1449    Mutex::Autolock _l(mLock);
1450
1451    outHwDev = findSuitableHwDev_l(module, *pDevices);
1452    if (outHwDev == NULL)
1453        return 0;
1454
1455    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1456    audio_io_handle_t id = nextUniqueId();
1457
1458    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1459
1460    status_t status = hwDevHal->open_output_stream(hwDevHal,
1461                                          id,
1462                                          *pDevices,
1463                                          (audio_output_flags_t)flags,
1464                                          &config,
1465                                          &outStream);
1466
1467    mHardwareStatus = AUDIO_HW_IDLE;
1468    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1469            "Channels %x, status %d",
1470            outStream,
1471            config.sample_rate,
1472            config.format,
1473            config.channel_mask,
1474            status);
1475
1476    if (status == NO_ERROR && outStream != NULL) {
1477        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1478
1479        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1480            thread = new OffloadThread(this, output, id, *pDevices);
1481            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1482        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1483            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1484            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1485            thread = new DirectOutputThread(this, output, id, *pDevices);
1486            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1487        } else {
1488            thread = new MixerThread(this, output, id, *pDevices);
1489            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1490        }
1491        mPlaybackThreads.add(id, thread);
1492
1493        if (pSamplingRate != NULL) {
1494            *pSamplingRate = config.sample_rate;
1495        }
1496        if (pFormat != NULL) {
1497            *pFormat = config.format;
1498        }
1499        if (pChannelMask != NULL) {
1500            *pChannelMask = config.channel_mask;
1501        }
1502        if (pLatencyMs != NULL) {
1503            *pLatencyMs = thread->latency();
1504        }
1505
1506        // notify client processes of the new output creation
1507        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1508
1509        // the first primary output opened designates the primary hw device
1510        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1511            ALOGI("Using module %d has the primary audio interface", module);
1512            mPrimaryHardwareDev = outHwDev;
1513
1514            AutoMutex lock(mHardwareLock);
1515            mHardwareStatus = AUDIO_HW_SET_MODE;
1516            hwDevHal->set_mode(hwDevHal, mMode);
1517            mHardwareStatus = AUDIO_HW_IDLE;
1518        }
1519        return id;
1520    }
1521
1522    return 0;
1523}
1524
1525audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1526        audio_io_handle_t output2)
1527{
1528    Mutex::Autolock _l(mLock);
1529    MixerThread *thread1 = checkMixerThread_l(output1);
1530    MixerThread *thread2 = checkMixerThread_l(output2);
1531
1532    if (thread1 == NULL || thread2 == NULL) {
1533        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1534                output2);
1535        return 0;
1536    }
1537
1538    audio_io_handle_t id = nextUniqueId();
1539    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1540    thread->addOutputTrack(thread2);
1541    mPlaybackThreads.add(id, thread);
1542    // notify client processes of the new output creation
1543    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1544    return id;
1545}
1546
1547status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1548{
1549    return closeOutput_nonvirtual(output);
1550}
1551
1552status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1553{
1554    // keep strong reference on the playback thread so that
1555    // it is not destroyed while exit() is executed
1556    sp<PlaybackThread> thread;
1557    {
1558        Mutex::Autolock _l(mLock);
1559        thread = checkPlaybackThread_l(output);
1560        if (thread == NULL) {
1561            return BAD_VALUE;
1562        }
1563
1564        ALOGV("closeOutput() %d", output);
1565
1566        if (thread->type() == ThreadBase::MIXER) {
1567            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1568                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1569                    DuplicatingThread *dupThread =
1570                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1571                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1572
1573                }
1574            }
1575        }
1576
1577
1578        mPlaybackThreads.removeItem(output);
1579        // save all effects to the default thread
1580        if (mPlaybackThreads.size()) {
1581            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1582            if (dstThread != NULL) {
1583                // audioflinger lock is held here so the acquisition order of thread locks does not
1584                // matter
1585                Mutex::Autolock _dl(dstThread->mLock);
1586                Mutex::Autolock _sl(thread->mLock);
1587                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1588                for (size_t i = 0; i < effectChains.size(); i ++) {
1589                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1590                }
1591            }
1592        }
1593        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1594    }
1595    thread->exit();
1596    // The thread entity (active unit of execution) is no longer running here,
1597    // but the ThreadBase container still exists.
1598
1599    if (thread->type() != ThreadBase::DUPLICATING) {
1600        AudioStreamOut *out = thread->clearOutput();
1601        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1602        // from now on thread->mOutput is NULL
1603        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1604        delete out;
1605    }
1606    return NO_ERROR;
1607}
1608
1609status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1610{
1611    Mutex::Autolock _l(mLock);
1612    PlaybackThread *thread = checkPlaybackThread_l(output);
1613
1614    if (thread == NULL) {
1615        return BAD_VALUE;
1616    }
1617
1618    ALOGV("suspendOutput() %d", output);
1619    thread->suspend();
1620
1621    return NO_ERROR;
1622}
1623
1624status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1625{
1626    Mutex::Autolock _l(mLock);
1627    PlaybackThread *thread = checkPlaybackThread_l(output);
1628
1629    if (thread == NULL) {
1630        return BAD_VALUE;
1631    }
1632
1633    ALOGV("restoreOutput() %d", output);
1634
1635    thread->restore();
1636
1637    return NO_ERROR;
1638}
1639
1640audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1641                                          audio_devices_t *pDevices,
1642                                          uint32_t *pSamplingRate,
1643                                          audio_format_t *pFormat,
1644                                          audio_channel_mask_t *pChannelMask)
1645{
1646    status_t status;
1647    RecordThread *thread = NULL;
1648    struct audio_config config;
1649    memset(&config, 0, sizeof(config));
1650    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1651    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1652    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1653
1654    uint32_t reqSamplingRate = config.sample_rate;
1655    audio_format_t reqFormat = config.format;
1656    audio_channel_mask_t reqChannelMask = config.channel_mask;
1657    audio_stream_in_t *inStream = NULL;
1658    AudioHwDevice *inHwDev;
1659
1660    if (pDevices == NULL || *pDevices == 0) {
1661        return 0;
1662    }
1663
1664    Mutex::Autolock _l(mLock);
1665
1666    inHwDev = findSuitableHwDev_l(module, *pDevices);
1667    if (inHwDev == NULL)
1668        return 0;
1669
1670    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1671    audio_io_handle_t id = nextUniqueId();
1672
1673    status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1674                                        &inStream);
1675    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1676            "status %d",
1677            inStream,
1678            config.sample_rate,
1679            config.format,
1680            config.channel_mask,
1681            status);
1682
1683    // If the input could not be opened with the requested parameters and we can handle the
1684    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1685    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1686    if (status == BAD_VALUE &&
1687        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1688        (config.sample_rate <= 2 * reqSamplingRate) &&
1689        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1690        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1691        inStream = NULL;
1692        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1693    }
1694
1695    if (status == NO_ERROR && inStream != NULL) {
1696
1697#ifdef TEE_SINK
1698        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1699        // or (re-)create if current Pipe is idle and does not match the new format
1700        sp<NBAIO_Sink> teeSink;
1701        enum {
1702            TEE_SINK_NO,    // don't copy input
1703            TEE_SINK_NEW,   // copy input using a new pipe
1704            TEE_SINK_OLD,   // copy input using an existing pipe
1705        } kind;
1706        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1707                                        popcount(inStream->common.get_channels(&inStream->common)));
1708        if (!mTeeSinkInputEnabled) {
1709            kind = TEE_SINK_NO;
1710        } else if (format == Format_Invalid) {
1711            kind = TEE_SINK_NO;
1712        } else if (mRecordTeeSink == 0) {
1713            kind = TEE_SINK_NEW;
1714        } else if (mRecordTeeSink->getStrongCount() != 1) {
1715            kind = TEE_SINK_NO;
1716        } else if (format == mRecordTeeSink->format()) {
1717            kind = TEE_SINK_OLD;
1718        } else {
1719            kind = TEE_SINK_NEW;
1720        }
1721        switch (kind) {
1722        case TEE_SINK_NEW: {
1723            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1724            size_t numCounterOffers = 0;
1725            const NBAIO_Format offers[1] = {format};
1726            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1727            ALOG_ASSERT(index == 0);
1728            PipeReader *pipeReader = new PipeReader(*pipe);
1729            numCounterOffers = 0;
1730            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1731            ALOG_ASSERT(index == 0);
1732            mRecordTeeSink = pipe;
1733            mRecordTeeSource = pipeReader;
1734            teeSink = pipe;
1735            }
1736            break;
1737        case TEE_SINK_OLD:
1738            teeSink = mRecordTeeSink;
1739            break;
1740        case TEE_SINK_NO:
1741        default:
1742            break;
1743        }
1744#endif
1745
1746        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1747
1748        // Start record thread
1749        // RecordThread requires both input and output device indication to forward to audio
1750        // pre processing modules
1751        thread = new RecordThread(this,
1752                                  input,
1753                                  reqSamplingRate,
1754                                  reqChannelMask,
1755                                  id,
1756                                  primaryOutputDevice_l(),
1757                                  *pDevices
1758#ifdef TEE_SINK
1759                                  , teeSink
1760#endif
1761                                  );
1762        mRecordThreads.add(id, thread);
1763        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1764        if (pSamplingRate != NULL) {
1765            *pSamplingRate = reqSamplingRate;
1766        }
1767        if (pFormat != NULL) {
1768            *pFormat = config.format;
1769        }
1770        if (pChannelMask != NULL) {
1771            *pChannelMask = reqChannelMask;
1772        }
1773
1774        // notify client processes of the new input creation
1775        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1776        return id;
1777    }
1778
1779    return 0;
1780}
1781
1782status_t AudioFlinger::closeInput(audio_io_handle_t input)
1783{
1784    return closeInput_nonvirtual(input);
1785}
1786
1787status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1788{
1789    // keep strong reference on the record thread so that
1790    // it is not destroyed while exit() is executed
1791    sp<RecordThread> thread;
1792    {
1793        Mutex::Autolock _l(mLock);
1794        thread = checkRecordThread_l(input);
1795        if (thread == 0) {
1796            return BAD_VALUE;
1797        }
1798
1799        ALOGV("closeInput() %d", input);
1800        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1801        mRecordThreads.removeItem(input);
1802    }
1803    thread->exit();
1804    // The thread entity (active unit of execution) is no longer running here,
1805    // but the ThreadBase container still exists.
1806
1807    AudioStreamIn *in = thread->clearInput();
1808    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1809    // from now on thread->mInput is NULL
1810    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1811    delete in;
1812
1813    return NO_ERROR;
1814}
1815
1816status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1817{
1818    Mutex::Autolock _l(mLock);
1819    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1820
1821    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1822        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1823        thread->invalidateTracks(stream);
1824    }
1825
1826    return NO_ERROR;
1827}
1828
1829
1830int AudioFlinger::newAudioSessionId()
1831{
1832    return nextUniqueId();
1833}
1834
1835void AudioFlinger::acquireAudioSessionId(int audioSession)
1836{
1837    Mutex::Autolock _l(mLock);
1838    pid_t caller = IPCThreadState::self()->getCallingPid();
1839    ALOGV("acquiring %d from %d", audioSession, caller);
1840    size_t num = mAudioSessionRefs.size();
1841    for (size_t i = 0; i< num; i++) {
1842        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1843        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1844            ref->mCnt++;
1845            ALOGV(" incremented refcount to %d", ref->mCnt);
1846            return;
1847        }
1848    }
1849    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1850    ALOGV(" added new entry for %d", audioSession);
1851}
1852
1853void AudioFlinger::releaseAudioSessionId(int audioSession)
1854{
1855    Mutex::Autolock _l(mLock);
1856    pid_t caller = IPCThreadState::self()->getCallingPid();
1857    ALOGV("releasing %d from %d", audioSession, caller);
1858    size_t num = mAudioSessionRefs.size();
1859    for (size_t i = 0; i< num; i++) {
1860        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1861        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1862            ref->mCnt--;
1863            ALOGV(" decremented refcount to %d", ref->mCnt);
1864            if (ref->mCnt == 0) {
1865                mAudioSessionRefs.removeAt(i);
1866                delete ref;
1867                purgeStaleEffects_l();
1868            }
1869            return;
1870        }
1871    }
1872    ALOGW("session id %d not found for pid %d", audioSession, caller);
1873}
1874
1875void AudioFlinger::purgeStaleEffects_l() {
1876
1877    ALOGV("purging stale effects");
1878
1879    Vector< sp<EffectChain> > chains;
1880
1881    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1882        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1883        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1884            sp<EffectChain> ec = t->mEffectChains[j];
1885            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1886                chains.push(ec);
1887            }
1888        }
1889    }
1890    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1891        sp<RecordThread> t = mRecordThreads.valueAt(i);
1892        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1893            sp<EffectChain> ec = t->mEffectChains[j];
1894            chains.push(ec);
1895        }
1896    }
1897
1898    for (size_t i = 0; i < chains.size(); i++) {
1899        sp<EffectChain> ec = chains[i];
1900        int sessionid = ec->sessionId();
1901        sp<ThreadBase> t = ec->mThread.promote();
1902        if (t == 0) {
1903            continue;
1904        }
1905        size_t numsessionrefs = mAudioSessionRefs.size();
1906        bool found = false;
1907        for (size_t k = 0; k < numsessionrefs; k++) {
1908            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1909            if (ref->mSessionid == sessionid) {
1910                ALOGV(" session %d still exists for %d with %d refs",
1911                    sessionid, ref->mPid, ref->mCnt);
1912                found = true;
1913                break;
1914            }
1915        }
1916        if (!found) {
1917            Mutex::Autolock _l(t->mLock);
1918            // remove all effects from the chain
1919            while (ec->mEffects.size()) {
1920                sp<EffectModule> effect = ec->mEffects[0];
1921                effect->unPin();
1922                t->removeEffect_l(effect);
1923                if (effect->purgeHandles()) {
1924                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
1925                }
1926                AudioSystem::unregisterEffect(effect->id());
1927            }
1928        }
1929    }
1930    return;
1931}
1932
1933// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
1934AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
1935{
1936    return mPlaybackThreads.valueFor(output).get();
1937}
1938
1939// checkMixerThread_l() must be called with AudioFlinger::mLock held
1940AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
1941{
1942    PlaybackThread *thread = checkPlaybackThread_l(output);
1943    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
1944}
1945
1946// checkRecordThread_l() must be called with AudioFlinger::mLock held
1947AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
1948{
1949    return mRecordThreads.valueFor(input).get();
1950}
1951
1952uint32_t AudioFlinger::nextUniqueId()
1953{
1954    return android_atomic_inc(&mNextUniqueId);
1955}
1956
1957AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
1958{
1959    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1960        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1961        AudioStreamOut *output = thread->getOutput();
1962        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
1963            return thread;
1964        }
1965    }
1966    return NULL;
1967}
1968
1969audio_devices_t AudioFlinger::primaryOutputDevice_l() const
1970{
1971    PlaybackThread *thread = primaryPlaybackThread_l();
1972
1973    if (thread == NULL) {
1974        return 0;
1975    }
1976
1977    return thread->outDevice();
1978}
1979
1980sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
1981                                    int triggerSession,
1982                                    int listenerSession,
1983                                    sync_event_callback_t callBack,
1984                                    void *cookie)
1985{
1986    Mutex::Autolock _l(mLock);
1987
1988    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
1989    status_t playStatus = NAME_NOT_FOUND;
1990    status_t recStatus = NAME_NOT_FOUND;
1991    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1992        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
1993        if (playStatus == NO_ERROR) {
1994            return event;
1995        }
1996    }
1997    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1998        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
1999        if (recStatus == NO_ERROR) {
2000            return event;
2001        }
2002    }
2003    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2004        mPendingSyncEvents.add(event);
2005    } else {
2006        ALOGV("createSyncEvent() invalid event %d", event->type());
2007        event.clear();
2008    }
2009    return event;
2010}
2011
2012// ----------------------------------------------------------------------------
2013//  Effect management
2014// ----------------------------------------------------------------------------
2015
2016
2017status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2018{
2019    Mutex::Autolock _l(mLock);
2020    return EffectQueryNumberEffects(numEffects);
2021}
2022
2023status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2024{
2025    Mutex::Autolock _l(mLock);
2026    return EffectQueryEffect(index, descriptor);
2027}
2028
2029status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2030        effect_descriptor_t *descriptor) const
2031{
2032    Mutex::Autolock _l(mLock);
2033    return EffectGetDescriptor(pUuid, descriptor);
2034}
2035
2036
2037sp<IEffect> AudioFlinger::createEffect(
2038        effect_descriptor_t *pDesc,
2039        const sp<IEffectClient>& effectClient,
2040        int32_t priority,
2041        audio_io_handle_t io,
2042        int sessionId,
2043        status_t *status,
2044        int *id,
2045        int *enabled)
2046{
2047    status_t lStatus = NO_ERROR;
2048    sp<EffectHandle> handle;
2049    effect_descriptor_t desc;
2050
2051    pid_t pid = IPCThreadState::self()->getCallingPid();
2052    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2053            pid, effectClient.get(), priority, sessionId, io);
2054
2055    if (pDesc == NULL) {
2056        lStatus = BAD_VALUE;
2057        goto Exit;
2058    }
2059
2060    // check audio settings permission for global effects
2061    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2062        lStatus = PERMISSION_DENIED;
2063        goto Exit;
2064    }
2065
2066    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2067    // that can only be created by audio policy manager (running in same process)
2068    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2069        lStatus = PERMISSION_DENIED;
2070        goto Exit;
2071    }
2072
2073    if (io == 0) {
2074        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2075            // output must be specified by AudioPolicyManager when using session
2076            // AUDIO_SESSION_OUTPUT_STAGE
2077            lStatus = BAD_VALUE;
2078            goto Exit;
2079        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2080            // if the output returned by getOutputForEffect() is removed before we lock the
2081            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2082            // and we will exit safely
2083            io = AudioSystem::getOutputForEffect(&desc);
2084        }
2085    }
2086
2087    {
2088        Mutex::Autolock _l(mLock);
2089
2090
2091        if (!EffectIsNullUuid(&pDesc->uuid)) {
2092            // if uuid is specified, request effect descriptor
2093            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2094            if (lStatus < 0) {
2095                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2096                goto Exit;
2097            }
2098        } else {
2099            // if uuid is not specified, look for an available implementation
2100            // of the required type in effect factory
2101            if (EffectIsNullUuid(&pDesc->type)) {
2102                ALOGW("createEffect() no effect type");
2103                lStatus = BAD_VALUE;
2104                goto Exit;
2105            }
2106            uint32_t numEffects = 0;
2107            effect_descriptor_t d;
2108            d.flags = 0; // prevent compiler warning
2109            bool found = false;
2110
2111            lStatus = EffectQueryNumberEffects(&numEffects);
2112            if (lStatus < 0) {
2113                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2114                goto Exit;
2115            }
2116            for (uint32_t i = 0; i < numEffects; i++) {
2117                lStatus = EffectQueryEffect(i, &desc);
2118                if (lStatus < 0) {
2119                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2120                    continue;
2121                }
2122                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2123                    // If matching type found save effect descriptor. If the session is
2124                    // 0 and the effect is not auxiliary, continue enumeration in case
2125                    // an auxiliary version of this effect type is available
2126                    found = true;
2127                    d = desc;
2128                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2129                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2130                        break;
2131                    }
2132                }
2133            }
2134            if (!found) {
2135                lStatus = BAD_VALUE;
2136                ALOGW("createEffect() effect not found");
2137                goto Exit;
2138            }
2139            // For same effect type, chose auxiliary version over insert version if
2140            // connect to output mix (Compliance to OpenSL ES)
2141            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2142                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2143                desc = d;
2144            }
2145        }
2146
2147        // Do not allow auxiliary effects on a session different from 0 (output mix)
2148        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2149             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2150            lStatus = INVALID_OPERATION;
2151            goto Exit;
2152        }
2153
2154        // check recording permission for visualizer
2155        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2156            !recordingAllowed()) {
2157            lStatus = PERMISSION_DENIED;
2158            goto Exit;
2159        }
2160
2161        // return effect descriptor
2162        *pDesc = desc;
2163
2164        // If output is not specified try to find a matching audio session ID in one of the
2165        // output threads.
2166        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2167        // because of code checking output when entering the function.
2168        // Note: io is never 0 when creating an effect on an input
2169        if (io == 0) {
2170            // look for the thread where the specified audio session is present
2171            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2172                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2173                    io = mPlaybackThreads.keyAt(i);
2174                    break;
2175                }
2176            }
2177            if (io == 0) {
2178                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2179                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2180                        io = mRecordThreads.keyAt(i);
2181                        break;
2182                    }
2183                }
2184            }
2185            // If no output thread contains the requested session ID, default to
2186            // first output. The effect chain will be moved to the correct output
2187            // thread when a track with the same session ID is created
2188            if (io == 0 && mPlaybackThreads.size()) {
2189                io = mPlaybackThreads.keyAt(0);
2190            }
2191            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2192        }
2193        ThreadBase *thread = checkRecordThread_l(io);
2194        if (thread == NULL) {
2195            thread = checkPlaybackThread_l(io);
2196            if (thread == NULL) {
2197                ALOGE("createEffect() unknown output thread");
2198                lStatus = BAD_VALUE;
2199                goto Exit;
2200            }
2201        }
2202
2203        sp<Client> client = registerPid_l(pid);
2204
2205        // create effect on selected output thread
2206        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2207                &desc, enabled, &lStatus);
2208        if (handle != 0 && id != NULL) {
2209            *id = handle->id();
2210        }
2211    }
2212
2213Exit:
2214    *status = lStatus;
2215    return handle;
2216}
2217
2218status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2219        audio_io_handle_t dstOutput)
2220{
2221    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2222            sessionId, srcOutput, dstOutput);
2223    Mutex::Autolock _l(mLock);
2224    if (srcOutput == dstOutput) {
2225        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2226        return NO_ERROR;
2227    }
2228    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2229    if (srcThread == NULL) {
2230        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2231        return BAD_VALUE;
2232    }
2233    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2234    if (dstThread == NULL) {
2235        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2236        return BAD_VALUE;
2237    }
2238
2239    Mutex::Autolock _dl(dstThread->mLock);
2240    Mutex::Autolock _sl(srcThread->mLock);
2241    moveEffectChain_l(sessionId, srcThread, dstThread, false);
2242
2243    return NO_ERROR;
2244}
2245
2246// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2247status_t AudioFlinger::moveEffectChain_l(int sessionId,
2248                                   AudioFlinger::PlaybackThread *srcThread,
2249                                   AudioFlinger::PlaybackThread *dstThread,
2250                                   bool reRegister)
2251{
2252    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2253            sessionId, srcThread, dstThread);
2254
2255    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2256    if (chain == 0) {
2257        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2258                sessionId, srcThread);
2259        return INVALID_OPERATION;
2260    }
2261
2262    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2263    // so that a new chain is created with correct parameters when first effect is added. This is
2264    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2265    // removed.
2266    srcThread->removeEffectChain_l(chain);
2267
2268    // transfer all effects one by one so that new effect chain is created on new thread with
2269    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2270    audio_io_handle_t dstOutput = dstThread->id();
2271    sp<EffectChain> dstChain;
2272    uint32_t strategy = 0; // prevent compiler warning
2273    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2274    while (effect != 0) {
2275        srcThread->removeEffect_l(effect);
2276        dstThread->addEffect_l(effect);
2277        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2278        if (effect->state() == EffectModule::ACTIVE ||
2279                effect->state() == EffectModule::STOPPING) {
2280            effect->start();
2281        }
2282        // if the move request is not received from audio policy manager, the effect must be
2283        // re-registered with the new strategy and output
2284        if (dstChain == 0) {
2285            dstChain = effect->chain().promote();
2286            if (dstChain == 0) {
2287                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2288                srcThread->addEffect_l(effect);
2289                return NO_INIT;
2290            }
2291            strategy = dstChain->strategy();
2292        }
2293        if (reRegister) {
2294            AudioSystem::unregisterEffect(effect->id());
2295            AudioSystem::registerEffect(&effect->desc(),
2296                                        dstOutput,
2297                                        strategy,
2298                                        sessionId,
2299                                        effect->id());
2300        }
2301        effect = chain->getEffectFromId_l(0);
2302    }
2303
2304    return NO_ERROR;
2305}
2306
2307struct Entry {
2308#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2309    char mName[MAX_NAME];
2310};
2311
2312int comparEntry(const void *p1, const void *p2)
2313{
2314    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2315}
2316
2317#ifdef TEE_SINK
2318void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2319{
2320    NBAIO_Source *teeSource = source.get();
2321    if (teeSource != NULL) {
2322        // .wav rotation
2323        // There is a benign race condition if 2 threads call this simultaneously.
2324        // They would both traverse the directory, but the result would simply be
2325        // failures at unlink() which are ignored.  It's also unlikely since
2326        // normally dumpsys is only done by bugreport or from the command line.
2327        char teePath[32+256];
2328        strcpy(teePath, "/data/misc/media");
2329        size_t teePathLen = strlen(teePath);
2330        DIR *dir = opendir(teePath);
2331        teePath[teePathLen++] = '/';
2332        if (dir != NULL) {
2333#define MAX_SORT 20 // number of entries to sort
2334#define MAX_KEEP 10 // number of entries to keep
2335            struct Entry entries[MAX_SORT];
2336            size_t entryCount = 0;
2337            while (entryCount < MAX_SORT) {
2338                struct dirent de;
2339                struct dirent *result = NULL;
2340                int rc = readdir_r(dir, &de, &result);
2341                if (rc != 0) {
2342                    ALOGW("readdir_r failed %d", rc);
2343                    break;
2344                }
2345                if (result == NULL) {
2346                    break;
2347                }
2348                if (result != &de) {
2349                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2350                    break;
2351                }
2352                // ignore non .wav file entries
2353                size_t nameLen = strlen(de.d_name);
2354                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2355                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2356                    continue;
2357                }
2358                strcpy(entries[entryCount++].mName, de.d_name);
2359            }
2360            (void) closedir(dir);
2361            if (entryCount > MAX_KEEP) {
2362                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2363                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2364                    strcpy(&teePath[teePathLen], entries[i].mName);
2365                    (void) unlink(teePath);
2366                }
2367            }
2368        } else {
2369            if (fd >= 0) {
2370                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2371            }
2372        }
2373        char teeTime[16];
2374        struct timeval tv;
2375        gettimeofday(&tv, NULL);
2376        struct tm tm;
2377        localtime_r(&tv.tv_sec, &tm);
2378        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2379        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2380        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2381        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2382        if (teeFd >= 0) {
2383            char wavHeader[44];
2384            memcpy(wavHeader,
2385                "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",
2386                sizeof(wavHeader));
2387            NBAIO_Format format = teeSource->format();
2388            unsigned channelCount = Format_channelCount(format);
2389            ALOG_ASSERT(channelCount <= FCC_2);
2390            uint32_t sampleRate = Format_sampleRate(format);
2391            wavHeader[22] = channelCount;       // number of channels
2392            wavHeader[24] = sampleRate;         // sample rate
2393            wavHeader[25] = sampleRate >> 8;
2394            wavHeader[32] = channelCount * 2;   // block alignment
2395            write(teeFd, wavHeader, sizeof(wavHeader));
2396            size_t total = 0;
2397            bool firstRead = true;
2398            for (;;) {
2399#define TEE_SINK_READ 1024
2400                short buffer[TEE_SINK_READ * FCC_2];
2401                size_t count = TEE_SINK_READ;
2402                ssize_t actual = teeSource->read(buffer, count,
2403                        AudioBufferProvider::kInvalidPTS);
2404                bool wasFirstRead = firstRead;
2405                firstRead = false;
2406                if (actual <= 0) {
2407                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2408                        continue;
2409                    }
2410                    break;
2411                }
2412                ALOG_ASSERT(actual <= (ssize_t)count);
2413                write(teeFd, buffer, actual * channelCount * sizeof(short));
2414                total += actual;
2415            }
2416            lseek(teeFd, (off_t) 4, SEEK_SET);
2417            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2418            write(teeFd, &temp, sizeof(temp));
2419            lseek(teeFd, (off_t) 40, SEEK_SET);
2420            temp =  total * channelCount * sizeof(short);
2421            write(teeFd, &temp, sizeof(temp));
2422            close(teeFd);
2423            if (fd >= 0) {
2424                fdprintf(fd, "tee copied to %s\n", teePath);
2425            }
2426        } else {
2427            if (fd >= 0) {
2428                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2429            }
2430        }
2431    }
2432}
2433#endif
2434
2435// ----------------------------------------------------------------------------
2436
2437status_t AudioFlinger::onTransact(
2438        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2439{
2440    return BnAudioFlinger::onTransact(code, data, reply, flags);
2441}
2442
2443}; // namespace android
2444