AudioFlinger.cpp revision da6ef1320d0161b1640dc84d7a9c5a25860c3619 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include <dirent.h>
23#include <math.h>
24#include <signal.h>
25#include <sys/time.h>
26#include <sys/resource.h>
27
28#include <binder/IPCThreadState.h>
29#include <binder/IServiceManager.h>
30#include <utils/Log.h>
31#include <utils/Trace.h>
32#include <binder/Parcel.h>
33#include <utils/String16.h>
34#include <utils/threads.h>
35#include <utils/Atomic.h>
36
37#include <cutils/bitops.h>
38#include <cutils/properties.h>
39#include <cutils/compiler.h>
40
41//#include <private/media/AudioTrackShared.h>
42//#include <private/media/AudioEffectShared.h>
43
44#include <system/audio.h>
45#include <hardware/audio.h>
46
47#include "AudioMixer.h"
48#include "AudioFlinger.h"
49#include "ServiceUtilities.h"
50
51#include <media/EffectsFactoryApi.h>
52#include <audio_effects/effect_visualizer.h>
53#include <audio_effects/effect_ns.h>
54#include <audio_effects/effect_aec.h>
55
56#include <audio_utils/primitives.h>
57
58#include <powermanager/PowerManager.h>
59
60#include <common_time/cc_helper.h>
61//#include <common_time/local_clock.h>
62
63#include <media/IMediaLogService.h>
64
65#include <media/nbaio/Pipe.h>
66#include <media/nbaio/PipeReader.h>
67
68// ----------------------------------------------------------------------------
69
70// Note: the following macro is used for extremely verbose logging message.  In
71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
72// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
73// are so verbose that we want to suppress them even when we have ALOG_ASSERT
74// turned on.  Do not uncomment the #def below unless you really know what you
75// are doing and want to see all of the extremely verbose messages.
76//#define VERY_VERY_VERBOSE_LOGGING
77#ifdef VERY_VERY_VERBOSE_LOGGING
78#define ALOGVV ALOGV
79#else
80#define ALOGVV(a...) do { } while(0)
81#endif
82
83namespace android {
84
85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
86static const char kHardwareLockedString[] = "Hardware lock is taken\n";
87
88
89nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
90
91uint32_t AudioFlinger::mScreenState;
92
93bool AudioFlinger::mTeeSinkInputEnabled = false;
94bool AudioFlinger::mTeeSinkOutputEnabled = false;
95bool AudioFlinger::mTeeSinkTrackEnabled = false;
96
97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
100
101// ----------------------------------------------------------------------------
102
103static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
104{
105    const hw_module_t *mod;
106    int rc;
107
108    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
109    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
110                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
111    if (rc) {
112        goto out;
113    }
114    rc = audio_hw_device_open(mod, dev);
115    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
116                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
117    if (rc) {
118        goto out;
119    }
120    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
121        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
122        rc = BAD_VALUE;
123        goto out;
124    }
125    return 0;
126
127out:
128    *dev = NULL;
129    return rc;
130}
131
132// ----------------------------------------------------------------------------
133
134AudioFlinger::AudioFlinger()
135    : BnAudioFlinger(),
136      mPrimaryHardwareDev(NULL),
137      mHardwareStatus(AUDIO_HW_IDLE),
138      mMasterVolume(1.0f),
139      mMasterMute(false),
140      mNextUniqueId(1),
141      mMode(AUDIO_MODE_INVALID),
142      mBtNrecIsOff(false)
143{
144    char value[PROPERTY_VALUE_MAX];
145    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
146    if (doLog) {
147        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
148    }
149    (void) property_get("ro.debuggable", value, "0");
150    int debuggable = atoi(value);
151    int teeEnabled = 0;
152    if (debuggable) {
153        (void) property_get("af.tee", value, "0");
154        teeEnabled = atoi(value);
155    }
156    if (teeEnabled & 1)
157        mTeeSinkInputEnabled = true;
158    if (teeEnabled & 2)
159        mTeeSinkOutputEnabled = true;
160    if (teeEnabled & 4)
161        mTeeSinkTrackEnabled = true;
162}
163
164void AudioFlinger::onFirstRef()
165{
166    int rc = 0;
167
168    Mutex::Autolock _l(mLock);
169
170    /* TODO: move all this work into an Init() function */
171    char val_str[PROPERTY_VALUE_MAX] = { 0 };
172    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
173        uint32_t int_val;
174        if (1 == sscanf(val_str, "%u", &int_val)) {
175            mStandbyTimeInNsecs = milliseconds(int_val);
176            ALOGI("Using %u mSec as standby time.", int_val);
177        } else {
178            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
179            ALOGI("Using default %u mSec as standby time.",
180                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
181        }
182    }
183
184    mMode = AUDIO_MODE_NORMAL;
185}
186
187AudioFlinger::~AudioFlinger()
188{
189    while (!mRecordThreads.isEmpty()) {
190        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
191        closeInput_nonvirtual(mRecordThreads.keyAt(0));
192    }
193    while (!mPlaybackThreads.isEmpty()) {
194        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
195        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
196    }
197
198    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
199        // no mHardwareLock needed, as there are no other references to this
200        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
201        delete mAudioHwDevs.valueAt(i);
202    }
203}
204
205static const char * const audio_interfaces[] = {
206    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
207    AUDIO_HARDWARE_MODULE_ID_A2DP,
208    AUDIO_HARDWARE_MODULE_ID_USB,
209};
210#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
211
212AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
213        audio_module_handle_t module,
214        audio_devices_t devices)
215{
216    // if module is 0, the request comes from an old policy manager and we should load
217    // well known modules
218    if (module == 0) {
219        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
220        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
221            loadHwModule_l(audio_interfaces[i]);
222        }
223        // then try to find a module supporting the requested device.
224        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
225            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
226            audio_hw_device_t *dev = audioHwDevice->hwDevice();
227            if ((dev->get_supported_devices != NULL) &&
228                    (dev->get_supported_devices(dev) & devices) == devices)
229                return audioHwDevice;
230        }
231    } else {
232        // check a match for the requested module handle
233        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
234        if (audioHwDevice != NULL) {
235            return audioHwDevice;
236        }
237    }
238
239    return NULL;
240}
241
242void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
243{
244    const size_t SIZE = 256;
245    char buffer[SIZE];
246    String8 result;
247
248    result.append("Clients:\n");
249    for (size_t i = 0; i < mClients.size(); ++i) {
250        sp<Client> client = mClients.valueAt(i).promote();
251        if (client != 0) {
252            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
253            result.append(buffer);
254        }
255    }
256
257    result.append("Global session refs:\n");
258    result.append(" session pid count\n");
259    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
260        AudioSessionRef *r = mAudioSessionRefs[i];
261        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
262        result.append(buffer);
263    }
264    write(fd, result.string(), result.size());
265}
266
267
268void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
269{
270    const size_t SIZE = 256;
271    char buffer[SIZE];
272    String8 result;
273    hardware_call_state hardwareStatus = mHardwareStatus;
274
275    snprintf(buffer, SIZE, "Hardware status: %d\n"
276                           "Standby Time mSec: %u\n",
277                            hardwareStatus,
278                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
279    result.append(buffer);
280    write(fd, result.string(), result.size());
281}
282
283void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
284{
285    const size_t SIZE = 256;
286    char buffer[SIZE];
287    String8 result;
288    snprintf(buffer, SIZE, "Permission Denial: "
289            "can't dump AudioFlinger from pid=%d, uid=%d\n",
290            IPCThreadState::self()->getCallingPid(),
291            IPCThreadState::self()->getCallingUid());
292    result.append(buffer);
293    write(fd, result.string(), result.size());
294}
295
296bool AudioFlinger::dumpTryLock(Mutex& mutex)
297{
298    bool locked = false;
299    for (int i = 0; i < kDumpLockRetries; ++i) {
300        if (mutex.tryLock() == NO_ERROR) {
301            locked = true;
302            break;
303        }
304        usleep(kDumpLockSleepUs);
305    }
306    return locked;
307}
308
309status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
310{
311    if (!dumpAllowed()) {
312        dumpPermissionDenial(fd, args);
313    } else {
314        // get state of hardware lock
315        bool hardwareLocked = dumpTryLock(mHardwareLock);
316        if (!hardwareLocked) {
317            String8 result(kHardwareLockedString);
318            write(fd, result.string(), result.size());
319        } else {
320            mHardwareLock.unlock();
321        }
322
323        bool locked = dumpTryLock(mLock);
324
325        // failed to lock - AudioFlinger is probably deadlocked
326        if (!locked) {
327            String8 result(kDeadlockedString);
328            write(fd, result.string(), result.size());
329        }
330
331        dumpClients(fd, args);
332        dumpInternals(fd, args);
333
334        // dump playback threads
335        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
336            mPlaybackThreads.valueAt(i)->dump(fd, args);
337        }
338
339        // dump record threads
340        for (size_t i = 0; i < mRecordThreads.size(); i++) {
341            mRecordThreads.valueAt(i)->dump(fd, args);
342        }
343
344        // dump all hardware devs
345        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
346            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
347            dev->dump(dev, fd);
348        }
349
350        // dump the serially shared record tee sink
351        if (mRecordTeeSource != 0) {
352            dumpTee(fd, mRecordTeeSource);
353        }
354
355        if (locked) {
356            mLock.unlock();
357        }
358
359        // append a copy of media.log here by forwarding fd to it, but don't attempt
360        // to lookup the service if it's not running, as it will block for a second
361        if (mLogMemoryDealer != 0) {
362            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
363            if (binder != 0) {
364                fdprintf(fd, "\nmedia.log:\n");
365                Vector<String16> args;
366                binder->dump(fd, args);
367            }
368        }
369    }
370    return NO_ERROR;
371}
372
373sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
374{
375    // If pid is already in the mClients wp<> map, then use that entry
376    // (for which promote() is always != 0), otherwise create a new entry and Client.
377    sp<Client> client = mClients.valueFor(pid).promote();
378    if (client == 0) {
379        client = new Client(this, pid);
380        mClients.add(pid, client);
381    }
382
383    return client;
384}
385
386sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
387{
388    if (mLogMemoryDealer == 0) {
389        return new NBLog::Writer();
390    }
391    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
392    sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
393    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
394    if (binder != 0) {
395        interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
396    }
397    return writer;
398}
399
400void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
401{
402    if (writer == 0) {
403        return;
404    }
405    sp<IMemory> iMemory(writer->getIMemory());
406    if (iMemory == 0) {
407        return;
408    }
409    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
410    if (binder != 0) {
411        interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
412        // Now the media.log remote reference to IMemory is gone.
413        // When our last local reference to IMemory also drops to zero,
414        // the IMemory destructor will deallocate the region from mMemoryDealer.
415    }
416}
417
418// IAudioFlinger interface
419
420
421sp<IAudioTrack> AudioFlinger::createTrack(
422        audio_stream_type_t streamType,
423        uint32_t sampleRate,
424        audio_format_t format,
425        audio_channel_mask_t channelMask,
426        size_t frameCount,
427        IAudioFlinger::track_flags_t *flags,
428        const sp<IMemory>& sharedBuffer,
429        audio_io_handle_t output,
430        pid_t tid,
431        int *sessionId,
432        status_t *status)
433{
434    sp<PlaybackThread::Track> track;
435    sp<TrackHandle> trackHandle;
436    sp<Client> client;
437    status_t lStatus;
438    int lSessionId;
439
440    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
441    // but if someone uses binder directly they could bypass that and cause us to crash
442    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
443        ALOGE("createTrack() invalid stream type %d", streamType);
444        lStatus = BAD_VALUE;
445        goto Exit;
446    }
447
448    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
449    // and we don't yet support 8.24 or 32-bit PCM
450    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
451        ALOGE("createTrack() invalid format %d", format);
452        lStatus = BAD_VALUE;
453        goto Exit;
454    }
455
456    {
457        Mutex::Autolock _l(mLock);
458        PlaybackThread *thread = checkPlaybackThread_l(output);
459        PlaybackThread *effectThread = NULL;
460        if (thread == NULL) {
461            ALOGE("unknown output thread");
462            lStatus = BAD_VALUE;
463            goto Exit;
464        }
465
466        pid_t pid = IPCThreadState::self()->getCallingPid();
467        client = registerPid_l(pid);
468
469        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
470        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
471            // check if an effect chain with the same session ID is present on another
472            // output thread and move it here.
473            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
474                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
475                if (mPlaybackThreads.keyAt(i) != output) {
476                    uint32_t sessions = t->hasAudioSession(*sessionId);
477                    if (sessions & PlaybackThread::EFFECT_SESSION) {
478                        effectThread = t.get();
479                        break;
480                    }
481                }
482            }
483            lSessionId = *sessionId;
484        } else {
485            // if no audio session id is provided, create one here
486            lSessionId = nextUniqueId();
487            if (sessionId != NULL) {
488                *sessionId = lSessionId;
489            }
490        }
491        ALOGV("createTrack() lSessionId: %d", lSessionId);
492
493        track = thread->createTrack_l(client, streamType, sampleRate, format,
494                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus);
495
496        // move effect chain to this output thread if an effect on same session was waiting
497        // for a track to be created
498        if (lStatus == NO_ERROR && effectThread != NULL) {
499            Mutex::Autolock _dl(thread->mLock);
500            Mutex::Autolock _sl(effectThread->mLock);
501            moveEffectChain_l(lSessionId, effectThread, thread, true);
502        }
503
504        // Look for sync events awaiting for a session to be used.
505        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
506            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
507                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
508                    if (lStatus == NO_ERROR) {
509                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
510                    } else {
511                        mPendingSyncEvents[i]->cancel();
512                    }
513                    mPendingSyncEvents.removeAt(i);
514                    i--;
515                }
516            }
517        }
518    }
519    if (lStatus == NO_ERROR) {
520        trackHandle = new TrackHandle(track);
521    } else {
522        // remove local strong reference to Client before deleting the Track so that the Client
523        // destructor is called by the TrackBase destructor with mLock held
524        client.clear();
525        track.clear();
526    }
527
528Exit:
529    if (status != NULL) {
530        *status = lStatus;
531    }
532    return trackHandle;
533}
534
535uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
536{
537    Mutex::Autolock _l(mLock);
538    PlaybackThread *thread = checkPlaybackThread_l(output);
539    if (thread == NULL) {
540        ALOGW("sampleRate() unknown thread %d", output);
541        return 0;
542    }
543    return thread->sampleRate();
544}
545
546int AudioFlinger::channelCount(audio_io_handle_t output) const
547{
548    Mutex::Autolock _l(mLock);
549    PlaybackThread *thread = checkPlaybackThread_l(output);
550    if (thread == NULL) {
551        ALOGW("channelCount() unknown thread %d", output);
552        return 0;
553    }
554    return thread->channelCount();
555}
556
557audio_format_t AudioFlinger::format(audio_io_handle_t output) const
558{
559    Mutex::Autolock _l(mLock);
560    PlaybackThread *thread = checkPlaybackThread_l(output);
561    if (thread == NULL) {
562        ALOGW("format() unknown thread %d", output);
563        return AUDIO_FORMAT_INVALID;
564    }
565    return thread->format();
566}
567
568size_t AudioFlinger::frameCount(audio_io_handle_t output) const
569{
570    Mutex::Autolock _l(mLock);
571    PlaybackThread *thread = checkPlaybackThread_l(output);
572    if (thread == NULL) {
573        ALOGW("frameCount() unknown thread %d", output);
574        return 0;
575    }
576    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
577    //       should examine all callers and fix them to handle smaller counts
578    return thread->frameCount();
579}
580
581uint32_t AudioFlinger::latency(audio_io_handle_t output) const
582{
583    Mutex::Autolock _l(mLock);
584    PlaybackThread *thread = checkPlaybackThread_l(output);
585    if (thread == NULL) {
586        ALOGW("latency() unknown thread %d", output);
587        return 0;
588    }
589    return thread->latency();
590}
591
592status_t AudioFlinger::setMasterVolume(float value)
593{
594    status_t ret = initCheck();
595    if (ret != NO_ERROR) {
596        return ret;
597    }
598
599    // check calling permissions
600    if (!settingsAllowed()) {
601        return PERMISSION_DENIED;
602    }
603
604    Mutex::Autolock _l(mLock);
605    mMasterVolume = value;
606
607    // Set master volume in the HALs which support it.
608    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
609        AutoMutex lock(mHardwareLock);
610        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
611
612        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
613        if (dev->canSetMasterVolume()) {
614            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
615        }
616        mHardwareStatus = AUDIO_HW_IDLE;
617    }
618
619    // Now set the master volume in each playback thread.  Playback threads
620    // assigned to HALs which do not have master volume support will apply
621    // master volume during the mix operation.  Threads with HALs which do
622    // support master volume will simply ignore the setting.
623    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
624        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
625
626    return NO_ERROR;
627}
628
629status_t AudioFlinger::setMode(audio_mode_t mode)
630{
631    status_t ret = initCheck();
632    if (ret != NO_ERROR) {
633        return ret;
634    }
635
636    // check calling permissions
637    if (!settingsAllowed()) {
638        return PERMISSION_DENIED;
639    }
640    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
641        ALOGW("Illegal value: setMode(%d)", mode);
642        return BAD_VALUE;
643    }
644
645    { // scope for the lock
646        AutoMutex lock(mHardwareLock);
647        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
648        mHardwareStatus = AUDIO_HW_SET_MODE;
649        ret = dev->set_mode(dev, mode);
650        mHardwareStatus = AUDIO_HW_IDLE;
651    }
652
653    if (NO_ERROR == ret) {
654        Mutex::Autolock _l(mLock);
655        mMode = mode;
656        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
657            mPlaybackThreads.valueAt(i)->setMode(mode);
658    }
659
660    return ret;
661}
662
663status_t AudioFlinger::setMicMute(bool state)
664{
665    status_t ret = initCheck();
666    if (ret != NO_ERROR) {
667        return ret;
668    }
669
670    // check calling permissions
671    if (!settingsAllowed()) {
672        return PERMISSION_DENIED;
673    }
674
675    AutoMutex lock(mHardwareLock);
676    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
677    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
678    ret = dev->set_mic_mute(dev, state);
679    mHardwareStatus = AUDIO_HW_IDLE;
680    return ret;
681}
682
683bool AudioFlinger::getMicMute() const
684{
685    status_t ret = initCheck();
686    if (ret != NO_ERROR) {
687        return false;
688    }
689
690    bool state = AUDIO_MODE_INVALID;
691    AutoMutex lock(mHardwareLock);
692    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
693    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
694    dev->get_mic_mute(dev, &state);
695    mHardwareStatus = AUDIO_HW_IDLE;
696    return state;
697}
698
699status_t AudioFlinger::setMasterMute(bool muted)
700{
701    status_t ret = initCheck();
702    if (ret != NO_ERROR) {
703        return ret;
704    }
705
706    // check calling permissions
707    if (!settingsAllowed()) {
708        return PERMISSION_DENIED;
709    }
710
711    Mutex::Autolock _l(mLock);
712    mMasterMute = muted;
713
714    // Set master mute in the HALs which support it.
715    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
716        AutoMutex lock(mHardwareLock);
717        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
718
719        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
720        if (dev->canSetMasterMute()) {
721            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
722        }
723        mHardwareStatus = AUDIO_HW_IDLE;
724    }
725
726    // Now set the master mute in each playback thread.  Playback threads
727    // assigned to HALs which do not have master mute support will apply master
728    // mute during the mix operation.  Threads with HALs which do support master
729    // mute will simply ignore the setting.
730    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
731        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
732
733    return NO_ERROR;
734}
735
736float AudioFlinger::masterVolume() const
737{
738    Mutex::Autolock _l(mLock);
739    return masterVolume_l();
740}
741
742bool AudioFlinger::masterMute() const
743{
744    Mutex::Autolock _l(mLock);
745    return masterMute_l();
746}
747
748float AudioFlinger::masterVolume_l() const
749{
750    return mMasterVolume;
751}
752
753bool AudioFlinger::masterMute_l() const
754{
755    return mMasterMute;
756}
757
758status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
759        audio_io_handle_t output)
760{
761    // check calling permissions
762    if (!settingsAllowed()) {
763        return PERMISSION_DENIED;
764    }
765
766    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
767        ALOGE("setStreamVolume() invalid stream %d", stream);
768        return BAD_VALUE;
769    }
770
771    AutoMutex lock(mLock);
772    PlaybackThread *thread = NULL;
773    if (output) {
774        thread = checkPlaybackThread_l(output);
775        if (thread == NULL) {
776            return BAD_VALUE;
777        }
778    }
779
780    mStreamTypes[stream].volume = value;
781
782    if (thread == NULL) {
783        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
784            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
785        }
786    } else {
787        thread->setStreamVolume(stream, value);
788    }
789
790    return NO_ERROR;
791}
792
793status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
794{
795    // check calling permissions
796    if (!settingsAllowed()) {
797        return PERMISSION_DENIED;
798    }
799
800    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
801        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
802        ALOGE("setStreamMute() invalid stream %d", stream);
803        return BAD_VALUE;
804    }
805
806    AutoMutex lock(mLock);
807    mStreamTypes[stream].mute = muted;
808    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
809        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
810
811    return NO_ERROR;
812}
813
814float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
815{
816    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
817        return 0.0f;
818    }
819
820    AutoMutex lock(mLock);
821    float volume;
822    if (output) {
823        PlaybackThread *thread = checkPlaybackThread_l(output);
824        if (thread == NULL) {
825            return 0.0f;
826        }
827        volume = thread->streamVolume(stream);
828    } else {
829        volume = streamVolume_l(stream);
830    }
831
832    return volume;
833}
834
835bool AudioFlinger::streamMute(audio_stream_type_t stream) const
836{
837    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
838        return true;
839    }
840
841    AutoMutex lock(mLock);
842    return streamMute_l(stream);
843}
844
845status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
846{
847    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
848            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
849
850    // check calling permissions
851    if (!settingsAllowed()) {
852        return PERMISSION_DENIED;
853    }
854
855    // ioHandle == 0 means the parameters are global to the audio hardware interface
856    if (ioHandle == 0) {
857        Mutex::Autolock _l(mLock);
858        status_t final_result = NO_ERROR;
859        {
860            AutoMutex lock(mHardwareLock);
861            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
862            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
863                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
864                status_t result = dev->set_parameters(dev, keyValuePairs.string());
865                final_result = result ?: final_result;
866            }
867            mHardwareStatus = AUDIO_HW_IDLE;
868        }
869        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
870        AudioParameter param = AudioParameter(keyValuePairs);
871        String8 value;
872        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
873            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
874            if (mBtNrecIsOff != btNrecIsOff) {
875                for (size_t i = 0; i < mRecordThreads.size(); i++) {
876                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
877                    audio_devices_t device = thread->inDevice();
878                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
879                    // collect all of the thread's session IDs
880                    KeyedVector<int, bool> ids = thread->sessionIds();
881                    // suspend effects associated with those session IDs
882                    for (size_t j = 0; j < ids.size(); ++j) {
883                        int sessionId = ids.keyAt(j);
884                        thread->setEffectSuspended(FX_IID_AEC,
885                                                   suspend,
886                                                   sessionId);
887                        thread->setEffectSuspended(FX_IID_NS,
888                                                   suspend,
889                                                   sessionId);
890                    }
891                }
892                mBtNrecIsOff = btNrecIsOff;
893            }
894        }
895        String8 screenState;
896        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
897            bool isOff = screenState == "off";
898            if (isOff != (AudioFlinger::mScreenState & 1)) {
899                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
900            }
901        }
902        return final_result;
903    }
904
905    // hold a strong ref on thread in case closeOutput() or closeInput() is called
906    // and the thread is exited once the lock is released
907    sp<ThreadBase> thread;
908    {
909        Mutex::Autolock _l(mLock);
910        thread = checkPlaybackThread_l(ioHandle);
911        if (thread == 0) {
912            thread = checkRecordThread_l(ioHandle);
913        } else if (thread == primaryPlaybackThread_l()) {
914            // indicate output device change to all input threads for pre processing
915            AudioParameter param = AudioParameter(keyValuePairs);
916            int value;
917            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
918                    (value != 0)) {
919                for (size_t i = 0; i < mRecordThreads.size(); i++) {
920                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
921                }
922            }
923        }
924    }
925    if (thread != 0) {
926        return thread->setParameters(keyValuePairs);
927    }
928    return BAD_VALUE;
929}
930
931String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
932{
933    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
934            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
935
936    Mutex::Autolock _l(mLock);
937
938    if (ioHandle == 0) {
939        String8 out_s8;
940
941        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
942            char *s;
943            {
944            AutoMutex lock(mHardwareLock);
945            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
946            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
947            s = dev->get_parameters(dev, keys.string());
948            mHardwareStatus = AUDIO_HW_IDLE;
949            }
950            out_s8 += String8(s ? s : "");
951            free(s);
952        }
953        return out_s8;
954    }
955
956    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
957    if (playbackThread != NULL) {
958        return playbackThread->getParameters(keys);
959    }
960    RecordThread *recordThread = checkRecordThread_l(ioHandle);
961    if (recordThread != NULL) {
962        return recordThread->getParameters(keys);
963    }
964    return String8("");
965}
966
967size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
968        audio_channel_mask_t channelMask) const
969{
970    status_t ret = initCheck();
971    if (ret != NO_ERROR) {
972        return 0;
973    }
974
975    AutoMutex lock(mHardwareLock);
976    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
977    struct audio_config config = {
978        sample_rate: sampleRate,
979        channel_mask: channelMask,
980        format: format,
981    };
982    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
983    size_t size = dev->get_input_buffer_size(dev, &config);
984    mHardwareStatus = AUDIO_HW_IDLE;
985    return size;
986}
987
988unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
989{
990    Mutex::Autolock _l(mLock);
991
992    RecordThread *recordThread = checkRecordThread_l(ioHandle);
993    if (recordThread != NULL) {
994        return recordThread->getInputFramesLost();
995    }
996    return 0;
997}
998
999status_t AudioFlinger::setVoiceVolume(float value)
1000{
1001    status_t ret = initCheck();
1002    if (ret != NO_ERROR) {
1003        return ret;
1004    }
1005
1006    // check calling permissions
1007    if (!settingsAllowed()) {
1008        return PERMISSION_DENIED;
1009    }
1010
1011    AutoMutex lock(mHardwareLock);
1012    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1013    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1014    ret = dev->set_voice_volume(dev, value);
1015    mHardwareStatus = AUDIO_HW_IDLE;
1016
1017    return ret;
1018}
1019
1020status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1021        audio_io_handle_t output) const
1022{
1023    status_t status;
1024
1025    Mutex::Autolock _l(mLock);
1026
1027    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1028    if (playbackThread != NULL) {
1029        return playbackThread->getRenderPosition(halFrames, dspFrames);
1030    }
1031
1032    return BAD_VALUE;
1033}
1034
1035void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1036{
1037
1038    Mutex::Autolock _l(mLock);
1039
1040    pid_t pid = IPCThreadState::self()->getCallingPid();
1041    if (mNotificationClients.indexOfKey(pid) < 0) {
1042        sp<NotificationClient> notificationClient = new NotificationClient(this,
1043                                                                            client,
1044                                                                            pid);
1045        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1046
1047        mNotificationClients.add(pid, notificationClient);
1048
1049        sp<IBinder> binder = client->asBinder();
1050        binder->linkToDeath(notificationClient);
1051
1052        // the config change is always sent from playback or record threads to avoid deadlock
1053        // with AudioSystem::gLock
1054        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1055            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1056        }
1057
1058        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1059            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1060        }
1061    }
1062}
1063
1064void AudioFlinger::removeNotificationClient(pid_t pid)
1065{
1066    Mutex::Autolock _l(mLock);
1067
1068    mNotificationClients.removeItem(pid);
1069
1070    ALOGV("%d died, releasing its sessions", pid);
1071    size_t num = mAudioSessionRefs.size();
1072    bool removed = false;
1073    for (size_t i = 0; i< num; ) {
1074        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1075        ALOGV(" pid %d @ %d", ref->mPid, i);
1076        if (ref->mPid == pid) {
1077            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1078            mAudioSessionRefs.removeAt(i);
1079            delete ref;
1080            removed = true;
1081            num--;
1082        } else {
1083            i++;
1084        }
1085    }
1086    if (removed) {
1087        purgeStaleEffects_l();
1088    }
1089}
1090
1091// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1092void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1093{
1094    size_t size = mNotificationClients.size();
1095    for (size_t i = 0; i < size; i++) {
1096        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1097                                                                               param2);
1098    }
1099}
1100
1101// removeClient_l() must be called with AudioFlinger::mLock held
1102void AudioFlinger::removeClient_l(pid_t pid)
1103{
1104    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1105            IPCThreadState::self()->getCallingPid());
1106    mClients.removeItem(pid);
1107}
1108
1109// getEffectThread_l() must be called with AudioFlinger::mLock held
1110sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1111{
1112    sp<PlaybackThread> thread;
1113
1114    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1115        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1116            ALOG_ASSERT(thread == 0);
1117            thread = mPlaybackThreads.valueAt(i);
1118        }
1119    }
1120
1121    return thread;
1122}
1123
1124
1125
1126// ----------------------------------------------------------------------------
1127
1128AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1129    :   RefBase(),
1130        mAudioFlinger(audioFlinger),
1131        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1132        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1133        mPid(pid),
1134        mTimedTrackCount(0)
1135{
1136    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1137}
1138
1139// Client destructor must be called with AudioFlinger::mLock held
1140AudioFlinger::Client::~Client()
1141{
1142    mAudioFlinger->removeClient_l(mPid);
1143}
1144
1145sp<MemoryDealer> AudioFlinger::Client::heap() const
1146{
1147    return mMemoryDealer;
1148}
1149
1150// Reserve one of the limited slots for a timed audio track associated
1151// with this client
1152bool AudioFlinger::Client::reserveTimedTrack()
1153{
1154    const int kMaxTimedTracksPerClient = 4;
1155
1156    Mutex::Autolock _l(mTimedTrackLock);
1157
1158    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1159        ALOGW("can not create timed track - pid %d has exceeded the limit",
1160             mPid);
1161        return false;
1162    }
1163
1164    mTimedTrackCount++;
1165    return true;
1166}
1167
1168// Release a slot for a timed audio track
1169void AudioFlinger::Client::releaseTimedTrack()
1170{
1171    Mutex::Autolock _l(mTimedTrackLock);
1172    mTimedTrackCount--;
1173}
1174
1175// ----------------------------------------------------------------------------
1176
1177AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1178                                                     const sp<IAudioFlingerClient>& client,
1179                                                     pid_t pid)
1180    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1181{
1182}
1183
1184AudioFlinger::NotificationClient::~NotificationClient()
1185{
1186}
1187
1188void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1189{
1190    sp<NotificationClient> keep(this);
1191    mAudioFlinger->removeNotificationClient(mPid);
1192}
1193
1194
1195// ----------------------------------------------------------------------------
1196
1197sp<IAudioRecord> AudioFlinger::openRecord(
1198        audio_io_handle_t input,
1199        uint32_t sampleRate,
1200        audio_format_t format,
1201        audio_channel_mask_t channelMask,
1202        size_t frameCount,
1203        IAudioFlinger::track_flags_t flags,
1204        pid_t tid,
1205        int *sessionId,
1206        status_t *status)
1207{
1208    sp<RecordThread::RecordTrack> recordTrack;
1209    sp<RecordHandle> recordHandle;
1210    sp<Client> client;
1211    status_t lStatus;
1212    RecordThread *thread;
1213    size_t inFrameCount;
1214    int lSessionId;
1215
1216    // check calling permissions
1217    if (!recordingAllowed()) {
1218        lStatus = PERMISSION_DENIED;
1219        goto Exit;
1220    }
1221
1222    // add client to list
1223    { // scope for mLock
1224        Mutex::Autolock _l(mLock);
1225        thread = checkRecordThread_l(input);
1226        if (thread == NULL) {
1227            lStatus = BAD_VALUE;
1228            goto Exit;
1229        }
1230
1231        pid_t pid = IPCThreadState::self()->getCallingPid();
1232        client = registerPid_l(pid);
1233
1234        // If no audio session id is provided, create one here
1235        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1236            lSessionId = *sessionId;
1237        } else {
1238            lSessionId = nextUniqueId();
1239            if (sessionId != NULL) {
1240                *sessionId = lSessionId;
1241            }
1242        }
1243        // create new record track.
1244        // The record track uses one track in mHardwareMixerThread by convention.
1245        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1246                                                  frameCount, lSessionId, flags, tid, &lStatus);
1247    }
1248    if (lStatus != NO_ERROR) {
1249        // remove local strong reference to Client before deleting the RecordTrack so that the
1250        // Client destructor is called by the TrackBase destructor with mLock held
1251        client.clear();
1252        recordTrack.clear();
1253        goto Exit;
1254    }
1255
1256    // return to handle to client
1257    recordHandle = new RecordHandle(recordTrack);
1258    lStatus = NO_ERROR;
1259
1260Exit:
1261    if (status) {
1262        *status = lStatus;
1263    }
1264    return recordHandle;
1265}
1266
1267
1268
1269// ----------------------------------------------------------------------------
1270
1271audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1272{
1273    if (!settingsAllowed()) {
1274        return 0;
1275    }
1276    Mutex::Autolock _l(mLock);
1277    return loadHwModule_l(name);
1278}
1279
1280// loadHwModule_l() must be called with AudioFlinger::mLock held
1281audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1282{
1283    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1284        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1285            ALOGW("loadHwModule() module %s already loaded", name);
1286            return mAudioHwDevs.keyAt(i);
1287        }
1288    }
1289
1290    audio_hw_device_t *dev;
1291
1292    int rc = load_audio_interface(name, &dev);
1293    if (rc) {
1294        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1295        return 0;
1296    }
1297
1298    mHardwareStatus = AUDIO_HW_INIT;
1299    rc = dev->init_check(dev);
1300    mHardwareStatus = AUDIO_HW_IDLE;
1301    if (rc) {
1302        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1303        return 0;
1304    }
1305
1306    // Check and cache this HAL's level of support for master mute and master
1307    // volume.  If this is the first HAL opened, and it supports the get
1308    // methods, use the initial values provided by the HAL as the current
1309    // master mute and volume settings.
1310
1311    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1312    {  // scope for auto-lock pattern
1313        AutoMutex lock(mHardwareLock);
1314
1315        if (0 == mAudioHwDevs.size()) {
1316            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1317            if (NULL != dev->get_master_volume) {
1318                float mv;
1319                if (OK == dev->get_master_volume(dev, &mv)) {
1320                    mMasterVolume = mv;
1321                }
1322            }
1323
1324            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1325            if (NULL != dev->get_master_mute) {
1326                bool mm;
1327                if (OK == dev->get_master_mute(dev, &mm)) {
1328                    mMasterMute = mm;
1329                }
1330            }
1331        }
1332
1333        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1334        if ((NULL != dev->set_master_volume) &&
1335            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1336            flags = static_cast<AudioHwDevice::Flags>(flags |
1337                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1338        }
1339
1340        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1341        if ((NULL != dev->set_master_mute) &&
1342            (OK == dev->set_master_mute(dev, mMasterMute))) {
1343            flags = static_cast<AudioHwDevice::Flags>(flags |
1344                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1345        }
1346
1347        mHardwareStatus = AUDIO_HW_IDLE;
1348    }
1349
1350    audio_module_handle_t handle = nextUniqueId();
1351    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1352
1353    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1354          name, dev->common.module->name, dev->common.module->id, handle);
1355
1356    return handle;
1357
1358}
1359
1360// ----------------------------------------------------------------------------
1361
1362uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1363{
1364    Mutex::Autolock _l(mLock);
1365    PlaybackThread *thread = primaryPlaybackThread_l();
1366    return thread != NULL ? thread->sampleRate() : 0;
1367}
1368
1369size_t AudioFlinger::getPrimaryOutputFrameCount()
1370{
1371    Mutex::Autolock _l(mLock);
1372    PlaybackThread *thread = primaryPlaybackThread_l();
1373    return thread != NULL ? thread->frameCountHAL() : 0;
1374}
1375
1376// ----------------------------------------------------------------------------
1377
1378audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1379                                           audio_devices_t *pDevices,
1380                                           uint32_t *pSamplingRate,
1381                                           audio_format_t *pFormat,
1382                                           audio_channel_mask_t *pChannelMask,
1383                                           uint32_t *pLatencyMs,
1384                                           audio_output_flags_t flags)
1385{
1386    status_t status;
1387    PlaybackThread *thread = NULL;
1388    struct audio_config config = {
1389        sample_rate: pSamplingRate ? *pSamplingRate : 0,
1390        channel_mask: pChannelMask ? *pChannelMask : 0,
1391        format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT,
1392    };
1393    audio_stream_out_t *outStream = NULL;
1394    AudioHwDevice *outHwDev;
1395
1396    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
1397              module,
1398              (pDevices != NULL) ? *pDevices : 0,
1399              config.sample_rate,
1400              config.format,
1401              config.channel_mask,
1402              flags);
1403
1404    if (pDevices == NULL || *pDevices == 0) {
1405        return 0;
1406    }
1407
1408    Mutex::Autolock _l(mLock);
1409
1410    outHwDev = findSuitableHwDev_l(module, *pDevices);
1411    if (outHwDev == NULL)
1412        return 0;
1413
1414    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1415    audio_io_handle_t id = nextUniqueId();
1416
1417    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1418
1419    status = hwDevHal->open_output_stream(hwDevHal,
1420                                          id,
1421                                          *pDevices,
1422                                          (audio_output_flags_t)flags,
1423                                          &config,
1424                                          &outStream);
1425
1426    mHardwareStatus = AUDIO_HW_IDLE;
1427    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, "
1428            "Channels %x, status %d",
1429            outStream,
1430            config.sample_rate,
1431            config.format,
1432            config.channel_mask,
1433            status);
1434
1435    if (status == NO_ERROR && outStream != NULL) {
1436        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
1437
1438        if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1439            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1440            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1441            thread = new DirectOutputThread(this, output, id, *pDevices);
1442            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1443        } else {
1444            thread = new MixerThread(this, output, id, *pDevices);
1445            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1446        }
1447        mPlaybackThreads.add(id, thread);
1448
1449        if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate;
1450        if (pFormat != NULL) *pFormat = config.format;
1451        if (pChannelMask != NULL) *pChannelMask = config.channel_mask;
1452        if (pLatencyMs != NULL) *pLatencyMs = thread->latency();
1453
1454        // notify client processes of the new output creation
1455        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1456
1457        // the first primary output opened designates the primary hw device
1458        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1459            ALOGI("Using module %d has the primary audio interface", module);
1460            mPrimaryHardwareDev = outHwDev;
1461
1462            AutoMutex lock(mHardwareLock);
1463            mHardwareStatus = AUDIO_HW_SET_MODE;
1464            hwDevHal->set_mode(hwDevHal, mMode);
1465            mHardwareStatus = AUDIO_HW_IDLE;
1466        }
1467        return id;
1468    }
1469
1470    return 0;
1471}
1472
1473audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1474        audio_io_handle_t output2)
1475{
1476    Mutex::Autolock _l(mLock);
1477    MixerThread *thread1 = checkMixerThread_l(output1);
1478    MixerThread *thread2 = checkMixerThread_l(output2);
1479
1480    if (thread1 == NULL || thread2 == NULL) {
1481        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1482                output2);
1483        return 0;
1484    }
1485
1486    audio_io_handle_t id = nextUniqueId();
1487    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1488    thread->addOutputTrack(thread2);
1489    mPlaybackThreads.add(id, thread);
1490    // notify client processes of the new output creation
1491    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1492    return id;
1493}
1494
1495status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1496{
1497    return closeOutput_nonvirtual(output);
1498}
1499
1500status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1501{
1502    // keep strong reference on the playback thread so that
1503    // it is not destroyed while exit() is executed
1504    sp<PlaybackThread> thread;
1505    {
1506        Mutex::Autolock _l(mLock);
1507        thread = checkPlaybackThread_l(output);
1508        if (thread == NULL) {
1509            return BAD_VALUE;
1510        }
1511
1512        ALOGV("closeOutput() %d", output);
1513
1514        if (thread->type() == ThreadBase::MIXER) {
1515            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1516                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1517                    DuplicatingThread *dupThread =
1518                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1519                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1520                }
1521            }
1522        }
1523        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1524        mPlaybackThreads.removeItem(output);
1525    }
1526    thread->exit();
1527    // The thread entity (active unit of execution) is no longer running here,
1528    // but the ThreadBase container still exists.
1529
1530    if (thread->type() != ThreadBase::DUPLICATING) {
1531        AudioStreamOut *out = thread->clearOutput();
1532        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1533        // from now on thread->mOutput is NULL
1534        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1535        delete out;
1536    }
1537    return NO_ERROR;
1538}
1539
1540status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1541{
1542    Mutex::Autolock _l(mLock);
1543    PlaybackThread *thread = checkPlaybackThread_l(output);
1544
1545    if (thread == NULL) {
1546        return BAD_VALUE;
1547    }
1548
1549    ALOGV("suspendOutput() %d", output);
1550    thread->suspend();
1551
1552    return NO_ERROR;
1553}
1554
1555status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1556{
1557    Mutex::Autolock _l(mLock);
1558    PlaybackThread *thread = checkPlaybackThread_l(output);
1559
1560    if (thread == NULL) {
1561        return BAD_VALUE;
1562    }
1563
1564    ALOGV("restoreOutput() %d", output);
1565
1566    thread->restore();
1567
1568    return NO_ERROR;
1569}
1570
1571audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1572                                          audio_devices_t *pDevices,
1573                                          uint32_t *pSamplingRate,
1574                                          audio_format_t *pFormat,
1575                                          audio_channel_mask_t *pChannelMask)
1576{
1577    status_t status;
1578    RecordThread *thread = NULL;
1579    struct audio_config config = {
1580        sample_rate: pSamplingRate ? *pSamplingRate : 0,
1581        channel_mask: pChannelMask ? *pChannelMask : 0,
1582        format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT,
1583    };
1584    uint32_t reqSamplingRate = config.sample_rate;
1585    audio_format_t reqFormat = config.format;
1586    audio_channel_mask_t reqChannels = config.channel_mask;
1587    audio_stream_in_t *inStream = NULL;
1588    AudioHwDevice *inHwDev;
1589
1590    if (pDevices == NULL || *pDevices == 0) {
1591        return 0;
1592    }
1593
1594    Mutex::Autolock _l(mLock);
1595
1596    inHwDev = findSuitableHwDev_l(module, *pDevices);
1597    if (inHwDev == NULL)
1598        return 0;
1599
1600    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1601    audio_io_handle_t id = nextUniqueId();
1602
1603    status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1604                                        &inStream);
1605    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1606            "status %d",
1607            inStream,
1608            config.sample_rate,
1609            config.format,
1610            config.channel_mask,
1611            status);
1612
1613    // If the input could not be opened with the requested parameters and we can handle the
1614    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1615    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1616    if (status == BAD_VALUE &&
1617        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1618        (config.sample_rate <= 2 * reqSamplingRate) &&
1619        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
1620        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1621        inStream = NULL;
1622        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1623    }
1624
1625    if (status == NO_ERROR && inStream != NULL) {
1626
1627        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1628        // or (re-)create if current Pipe is idle and does not match the new format
1629        sp<NBAIO_Sink> teeSink;
1630        enum {
1631            TEE_SINK_NO,    // don't copy input
1632            TEE_SINK_NEW,   // copy input using a new pipe
1633            TEE_SINK_OLD,   // copy input using an existing pipe
1634        } kind;
1635        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1636                                        popcount(inStream->common.get_channels(&inStream->common)));
1637        if (!mTeeSinkInputEnabled) {
1638            kind = TEE_SINK_NO;
1639        } else if (format == Format_Invalid) {
1640            kind = TEE_SINK_NO;
1641        } else if (mRecordTeeSink == 0) {
1642            kind = TEE_SINK_NEW;
1643        } else if (mRecordTeeSink->getStrongCount() != 1) {
1644            kind = TEE_SINK_NO;
1645        } else if (format == mRecordTeeSink->format()) {
1646            kind = TEE_SINK_OLD;
1647        } else {
1648            kind = TEE_SINK_NEW;
1649        }
1650        switch (kind) {
1651        case TEE_SINK_NEW: {
1652            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1653            size_t numCounterOffers = 0;
1654            const NBAIO_Format offers[1] = {format};
1655            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1656            ALOG_ASSERT(index == 0);
1657            PipeReader *pipeReader = new PipeReader(*pipe);
1658            numCounterOffers = 0;
1659            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1660            ALOG_ASSERT(index == 0);
1661            mRecordTeeSink = pipe;
1662            mRecordTeeSource = pipeReader;
1663            teeSink = pipe;
1664            }
1665            break;
1666        case TEE_SINK_OLD:
1667            teeSink = mRecordTeeSink;
1668            break;
1669        case TEE_SINK_NO:
1670        default:
1671            break;
1672        }
1673
1674        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1675
1676        // Start record thread
1677        // RecorThread require both input and output device indication to forward to audio
1678        // pre processing modules
1679        thread = new RecordThread(this,
1680                                  input,
1681                                  reqSamplingRate,
1682                                  reqChannels,
1683                                  id,
1684                                  primaryOutputDevice_l(),
1685                                  *pDevices,
1686                                  teeSink);
1687        mRecordThreads.add(id, thread);
1688        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1689        if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
1690        if (pFormat != NULL) *pFormat = config.format;
1691        if (pChannelMask != NULL) *pChannelMask = reqChannels;
1692
1693        // notify client processes of the new input creation
1694        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1695        return id;
1696    }
1697
1698    return 0;
1699}
1700
1701status_t AudioFlinger::closeInput(audio_io_handle_t input)
1702{
1703    return closeInput_nonvirtual(input);
1704}
1705
1706status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1707{
1708    // keep strong reference on the record thread so that
1709    // it is not destroyed while exit() is executed
1710    sp<RecordThread> thread;
1711    {
1712        Mutex::Autolock _l(mLock);
1713        thread = checkRecordThread_l(input);
1714        if (thread == 0) {
1715            return BAD_VALUE;
1716        }
1717
1718        ALOGV("closeInput() %d", input);
1719        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1720        mRecordThreads.removeItem(input);
1721    }
1722    thread->exit();
1723    // The thread entity (active unit of execution) is no longer running here,
1724    // but the ThreadBase container still exists.
1725
1726    AudioStreamIn *in = thread->clearInput();
1727    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1728    // from now on thread->mInput is NULL
1729    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1730    delete in;
1731
1732    return NO_ERROR;
1733}
1734
1735status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1736{
1737    Mutex::Autolock _l(mLock);
1738    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1739
1740    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1741        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1742        thread->invalidateTracks(stream);
1743    }
1744
1745    return NO_ERROR;
1746}
1747
1748
1749int AudioFlinger::newAudioSessionId()
1750{
1751    return nextUniqueId();
1752}
1753
1754void AudioFlinger::acquireAudioSessionId(int audioSession)
1755{
1756    Mutex::Autolock _l(mLock);
1757    pid_t caller = IPCThreadState::self()->getCallingPid();
1758    ALOGV("acquiring %d from %d", audioSession, caller);
1759    size_t num = mAudioSessionRefs.size();
1760    for (size_t i = 0; i< num; i++) {
1761        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1762        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1763            ref->mCnt++;
1764            ALOGV(" incremented refcount to %d", ref->mCnt);
1765            return;
1766        }
1767    }
1768    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1769    ALOGV(" added new entry for %d", audioSession);
1770}
1771
1772void AudioFlinger::releaseAudioSessionId(int audioSession)
1773{
1774    Mutex::Autolock _l(mLock);
1775    pid_t caller = IPCThreadState::self()->getCallingPid();
1776    ALOGV("releasing %d from %d", audioSession, caller);
1777    size_t num = mAudioSessionRefs.size();
1778    for (size_t i = 0; i< num; i++) {
1779        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1780        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1781            ref->mCnt--;
1782            ALOGV(" decremented refcount to %d", ref->mCnt);
1783            if (ref->mCnt == 0) {
1784                mAudioSessionRefs.removeAt(i);
1785                delete ref;
1786                purgeStaleEffects_l();
1787            }
1788            return;
1789        }
1790    }
1791    ALOGW("session id %d not found for pid %d", audioSession, caller);
1792}
1793
1794void AudioFlinger::purgeStaleEffects_l() {
1795
1796    ALOGV("purging stale effects");
1797
1798    Vector< sp<EffectChain> > chains;
1799
1800    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1801        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1802        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1803            sp<EffectChain> ec = t->mEffectChains[j];
1804            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1805                chains.push(ec);
1806            }
1807        }
1808    }
1809    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1810        sp<RecordThread> t = mRecordThreads.valueAt(i);
1811        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1812            sp<EffectChain> ec = t->mEffectChains[j];
1813            chains.push(ec);
1814        }
1815    }
1816
1817    for (size_t i = 0; i < chains.size(); i++) {
1818        sp<EffectChain> ec = chains[i];
1819        int sessionid = ec->sessionId();
1820        sp<ThreadBase> t = ec->mThread.promote();
1821        if (t == 0) {
1822            continue;
1823        }
1824        size_t numsessionrefs = mAudioSessionRefs.size();
1825        bool found = false;
1826        for (size_t k = 0; k < numsessionrefs; k++) {
1827            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1828            if (ref->mSessionid == sessionid) {
1829                ALOGV(" session %d still exists for %d with %d refs",
1830                    sessionid, ref->mPid, ref->mCnt);
1831                found = true;
1832                break;
1833            }
1834        }
1835        if (!found) {
1836            Mutex::Autolock _l (t->mLock);
1837            // remove all effects from the chain
1838            while (ec->mEffects.size()) {
1839                sp<EffectModule> effect = ec->mEffects[0];
1840                effect->unPin();
1841                t->removeEffect_l(effect);
1842                if (effect->purgeHandles()) {
1843                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
1844                }
1845                AudioSystem::unregisterEffect(effect->id());
1846            }
1847        }
1848    }
1849    return;
1850}
1851
1852// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
1853AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
1854{
1855    return mPlaybackThreads.valueFor(output).get();
1856}
1857
1858// checkMixerThread_l() must be called with AudioFlinger::mLock held
1859AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
1860{
1861    PlaybackThread *thread = checkPlaybackThread_l(output);
1862    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
1863}
1864
1865// checkRecordThread_l() must be called with AudioFlinger::mLock held
1866AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
1867{
1868    return mRecordThreads.valueFor(input).get();
1869}
1870
1871uint32_t AudioFlinger::nextUniqueId()
1872{
1873    return android_atomic_inc(&mNextUniqueId);
1874}
1875
1876AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
1877{
1878    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1879        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1880        AudioStreamOut *output = thread->getOutput();
1881        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
1882            return thread;
1883        }
1884    }
1885    return NULL;
1886}
1887
1888audio_devices_t AudioFlinger::primaryOutputDevice_l() const
1889{
1890    PlaybackThread *thread = primaryPlaybackThread_l();
1891
1892    if (thread == NULL) {
1893        return 0;
1894    }
1895
1896    return thread->outDevice();
1897}
1898
1899sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
1900                                    int triggerSession,
1901                                    int listenerSession,
1902                                    sync_event_callback_t callBack,
1903                                    void *cookie)
1904{
1905    Mutex::Autolock _l(mLock);
1906
1907    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
1908    status_t playStatus = NAME_NOT_FOUND;
1909    status_t recStatus = NAME_NOT_FOUND;
1910    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1911        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
1912        if (playStatus == NO_ERROR) {
1913            return event;
1914        }
1915    }
1916    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1917        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
1918        if (recStatus == NO_ERROR) {
1919            return event;
1920        }
1921    }
1922    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
1923        mPendingSyncEvents.add(event);
1924    } else {
1925        ALOGV("createSyncEvent() invalid event %d", event->type());
1926        event.clear();
1927    }
1928    return event;
1929}
1930
1931// ----------------------------------------------------------------------------
1932//  Effect management
1933// ----------------------------------------------------------------------------
1934
1935
1936status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
1937{
1938    Mutex::Autolock _l(mLock);
1939    return EffectQueryNumberEffects(numEffects);
1940}
1941
1942status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
1943{
1944    Mutex::Autolock _l(mLock);
1945    return EffectQueryEffect(index, descriptor);
1946}
1947
1948status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
1949        effect_descriptor_t *descriptor) const
1950{
1951    Mutex::Autolock _l(mLock);
1952    return EffectGetDescriptor(pUuid, descriptor);
1953}
1954
1955
1956sp<IEffect> AudioFlinger::createEffect(
1957        effect_descriptor_t *pDesc,
1958        const sp<IEffectClient>& effectClient,
1959        int32_t priority,
1960        audio_io_handle_t io,
1961        int sessionId,
1962        status_t *status,
1963        int *id,
1964        int *enabled)
1965{
1966    status_t lStatus = NO_ERROR;
1967    sp<EffectHandle> handle;
1968    effect_descriptor_t desc;
1969
1970    pid_t pid = IPCThreadState::self()->getCallingPid();
1971    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
1972            pid, effectClient.get(), priority, sessionId, io);
1973
1974    if (pDesc == NULL) {
1975        lStatus = BAD_VALUE;
1976        goto Exit;
1977    }
1978
1979    // check audio settings permission for global effects
1980    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
1981        lStatus = PERMISSION_DENIED;
1982        goto Exit;
1983    }
1984
1985    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
1986    // that can only be created by audio policy manager (running in same process)
1987    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
1988        lStatus = PERMISSION_DENIED;
1989        goto Exit;
1990    }
1991
1992    if (io == 0) {
1993        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
1994            // output must be specified by AudioPolicyManager when using session
1995            // AUDIO_SESSION_OUTPUT_STAGE
1996            lStatus = BAD_VALUE;
1997            goto Exit;
1998        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1999            // if the output returned by getOutputForEffect() is removed before we lock the
2000            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2001            // and we will exit safely
2002            io = AudioSystem::getOutputForEffect(&desc);
2003        }
2004    }
2005
2006    {
2007        Mutex::Autolock _l(mLock);
2008
2009
2010        if (!EffectIsNullUuid(&pDesc->uuid)) {
2011            // if uuid is specified, request effect descriptor
2012            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2013            if (lStatus < 0) {
2014                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2015                goto Exit;
2016            }
2017        } else {
2018            // if uuid is not specified, look for an available implementation
2019            // of the required type in effect factory
2020            if (EffectIsNullUuid(&pDesc->type)) {
2021                ALOGW("createEffect() no effect type");
2022                lStatus = BAD_VALUE;
2023                goto Exit;
2024            }
2025            uint32_t numEffects = 0;
2026            effect_descriptor_t d;
2027            d.flags = 0; // prevent compiler warning
2028            bool found = false;
2029
2030            lStatus = EffectQueryNumberEffects(&numEffects);
2031            if (lStatus < 0) {
2032                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2033                goto Exit;
2034            }
2035            for (uint32_t i = 0; i < numEffects; i++) {
2036                lStatus = EffectQueryEffect(i, &desc);
2037                if (lStatus < 0) {
2038                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2039                    continue;
2040                }
2041                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2042                    // If matching type found save effect descriptor. If the session is
2043                    // 0 and the effect is not auxiliary, continue enumeration in case
2044                    // an auxiliary version of this effect type is available
2045                    found = true;
2046                    d = desc;
2047                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2048                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2049                        break;
2050                    }
2051                }
2052            }
2053            if (!found) {
2054                lStatus = BAD_VALUE;
2055                ALOGW("createEffect() effect not found");
2056                goto Exit;
2057            }
2058            // For same effect type, chose auxiliary version over insert version if
2059            // connect to output mix (Compliance to OpenSL ES)
2060            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2061                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2062                desc = d;
2063            }
2064        }
2065
2066        // Do not allow auxiliary effects on a session different from 0 (output mix)
2067        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2068             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2069            lStatus = INVALID_OPERATION;
2070            goto Exit;
2071        }
2072
2073        // check recording permission for visualizer
2074        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2075            !recordingAllowed()) {
2076            lStatus = PERMISSION_DENIED;
2077            goto Exit;
2078        }
2079
2080        // return effect descriptor
2081        *pDesc = desc;
2082
2083        // If output is not specified try to find a matching audio session ID in one of the
2084        // output threads.
2085        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2086        // because of code checking output when entering the function.
2087        // Note: io is never 0 when creating an effect on an input
2088        if (io == 0) {
2089            // look for the thread where the specified audio session is present
2090            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2091                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2092                    io = mPlaybackThreads.keyAt(i);
2093                    break;
2094                }
2095            }
2096            if (io == 0) {
2097                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2098                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2099                        io = mRecordThreads.keyAt(i);
2100                        break;
2101                    }
2102                }
2103            }
2104            // If no output thread contains the requested session ID, default to
2105            // first output. The effect chain will be moved to the correct output
2106            // thread when a track with the same session ID is created
2107            if (io == 0 && mPlaybackThreads.size()) {
2108                io = mPlaybackThreads.keyAt(0);
2109            }
2110            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2111        }
2112        ThreadBase *thread = checkRecordThread_l(io);
2113        if (thread == NULL) {
2114            thread = checkPlaybackThread_l(io);
2115            if (thread == NULL) {
2116                ALOGE("createEffect() unknown output thread");
2117                lStatus = BAD_VALUE;
2118                goto Exit;
2119            }
2120        }
2121
2122        sp<Client> client = registerPid_l(pid);
2123
2124        // create effect on selected output thread
2125        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2126                &desc, enabled, &lStatus);
2127        if (handle != 0 && id != NULL) {
2128            *id = handle->id();
2129        }
2130    }
2131
2132Exit:
2133    if (status != NULL) {
2134        *status = lStatus;
2135    }
2136    return handle;
2137}
2138
2139status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2140        audio_io_handle_t dstOutput)
2141{
2142    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2143            sessionId, srcOutput, dstOutput);
2144    Mutex::Autolock _l(mLock);
2145    if (srcOutput == dstOutput) {
2146        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2147        return NO_ERROR;
2148    }
2149    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2150    if (srcThread == NULL) {
2151        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2152        return BAD_VALUE;
2153    }
2154    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2155    if (dstThread == NULL) {
2156        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2157        return BAD_VALUE;
2158    }
2159
2160    Mutex::Autolock _dl(dstThread->mLock);
2161    Mutex::Autolock _sl(srcThread->mLock);
2162    moveEffectChain_l(sessionId, srcThread, dstThread, false);
2163
2164    return NO_ERROR;
2165}
2166
2167// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2168status_t AudioFlinger::moveEffectChain_l(int sessionId,
2169                                   AudioFlinger::PlaybackThread *srcThread,
2170                                   AudioFlinger::PlaybackThread *dstThread,
2171                                   bool reRegister)
2172{
2173    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2174            sessionId, srcThread, dstThread);
2175
2176    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2177    if (chain == 0) {
2178        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2179                sessionId, srcThread);
2180        return INVALID_OPERATION;
2181    }
2182
2183    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2184    // so that a new chain is created with correct parameters when first effect is added. This is
2185    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2186    // removed.
2187    srcThread->removeEffectChain_l(chain);
2188
2189    // transfer all effects one by one so that new effect chain is created on new thread with
2190    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2191    audio_io_handle_t dstOutput = dstThread->id();
2192    sp<EffectChain> dstChain;
2193    uint32_t strategy = 0; // prevent compiler warning
2194    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2195    while (effect != 0) {
2196        srcThread->removeEffect_l(effect);
2197        dstThread->addEffect_l(effect);
2198        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2199        if (effect->state() == EffectModule::ACTIVE ||
2200                effect->state() == EffectModule::STOPPING) {
2201            effect->start();
2202        }
2203        // if the move request is not received from audio policy manager, the effect must be
2204        // re-registered with the new strategy and output
2205        if (dstChain == 0) {
2206            dstChain = effect->chain().promote();
2207            if (dstChain == 0) {
2208                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2209                srcThread->addEffect_l(effect);
2210                return NO_INIT;
2211            }
2212            strategy = dstChain->strategy();
2213        }
2214        if (reRegister) {
2215            AudioSystem::unregisterEffect(effect->id());
2216            AudioSystem::registerEffect(&effect->desc(),
2217                                        dstOutput,
2218                                        strategy,
2219                                        sessionId,
2220                                        effect->id());
2221        }
2222        effect = chain->getEffectFromId_l(0);
2223    }
2224
2225    return NO_ERROR;
2226}
2227
2228struct Entry {
2229#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2230    char mName[MAX_NAME];
2231};
2232
2233int comparEntry(const void *p1, const void *p2)
2234{
2235    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2236}
2237
2238void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2239{
2240    NBAIO_Source *teeSource = source.get();
2241    if (teeSource != NULL) {
2242        // .wav rotation
2243        // There is a benign race condition if 2 threads call this simultaneously.
2244        // They would both traverse the directory, but the result would simply be
2245        // failures at unlink() which are ignored.  It's also unlikely since
2246        // normally dumpsys is only done by bugreport or from the command line.
2247        char teePath[32+256];
2248        strcpy(teePath, "/data/misc/media");
2249        size_t teePathLen = strlen(teePath);
2250        DIR *dir = opendir(teePath);
2251        teePath[teePathLen++] = '/';
2252        if (dir != NULL) {
2253#define MAX_SORT 20 // number of entries to sort
2254#define MAX_KEEP 10 // number of entries to keep
2255            struct Entry entries[MAX_SORT];
2256            size_t entryCount = 0;
2257            while (entryCount < MAX_SORT) {
2258                struct dirent de;
2259                struct dirent *result = NULL;
2260                int rc = readdir_r(dir, &de, &result);
2261                if (rc != 0) {
2262                    ALOGW("readdir_r failed %d", rc);
2263                    break;
2264                }
2265                if (result == NULL) {
2266                    break;
2267                }
2268                if (result != &de) {
2269                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2270                    break;
2271                }
2272                // ignore non .wav file entries
2273                size_t nameLen = strlen(de.d_name);
2274                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2275                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2276                    continue;
2277                }
2278                strcpy(entries[entryCount++].mName, de.d_name);
2279            }
2280            (void) closedir(dir);
2281            if (entryCount > MAX_KEEP) {
2282                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2283                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2284                    strcpy(&teePath[teePathLen], entries[i].mName);
2285                    (void) unlink(teePath);
2286                }
2287            }
2288        } else {
2289            if (fd >= 0) {
2290                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2291            }
2292        }
2293        char teeTime[16];
2294        struct timeval tv;
2295        gettimeofday(&tv, NULL);
2296        struct tm tm;
2297        localtime_r(&tv.tv_sec, &tm);
2298        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2299        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2300        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2301        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2302        if (teeFd >= 0) {
2303            char wavHeader[44];
2304            memcpy(wavHeader,
2305                "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",
2306                sizeof(wavHeader));
2307            NBAIO_Format format = teeSource->format();
2308            unsigned channelCount = Format_channelCount(format);
2309            ALOG_ASSERT(channelCount <= FCC_2);
2310            uint32_t sampleRate = Format_sampleRate(format);
2311            wavHeader[22] = channelCount;       // number of channels
2312            wavHeader[24] = sampleRate;         // sample rate
2313            wavHeader[25] = sampleRate >> 8;
2314            wavHeader[32] = channelCount * 2;   // block alignment
2315            write(teeFd, wavHeader, sizeof(wavHeader));
2316            size_t total = 0;
2317            bool firstRead = true;
2318            for (;;) {
2319#define TEE_SINK_READ 1024
2320                short buffer[TEE_SINK_READ * FCC_2];
2321                size_t count = TEE_SINK_READ;
2322                ssize_t actual = teeSource->read(buffer, count,
2323                        AudioBufferProvider::kInvalidPTS);
2324                bool wasFirstRead = firstRead;
2325                firstRead = false;
2326                if (actual <= 0) {
2327                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2328                        continue;
2329                    }
2330                    break;
2331                }
2332                ALOG_ASSERT(actual <= (ssize_t)count);
2333                write(teeFd, buffer, actual * channelCount * sizeof(short));
2334                total += actual;
2335            }
2336            lseek(teeFd, (off_t) 4, SEEK_SET);
2337            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2338            write(teeFd, &temp, sizeof(temp));
2339            lseek(teeFd, (off_t) 40, SEEK_SET);
2340            temp =  total * channelCount * sizeof(short);
2341            write(teeFd, &temp, sizeof(temp));
2342            close(teeFd);
2343            if (fd >= 0) {
2344                fdprintf(fd, "tee copied to %s\n", teePath);
2345            }
2346        } else {
2347            if (fd >= 0) {
2348                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2349            }
2350        }
2351    }
2352}
2353
2354// ----------------------------------------------------------------------------
2355
2356status_t AudioFlinger::onTransact(
2357        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2358{
2359    return BnAudioFlinger::onTransact(code, data, reply, flags);
2360}
2361
2362}; // namespace android
2363