AudioFlinger.cpp revision 8136cfae9c22ae8ff42eec9ed751833dda605444 
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//TODO: remove when effect offload is implemented
102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
103// we define a minimum time during which a global effect is considered enabled.
104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
105
106// ----------------------------------------------------------------------------
107
108static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
109{
110    const hw_module_t *mod;
111    int rc;
112
113    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
114    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
115                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
116    if (rc) {
117        goto out;
118    }
119    rc = audio_hw_device_open(mod, dev);
120    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
121                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
122    if (rc) {
123        goto out;
124    }
125    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
126        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
127        rc = BAD_VALUE;
128        goto out;
129    }
130    return 0;
131
132out:
133    *dev = NULL;
134    return rc;
135}
136
137// ----------------------------------------------------------------------------
138
139AudioFlinger::AudioFlinger()
140    : BnAudioFlinger(),
141      mPrimaryHardwareDev(NULL),
142      mHardwareStatus(AUDIO_HW_IDLE),
143      mMasterVolume(1.0f),
144      mMasterMute(false),
145      mNextUniqueId(1),
146      mMode(AUDIO_MODE_INVALID),
147      mBtNrecIsOff(false),
148      mIsLowRamDevice(true),
149      mIsDeviceTypeKnown(false),
150      mGlobalEffectEnableTime(0)
151{
152    getpid_cached = getpid();
153    char value[PROPERTY_VALUE_MAX];
154    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
155    if (doLog) {
156        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
157    }
158#ifdef TEE_SINK
159    (void) property_get("ro.debuggable", value, "0");
160    int debuggable = atoi(value);
161    int teeEnabled = 0;
162    if (debuggable) {
163        (void) property_get("af.tee", value, "0");
164        teeEnabled = atoi(value);
165    }
166    if (teeEnabled & 1) {
167        mTeeSinkInputEnabled = true;
168    }
169    if (teeEnabled & 2) {
170        mTeeSinkOutputEnabled = true;
171    }
172    if (teeEnabled & 4) {
173        mTeeSinkTrackEnabled = true;
174    }
175#endif
176}
177
178void AudioFlinger::onFirstRef()
179{
180    int rc = 0;
181
182    Mutex::Autolock _l(mLock);
183
184    /* TODO: move all this work into an Init() function */
185    char val_str[PROPERTY_VALUE_MAX] = { 0 };
186    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
187        uint32_t int_val;
188        if (1 == sscanf(val_str, "%u", &int_val)) {
189            mStandbyTimeInNsecs = milliseconds(int_val);
190            ALOGI("Using %u mSec as standby time.", int_val);
191        } else {
192            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
193            ALOGI("Using default %u mSec as standby time.",
194                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
195        }
196    }
197
198    mMode = AUDIO_MODE_NORMAL;
199}
200
201AudioFlinger::~AudioFlinger()
202{
203    while (!mRecordThreads.isEmpty()) {
204        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
205        closeInput_nonvirtual(mRecordThreads.keyAt(0));
206    }
207    while (!mPlaybackThreads.isEmpty()) {
208        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
209        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
210    }
211
212    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
213        // no mHardwareLock needed, as there are no other references to this
214        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
215        delete mAudioHwDevs.valueAt(i);
216    }
217}
218
219static const char * const audio_interfaces[] = {
220    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
221    AUDIO_HARDWARE_MODULE_ID_A2DP,
222    AUDIO_HARDWARE_MODULE_ID_USB,
223};
224#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
225
226AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
227        audio_module_handle_t module,
228        audio_devices_t devices)
229{
230    // if module is 0, the request comes from an old policy manager and we should load
231    // well known modules
232    if (module == 0) {
233        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
234        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
235            loadHwModule_l(audio_interfaces[i]);
236        }
237        // then try to find a module supporting the requested device.
238        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
239            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
240            audio_hw_device_t *dev = audioHwDevice->hwDevice();
241            if ((dev->get_supported_devices != NULL) &&
242                    (dev->get_supported_devices(dev) & devices) == devices)
243                return audioHwDevice;
244        }
245    } else {
246        // check a match for the requested module handle
247        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
248        if (audioHwDevice != NULL) {
249            return audioHwDevice;
250        }
251    }
252
253    return NULL;
254}
255
256void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
257{
258    const size_t SIZE = 256;
259    char buffer[SIZE];
260    String8 result;
261
262    result.append("Clients:\n");
263    for (size_t i = 0; i < mClients.size(); ++i) {
264        sp<Client> client = mClients.valueAt(i).promote();
265        if (client != 0) {
266            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
267            result.append(buffer);
268        }
269    }
270
271    result.append("Global session refs:\n");
272    result.append(" session pid count\n");
273    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
274        AudioSessionRef *r = mAudioSessionRefs[i];
275        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
276        result.append(buffer);
277    }
278    write(fd, result.string(), result.size());
279}
280
281
282void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
283{
284    const size_t SIZE = 256;
285    char buffer[SIZE];
286    String8 result;
287    hardware_call_state hardwareStatus = mHardwareStatus;
288
289    snprintf(buffer, SIZE, "Hardware status: %d\n"
290                           "Standby Time mSec: %u\n",
291                            hardwareStatus,
292                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
293    result.append(buffer);
294    write(fd, result.string(), result.size());
295}
296
297void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
298{
299    const size_t SIZE = 256;
300    char buffer[SIZE];
301    String8 result;
302    snprintf(buffer, SIZE, "Permission Denial: "
303            "can't dump AudioFlinger from pid=%d, uid=%d\n",
304            IPCThreadState::self()->getCallingPid(),
305            IPCThreadState::self()->getCallingUid());
306    result.append(buffer);
307    write(fd, result.string(), result.size());
308}
309
310bool AudioFlinger::dumpTryLock(Mutex& mutex)
311{
312    bool locked = false;
313    for (int i = 0; i < kDumpLockRetries; ++i) {
314        if (mutex.tryLock() == NO_ERROR) {
315            locked = true;
316            break;
317        }
318        usleep(kDumpLockSleepUs);
319    }
320    return locked;
321}
322
323status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
324{
325    if (!dumpAllowed()) {
326        dumpPermissionDenial(fd, args);
327    } else {
328        // get state of hardware lock
329        bool hardwareLocked = dumpTryLock(mHardwareLock);
330        if (!hardwareLocked) {
331            String8 result(kHardwareLockedString);
332            write(fd, result.string(), result.size());
333        } else {
334            mHardwareLock.unlock();
335        }
336
337        bool locked = dumpTryLock(mLock);
338
339        // failed to lock - AudioFlinger is probably deadlocked
340        if (!locked) {
341            String8 result(kDeadlockedString);
342            write(fd, result.string(), result.size());
343        }
344
345        dumpClients(fd, args);
346        dumpInternals(fd, args);
347
348        // dump playback threads
349        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
350            mPlaybackThreads.valueAt(i)->dump(fd, args);
351        }
352
353        // dump record threads
354        for (size_t i = 0; i < mRecordThreads.size(); i++) {
355            mRecordThreads.valueAt(i)->dump(fd, args);
356        }
357
358        // dump all hardware devs
359        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
360            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
361            dev->dump(dev, fd);
362        }
363
364#ifdef TEE_SINK
365        // dump the serially shared record tee sink
366        if (mRecordTeeSource != 0) {
367            dumpTee(fd, mRecordTeeSource);
368        }
369#endif
370
371        if (locked) {
372            mLock.unlock();
373        }
374
375        // append a copy of media.log here by forwarding fd to it, but don't attempt
376        // to lookup the service if it's not running, as it will block for a second
377        if (mLogMemoryDealer != 0) {
378            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
379            if (binder != 0) {
380                fdprintf(fd, "\nmedia.log:\n");
381                Vector<String16> args;
382                binder->dump(fd, args);
383            }
384        }
385    }
386    return NO_ERROR;
387}
388
389sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
390{
391    // If pid is already in the mClients wp<> map, then use that entry
392    // (for which promote() is always != 0), otherwise create a new entry and Client.
393    sp<Client> client = mClients.valueFor(pid).promote();
394    if (client == 0) {
395        client = new Client(this, pid);
396        mClients.add(pid, client);
397    }
398
399    return client;
400}
401
402sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
403{
404    if (mLogMemoryDealer == 0) {
405        return new NBLog::Writer();
406    }
407    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
408    sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
409    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
410    if (binder != 0) {
411        interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
412    }
413    return writer;
414}
415
416void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
417{
418    if (writer == 0) {
419        return;
420    }
421    sp<IMemory> iMemory(writer->getIMemory());
422    if (iMemory == 0) {
423        return;
424    }
425    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
426    if (binder != 0) {
427        interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
428        // Now the media.log remote reference to IMemory is gone.
429        // When our last local reference to IMemory also drops to zero,
430        // the IMemory destructor will deallocate the region from mMemoryDealer.
431    }
432}
433
434// IAudioFlinger interface
435
436
437sp<IAudioTrack> AudioFlinger::createTrack(
438        audio_stream_type_t streamType,
439        uint32_t sampleRate,
440        audio_format_t format,
441        audio_channel_mask_t channelMask,
442        size_t frameCount,
443        IAudioFlinger::track_flags_t *flags,
444        const sp<IMemory>& sharedBuffer,
445        audio_io_handle_t output,
446        pid_t tid,
447        int *sessionId,
448        String8& name,
449        status_t *status)
450{
451    sp<PlaybackThread::Track> track;
452    sp<TrackHandle> trackHandle;
453    sp<Client> client;
454    status_t lStatus;
455    int lSessionId;
456
457    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
458    // but if someone uses binder directly they could bypass that and cause us to crash
459    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
460        ALOGE("createTrack() invalid stream type %d", streamType);
461        lStatus = BAD_VALUE;
462        goto Exit;
463    }
464
465    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
466    // and we don't yet support 8.24 or 32-bit PCM
467    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
468        ALOGE("createTrack() invalid format %d", format);
469        lStatus = BAD_VALUE;
470        goto Exit;
471    }
472
473    {
474        Mutex::Autolock _l(mLock);
475        PlaybackThread *thread = checkPlaybackThread_l(output);
476        PlaybackThread *effectThread = NULL;
477        if (thread == NULL) {
478            ALOGE("no playback thread found for output handle %d", output);
479            lStatus = BAD_VALUE;
480            goto Exit;
481        }
482
483        pid_t pid = IPCThreadState::self()->getCallingPid();
484        client = registerPid_l(pid);
485
486        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
487        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
488            // check if an effect chain with the same session ID is present on another
489            // output thread and move it here.
490            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
491                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
492                if (mPlaybackThreads.keyAt(i) != output) {
493                    uint32_t sessions = t->hasAudioSession(*sessionId);
494                    if (sessions & PlaybackThread::EFFECT_SESSION) {
495                        effectThread = t.get();
496                        break;
497                    }
498                }
499            }
500            lSessionId = *sessionId;
501        } else {
502            // if no audio session id is provided, create one here
503            lSessionId = nextUniqueId();
504            if (sessionId != NULL) {
505                *sessionId = lSessionId;
506            }
507        }
508        ALOGV("createTrack() lSessionId: %d", lSessionId);
509
510        track = thread->createTrack_l(client, streamType, sampleRate, format,
511                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus);
512        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
513
514        // move effect chain to this output thread if an effect on same session was waiting
515        // for a track to be created
516        if (lStatus == NO_ERROR && effectThread != NULL) {
517            // no risk of deadlock because AudioFlinger::mLock is held
518            Mutex::Autolock _dl(thread->mLock);
519            Mutex::Autolock _sl(effectThread->mLock);
520            moveEffectChain_l(lSessionId, effectThread, thread, true);
521        }
522
523        // Look for sync events awaiting for a session to be used.
524        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
525            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
526                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
527                    if (lStatus == NO_ERROR) {
528                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
529                    } else {
530                        mPendingSyncEvents[i]->cancel();
531                    }
532                    mPendingSyncEvents.removeAt(i);
533                    i--;
534                }
535            }
536        }
537
538    }
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    *status = lStatus;
554    return trackHandle;
555}
556
557uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
558{
559    Mutex::Autolock _l(mLock);
560    PlaybackThread *thread = checkPlaybackThread_l(output);
561    if (thread == NULL) {
562        ALOGW("sampleRate() unknown thread %d", output);
563        return 0;
564    }
565    return thread->sampleRate();
566}
567
568int AudioFlinger::channelCount(audio_io_handle_t output) const
569{
570    Mutex::Autolock _l(mLock);
571    PlaybackThread *thread = checkPlaybackThread_l(output);
572    if (thread == NULL) {
573        ALOGW("channelCount() unknown thread %d", output);
574        return 0;
575    }
576    return thread->channelCount();
577}
578
579audio_format_t AudioFlinger::format(audio_io_handle_t output) const
580{
581    Mutex::Autolock _l(mLock);
582    PlaybackThread *thread = checkPlaybackThread_l(output);
583    if (thread == NULL) {
584        ALOGW("format() unknown thread %d", output);
585        return AUDIO_FORMAT_INVALID;
586    }
587    return thread->format();
588}
589
590size_t AudioFlinger::frameCount(audio_io_handle_t output) const
591{
592    Mutex::Autolock _l(mLock);
593    PlaybackThread *thread = checkPlaybackThread_l(output);
594    if (thread == NULL) {
595        ALOGW("frameCount() unknown thread %d", output);
596        return 0;
597    }
598    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
599    //       should examine all callers and fix them to handle smaller counts
600    return thread->frameCount();
601}
602
603uint32_t AudioFlinger::latency(audio_io_handle_t output) const
604{
605    Mutex::Autolock _l(mLock);
606    PlaybackThread *thread = checkPlaybackThread_l(output);
607    if (thread == NULL) {
608        ALOGW("latency(): no playback thread found for output handle %d", output);
609        return 0;
610    }
611    return thread->latency();
612}
613
614status_t AudioFlinger::setMasterVolume(float value)
615{
616    status_t ret = initCheck();
617    if (ret != NO_ERROR) {
618        return ret;
619    }
620
621    // check calling permissions
622    if (!settingsAllowed()) {
623        return PERMISSION_DENIED;
624    }
625
626    Mutex::Autolock _l(mLock);
627    mMasterVolume = value;
628
629    // Set master volume in the HALs which support it.
630    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
631        AutoMutex lock(mHardwareLock);
632        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
633
634        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
635        if (dev->canSetMasterVolume()) {
636            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
637        }
638        mHardwareStatus = AUDIO_HW_IDLE;
639    }
640
641    // Now set the master volume in each playback thread.  Playback threads
642    // assigned to HALs which do not have master volume support will apply
643    // master volume during the mix operation.  Threads with HALs which do
644    // support master volume will simply ignore the setting.
645    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
646        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
647
648    return NO_ERROR;
649}
650
651status_t AudioFlinger::setMode(audio_mode_t mode)
652{
653    status_t ret = initCheck();
654    if (ret != NO_ERROR) {
655        return ret;
656    }
657
658    // check calling permissions
659    if (!settingsAllowed()) {
660        return PERMISSION_DENIED;
661    }
662    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
663        ALOGW("Illegal value: setMode(%d)", mode);
664        return BAD_VALUE;
665    }
666
667    { // scope for the lock
668        AutoMutex lock(mHardwareLock);
669        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
670        mHardwareStatus = AUDIO_HW_SET_MODE;
671        ret = dev->set_mode(dev, mode);
672        mHardwareStatus = AUDIO_HW_IDLE;
673    }
674
675    if (NO_ERROR == ret) {
676        Mutex::Autolock _l(mLock);
677        mMode = mode;
678        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
679            mPlaybackThreads.valueAt(i)->setMode(mode);
680    }
681
682    return ret;
683}
684
685status_t AudioFlinger::setMicMute(bool state)
686{
687    status_t ret = initCheck();
688    if (ret != NO_ERROR) {
689        return ret;
690    }
691
692    // check calling permissions
693    if (!settingsAllowed()) {
694        return PERMISSION_DENIED;
695    }
696
697    AutoMutex lock(mHardwareLock);
698    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
699    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
700    ret = dev->set_mic_mute(dev, state);
701    mHardwareStatus = AUDIO_HW_IDLE;
702    return ret;
703}
704
705bool AudioFlinger::getMicMute() const
706{
707    status_t ret = initCheck();
708    if (ret != NO_ERROR) {
709        return false;
710    }
711
712    bool state = AUDIO_MODE_INVALID;
713    AutoMutex lock(mHardwareLock);
714    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
715    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
716    dev->get_mic_mute(dev, &state);
717    mHardwareStatus = AUDIO_HW_IDLE;
718    return state;
719}
720
721status_t AudioFlinger::setMasterMute(bool muted)
722{
723    status_t ret = initCheck();
724    if (ret != NO_ERROR) {
725        return ret;
726    }
727
728    // check calling permissions
729    if (!settingsAllowed()) {
730        return PERMISSION_DENIED;
731    }
732
733    Mutex::Autolock _l(mLock);
734    mMasterMute = muted;
735
736    // Set master mute in the HALs which support it.
737    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
738        AutoMutex lock(mHardwareLock);
739        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
740
741        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
742        if (dev->canSetMasterMute()) {
743            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
744        }
745        mHardwareStatus = AUDIO_HW_IDLE;
746    }
747
748    // Now set the master mute in each playback thread.  Playback threads
749    // assigned to HALs which do not have master mute support will apply master
750    // mute during the mix operation.  Threads with HALs which do support master
751    // mute will simply ignore the setting.
752    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
753        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
754
755    return NO_ERROR;
756}
757
758float AudioFlinger::masterVolume() const
759{
760    Mutex::Autolock _l(mLock);
761    return masterVolume_l();
762}
763
764bool AudioFlinger::masterMute() const
765{
766    Mutex::Autolock _l(mLock);
767    return masterMute_l();
768}
769
770float AudioFlinger::masterVolume_l() const
771{
772    return mMasterVolume;
773}
774
775bool AudioFlinger::masterMute_l() const
776{
777    return mMasterMute;
778}
779
780status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
781        audio_io_handle_t output)
782{
783    // check calling permissions
784    if (!settingsAllowed()) {
785        return PERMISSION_DENIED;
786    }
787
788    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
789        ALOGE("setStreamVolume() invalid stream %d", stream);
790        return BAD_VALUE;
791    }
792
793    AutoMutex lock(mLock);
794    PlaybackThread *thread = NULL;
795    if (output) {
796        thread = checkPlaybackThread_l(output);
797        if (thread == NULL) {
798            return BAD_VALUE;
799        }
800    }
801
802    mStreamTypes[stream].volume = value;
803
804    if (thread == NULL) {
805        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
806            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
807        }
808    } else {
809        thread->setStreamVolume(stream, value);
810    }
811
812    return NO_ERROR;
813}
814
815status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
816{
817    // check calling permissions
818    if (!settingsAllowed()) {
819        return PERMISSION_DENIED;
820    }
821
822    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
823        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
824        ALOGE("setStreamMute() invalid stream %d", stream);
825        return BAD_VALUE;
826    }
827
828    AutoMutex lock(mLock);
829    mStreamTypes[stream].mute = muted;
830    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
831        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
832
833    return NO_ERROR;
834}
835
836float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
837{
838    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
839        return 0.0f;
840    }
841
842    AutoMutex lock(mLock);
843    float volume;
844    if (output) {
845        PlaybackThread *thread = checkPlaybackThread_l(output);
846        if (thread == NULL) {
847            return 0.0f;
848        }
849        volume = thread->streamVolume(stream);
850    } else {
851        volume = streamVolume_l(stream);
852    }
853
854    return volume;
855}
856
857bool AudioFlinger::streamMute(audio_stream_type_t stream) const
858{
859    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
860        return true;
861    }
862
863    AutoMutex lock(mLock);
864    return streamMute_l(stream);
865}
866
867status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
868{
869    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
870            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
871
872    // check calling permissions
873    if (!settingsAllowed()) {
874        return PERMISSION_DENIED;
875    }
876
877    // ioHandle == 0 means the parameters are global to the audio hardware interface
878    if (ioHandle == 0) {
879        Mutex::Autolock _l(mLock);
880        status_t final_result = NO_ERROR;
881        {
882            AutoMutex lock(mHardwareLock);
883            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
884            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
885                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
886                status_t result = dev->set_parameters(dev, keyValuePairs.string());
887                final_result = result ?: final_result;
888            }
889            mHardwareStatus = AUDIO_HW_IDLE;
890        }
891        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
892        AudioParameter param = AudioParameter(keyValuePairs);
893        String8 value;
894        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
895            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
896            if (mBtNrecIsOff != btNrecIsOff) {
897                for (size_t i = 0; i < mRecordThreads.size(); i++) {
898                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
899                    audio_devices_t device = thread->inDevice();
900                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
901                    // collect all of the thread's session IDs
902                    KeyedVector<int, bool> ids = thread->sessionIds();
903                    // suspend effects associated with those session IDs
904                    for (size_t j = 0; j < ids.size(); ++j) {
905                        int sessionId = ids.keyAt(j);
906                        thread->setEffectSuspended(FX_IID_AEC,
907                                                   suspend,
908                                                   sessionId);
909                        thread->setEffectSuspended(FX_IID_NS,
910                                                   suspend,
911                                                   sessionId);
912                    }
913                }
914                mBtNrecIsOff = btNrecIsOff;
915            }
916        }
917        String8 screenState;
918        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
919            bool isOff = screenState == "off";
920            if (isOff != (AudioFlinger::mScreenState & 1)) {
921                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
922            }
923        }
924        return final_result;
925    }
926
927    // hold a strong ref on thread in case closeOutput() or closeInput() is called
928    // and the thread is exited once the lock is released
929    sp<ThreadBase> thread;
930    {
931        Mutex::Autolock _l(mLock);
932        thread = checkPlaybackThread_l(ioHandle);
933        if (thread == 0) {
934            thread = checkRecordThread_l(ioHandle);
935        } else if (thread == primaryPlaybackThread_l()) {
936            // indicate output device change to all input threads for pre processing
937            AudioParameter param = AudioParameter(keyValuePairs);
938            int value;
939            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
940                    (value != 0)) {
941                for (size_t i = 0; i < mRecordThreads.size(); i++) {
942                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
943                }
944            }
945        }
946    }
947    if (thread != 0) {
948        return thread->setParameters(keyValuePairs);
949    }
950    return BAD_VALUE;
951}
952
953String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
954{
955    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
956            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
957
958    Mutex::Autolock _l(mLock);
959
960    if (ioHandle == 0) {
961        String8 out_s8;
962
963        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
964            char *s;
965            {
966            AutoMutex lock(mHardwareLock);
967            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
968            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
969            s = dev->get_parameters(dev, keys.string());
970            mHardwareStatus = AUDIO_HW_IDLE;
971            }
972            out_s8 += String8(s ? s : "");
973            free(s);
974        }
975        return out_s8;
976    }
977
978    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
979    if (playbackThread != NULL) {
980        return playbackThread->getParameters(keys);
981    }
982    RecordThread *recordThread = checkRecordThread_l(ioHandle);
983    if (recordThread != NULL) {
984        return recordThread->getParameters(keys);
985    }
986    return String8("");
987}
988
989size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
990        audio_channel_mask_t channelMask) const
991{
992    status_t ret = initCheck();
993    if (ret != NO_ERROR) {
994        return 0;
995    }
996
997    AutoMutex lock(mHardwareLock);
998    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
999    struct audio_config config;
1000    memset(&config, 0, sizeof(config));
1001    config.sample_rate = sampleRate;
1002    config.channel_mask = channelMask;
1003    config.format = format;
1004
1005    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1006    size_t size = dev->get_input_buffer_size(dev, &config);
1007    mHardwareStatus = AUDIO_HW_IDLE;
1008    return size;
1009}
1010
1011unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1012{
1013    Mutex::Autolock _l(mLock);
1014
1015    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1016    if (recordThread != NULL) {
1017        return recordThread->getInputFramesLost();
1018    }
1019    return 0;
1020}
1021
1022status_t AudioFlinger::setVoiceVolume(float value)
1023{
1024    status_t ret = initCheck();
1025    if (ret != NO_ERROR) {
1026        return ret;
1027    }
1028
1029    // check calling permissions
1030    if (!settingsAllowed()) {
1031        return PERMISSION_DENIED;
1032    }
1033
1034    AutoMutex lock(mHardwareLock);
1035    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1036    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1037    ret = dev->set_voice_volume(dev, value);
1038    mHardwareStatus = AUDIO_HW_IDLE;
1039
1040    return ret;
1041}
1042
1043status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1044        audio_io_handle_t output) const
1045{
1046    status_t status;
1047
1048    Mutex::Autolock _l(mLock);
1049
1050    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1051    if (playbackThread != NULL) {
1052        return playbackThread->getRenderPosition(halFrames, dspFrames);
1053    }
1054
1055    return BAD_VALUE;
1056}
1057
1058void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1059{
1060
1061    Mutex::Autolock _l(mLock);
1062
1063    pid_t pid = IPCThreadState::self()->getCallingPid();
1064    if (mNotificationClients.indexOfKey(pid) < 0) {
1065        sp<NotificationClient> notificationClient = new NotificationClient(this,
1066                                                                            client,
1067                                                                            pid);
1068        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1069
1070        mNotificationClients.add(pid, notificationClient);
1071
1072        sp<IBinder> binder = client->asBinder();
1073        binder->linkToDeath(notificationClient);
1074
1075        // the config change is always sent from playback or record threads to avoid deadlock
1076        // with AudioSystem::gLock
1077        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1078            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1079        }
1080
1081        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1082            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1083        }
1084    }
1085}
1086
1087void AudioFlinger::removeNotificationClient(pid_t pid)
1088{
1089    Mutex::Autolock _l(mLock);
1090
1091    mNotificationClients.removeItem(pid);
1092
1093    ALOGV("%d died, releasing its sessions", pid);
1094    size_t num = mAudioSessionRefs.size();
1095    bool removed = false;
1096    for (size_t i = 0; i< num; ) {
1097        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1098        ALOGV(" pid %d @ %d", ref->mPid, i);
1099        if (ref->mPid == pid) {
1100            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1101            mAudioSessionRefs.removeAt(i);
1102            delete ref;
1103            removed = true;
1104            num--;
1105        } else {
1106            i++;
1107        }
1108    }
1109    if (removed) {
1110        purgeStaleEffects_l();
1111    }
1112}
1113
1114// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1115void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1116{
1117    size_t size = mNotificationClients.size();
1118    for (size_t i = 0; i < size; i++) {
1119        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1120                                                                               param2);
1121    }
1122}
1123
1124// removeClient_l() must be called with AudioFlinger::mLock held
1125void AudioFlinger::removeClient_l(pid_t pid)
1126{
1127    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1128            IPCThreadState::self()->getCallingPid());
1129    mClients.removeItem(pid);
1130}
1131
1132// getEffectThread_l() must be called with AudioFlinger::mLock held
1133sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1134{
1135    sp<PlaybackThread> thread;
1136
1137    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1138        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1139            ALOG_ASSERT(thread == 0);
1140            thread = mPlaybackThreads.valueAt(i);
1141        }
1142    }
1143
1144    return thread;
1145}
1146
1147
1148
1149// ----------------------------------------------------------------------------
1150
1151AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1152    :   RefBase(),
1153        mAudioFlinger(audioFlinger),
1154        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1155        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1156        mPid(pid),
1157        mTimedTrackCount(0)
1158{
1159    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1160}
1161
1162// Client destructor must be called with AudioFlinger::mLock held
1163AudioFlinger::Client::~Client()
1164{
1165    mAudioFlinger->removeClient_l(mPid);
1166}
1167
1168sp<MemoryDealer> AudioFlinger::Client::heap() const
1169{
1170    return mMemoryDealer;
1171}
1172
1173// Reserve one of the limited slots for a timed audio track associated
1174// with this client
1175bool AudioFlinger::Client::reserveTimedTrack()
1176{
1177    const int kMaxTimedTracksPerClient = 4;
1178
1179    Mutex::Autolock _l(mTimedTrackLock);
1180
1181    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1182        ALOGW("can not create timed track - pid %d has exceeded the limit",
1183             mPid);
1184        return false;
1185    }
1186
1187    mTimedTrackCount++;
1188    return true;
1189}
1190
1191// Release a slot for a timed audio track
1192void AudioFlinger::Client::releaseTimedTrack()
1193{
1194    Mutex::Autolock _l(mTimedTrackLock);
1195    mTimedTrackCount--;
1196}
1197
1198// ----------------------------------------------------------------------------
1199
1200AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1201                                                     const sp<IAudioFlingerClient>& client,
1202                                                     pid_t pid)
1203    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1204{
1205}
1206
1207AudioFlinger::NotificationClient::~NotificationClient()
1208{
1209}
1210
1211void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1212{
1213    sp<NotificationClient> keep(this);
1214    mAudioFlinger->removeNotificationClient(mPid);
1215}
1216
1217
1218// ----------------------------------------------------------------------------
1219
1220static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1221    return audio_is_remote_submix_device(inDevice);
1222}
1223
1224sp<IAudioRecord> AudioFlinger::openRecord(
1225        audio_io_handle_t input,
1226        uint32_t sampleRate,
1227        audio_format_t format,
1228        audio_channel_mask_t channelMask,
1229        size_t frameCount,
1230        IAudioFlinger::track_flags_t *flags,
1231        pid_t tid,
1232        int *sessionId,
1233        status_t *status)
1234{
1235    sp<RecordThread::RecordTrack> recordTrack;
1236    sp<RecordHandle> recordHandle;
1237    sp<Client> client;
1238    status_t lStatus;
1239    RecordThread *thread;
1240    size_t inFrameCount;
1241    int lSessionId;
1242
1243    // check calling permissions
1244    if (!recordingAllowed()) {
1245        lStatus = PERMISSION_DENIED;
1246        goto Exit;
1247    }
1248
1249    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1250        ALOGE("openRecord() invalid format %d", format);
1251        lStatus = BAD_VALUE;
1252        goto Exit;
1253    }
1254
1255    // add client to list
1256    { // scope for mLock
1257        Mutex::Autolock _l(mLock);
1258        thread = checkRecordThread_l(input);
1259        if (thread == NULL) {
1260            lStatus = BAD_VALUE;
1261            goto Exit;
1262        }
1263
1264        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1265                && !captureAudioOutputAllowed()) {
1266            lStatus = PERMISSION_DENIED;
1267            goto Exit;
1268        }
1269
1270        pid_t pid = IPCThreadState::self()->getCallingPid();
1271        client = registerPid_l(pid);
1272
1273        // If no audio session id is provided, create one here
1274        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1275            lSessionId = *sessionId;
1276        } else {
1277            lSessionId = nextUniqueId();
1278            if (sessionId != NULL) {
1279                *sessionId = lSessionId;
1280            }
1281        }
1282        // create new record track.
1283        // The record track uses one track in mHardwareMixerThread by convention.
1284        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1285                                                  frameCount, lSessionId, flags, tid, &lStatus);
1286    }
1287
1288    if (lStatus != NO_ERROR) {
1289        // remove local strong reference to Client before deleting the RecordTrack so that the
1290        // Client destructor is called by the TrackBase destructor with mLock held
1291        client.clear();
1292        recordTrack.clear();
1293        goto Exit;
1294    }
1295
1296    // return handle to client
1297    recordHandle = new RecordHandle(recordTrack);
1298
1299Exit:
1300    *status = lStatus;
1301    return recordHandle;
1302}
1303
1304
1305
1306// ----------------------------------------------------------------------------
1307
1308audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1309{
1310    if (!settingsAllowed()) {
1311        return 0;
1312    }
1313    Mutex::Autolock _l(mLock);
1314    return loadHwModule_l(name);
1315}
1316
1317// loadHwModule_l() must be called with AudioFlinger::mLock held
1318audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1319{
1320    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1321        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1322            ALOGW("loadHwModule() module %s already loaded", name);
1323            return mAudioHwDevs.keyAt(i);
1324        }
1325    }
1326
1327    audio_hw_device_t *dev;
1328
1329    int rc = load_audio_interface(name, &dev);
1330    if (rc) {
1331        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1332        return 0;
1333    }
1334
1335    mHardwareStatus = AUDIO_HW_INIT;
1336    rc = dev->init_check(dev);
1337    mHardwareStatus = AUDIO_HW_IDLE;
1338    if (rc) {
1339        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1340        return 0;
1341    }
1342
1343    // Check and cache this HAL's level of support for master mute and master
1344    // volume.  If this is the first HAL opened, and it supports the get
1345    // methods, use the initial values provided by the HAL as the current
1346    // master mute and volume settings.
1347
1348    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1349    {  // scope for auto-lock pattern
1350        AutoMutex lock(mHardwareLock);
1351
1352        if (0 == mAudioHwDevs.size()) {
1353            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1354            if (NULL != dev->get_master_volume) {
1355                float mv;
1356                if (OK == dev->get_master_volume(dev, &mv)) {
1357                    mMasterVolume = mv;
1358                }
1359            }
1360
1361            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1362            if (NULL != dev->get_master_mute) {
1363                bool mm;
1364                if (OK == dev->get_master_mute(dev, &mm)) {
1365                    mMasterMute = mm;
1366                }
1367            }
1368        }
1369
1370        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1371        if ((NULL != dev->set_master_volume) &&
1372            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1373            flags = static_cast<AudioHwDevice::Flags>(flags |
1374                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1375        }
1376
1377        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1378        if ((NULL != dev->set_master_mute) &&
1379            (OK == dev->set_master_mute(dev, mMasterMute))) {
1380            flags = static_cast<AudioHwDevice::Flags>(flags |
1381                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1382        }
1383
1384        mHardwareStatus = AUDIO_HW_IDLE;
1385    }
1386
1387    audio_module_handle_t handle = nextUniqueId();
1388    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1389
1390    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1391          name, dev->common.module->name, dev->common.module->id, handle);
1392
1393    return handle;
1394
1395}
1396
1397// ----------------------------------------------------------------------------
1398
1399uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1400{
1401    Mutex::Autolock _l(mLock);
1402    PlaybackThread *thread = primaryPlaybackThread_l();
1403    return thread != NULL ? thread->sampleRate() : 0;
1404}
1405
1406size_t AudioFlinger::getPrimaryOutputFrameCount()
1407{
1408    Mutex::Autolock _l(mLock);
1409    PlaybackThread *thread = primaryPlaybackThread_l();
1410    return thread != NULL ? thread->frameCountHAL() : 0;
1411}
1412
1413// ----------------------------------------------------------------------------
1414
1415status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1416{
1417    uid_t uid = IPCThreadState::self()->getCallingUid();
1418    if (uid != AID_SYSTEM) {
1419        return PERMISSION_DENIED;
1420    }
1421    Mutex::Autolock _l(mLock);
1422    if (mIsDeviceTypeKnown) {
1423        return INVALID_OPERATION;
1424    }
1425    mIsLowRamDevice = isLowRamDevice;
1426    mIsDeviceTypeKnown = true;
1427    return NO_ERROR;
1428}
1429
1430// ----------------------------------------------------------------------------
1431
1432audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1433                                           audio_devices_t *pDevices,
1434                                           uint32_t *pSamplingRate,
1435                                           audio_format_t *pFormat,
1436                                           audio_channel_mask_t *pChannelMask,
1437                                           uint32_t *pLatencyMs,
1438                                           audio_output_flags_t flags,
1439                                           const audio_offload_info_t *offloadInfo)
1440{
1441    struct audio_config config;
1442    memset(&config, 0, sizeof(config));
1443    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1444    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1445    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1446    if (offloadInfo != NULL) {
1447        config.offload_info = *offloadInfo;
1448    }
1449
1450    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1451              module,
1452              (pDevices != NULL) ? *pDevices : 0,
1453              config.sample_rate,
1454              config.format,
1455              config.channel_mask,
1456              flags);
1457    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1458          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1459
1460    if (pDevices == NULL || *pDevices == 0) {
1461        return 0;
1462    }
1463
1464    Mutex::Autolock _l(mLock);
1465
1466    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1467    if (outHwDev == NULL) {
1468        return 0;
1469    }
1470
1471    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1472    audio_io_handle_t id = nextUniqueId();
1473
1474    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1475
1476    audio_stream_out_t *outStream = NULL;
1477    status_t status = hwDevHal->open_output_stream(hwDevHal,
1478                                          id,
1479                                          *pDevices,
1480                                          (audio_output_flags_t)flags,
1481                                          &config,
1482                                          &outStream);
1483
1484    mHardwareStatus = AUDIO_HW_IDLE;
1485    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1486            "Channels %x, status %d",
1487            outStream,
1488            config.sample_rate,
1489            config.format,
1490            config.channel_mask,
1491            status);
1492
1493    if (status == NO_ERROR && outStream != NULL) {
1494        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1495
1496        PlaybackThread *thread;
1497        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1498            thread = new OffloadThread(this, output, id, *pDevices);
1499            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1500        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1501            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1502            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1503            thread = new DirectOutputThread(this, output, id, *pDevices);
1504            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1505        } else {
1506            thread = new MixerThread(this, output, id, *pDevices);
1507            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1508        }
1509        mPlaybackThreads.add(id, thread);
1510
1511        if (pSamplingRate != NULL) {
1512            *pSamplingRate = config.sample_rate;
1513        }
1514        if (pFormat != NULL) {
1515            *pFormat = config.format;
1516        }
1517        if (pChannelMask != NULL) {
1518            *pChannelMask = config.channel_mask;
1519        }
1520        if (pLatencyMs != NULL) {
1521            *pLatencyMs = thread->latency();
1522        }
1523
1524        // notify client processes of the new output creation
1525        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1526
1527        // the first primary output opened designates the primary hw device
1528        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1529            ALOGI("Using module %d has the primary audio interface", module);
1530            mPrimaryHardwareDev = outHwDev;
1531
1532            AutoMutex lock(mHardwareLock);
1533            mHardwareStatus = AUDIO_HW_SET_MODE;
1534            hwDevHal->set_mode(hwDevHal, mMode);
1535            mHardwareStatus = AUDIO_HW_IDLE;
1536        }
1537        return id;
1538    }
1539
1540    return 0;
1541}
1542
1543audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1544        audio_io_handle_t output2)
1545{
1546    Mutex::Autolock _l(mLock);
1547    MixerThread *thread1 = checkMixerThread_l(output1);
1548    MixerThread *thread2 = checkMixerThread_l(output2);
1549
1550    if (thread1 == NULL || thread2 == NULL) {
1551        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1552                output2);
1553        return 0;
1554    }
1555
1556    audio_io_handle_t id = nextUniqueId();
1557    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1558    thread->addOutputTrack(thread2);
1559    mPlaybackThreads.add(id, thread);
1560    // notify client processes of the new output creation
1561    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1562    return id;
1563}
1564
1565status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1566{
1567    return closeOutput_nonvirtual(output);
1568}
1569
1570status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1571{
1572    // keep strong reference on the playback thread so that
1573    // it is not destroyed while exit() is executed
1574    sp<PlaybackThread> thread;
1575    {
1576        Mutex::Autolock _l(mLock);
1577        thread = checkPlaybackThread_l(output);
1578        if (thread == NULL) {
1579            return BAD_VALUE;
1580        }
1581
1582        ALOGV("closeOutput() %d", output);
1583
1584        if (thread->type() == ThreadBase::MIXER) {
1585            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1586                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1587                    DuplicatingThread *dupThread =
1588                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1589                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1590
1591                }
1592            }
1593        }
1594
1595
1596        mPlaybackThreads.removeItem(output);
1597        // save all effects to the default thread
1598        if (mPlaybackThreads.size()) {
1599            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1600            if (dstThread != NULL) {
1601                // audioflinger lock is held here so the acquisition order of thread locks does not
1602                // matter
1603                Mutex::Autolock _dl(dstThread->mLock);
1604                Mutex::Autolock _sl(thread->mLock);
1605                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1606                for (size_t i = 0; i < effectChains.size(); i ++) {
1607                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1608                }
1609            }
1610        }
1611        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1612    }
1613    thread->exit();
1614    // The thread entity (active unit of execution) is no longer running here,
1615    // but the ThreadBase container still exists.
1616
1617    if (thread->type() != ThreadBase::DUPLICATING) {
1618        AudioStreamOut *out = thread->clearOutput();
1619        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1620        // from now on thread->mOutput is NULL
1621        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1622        delete out;
1623    }
1624    return NO_ERROR;
1625}
1626
1627status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1628{
1629    Mutex::Autolock _l(mLock);
1630    PlaybackThread *thread = checkPlaybackThread_l(output);
1631
1632    if (thread == NULL) {
1633        return BAD_VALUE;
1634    }
1635
1636    ALOGV("suspendOutput() %d", output);
1637    thread->suspend();
1638
1639    return NO_ERROR;
1640}
1641
1642status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1643{
1644    Mutex::Autolock _l(mLock);
1645    PlaybackThread *thread = checkPlaybackThread_l(output);
1646
1647    if (thread == NULL) {
1648        return BAD_VALUE;
1649    }
1650
1651    ALOGV("restoreOutput() %d", output);
1652
1653    thread->restore();
1654
1655    return NO_ERROR;
1656}
1657
1658audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1659                                          audio_devices_t *pDevices,
1660                                          uint32_t *pSamplingRate,
1661                                          audio_format_t *pFormat,
1662                                          audio_channel_mask_t *pChannelMask)
1663{
1664    struct audio_config config;
1665    memset(&config, 0, sizeof(config));
1666    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1667    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1668    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1669
1670    uint32_t reqSamplingRate = config.sample_rate;
1671    audio_format_t reqFormat = config.format;
1672    audio_channel_mask_t reqChannelMask = config.channel_mask;
1673
1674    if (pDevices == NULL || *pDevices == 0) {
1675        return 0;
1676    }
1677
1678    Mutex::Autolock _l(mLock);
1679
1680    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1681    if (inHwDev == NULL) {
1682        return 0;
1683    }
1684
1685    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1686    audio_io_handle_t id = nextUniqueId();
1687
1688    audio_stream_in_t *inStream = NULL;
1689    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1690                                        &inStream);
1691    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1692            "status %d",
1693            inStream,
1694            config.sample_rate,
1695            config.format,
1696            config.channel_mask,
1697            status);
1698
1699    // If the input could not be opened with the requested parameters and we can handle the
1700    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1701    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1702    if (status == BAD_VALUE &&
1703        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1704        (config.sample_rate <= 2 * reqSamplingRate) &&
1705        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1706        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1707        inStream = NULL;
1708        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1709    }
1710
1711    if (status == NO_ERROR && inStream != NULL) {
1712
1713#ifdef TEE_SINK
1714        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1715        // or (re-)create if current Pipe is idle and does not match the new format
1716        sp<NBAIO_Sink> teeSink;
1717        enum {
1718            TEE_SINK_NO,    // don't copy input
1719            TEE_SINK_NEW,   // copy input using a new pipe
1720            TEE_SINK_OLD,   // copy input using an existing pipe
1721        } kind;
1722        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1723                                        popcount(inStream->common.get_channels(&inStream->common)));
1724        if (!mTeeSinkInputEnabled) {
1725            kind = TEE_SINK_NO;
1726        } else if (format == Format_Invalid) {
1727            kind = TEE_SINK_NO;
1728        } else if (mRecordTeeSink == 0) {
1729            kind = TEE_SINK_NEW;
1730        } else if (mRecordTeeSink->getStrongCount() != 1) {
1731            kind = TEE_SINK_NO;
1732        } else if (format == mRecordTeeSink->format()) {
1733            kind = TEE_SINK_OLD;
1734        } else {
1735            kind = TEE_SINK_NEW;
1736        }
1737        switch (kind) {
1738        case TEE_SINK_NEW: {
1739            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1740            size_t numCounterOffers = 0;
1741            const NBAIO_Format offers[1] = {format};
1742            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1743            ALOG_ASSERT(index == 0);
1744            PipeReader *pipeReader = new PipeReader(*pipe);
1745            numCounterOffers = 0;
1746            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1747            ALOG_ASSERT(index == 0);
1748            mRecordTeeSink = pipe;
1749            mRecordTeeSource = pipeReader;
1750            teeSink = pipe;
1751            }
1752            break;
1753        case TEE_SINK_OLD:
1754            teeSink = mRecordTeeSink;
1755            break;
1756        case TEE_SINK_NO:
1757        default:
1758            break;
1759        }
1760#endif
1761
1762        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1763
1764        // Start record thread
1765        // RecordThread requires both input and output device indication to forward to audio
1766        // pre processing modules
1767        RecordThread *thread = new RecordThread(this,
1768                                  input,
1769                                  reqSamplingRate,
1770                                  reqChannelMask,
1771                                  id,
1772                                  primaryOutputDevice_l(),
1773                                  *pDevices
1774#ifdef TEE_SINK
1775                                  , teeSink
1776#endif
1777                                  );
1778        mRecordThreads.add(id, thread);
1779        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1780        if (pSamplingRate != NULL) {
1781            *pSamplingRate = reqSamplingRate;
1782        }
1783        if (pFormat != NULL) {
1784            *pFormat = config.format;
1785        }
1786        if (pChannelMask != NULL) {
1787            *pChannelMask = reqChannelMask;
1788        }
1789
1790        // notify client processes of the new input creation
1791        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1792        return id;
1793    }
1794
1795    return 0;
1796}
1797
1798status_t AudioFlinger::closeInput(audio_io_handle_t input)
1799{
1800    return closeInput_nonvirtual(input);
1801}
1802
1803status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1804{
1805    // keep strong reference on the record thread so that
1806    // it is not destroyed while exit() is executed
1807    sp<RecordThread> thread;
1808    {
1809        Mutex::Autolock _l(mLock);
1810        thread = checkRecordThread_l(input);
1811        if (thread == 0) {
1812            return BAD_VALUE;
1813        }
1814
1815        ALOGV("closeInput() %d", input);
1816        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1817        mRecordThreads.removeItem(input);
1818    }
1819    thread->exit();
1820    // The thread entity (active unit of execution) is no longer running here,
1821    // but the ThreadBase container still exists.
1822
1823    AudioStreamIn *in = thread->clearInput();
1824    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1825    // from now on thread->mInput is NULL
1826    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1827    delete in;
1828
1829    return NO_ERROR;
1830}
1831
1832status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1833{
1834    Mutex::Autolock _l(mLock);
1835    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1836
1837    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1838        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1839        thread->invalidateTracks(stream);
1840    }
1841
1842    return NO_ERROR;
1843}
1844
1845
1846int AudioFlinger::newAudioSessionId()
1847{
1848    return nextUniqueId();
1849}
1850
1851void AudioFlinger::acquireAudioSessionId(int audioSession)
1852{
1853    Mutex::Autolock _l(mLock);
1854    pid_t caller = IPCThreadState::self()->getCallingPid();
1855    ALOGV("acquiring %d from %d", audioSession, caller);
1856    size_t num = mAudioSessionRefs.size();
1857    for (size_t i = 0; i< num; i++) {
1858        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1859        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1860            ref->mCnt++;
1861            ALOGV(" incremented refcount to %d", ref->mCnt);
1862            return;
1863        }
1864    }
1865    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1866    ALOGV(" added new entry for %d", audioSession);
1867}
1868
1869void AudioFlinger::releaseAudioSessionId(int audioSession)
1870{
1871    Mutex::Autolock _l(mLock);
1872    pid_t caller = IPCThreadState::self()->getCallingPid();
1873    ALOGV("releasing %d from %d", audioSession, caller);
1874    size_t num = mAudioSessionRefs.size();
1875    for (size_t i = 0; i< num; i++) {
1876        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1877        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1878            ref->mCnt--;
1879            ALOGV(" decremented refcount to %d", ref->mCnt);
1880            if (ref->mCnt == 0) {
1881                mAudioSessionRefs.removeAt(i);
1882                delete ref;
1883                purgeStaleEffects_l();
1884            }
1885            return;
1886        }
1887    }
1888    ALOGW("session id %d not found for pid %d", audioSession, caller);
1889}
1890
1891void AudioFlinger::purgeStaleEffects_l() {
1892
1893    ALOGV("purging stale effects");
1894
1895    Vector< sp<EffectChain> > chains;
1896
1897    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1898        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1899        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1900            sp<EffectChain> ec = t->mEffectChains[j];
1901            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1902                chains.push(ec);
1903            }
1904        }
1905    }
1906    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1907        sp<RecordThread> t = mRecordThreads.valueAt(i);
1908        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1909            sp<EffectChain> ec = t->mEffectChains[j];
1910            chains.push(ec);
1911        }
1912    }
1913
1914    for (size_t i = 0; i < chains.size(); i++) {
1915        sp<EffectChain> ec = chains[i];
1916        int sessionid = ec->sessionId();
1917        sp<ThreadBase> t = ec->mThread.promote();
1918        if (t == 0) {
1919            continue;
1920        }
1921        size_t numsessionrefs = mAudioSessionRefs.size();
1922        bool found = false;
1923        for (size_t k = 0; k < numsessionrefs; k++) {
1924            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1925            if (ref->mSessionid == sessionid) {
1926                ALOGV(" session %d still exists for %d with %d refs",
1927                    sessionid, ref->mPid, ref->mCnt);
1928                found = true;
1929                break;
1930            }
1931        }
1932        if (!found) {
1933            Mutex::Autolock _l(t->mLock);
1934            // remove all effects from the chain
1935            while (ec->mEffects.size()) {
1936                sp<EffectModule> effect = ec->mEffects[0];
1937                effect->unPin();
1938                t->removeEffect_l(effect);
1939                if (effect->purgeHandles()) {
1940                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
1941                }
1942                AudioSystem::unregisterEffect(effect->id());
1943            }
1944        }
1945    }
1946    return;
1947}
1948
1949// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
1950AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
1951{
1952    return mPlaybackThreads.valueFor(output).get();
1953}
1954
1955// checkMixerThread_l() must be called with AudioFlinger::mLock held
1956AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
1957{
1958    PlaybackThread *thread = checkPlaybackThread_l(output);
1959    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
1960}
1961
1962// checkRecordThread_l() must be called with AudioFlinger::mLock held
1963AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
1964{
1965    return mRecordThreads.valueFor(input).get();
1966}
1967
1968uint32_t AudioFlinger::nextUniqueId()
1969{
1970    return android_atomic_inc(&mNextUniqueId);
1971}
1972
1973AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
1974{
1975    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1976        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1977        AudioStreamOut *output = thread->getOutput();
1978        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
1979            return thread;
1980        }
1981    }
1982    return NULL;
1983}
1984
1985audio_devices_t AudioFlinger::primaryOutputDevice_l() const
1986{
1987    PlaybackThread *thread = primaryPlaybackThread_l();
1988
1989    if (thread == NULL) {
1990        return 0;
1991    }
1992
1993    return thread->outDevice();
1994}
1995
1996sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
1997                                    int triggerSession,
1998                                    int listenerSession,
1999                                    sync_event_callback_t callBack,
2000                                    void *cookie)
2001{
2002    Mutex::Autolock _l(mLock);
2003
2004    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2005    status_t playStatus = NAME_NOT_FOUND;
2006    status_t recStatus = NAME_NOT_FOUND;
2007    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2008        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2009        if (playStatus == NO_ERROR) {
2010            return event;
2011        }
2012    }
2013    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2014        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2015        if (recStatus == NO_ERROR) {
2016            return event;
2017        }
2018    }
2019    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2020        mPendingSyncEvents.add(event);
2021    } else {
2022        ALOGV("createSyncEvent() invalid event %d", event->type());
2023        event.clear();
2024    }
2025    return event;
2026}
2027
2028// ----------------------------------------------------------------------------
2029//  Effect management
2030// ----------------------------------------------------------------------------
2031
2032
2033status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2034{
2035    Mutex::Autolock _l(mLock);
2036    return EffectQueryNumberEffects(numEffects);
2037}
2038
2039status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2040{
2041    Mutex::Autolock _l(mLock);
2042    return EffectQueryEffect(index, descriptor);
2043}
2044
2045status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2046        effect_descriptor_t *descriptor) const
2047{
2048    Mutex::Autolock _l(mLock);
2049    return EffectGetDescriptor(pUuid, descriptor);
2050}
2051
2052
2053sp<IEffect> AudioFlinger::createEffect(
2054        effect_descriptor_t *pDesc,
2055        const sp<IEffectClient>& effectClient,
2056        int32_t priority,
2057        audio_io_handle_t io,
2058        int sessionId,
2059        status_t *status,
2060        int *id,
2061        int *enabled)
2062{
2063    status_t lStatus = NO_ERROR;
2064    sp<EffectHandle> handle;
2065    effect_descriptor_t desc;
2066
2067    pid_t pid = IPCThreadState::self()->getCallingPid();
2068    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2069            pid, effectClient.get(), priority, sessionId, io);
2070
2071    if (pDesc == NULL) {
2072        lStatus = BAD_VALUE;
2073        goto Exit;
2074    }
2075
2076    // check audio settings permission for global effects
2077    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2078        lStatus = PERMISSION_DENIED;
2079        goto Exit;
2080    }
2081
2082    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2083    // that can only be created by audio policy manager (running in same process)
2084    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2085        lStatus = PERMISSION_DENIED;
2086        goto Exit;
2087    }
2088
2089    if (io == 0) {
2090        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2091            // output must be specified by AudioPolicyManager when using session
2092            // AUDIO_SESSION_OUTPUT_STAGE
2093            lStatus = BAD_VALUE;
2094            goto Exit;
2095        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2096            // if the output returned by getOutputForEffect() is removed before we lock the
2097            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2098            // and we will exit safely
2099            io = AudioSystem::getOutputForEffect(&desc);
2100        }
2101    }
2102
2103    {
2104        Mutex::Autolock _l(mLock);
2105
2106
2107        if (!EffectIsNullUuid(&pDesc->uuid)) {
2108            // if uuid is specified, request effect descriptor
2109            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2110            if (lStatus < 0) {
2111                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2112                goto Exit;
2113            }
2114        } else {
2115            // if uuid is not specified, look for an available implementation
2116            // of the required type in effect factory
2117            if (EffectIsNullUuid(&pDesc->type)) {
2118                ALOGW("createEffect() no effect type");
2119                lStatus = BAD_VALUE;
2120                goto Exit;
2121            }
2122            uint32_t numEffects = 0;
2123            effect_descriptor_t d;
2124            d.flags = 0; // prevent compiler warning
2125            bool found = false;
2126
2127            lStatus = EffectQueryNumberEffects(&numEffects);
2128            if (lStatus < 0) {
2129                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2130                goto Exit;
2131            }
2132            for (uint32_t i = 0; i < numEffects; i++) {
2133                lStatus = EffectQueryEffect(i, &desc);
2134                if (lStatus < 0) {
2135                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2136                    continue;
2137                }
2138                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2139                    // If matching type found save effect descriptor. If the session is
2140                    // 0 and the effect is not auxiliary, continue enumeration in case
2141                    // an auxiliary version of this effect type is available
2142                    found = true;
2143                    d = desc;
2144                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2145                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2146                        break;
2147                    }
2148                }
2149            }
2150            if (!found) {
2151                lStatus = BAD_VALUE;
2152                ALOGW("createEffect() effect not found");
2153                goto Exit;
2154            }
2155            // For same effect type, chose auxiliary version over insert version if
2156            // connect to output mix (Compliance to OpenSL ES)
2157            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2158                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2159                desc = d;
2160            }
2161        }
2162
2163        // Do not allow auxiliary effects on a session different from 0 (output mix)
2164        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2165             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2166            lStatus = INVALID_OPERATION;
2167            goto Exit;
2168        }
2169
2170        // check recording permission for visualizer
2171        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2172            !recordingAllowed()) {
2173            lStatus = PERMISSION_DENIED;
2174            goto Exit;
2175        }
2176
2177        // return effect descriptor
2178        *pDesc = desc;
2179
2180        // If output is not specified try to find a matching audio session ID in one of the
2181        // output threads.
2182        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2183        // because of code checking output when entering the function.
2184        // Note: io is never 0 when creating an effect on an input
2185        if (io == 0) {
2186            // look for the thread where the specified audio session is present
2187            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2188                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2189                    io = mPlaybackThreads.keyAt(i);
2190                    break;
2191                }
2192            }
2193            if (io == 0) {
2194                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2195                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2196                        io = mRecordThreads.keyAt(i);
2197                        break;
2198                    }
2199                }
2200            }
2201            // If no output thread contains the requested session ID, default to
2202            // first output. The effect chain will be moved to the correct output
2203            // thread when a track with the same session ID is created
2204            if (io == 0 && mPlaybackThreads.size()) {
2205                io = mPlaybackThreads.keyAt(0);
2206            }
2207            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2208        }
2209        ThreadBase *thread = checkRecordThread_l(io);
2210        if (thread == NULL) {
2211            thread = checkPlaybackThread_l(io);
2212            if (thread == NULL) {
2213                ALOGE("createEffect() unknown output thread");
2214                lStatus = BAD_VALUE;
2215                goto Exit;
2216            }
2217        }
2218
2219        sp<Client> client = registerPid_l(pid);
2220
2221        // create effect on selected output thread
2222        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2223                &desc, enabled, &lStatus);
2224        if (handle != 0 && id != NULL) {
2225            *id = handle->id();
2226        }
2227    }
2228
2229Exit:
2230    *status = lStatus;
2231    return handle;
2232}
2233
2234status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2235        audio_io_handle_t dstOutput)
2236{
2237    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2238            sessionId, srcOutput, dstOutput);
2239    Mutex::Autolock _l(mLock);
2240    if (srcOutput == dstOutput) {
2241        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2242        return NO_ERROR;
2243    }
2244    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2245    if (srcThread == NULL) {
2246        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2247        return BAD_VALUE;
2248    }
2249    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2250    if (dstThread == NULL) {
2251        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2252        return BAD_VALUE;
2253    }
2254
2255    Mutex::Autolock _dl(dstThread->mLock);
2256    Mutex::Autolock _sl(srcThread->mLock);
2257    moveEffectChain_l(sessionId, srcThread, dstThread, false);
2258
2259    return NO_ERROR;
2260}
2261
2262// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2263status_t AudioFlinger::moveEffectChain_l(int sessionId,
2264                                   AudioFlinger::PlaybackThread *srcThread,
2265                                   AudioFlinger::PlaybackThread *dstThread,
2266                                   bool reRegister)
2267{
2268    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2269            sessionId, srcThread, dstThread);
2270
2271    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2272    if (chain == 0) {
2273        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2274                sessionId, srcThread);
2275        return INVALID_OPERATION;
2276    }
2277
2278    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2279    // so that a new chain is created with correct parameters when first effect is added. This is
2280    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2281    // removed.
2282    srcThread->removeEffectChain_l(chain);
2283
2284    // transfer all effects one by one so that new effect chain is created on new thread with
2285    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2286    audio_io_handle_t dstOutput = dstThread->id();
2287    sp<EffectChain> dstChain;
2288    uint32_t strategy = 0; // prevent compiler warning
2289    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2290    while (effect != 0) {
2291        srcThread->removeEffect_l(effect);
2292        dstThread->addEffect_l(effect);
2293        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2294        if (effect->state() == EffectModule::ACTIVE ||
2295                effect->state() == EffectModule::STOPPING) {
2296            effect->start();
2297        }
2298        // if the move request is not received from audio policy manager, the effect must be
2299        // re-registered with the new strategy and output
2300        if (dstChain == 0) {
2301            dstChain = effect->chain().promote();
2302            if (dstChain == 0) {
2303                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2304                srcThread->addEffect_l(effect);
2305                return NO_INIT;
2306            }
2307            strategy = dstChain->strategy();
2308        }
2309        if (reRegister) {
2310            AudioSystem::unregisterEffect(effect->id());
2311            AudioSystem::registerEffect(&effect->desc(),
2312                                        dstOutput,
2313                                        strategy,
2314                                        sessionId,
2315                                        effect->id());
2316        }
2317        effect = chain->getEffectFromId_l(0);
2318    }
2319
2320    return NO_ERROR;
2321}
2322
2323bool AudioFlinger::isGlobalEffectEnabled_l()
2324{
2325    if (mGlobalEffectEnableTime != 0 &&
2326            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2327        return true;
2328    }
2329
2330    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2331        sp<EffectChain> ec =
2332                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2333        if (ec != 0 && ec->isEnabled()) {
2334            return true;
2335        }
2336    }
2337    return false;
2338}
2339
2340void AudioFlinger::onGlobalEffectEnable()
2341{
2342    Mutex::Autolock _l(mLock);
2343
2344    mGlobalEffectEnableTime = systemTime();
2345
2346    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2347        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2348        if (t->mType == ThreadBase::OFFLOAD) {
2349            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2350        }
2351    }
2352
2353}
2354
2355struct Entry {
2356#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2357    char mName[MAX_NAME];
2358};
2359
2360int comparEntry(const void *p1, const void *p2)
2361{
2362    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2363}
2364
2365#ifdef TEE_SINK
2366void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2367{
2368    NBAIO_Source *teeSource = source.get();
2369    if (teeSource != NULL) {
2370        // .wav rotation
2371        // There is a benign race condition if 2 threads call this simultaneously.
2372        // They would both traverse the directory, but the result would simply be
2373        // failures at unlink() which are ignored.  It's also unlikely since
2374        // normally dumpsys is only done by bugreport or from the command line.
2375        char teePath[32+256];
2376        strcpy(teePath, "/data/misc/media");
2377        size_t teePathLen = strlen(teePath);
2378        DIR *dir = opendir(teePath);
2379        teePath[teePathLen++] = '/';
2380        if (dir != NULL) {
2381#define MAX_SORT 20 // number of entries to sort
2382#define MAX_KEEP 10 // number of entries to keep
2383            struct Entry entries[MAX_SORT];
2384            size_t entryCount = 0;
2385            while (entryCount < MAX_SORT) {
2386                struct dirent de;
2387                struct dirent *result = NULL;
2388                int rc = readdir_r(dir, &de, &result);
2389                if (rc != 0) {
2390                    ALOGW("readdir_r failed %d", rc);
2391                    break;
2392                }
2393                if (result == NULL) {
2394                    break;
2395                }
2396                if (result != &de) {
2397                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2398                    break;
2399                }
2400                // ignore non .wav file entries
2401                size_t nameLen = strlen(de.d_name);
2402                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2403                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2404                    continue;
2405                }
2406                strcpy(entries[entryCount++].mName, de.d_name);
2407            }
2408            (void) closedir(dir);
2409            if (entryCount > MAX_KEEP) {
2410                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2411                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2412                    strcpy(&teePath[teePathLen], entries[i].mName);
2413                    (void) unlink(teePath);
2414                }
2415            }
2416        } else {
2417            if (fd >= 0) {
2418                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2419            }
2420        }
2421        char teeTime[16];
2422        struct timeval tv;
2423        gettimeofday(&tv, NULL);
2424        struct tm tm;
2425        localtime_r(&tv.tv_sec, &tm);
2426        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2427        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2428        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2429        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2430        if (teeFd >= 0) {
2431            char wavHeader[44];
2432            memcpy(wavHeader,
2433                "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",
2434                sizeof(wavHeader));
2435            NBAIO_Format format = teeSource->format();
2436            unsigned channelCount = Format_channelCount(format);
2437            ALOG_ASSERT(channelCount <= FCC_2);
2438            uint32_t sampleRate = Format_sampleRate(format);
2439            wavHeader[22] = channelCount;       // number of channels
2440            wavHeader[24] = sampleRate;         // sample rate
2441            wavHeader[25] = sampleRate >> 8;
2442            wavHeader[32] = channelCount * 2;   // block alignment
2443            write(teeFd, wavHeader, sizeof(wavHeader));
2444            size_t total = 0;
2445            bool firstRead = true;
2446            for (;;) {
2447#define TEE_SINK_READ 1024
2448                short buffer[TEE_SINK_READ * FCC_2];
2449                size_t count = TEE_SINK_READ;
2450                ssize_t actual = teeSource->read(buffer, count,
2451                        AudioBufferProvider::kInvalidPTS);
2452                bool wasFirstRead = firstRead;
2453                firstRead = false;
2454                if (actual <= 0) {
2455                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2456                        continue;
2457                    }
2458                    break;
2459                }
2460                ALOG_ASSERT(actual <= (ssize_t)count);
2461                write(teeFd, buffer, actual * channelCount * sizeof(short));
2462                total += actual;
2463            }
2464            lseek(teeFd, (off_t) 4, SEEK_SET);
2465            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2466            write(teeFd, &temp, sizeof(temp));
2467            lseek(teeFd, (off_t) 40, SEEK_SET);
2468            temp =  total * channelCount * sizeof(short);
2469            write(teeFd, &temp, sizeof(temp));
2470            close(teeFd);
2471            if (fd >= 0) {
2472                fdprintf(fd, "tee copied to %s\n", teePath);
2473            }
2474        } else {
2475            if (fd >= 0) {
2476                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2477            }
2478        }
2479    }
2480}
2481#endif
2482
2483// ----------------------------------------------------------------------------
2484
2485status_t AudioFlinger::onTransact(
2486        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2487{
2488    return BnAudioFlinger::onTransact(code, data, reply, flags);
2489}
2490
2491}; // namespace android
2492