AudioFlinger.cpp revision 535235cf879728dca680279c21b37d5b0be5b10f 
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp
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 <math.h>
23#include <signal.h>
24#include <sys/time.h>
25#include <sys/resource.h>
26
27#include <binder/IPCThreadState.h>
28#include <binder/IServiceManager.h>
29#include <utils/Log.h>
30#include <binder/Parcel.h>
31#include <binder/IPCThreadState.h>
32#include <utils/String16.h>
33#include <utils/threads.h>
34#include <utils/Atomic.h>
35
36#include <cutils/bitops.h>
37#include <cutils/properties.h>
38
39#include <media/AudioTrack.h>
40#include <media/AudioRecord.h>
41#include <media/IMediaPlayerService.h>
42
43#include <private/media/AudioTrackShared.h>
44#include <private/media/AudioEffectShared.h>
45
46#include <system/audio.h>
47#include <hardware/audio.h>
48
49#include "AudioMixer.h"
50#include "AudioFlinger.h"
51
52#include <media/EffectsFactoryApi.h>
53#include <audio_effects/effect_visualizer.h>
54#include <audio_effects/effect_ns.h>
55#include <audio_effects/effect_aec.h>
56
57#include <cpustats/ThreadCpuUsage.h>
58#include <powermanager/PowerManager.h>
59// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
60
61#include <aah_timesrv/cc_helper.h>
62#include <aah_timesrv/local_clock.h>
63
64// ----------------------------------------------------------------------------
65
66
67namespace android {
68
69static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
70static const char* kHardwareLockedString = "Hardware lock is taken\n";
71
72//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
73static const float MAX_GAIN = 4096.0f;
74static const float MAX_GAIN_INT = 0x1000;
75
76// retry counts for buffer fill timeout
77// 50 * ~20msecs = 1 second
78static const int8_t kMaxTrackRetries = 50;
79static const int8_t kMaxTrackStartupRetries = 50;
80// allow less retry attempts on direct output thread.
81// direct outputs can be a scarce resource in audio hardware and should
82// be released as quickly as possible.
83static const int8_t kMaxTrackRetriesDirect = 2;
84
85static const int kDumpLockRetries = 50;
86static const int kDumpLockSleep = 20000;
87
88static const nsecs_t kWarningThrottle = seconds(5);
89
90// RecordThread loop sleep time upon application overrun or audio HAL read error
91static const int kRecordThreadSleepUs = 5000;
92
93// ----------------------------------------------------------------------------
94
95static bool recordingAllowed() {
96    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
97    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
98    if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
99    return ok;
100}
101
102static bool settingsAllowed() {
103    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
104    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
105    if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
106    return ok;
107}
108
109// To collect the amplifier usage
110static void addBatteryData(uint32_t params) {
111    sp<IBinder> binder =
112        defaultServiceManager()->getService(String16("media.player"));
113    sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder);
114    if (service.get() == NULL) {
115        LOGW("Cannot connect to the MediaPlayerService for battery tracking");
116        return;
117    }
118
119    service->addBatteryData(params);
120}
121
122static int load_audio_interface(const char *if_name, const hw_module_t **mod,
123                                audio_hw_device_t **dev)
124{
125    int rc;
126
127    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
128    if (rc)
129        goto out;
130
131    rc = audio_hw_device_open(*mod, dev);
132    LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
133            AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
134    if (rc)
135        goto out;
136
137    return 0;
138
139out:
140    *mod = NULL;
141    *dev = NULL;
142    return rc;
143}
144
145static const char *audio_interfaces[] = {
146    "primary",
147    "a2dp",
148    "usb",
149};
150#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
151
152// ----------------------------------------------------------------------------
153
154AudioFlinger::AudioFlinger()
155    : BnAudioFlinger(),
156      mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterVolumeSW(1.0f),
157      mMasterVolumeSupportLvl(MVS_NONE), mMasterMute(false), mNextUniqueId(1),
158      mBtNrec(false)
159{
160}
161
162void AudioFlinger::onFirstRef()
163{
164    int rc = 0;
165
166    Mutex::Autolock _l(mLock);
167
168    /* TODO: move all this work into an Init() function */
169    mHardwareStatus = AUDIO_HW_IDLE;
170
171    for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
172        const hw_module_t *mod;
173        audio_hw_device_t *dev;
174
175        rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
176        if (rc)
177            continue;
178
179        LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
180             mod->name, mod->id);
181        mAudioHwDevs.push(dev);
182
183        if (!mPrimaryHardwareDev) {
184            mPrimaryHardwareDev = dev;
185            LOGI("Using '%s' (%s.%s) as the primary audio interface",
186                 mod->name, mod->id, audio_interfaces[i]);
187        }
188    }
189
190    mHardwareStatus = AUDIO_HW_INIT;
191
192    if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
193        LOGE("Primary audio interface not found");
194        return;
195    }
196
197    // Determine the level of master volume support the primary audio HAL has,
198    // and set the initial master volume at the same time.
199    float initialVolume = 1.0;
200    mMasterVolumeSupportLvl = MVS_NONE;
201    if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) {
202        AutoMutex lock(mHardwareLock);
203        audio_hw_device_t *dev = mPrimaryHardwareDev;
204
205        mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
206        if ((NULL != dev->get_master_volume) &&
207            (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) {
208            mMasterVolumeSupportLvl = MVS_FULL;
209        } else {
210            mMasterVolumeSupportLvl = MVS_SETONLY;
211            initialVolume = 1.0;
212        }
213
214        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
215        if ((NULL == dev->set_master_volume) ||
216            (NO_ERROR != dev->set_master_volume(dev, initialVolume))) {
217            mMasterVolumeSupportLvl = MVS_NONE;
218        }
219        mHardwareStatus = AUDIO_HW_INIT;
220    }
221
222    // Set the mode for each audio HAL, and try to set the initial volume (if
223    // supported) for all of the non-primary audio HALs.
224    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
225        audio_hw_device_t *dev = mAudioHwDevs[i];
226
227        mHardwareStatus = AUDIO_HW_INIT;
228        rc = dev->init_check(dev);
229        if (rc == 0) {
230            AutoMutex lock(mHardwareLock);
231
232            mMode = AUDIO_MODE_NORMAL;
233            mHardwareStatus = AUDIO_HW_SET_MODE;
234            dev->set_mode(dev, mMode);
235
236            if ((dev != mPrimaryHardwareDev) &&
237                (NULL != dev->set_master_volume)) {
238                mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
239                dev->set_master_volume(dev, initialVolume);
240            }
241
242            mHardwareStatus = AUDIO_HW_INIT;
243        }
244    }
245
246    mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl)
247                    ? initialVolume
248                    : 1.0;
249    mMasterVolume   = initialVolume;
250    mHardwareStatus = AUDIO_HW_IDLE;
251}
252
253status_t AudioFlinger::initCheck() const
254{
255    Mutex::Autolock _l(mLock);
256    if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
257        return NO_INIT;
258    return NO_ERROR;
259}
260
261AudioFlinger::~AudioFlinger()
262{
263    int num_devs = mAudioHwDevs.size();
264
265    while (!mRecordThreads.isEmpty()) {
266        // closeInput() will remove first entry from mRecordThreads
267        closeInput(mRecordThreads.keyAt(0));
268    }
269    while (!mPlaybackThreads.isEmpty()) {
270        // closeOutput() will remove first entry from mPlaybackThreads
271        closeOutput(mPlaybackThreads.keyAt(0));
272    }
273
274    for (int i = 0; i < num_devs; i++) {
275        audio_hw_device_t *dev = mAudioHwDevs[i];
276        audio_hw_device_close(dev);
277    }
278    mAudioHwDevs.clear();
279}
280
281audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
282{
283    /* first matching HW device is returned */
284    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
285        audio_hw_device_t *dev = mAudioHwDevs[i];
286        if ((dev->get_supported_devices(dev) & devices) == devices)
287            return dev;
288    }
289    return NULL;
290}
291
292status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
293{
294    const size_t SIZE = 256;
295    char buffer[SIZE];
296    String8 result;
297
298    result.append("Clients:\n");
299    for (size_t i = 0; i < mClients.size(); ++i) {
300        wp<Client> wClient = mClients.valueAt(i);
301        if (wClient != 0) {
302            sp<Client> client = wClient.promote();
303            if (client != 0) {
304                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
305                result.append(buffer);
306            }
307        }
308    }
309
310    result.append("Global session refs:\n");
311    result.append(" session pid cnt\n");
312    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
313        AudioSessionRef *r = mAudioSessionRefs[i];
314        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
315        result.append(buffer);
316    }
317    write(fd, result.string(), result.size());
318    return NO_ERROR;
319}
320
321
322status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
323{
324    const size_t SIZE = 256;
325    char buffer[SIZE];
326    String8 result;
327    int hardwareStatus = mHardwareStatus;
328
329    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
330    result.append(buffer);
331    write(fd, result.string(), result.size());
332    return NO_ERROR;
333}
334
335status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
336{
337    const size_t SIZE = 256;
338    char buffer[SIZE];
339    String8 result;
340    snprintf(buffer, SIZE, "Permission Denial: "
341            "can't dump AudioFlinger from pid=%d, uid=%d\n",
342            IPCThreadState::self()->getCallingPid(),
343            IPCThreadState::self()->getCallingUid());
344    result.append(buffer);
345    write(fd, result.string(), result.size());
346    return NO_ERROR;
347}
348
349static bool tryLock(Mutex& mutex)
350{
351    bool locked = false;
352    for (int i = 0; i < kDumpLockRetries; ++i) {
353        if (mutex.tryLock() == NO_ERROR) {
354            locked = true;
355            break;
356        }
357        usleep(kDumpLockSleep);
358    }
359    return locked;
360}
361
362status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
363{
364    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
365        dumpPermissionDenial(fd, args);
366    } else {
367        // get state of hardware lock
368        bool hardwareLocked = tryLock(mHardwareLock);
369        if (!hardwareLocked) {
370            String8 result(kHardwareLockedString);
371            write(fd, result.string(), result.size());
372        } else {
373            mHardwareLock.unlock();
374        }
375
376        bool locked = tryLock(mLock);
377
378        // failed to lock - AudioFlinger is probably deadlocked
379        if (!locked) {
380            String8 result(kDeadlockedString);
381            write(fd, result.string(), result.size());
382        }
383
384        dumpClients(fd, args);
385        dumpInternals(fd, args);
386
387        // dump playback threads
388        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
389            mPlaybackThreads.valueAt(i)->dump(fd, args);
390        }
391
392        // dump record threads
393        for (size_t i = 0; i < mRecordThreads.size(); i++) {
394            mRecordThreads.valueAt(i)->dump(fd, args);
395        }
396
397        // dump all hardware devs
398        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
399            audio_hw_device_t *dev = mAudioHwDevs[i];
400            dev->dump(dev, fd);
401        }
402        if (locked) mLock.unlock();
403    }
404    return NO_ERROR;
405}
406
407
408// IAudioFlinger interface
409
410
411sp<IAudioTrack> AudioFlinger::createTrack(
412        pid_t pid,
413        int streamType,
414        uint32_t sampleRate,
415        uint32_t format,
416        uint32_t channelMask,
417        int frameCount,
418        uint32_t flags,
419        const sp<IMemory>& sharedBuffer,
420        int output,
421        bool isTimed,
422        int *sessionId,
423        status_t *status)
424{
425    sp<PlaybackThread::Track> track;
426    sp<TrackHandle> trackHandle;
427    sp<Client> client;
428    wp<Client> wclient;
429    status_t lStatus;
430    int lSessionId;
431
432    if (streamType >= AUDIO_STREAM_CNT) {
433        LOGE("invalid stream type");
434        lStatus = BAD_VALUE;
435        goto Exit;
436    }
437
438    {
439        Mutex::Autolock _l(mLock);
440        PlaybackThread *thread = checkPlaybackThread_l(output);
441        PlaybackThread *effectThread = NULL;
442        if (thread == NULL) {
443            LOGE("unknown output thread");
444            lStatus = BAD_VALUE;
445            goto Exit;
446        }
447
448        wclient = mClients.valueFor(pid);
449
450        if (wclient != NULL) {
451            client = wclient.promote();
452        } else {
453            client = new Client(this, pid);
454            mClients.add(pid, client);
455        }
456
457        LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
458        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
459            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
460                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
461                if (mPlaybackThreads.keyAt(i) != output) {
462                    // prevent same audio session on different output threads
463                    uint32_t sessions = t->hasAudioSession(*sessionId);
464                    if (sessions & PlaybackThread::TRACK_SESSION) {
465                        lStatus = BAD_VALUE;
466                        goto Exit;
467                    }
468                    // check if an effect with same session ID is waiting for a track to be created
469                    if (sessions & PlaybackThread::EFFECT_SESSION) {
470                        effectThread = t.get();
471                    }
472                }
473            }
474            lSessionId = *sessionId;
475        } else {
476            // if no audio session id is provided, create one here
477            lSessionId = nextUniqueId();
478            if (sessionId != NULL) {
479                *sessionId = lSessionId;
480            }
481        }
482        LOGV("createTrack() lSessionId: %d", lSessionId);
483
484        track = thread->createTrack_l(client, streamType, sampleRate, format,
485                channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus);
486
487        // move effect chain to this output thread if an effect on same session was waiting
488        // for a track to be created
489        if (lStatus == NO_ERROR && effectThread != NULL) {
490            Mutex::Autolock _dl(thread->mLock);
491            Mutex::Autolock _sl(effectThread->mLock);
492            moveEffectChain_l(lSessionId, effectThread, thread, true);
493        }
494    }
495    if (lStatus == NO_ERROR) {
496        trackHandle = new TrackHandle(track);
497    } else {
498        // remove local strong reference to Client before deleting the Track so that the Client
499        // destructor is called by the TrackBase destructor with mLock held
500        client.clear();
501        track.clear();
502    }
503
504Exit:
505    if(status) {
506        *status = lStatus;
507    }
508    return trackHandle;
509}
510
511uint32_t AudioFlinger::sampleRate(int output) const
512{
513    Mutex::Autolock _l(mLock);
514    PlaybackThread *thread = checkPlaybackThread_l(output);
515    if (thread == NULL) {
516        LOGW("sampleRate() unknown thread %d", output);
517        return 0;
518    }
519    return thread->sampleRate();
520}
521
522int AudioFlinger::channelCount(int output) const
523{
524    Mutex::Autolock _l(mLock);
525    PlaybackThread *thread = checkPlaybackThread_l(output);
526    if (thread == NULL) {
527        LOGW("channelCount() unknown thread %d", output);
528        return 0;
529    }
530    return thread->channelCount();
531}
532
533uint32_t AudioFlinger::format(int output) const
534{
535    Mutex::Autolock _l(mLock);
536    PlaybackThread *thread = checkPlaybackThread_l(output);
537    if (thread == NULL) {
538        LOGW("format() unknown thread %d", output);
539        return 0;
540    }
541    return thread->format();
542}
543
544size_t AudioFlinger::frameCount(int output) const
545{
546    Mutex::Autolock _l(mLock);
547    PlaybackThread *thread = checkPlaybackThread_l(output);
548    if (thread == NULL) {
549        LOGW("frameCount() unknown thread %d", output);
550        return 0;
551    }
552    return thread->frameCount();
553}
554
555uint32_t AudioFlinger::latency(int output) const
556{
557    Mutex::Autolock _l(mLock);
558    PlaybackThread *thread = checkPlaybackThread_l(output);
559    if (thread == NULL) {
560        LOGW("latency() unknown thread %d", output);
561        return 0;
562    }
563    return thread->latency();
564}
565
566status_t AudioFlinger::setMasterVolume(float value)
567{
568    status_t ret = initCheck();
569    if (ret != NO_ERROR) {
570        return ret;
571    }
572
573    // check calling permissions
574    if (!settingsAllowed()) {
575        return PERMISSION_DENIED;
576    }
577
578    float swmv = value;
579
580    // when hw supports master volume, don't scale in sw mixer
581    if (MVS_NONE != mMasterVolumeSupportLvl) {
582        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
583            AutoMutex lock(mHardwareLock);
584            audio_hw_device_t *dev = mAudioHwDevs[i];
585
586            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
587            if (NULL != dev->set_master_volume) {
588                dev->set_master_volume(dev, value);
589            }
590            mHardwareStatus = AUDIO_HW_IDLE;
591        }
592
593        swmv = 1.0;
594    }
595
596    Mutex::Autolock _l(mLock);
597    mMasterVolume   = value;
598    mMasterVolumeSW = swmv;
599    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
600       mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
601
602    return NO_ERROR;
603}
604
605status_t AudioFlinger::setMode(int mode)
606{
607    status_t ret = initCheck();
608    if (ret != NO_ERROR) {
609        return ret;
610    }
611
612    // check calling permissions
613    if (!settingsAllowed()) {
614        return PERMISSION_DENIED;
615    }
616    if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) {
617        LOGW("Illegal value: setMode(%d)", mode);
618        return BAD_VALUE;
619    }
620
621    { // scope for the lock
622        AutoMutex lock(mHardwareLock);
623        mHardwareStatus = AUDIO_HW_SET_MODE;
624        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
625        mHardwareStatus = AUDIO_HW_IDLE;
626    }
627
628    if (NO_ERROR == ret) {
629        Mutex::Autolock _l(mLock);
630        mMode = mode;
631        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
632           mPlaybackThreads.valueAt(i)->setMode(mode);
633    }
634
635    return ret;
636}
637
638status_t AudioFlinger::setMicMute(bool state)
639{
640    status_t ret = initCheck();
641    if (ret != NO_ERROR) {
642        return ret;
643    }
644
645    // check calling permissions
646    if (!settingsAllowed()) {
647        return PERMISSION_DENIED;
648    }
649
650    AutoMutex lock(mHardwareLock);
651    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
652    ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
653    mHardwareStatus = AUDIO_HW_IDLE;
654    return ret;
655}
656
657bool AudioFlinger::getMicMute() const
658{
659    status_t ret = initCheck();
660    if (ret != NO_ERROR) {
661        return false;
662    }
663
664    bool state = AUDIO_MODE_INVALID;
665    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
666    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
667    mHardwareStatus = AUDIO_HW_IDLE;
668    return state;
669}
670
671status_t AudioFlinger::setMasterMute(bool muted)
672{
673    // check calling permissions
674    if (!settingsAllowed()) {
675        return PERMISSION_DENIED;
676    }
677
678    Mutex::Autolock _l(mLock);
679    mMasterMute = muted;
680    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
681       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
682
683    return NO_ERROR;
684}
685
686float AudioFlinger::masterVolume() const
687{
688    if (MVS_FULL == mMasterVolumeSupportLvl) {
689        float ret_val;
690        AutoMutex lock(mHardwareLock);
691
692        mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
693        assert(NULL != mPrimaryHardwareDev);
694        assert(NULL != mPrimaryHardwareDev->get_master_volume);
695
696        mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val);
697        mHardwareStatus = AUDIO_HW_IDLE;
698        return ret_val;
699    }
700
701    return mMasterVolume;
702}
703
704float AudioFlinger::masterVolumeSW() const
705{
706    return mMasterVolumeSW;
707}
708
709bool AudioFlinger::masterMute() const
710{
711    return mMasterMute;
712}
713
714status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
715{
716    // check calling permissions
717    if (!settingsAllowed()) {
718        return PERMISSION_DENIED;
719    }
720
721    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
722        return BAD_VALUE;
723    }
724
725    AutoMutex lock(mLock);
726    PlaybackThread *thread = NULL;
727    if (output) {
728        thread = checkPlaybackThread_l(output);
729        if (thread == NULL) {
730            return BAD_VALUE;
731        }
732    }
733
734    mStreamTypes[stream].volume = value;
735
736    if (thread == NULL) {
737        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
738           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
739        }
740    } else {
741        thread->setStreamVolume(stream, value);
742    }
743
744    return NO_ERROR;
745}
746
747status_t AudioFlinger::setStreamMute(int stream, bool muted)
748{
749    // check calling permissions
750    if (!settingsAllowed()) {
751        return PERMISSION_DENIED;
752    }
753
754    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
755        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
756        return BAD_VALUE;
757    }
758
759    AutoMutex lock(mLock);
760    mStreamTypes[stream].mute = muted;
761    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
762       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
763
764    return NO_ERROR;
765}
766
767float AudioFlinger::streamVolume(int stream, int output) const
768{
769    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
770        return 0.0f;
771    }
772
773    AutoMutex lock(mLock);
774    float volume;
775    if (output) {
776        PlaybackThread *thread = checkPlaybackThread_l(output);
777        if (thread == NULL) {
778            return 0.0f;
779        }
780        volume = thread->streamVolume(stream);
781    } else {
782        volume = mStreamTypes[stream].volume;
783    }
784
785    return volume;
786}
787
788bool AudioFlinger::streamMute(int stream) const
789{
790    if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
791        return true;
792    }
793
794    return mStreamTypes[stream].mute;
795}
796
797status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
798{
799    status_t result;
800
801    LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
802            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
803    // check calling permissions
804    if (!settingsAllowed()) {
805        return PERMISSION_DENIED;
806    }
807
808    // ioHandle == 0 means the parameters are global to the audio hardware interface
809    if (ioHandle == 0) {
810        AutoMutex lock(mHardwareLock);
811        mHardwareStatus = AUDIO_SET_PARAMETER;
812        status_t final_result = NO_ERROR;
813        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
814            audio_hw_device_t *dev = mAudioHwDevs[i];
815            result = dev->set_parameters(dev, keyValuePairs.string());
816            final_result = result ?: final_result;
817        }
818        mHardwareStatus = AUDIO_HW_IDLE;
819        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
820        AudioParameter param = AudioParameter(keyValuePairs);
821        String8 value;
822        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
823            Mutex::Autolock _l(mLock);
824            bool btNrec = (value == AUDIO_PARAMETER_VALUE_ON);
825            if (mBtNrec != btNrec) {
826                for (size_t i = 0; i < mRecordThreads.size(); i++) {
827                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
828                    RecordThread::RecordTrack *track = thread->track();
829                    if (track != NULL) {
830                        audio_devices_t device = (audio_devices_t)(
831                                thread->device() & AUDIO_DEVICE_IN_ALL);
832                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrec;
833                        thread->setEffectSuspended(FX_IID_AEC,
834                                                   suspend,
835                                                   track->sessionId());
836                        thread->setEffectSuspended(FX_IID_NS,
837                                                   suspend,
838                                                   track->sessionId());
839                    }
840                }
841                mBtNrec = btNrec;
842            }
843        }
844        return final_result;
845    }
846
847    // hold a strong ref on thread in case closeOutput() or closeInput() is called
848    // and the thread is exited once the lock is released
849    sp<ThreadBase> thread;
850    {
851        Mutex::Autolock _l(mLock);
852        thread = checkPlaybackThread_l(ioHandle);
853        if (thread == NULL) {
854            thread = checkRecordThread_l(ioHandle);
855        } else if (thread.get() == primaryPlaybackThread_l()) {
856            // indicate output device change to all input threads for pre processing
857            AudioParameter param = AudioParameter(keyValuePairs);
858            int value;
859            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
860                for (size_t i = 0; i < mRecordThreads.size(); i++) {
861                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
862                }
863            }
864        }
865    }
866    if (thread != NULL) {
867        result = thread->setParameters(keyValuePairs);
868        return result;
869    }
870    return BAD_VALUE;
871}
872
873String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
874{
875//    LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
876//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
877
878    if (ioHandle == 0) {
879        String8 out_s8;
880
881        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
882            audio_hw_device_t *dev = mAudioHwDevs[i];
883            char *s = dev->get_parameters(dev, keys.string());
884            out_s8 += String8(s);
885            free(s);
886        }
887        return out_s8;
888    }
889
890    Mutex::Autolock _l(mLock);
891
892    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
893    if (playbackThread != NULL) {
894        return playbackThread->getParameters(keys);
895    }
896    RecordThread *recordThread = checkRecordThread_l(ioHandle);
897    if (recordThread != NULL) {
898        return recordThread->getParameters(keys);
899    }
900    return String8("");
901}
902
903size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
904{
905    status_t ret = initCheck();
906    if (ret != NO_ERROR) {
907        return 0;
908    }
909
910    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
911}
912
913unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
914{
915    if (ioHandle == 0) {
916        return 0;
917    }
918
919    Mutex::Autolock _l(mLock);
920
921    RecordThread *recordThread = checkRecordThread_l(ioHandle);
922    if (recordThread != NULL) {
923        return recordThread->getInputFramesLost();
924    }
925    return 0;
926}
927
928status_t AudioFlinger::setVoiceVolume(float value)
929{
930    status_t ret = initCheck();
931    if (ret != NO_ERROR) {
932        return ret;
933    }
934
935    // check calling permissions
936    if (!settingsAllowed()) {
937        return PERMISSION_DENIED;
938    }
939
940    AutoMutex lock(mHardwareLock);
941    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
942    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
943    mHardwareStatus = AUDIO_HW_IDLE;
944
945    return ret;
946}
947
948status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
949{
950    status_t status;
951
952    Mutex::Autolock _l(mLock);
953
954    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
955    if (playbackThread != NULL) {
956        return playbackThread->getRenderPosition(halFrames, dspFrames);
957    }
958
959    return BAD_VALUE;
960}
961
962void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
963{
964
965    Mutex::Autolock _l(mLock);
966
967    int pid = IPCThreadState::self()->getCallingPid();
968    if (mNotificationClients.indexOfKey(pid) < 0) {
969        sp<NotificationClient> notificationClient = new NotificationClient(this,
970                                                                            client,
971                                                                            pid);
972        LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
973
974        mNotificationClients.add(pid, notificationClient);
975
976        sp<IBinder> binder = client->asBinder();
977        binder->linkToDeath(notificationClient);
978
979        // the config change is always sent from playback or record threads to avoid deadlock
980        // with AudioSystem::gLock
981        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
982            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
983        }
984
985        for (size_t i = 0; i < mRecordThreads.size(); i++) {
986            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
987        }
988    }
989}
990
991void AudioFlinger::removeNotificationClient(pid_t pid)
992{
993    Mutex::Autolock _l(mLock);
994
995    int index = mNotificationClients.indexOfKey(pid);
996    if (index >= 0) {
997        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
998        LOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
999        mNotificationClients.removeItem(pid);
1000    }
1001
1002    LOGV("%d died, releasing its sessions", pid);
1003    int num = mAudioSessionRefs.size();
1004    bool removed = false;
1005    for (int i = 0; i< num; i++) {
1006        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1007        LOGV(" pid %d @ %d", ref->pid, i);
1008        if (ref->pid == pid) {
1009            LOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
1010            mAudioSessionRefs.removeAt(i);
1011            delete ref;
1012            removed = true;
1013            i--;
1014            num--;
1015        }
1016    }
1017    if (removed) {
1018        purgeStaleEffects_l();
1019    }
1020}
1021
1022// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1023void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
1024{
1025    size_t size = mNotificationClients.size();
1026    for (size_t i = 0; i < size; i++) {
1027        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
1028    }
1029}
1030
1031// removeClient_l() must be called with AudioFlinger::mLock held
1032void AudioFlinger::removeClient_l(pid_t pid)
1033{
1034    LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
1035    mClients.removeItem(pid);
1036}
1037
1038
1039// ----------------------------------------------------------------------------
1040
1041AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
1042    :   Thread(false),
1043        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
1044        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
1045        mDevice(device)
1046{
1047    mDeathRecipient = new PMDeathRecipient(this);
1048}
1049
1050AudioFlinger::ThreadBase::~ThreadBase()
1051{
1052    mParamCond.broadcast();
1053    mNewParameters.clear();
1054    // do not lock the mutex in destructor
1055    releaseWakeLock_l();
1056}
1057
1058void AudioFlinger::ThreadBase::exit()
1059{
1060    // keep a strong ref on ourself so that we wont get
1061    // destroyed in the middle of requestExitAndWait()
1062    sp <ThreadBase> strongMe = this;
1063
1064    LOGV("ThreadBase::exit");
1065    {
1066        AutoMutex lock(&mLock);
1067        mExiting = true;
1068        requestExit();
1069        mWaitWorkCV.signal();
1070    }
1071    requestExitAndWait();
1072}
1073
1074uint32_t AudioFlinger::ThreadBase::sampleRate() const
1075{
1076    return mSampleRate;
1077}
1078
1079int AudioFlinger::ThreadBase::channelCount() const
1080{
1081    return (int)mChannelCount;
1082}
1083
1084uint32_t AudioFlinger::ThreadBase::format() const
1085{
1086    return mFormat;
1087}
1088
1089size_t AudioFlinger::ThreadBase::frameCount() const
1090{
1091    return mFrameCount;
1092}
1093
1094status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1095{
1096    status_t status;
1097
1098    LOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1099    Mutex::Autolock _l(mLock);
1100
1101    mNewParameters.add(keyValuePairs);
1102    mWaitWorkCV.signal();
1103    // wait condition with timeout in case the thread loop has exited
1104    // before the request could be processed
1105    if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) {
1106        status = mParamStatus;
1107        mWaitWorkCV.signal();
1108    } else {
1109        status = TIMED_OUT;
1110    }
1111    return status;
1112}
1113
1114void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1115{
1116    Mutex::Autolock _l(mLock);
1117    sendConfigEvent_l(event, param);
1118}
1119
1120// sendConfigEvent_l() must be called with ThreadBase::mLock held
1121void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1122{
1123    ConfigEvent *configEvent = new ConfigEvent();
1124    configEvent->mEvent = event;
1125    configEvent->mParam = param;
1126    mConfigEvents.add(configEvent);
1127    LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1128    mWaitWorkCV.signal();
1129}
1130
1131void AudioFlinger::ThreadBase::processConfigEvents()
1132{
1133    mLock.lock();
1134    while(!mConfigEvents.isEmpty()) {
1135        LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1136        ConfigEvent *configEvent = mConfigEvents[0];
1137        mConfigEvents.removeAt(0);
1138        // release mLock before locking AudioFlinger mLock: lock order is always
1139        // AudioFlinger then ThreadBase to avoid cross deadlock
1140        mLock.unlock();
1141        mAudioFlinger->mLock.lock();
1142        audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
1143        mAudioFlinger->mLock.unlock();
1144        delete configEvent;
1145        mLock.lock();
1146    }
1147    mLock.unlock();
1148}
1149
1150status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1151{
1152    const size_t SIZE = 256;
1153    char buffer[SIZE];
1154    String8 result;
1155
1156    bool locked = tryLock(mLock);
1157    if (!locked) {
1158        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1159        write(fd, buffer, strlen(buffer));
1160    }
1161
1162    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1163    result.append(buffer);
1164    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1165    result.append(buffer);
1166    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1167    result.append(buffer);
1168    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1169    result.append(buffer);
1170    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1171    result.append(buffer);
1172    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1173    result.append(buffer);
1174    snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
1175    result.append(buffer);
1176
1177    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1178    result.append(buffer);
1179    result.append(" Index Command");
1180    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1181        snprintf(buffer, SIZE, "\n %02d    ", i);
1182        result.append(buffer);
1183        result.append(mNewParameters[i]);
1184    }
1185
1186    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1187    result.append(buffer);
1188    snprintf(buffer, SIZE, " Index event param\n");
1189    result.append(buffer);
1190    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1191        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
1192        result.append(buffer);
1193    }
1194    result.append("\n");
1195
1196    write(fd, result.string(), result.size());
1197
1198    if (locked) {
1199        mLock.unlock();
1200    }
1201    return NO_ERROR;
1202}
1203
1204status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1205{
1206    const size_t SIZE = 256;
1207    char buffer[SIZE];
1208    String8 result;
1209
1210    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1211    write(fd, buffer, strlen(buffer));
1212
1213    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1214        sp<EffectChain> chain = mEffectChains[i];
1215        if (chain != 0) {
1216            chain->dump(fd, args);
1217        }
1218    }
1219    return NO_ERROR;
1220}
1221
1222void AudioFlinger::ThreadBase::acquireWakeLock()
1223{
1224    Mutex::Autolock _l(mLock);
1225    acquireWakeLock_l();
1226}
1227
1228void AudioFlinger::ThreadBase::acquireWakeLock_l()
1229{
1230    if (mPowerManager == 0) {
1231        // use checkService() to avoid blocking if power service is not up yet
1232        sp<IBinder> binder =
1233            defaultServiceManager()->checkService(String16("power"));
1234        if (binder == 0) {
1235            LOGW("Thread %s cannot connect to the power manager service", mName);
1236        } else {
1237            mPowerManager = interface_cast<IPowerManager>(binder);
1238            binder->linkToDeath(mDeathRecipient);
1239        }
1240    }
1241    if (mPowerManager != 0) {
1242        sp<IBinder> binder = new BBinder();
1243        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
1244                                                         binder,
1245                                                         String16(mName));
1246        if (status == NO_ERROR) {
1247            mWakeLockToken = binder;
1248        }
1249        LOGV("acquireWakeLock_l() %s status %d", mName, status);
1250    }
1251}
1252
1253void AudioFlinger::ThreadBase::releaseWakeLock()
1254{
1255    Mutex::Autolock _l(mLock);
1256    releaseWakeLock_l();
1257}
1258
1259void AudioFlinger::ThreadBase::releaseWakeLock_l()
1260{
1261    if (mWakeLockToken != 0) {
1262        LOGV("releaseWakeLock_l() %s", mName);
1263        if (mPowerManager != 0) {
1264            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
1265        }
1266        mWakeLockToken.clear();
1267    }
1268}
1269
1270void AudioFlinger::ThreadBase::clearPowerManager()
1271{
1272    Mutex::Autolock _l(mLock);
1273    releaseWakeLock_l();
1274    mPowerManager.clear();
1275}
1276
1277void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
1278{
1279    sp<ThreadBase> thread = mThread.promote();
1280    if (thread != 0) {
1281        thread->clearPowerManager();
1282    }
1283    LOGW("power manager service died !!!");
1284}
1285
1286void AudioFlinger::ThreadBase::setEffectSuspended(
1287        const effect_uuid_t *type, bool suspend, int sessionId)
1288{
1289    Mutex::Autolock _l(mLock);
1290    setEffectSuspended_l(type, suspend, sessionId);
1291}
1292
1293void AudioFlinger::ThreadBase::setEffectSuspended_l(
1294        const effect_uuid_t *type, bool suspend, int sessionId)
1295{
1296    sp<EffectChain> chain;
1297    chain = getEffectChain_l(sessionId);
1298    if (chain != 0) {
1299        if (type != NULL) {
1300            chain->setEffectSuspended_l(type, suspend);
1301        } else {
1302            chain->setEffectSuspendedAll_l(suspend);
1303        }
1304    }
1305
1306    updateSuspendedSessions_l(type, suspend, sessionId);
1307}
1308
1309void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1310{
1311    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1312    if (index < 0) {
1313        return;
1314    }
1315
1316    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1317            mSuspendedSessions.editValueAt(index);
1318
1319    for (size_t i = 0; i < sessionEffects.size(); i++) {
1320        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1321        for (int j = 0; j < desc->mRefCount; j++) {
1322            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1323                chain->setEffectSuspendedAll_l(true);
1324            } else {
1325                LOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1326                     desc->mType.timeLow);
1327                chain->setEffectSuspended_l(&desc->mType, true);
1328            }
1329        }
1330    }
1331}
1332
1333void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1334                                                         bool suspend,
1335                                                         int sessionId)
1336{
1337    int index = mSuspendedSessions.indexOfKey(sessionId);
1338
1339    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1340
1341    if (suspend) {
1342        if (index >= 0) {
1343            sessionEffects = mSuspendedSessions.editValueAt(index);
1344        } else {
1345            mSuspendedSessions.add(sessionId, sessionEffects);
1346        }
1347    } else {
1348        if (index < 0) {
1349            return;
1350        }
1351        sessionEffects = mSuspendedSessions.editValueAt(index);
1352    }
1353
1354
1355    int key = EffectChain::kKeyForSuspendAll;
1356    if (type != NULL) {
1357        key = type->timeLow;
1358    }
1359    index = sessionEffects.indexOfKey(key);
1360
1361    sp <SuspendedSessionDesc> desc;
1362    if (suspend) {
1363        if (index >= 0) {
1364            desc = sessionEffects.valueAt(index);
1365        } else {
1366            desc = new SuspendedSessionDesc();
1367            if (type != NULL) {
1368                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1369            }
1370            sessionEffects.add(key, desc);
1371            LOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1372        }
1373        desc->mRefCount++;
1374    } else {
1375        if (index < 0) {
1376            return;
1377        }
1378        desc = sessionEffects.valueAt(index);
1379        if (--desc->mRefCount == 0) {
1380            LOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1381            sessionEffects.removeItemsAt(index);
1382            if (sessionEffects.isEmpty()) {
1383                LOGV("updateSuspendedSessions_l() restore removing session %d",
1384                                 sessionId);
1385                mSuspendedSessions.removeItem(sessionId);
1386            }
1387        }
1388    }
1389    if (!sessionEffects.isEmpty()) {
1390        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1391    }
1392}
1393
1394void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1395                                                            bool enabled,
1396                                                            int sessionId)
1397{
1398    Mutex::Autolock _l(mLock);
1399
1400    if (mType != RECORD) {
1401        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1402        // another session. This gives the priority to well behaved effect control panels
1403        // and applications not using global effects.
1404        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1405            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1406        }
1407    }
1408
1409    sp<EffectChain> chain = getEffectChain_l(sessionId);
1410    if (chain != 0) {
1411        chain->checkSuspendOnEffectEnabled(effect, enabled);
1412    }
1413}
1414
1415// ----------------------------------------------------------------------------
1416
1417AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1418                                             AudioStreamOut* output,
1419                                             int id,
1420                                             uint32_t device)
1421    :   ThreadBase(audioFlinger, id, device),
1422        mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
1423        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1424{
1425    snprintf(mName, kNameLength, "AudioOut_%d", id);
1426
1427    readOutputParameters();
1428
1429    mMasterVolume = mAudioFlinger->masterVolumeSW();
1430
1431    mMasterMute = mAudioFlinger->masterMute();
1432
1433    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
1434        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1435        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1436    }
1437}
1438
1439AudioFlinger::PlaybackThread::~PlaybackThread()
1440{
1441    delete [] mMixBuffer;
1442}
1443
1444status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1445{
1446    dumpInternals(fd, args);
1447    dumpTracks(fd, args);
1448    dumpEffectChains(fd, args);
1449    return NO_ERROR;
1450}
1451
1452status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1453{
1454    const size_t SIZE = 256;
1455    char buffer[SIZE];
1456    String8 result;
1457
1458    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1459    result.append(buffer);
1460    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1461    for (size_t i = 0; i < mTracks.size(); ++i) {
1462        sp<Track> track = mTracks[i];
1463        if (track != 0) {
1464            track->dump(buffer, SIZE);
1465            result.append(buffer);
1466        }
1467    }
1468
1469    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1470    result.append(buffer);
1471    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1472    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1473        wp<Track> wTrack = mActiveTracks[i];
1474        if (wTrack != 0) {
1475            sp<Track> track = wTrack.promote();
1476            if (track != 0) {
1477                track->dump(buffer, SIZE);
1478                result.append(buffer);
1479            }
1480        }
1481    }
1482    write(fd, result.string(), result.size());
1483    return NO_ERROR;
1484}
1485
1486status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1487{
1488    const size_t SIZE = 256;
1489    char buffer[SIZE];
1490    String8 result;
1491
1492    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1493    result.append(buffer);
1494    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1495    result.append(buffer);
1496    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1497    result.append(buffer);
1498    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1499    result.append(buffer);
1500    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1501    result.append(buffer);
1502    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1503    result.append(buffer);
1504    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1505    result.append(buffer);
1506    write(fd, result.string(), result.size());
1507
1508    dumpBase(fd, args);
1509
1510    return NO_ERROR;
1511}
1512
1513// Thread virtuals
1514status_t AudioFlinger::PlaybackThread::readyToRun()
1515{
1516    status_t status = initCheck();
1517    if (status == NO_ERROR) {
1518        LOGI("AudioFlinger's thread %p ready to run", this);
1519    } else {
1520        LOGE("No working audio driver found.");
1521    }
1522    return status;
1523}
1524
1525void AudioFlinger::PlaybackThread::onFirstRef()
1526{
1527    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1528}
1529
1530// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1531sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1532        const sp<AudioFlinger::Client>& client,
1533        int streamType,
1534        uint32_t sampleRate,
1535        uint32_t format,
1536        uint32_t channelMask,
1537        int frameCount,
1538        const sp<IMemory>& sharedBuffer,
1539        int sessionId,
1540        bool isTimed,
1541        status_t *status)
1542{
1543    sp<Track> track;
1544    status_t lStatus;
1545
1546    if (mType == DIRECT) {
1547        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1548            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1549                LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1550                        "for output %p with format %d",
1551                        sampleRate, format, channelMask, mOutput, mFormat);
1552                lStatus = BAD_VALUE;
1553                goto Exit;
1554            }
1555        }
1556    } else {
1557        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1558        if (sampleRate > mSampleRate*2) {
1559            LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1560            lStatus = BAD_VALUE;
1561            goto Exit;
1562        }
1563    }
1564
1565    lStatus = initCheck();
1566    if (lStatus != NO_ERROR) {
1567        LOGE("Audio driver not initialized.");
1568        goto Exit;
1569    }
1570
1571    { // scope for mLock
1572        Mutex::Autolock _l(mLock);
1573
1574        // all tracks in same audio session must share the same routing strategy otherwise
1575        // conflicts will happen when tracks are moved from one output to another by audio policy
1576        // manager
1577        uint32_t strategy =
1578                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1579        for (size_t i = 0; i < mTracks.size(); ++i) {
1580            sp<Track> t = mTracks[i];
1581            if (t != 0) {
1582                if (sessionId == t->sessionId() &&
1583                        strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
1584                    lStatus = BAD_VALUE;
1585                    goto Exit;
1586                }
1587            }
1588        }
1589
1590        if (!isTimed) {
1591            track = new Track(this, client, streamType, sampleRate, format,
1592                    channelMask, frameCount, sharedBuffer, sessionId);
1593        } else {
1594            track = TimedTrack::create(this, client, streamType, sampleRate, format,
1595                    channelMask, frameCount, sharedBuffer, sessionId);
1596        }
1597        if (track == NULL || track->getCblk() == NULL || track->name() < 0) {
1598            lStatus = NO_MEMORY;
1599            goto Exit;
1600        }
1601        mTracks.add(track);
1602
1603        sp<EffectChain> chain = getEffectChain_l(sessionId);
1604        if (chain != 0) {
1605            LOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1606            track->setMainBuffer(chain->inBuffer());
1607            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1608            chain->incTrackCnt();
1609        }
1610    }
1611    lStatus = NO_ERROR;
1612
1613Exit:
1614    if(status) {
1615        *status = lStatus;
1616    }
1617    return track;
1618}
1619
1620uint32_t AudioFlinger::PlaybackThread::latency() const
1621{
1622    Mutex::Autolock _l(mLock);
1623    if (initCheck() == NO_ERROR) {
1624        return mOutput->stream->get_latency(mOutput->stream);
1625    } else {
1626        return 0;
1627    }
1628}
1629
1630status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1631{
1632    mMasterVolume = value;
1633    return NO_ERROR;
1634}
1635
1636status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1637{
1638    mMasterMute = muted;
1639    return NO_ERROR;
1640}
1641
1642float AudioFlinger::PlaybackThread::masterVolume() const
1643{
1644    return mMasterVolume;
1645}
1646
1647bool AudioFlinger::PlaybackThread::masterMute() const
1648{
1649    return mMasterMute;
1650}
1651
1652status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
1653{
1654    mStreamTypes[stream].volume = value;
1655    return NO_ERROR;
1656}
1657
1658status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
1659{
1660    mStreamTypes[stream].mute = muted;
1661    return NO_ERROR;
1662}
1663
1664float AudioFlinger::PlaybackThread::streamVolume(int stream) const
1665{
1666    return mStreamTypes[stream].volume;
1667}
1668
1669bool AudioFlinger::PlaybackThread::streamMute(int stream) const
1670{
1671    return mStreamTypes[stream].mute;
1672}
1673
1674// addTrack_l() must be called with ThreadBase::mLock held
1675status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1676{
1677    status_t status = ALREADY_EXISTS;
1678
1679    // set retry count for buffer fill
1680    track->mRetryCount = kMaxTrackStartupRetries;
1681    if (mActiveTracks.indexOf(track) < 0) {
1682        // the track is newly added, make sure it fills up all its
1683        // buffers before playing. This is to ensure the client will
1684        // effectively get the latency it requested.
1685        track->mFillingUpStatus = Track::FS_FILLING;
1686        track->mResetDone = false;
1687        mActiveTracks.add(track);
1688        if (track->mainBuffer() != mMixBuffer) {
1689            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1690            if (chain != 0) {
1691                LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1692                chain->incActiveTrackCnt();
1693            }
1694        }
1695
1696        status = NO_ERROR;
1697    }
1698
1699    LOGV("mWaitWorkCV.broadcast");
1700    mWaitWorkCV.broadcast();
1701
1702    return status;
1703}
1704
1705// destroyTrack_l() must be called with ThreadBase::mLock held
1706void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1707{
1708    track->mState = TrackBase::TERMINATED;
1709    if (mActiveTracks.indexOf(track) < 0) {
1710        removeTrack_l(track);
1711    }
1712}
1713
1714void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1715{
1716    mTracks.remove(track);
1717    deleteTrackName_l(track->name());
1718    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1719    if (chain != 0) {
1720        chain->decTrackCnt();
1721    }
1722}
1723
1724String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1725{
1726    String8 out_s8 = String8("");
1727    char *s;
1728
1729    Mutex::Autolock _l(mLock);
1730    if (initCheck() != NO_ERROR) {
1731        return out_s8;
1732    }
1733
1734    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1735    out_s8 = String8(s);
1736    free(s);
1737    return out_s8;
1738}
1739
1740// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1741void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1742    AudioSystem::OutputDescriptor desc;
1743    void *param2 = 0;
1744
1745    LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1746
1747    switch (event) {
1748    case AudioSystem::OUTPUT_OPENED:
1749    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1750        desc.channels = mChannelMask;
1751        desc.samplingRate = mSampleRate;
1752        desc.format = mFormat;
1753        desc.frameCount = mFrameCount;
1754        desc.latency = latency();
1755        param2 = &desc;
1756        break;
1757
1758    case AudioSystem::STREAM_CONFIG_CHANGED:
1759        param2 = &param;
1760    case AudioSystem::OUTPUT_CLOSED:
1761    default:
1762        break;
1763    }
1764    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1765}
1766
1767void AudioFlinger::PlaybackThread::readOutputParameters()
1768{
1769    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1770    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1771    mChannelCount = (uint16_t)popcount(mChannelMask);
1772    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1773    mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
1774    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1775
1776    // FIXME - Current mixer implementation only supports stereo output: Always
1777    // Allocate a stereo buffer even if HW output is mono.
1778    if (mMixBuffer != NULL) delete[] mMixBuffer;
1779    mMixBuffer = new int16_t[mFrameCount * 2];
1780    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1781
1782    // force reconfiguration of effect chains and engines to take new buffer size and audio
1783    // parameters into account
1784    // Note that mLock is not held when readOutputParameters() is called from the constructor
1785    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1786    // matter.
1787    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1788    Vector< sp<EffectChain> > effectChains = mEffectChains;
1789    for (size_t i = 0; i < effectChains.size(); i ++) {
1790        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1791    }
1792}
1793
1794status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1795{
1796    if (halFrames == 0 || dspFrames == 0) {
1797        return BAD_VALUE;
1798    }
1799    Mutex::Autolock _l(mLock);
1800    if (initCheck() != NO_ERROR) {
1801        return INVALID_OPERATION;
1802    }
1803    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1804
1805    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1806}
1807
1808uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1809{
1810    Mutex::Autolock _l(mLock);
1811    uint32_t result = 0;
1812    if (getEffectChain_l(sessionId) != 0) {
1813        result = EFFECT_SESSION;
1814    }
1815
1816    for (size_t i = 0; i < mTracks.size(); ++i) {
1817        sp<Track> track = mTracks[i];
1818        if (sessionId == track->sessionId() &&
1819                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1820            result |= TRACK_SESSION;
1821            break;
1822        }
1823    }
1824
1825    return result;
1826}
1827
1828uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1829{
1830    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1831    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1832    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1833        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1834    }
1835    for (size_t i = 0; i < mTracks.size(); i++) {
1836        sp<Track> track = mTracks[i];
1837        if (sessionId == track->sessionId() &&
1838                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1839            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1840        }
1841    }
1842    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1843}
1844
1845
1846AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1847{
1848    Mutex::Autolock _l(mLock);
1849    return mOutput;
1850}
1851
1852AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1853{
1854    Mutex::Autolock _l(mLock);
1855    AudioStreamOut *output = mOutput;
1856    mOutput = NULL;
1857    return output;
1858}
1859
1860// this method must always be called either with ThreadBase mLock held or inside the thread loop
1861audio_stream_t* AudioFlinger::PlaybackThread::stream()
1862{
1863    if (mOutput == NULL) {
1864        return NULL;
1865    }
1866    return &mOutput->stream->common;
1867}
1868
1869// ----------------------------------------------------------------------------
1870
1871AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1872    :   PlaybackThread(audioFlinger, output, id, device),
1873        mAudioMixer(0)
1874{
1875    mType = ThreadBase::MIXER;
1876    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1877
1878    // FIXME - Current mixer implementation only supports stereo output
1879    if (mChannelCount == 1) {
1880        LOGE("Invalid audio hardware channel count");
1881    }
1882}
1883
1884AudioFlinger::MixerThread::~MixerThread()
1885{
1886    delete mAudioMixer;
1887}
1888
1889bool AudioFlinger::MixerThread::threadLoop()
1890{
1891    Vector< sp<Track> > tracksToRemove;
1892    uint32_t mixerStatus = MIXER_IDLE;
1893    nsecs_t standbyTime = systemTime();
1894    size_t mixBufferSize = mFrameCount * mFrameSize;
1895    // FIXME: Relaxed timing because of a certain device that can't meet latency
1896    // Should be reduced to 2x after the vendor fixes the driver issue
1897    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
1898    nsecs_t lastWarning = 0;
1899    bool longStandbyExit = false;
1900    uint32_t activeSleepTime = activeSleepTimeUs();
1901    uint32_t idleSleepTime = idleSleepTimeUs();
1902    uint32_t sleepTime = idleSleepTime;
1903    Vector< sp<EffectChain> > effectChains;
1904#ifdef DEBUG_CPU_USAGE
1905    ThreadCpuUsage cpu;
1906    const CentralTendencyStatistics& stats = cpu.statistics();
1907#endif
1908
1909    acquireWakeLock();
1910
1911    LocalClock lc;
1912    uint64_t localTimeFreq;
1913    localTimeFreq = lc.getLocalFreq();
1914
1915    while (!exitPending())
1916    {
1917#ifdef DEBUG_CPU_USAGE
1918        cpu.sampleAndEnable();
1919        unsigned n = stats.n();
1920        // cpu.elapsed() is expensive, so don't call it every loop
1921        if ((n & 127) == 1) {
1922            long long elapsed = cpu.elapsed();
1923            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1924                double perLoop = elapsed / (double) n;
1925                double perLoop100 = perLoop * 0.01;
1926                double mean = stats.mean();
1927                double stddev = stats.stddev();
1928                double minimum = stats.minimum();
1929                double maximum = stats.maximum();
1930                cpu.resetStatistics();
1931                LOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
1932                        elapsed * .000000001, n, perLoop * .000001,
1933                        mean * .001,
1934                        stddev * .001,
1935                        minimum * .001,
1936                        maximum * .001,
1937                        mean / perLoop100,
1938                        stddev / perLoop100,
1939                        minimum / perLoop100,
1940                        maximum / perLoop100);
1941            }
1942        }
1943#endif
1944        processConfigEvents();
1945
1946        mixerStatus = MIXER_IDLE;
1947        { // scope for mLock
1948
1949            Mutex::Autolock _l(mLock);
1950
1951            if (checkForNewParameters_l()) {
1952                mixBufferSize = mFrameCount * mFrameSize;
1953                // FIXME: Relaxed timing because of a certain device that can't meet latency
1954                // Should be reduced to 2x after the vendor fixes the driver issue
1955                maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
1956                activeSleepTime = activeSleepTimeUs();
1957                idleSleepTime = idleSleepTimeUs();
1958            }
1959
1960            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1961
1962            // put audio hardware into standby after short delay
1963            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1964                        mSuspended) {
1965                if (!mStandby) {
1966                    LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1967                    mOutput->stream->common.standby(&mOutput->stream->common);
1968                    mStandby = true;
1969                    mBytesWritten = 0;
1970                }
1971
1972                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1973                    // we're about to wait, flush the binder command buffer
1974                    IPCThreadState::self()->flushCommands();
1975
1976                    if (exitPending()) break;
1977
1978                    releaseWakeLock_l();
1979                    // wait until we have something to do...
1980                    LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1981                    mWaitWorkCV.wait(mLock);
1982                    LOGV("MixerThread %p TID %d waking up\n", this, gettid());
1983                    acquireWakeLock_l();
1984
1985                    if (mMasterMute == false) {
1986                        char value[PROPERTY_VALUE_MAX];
1987                        property_get("ro.audio.silent", value, "0");
1988                        if (atoi(value)) {
1989                            LOGD("Silence is golden");
1990                            setMasterMute(true);
1991                        }
1992                    }
1993
1994                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1995                    sleepTime = idleSleepTime;
1996                    continue;
1997                }
1998            }
1999
2000            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
2001
2002            // prevent any changes in effect chain list and in each effect chain
2003            // during mixing and effect process as the audio buffers could be deleted
2004            // or modified if an effect is created or deleted
2005            lockEffectChains_l(effectChains);
2006       }
2007
2008        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2009            // obtain the presentation timestamp of the next output buffer
2010            int64_t pts;
2011            status_t status = mOutput->stream->get_next_write_timestamp(
2012                mOutput->stream, &pts);
2013            if (status != NO_ERROR) {
2014                pts = AudioBufferProvider::kInvalidPTS;
2015            }
2016
2017            // mix buffers...
2018            mAudioMixer->process(pts);
2019            sleepTime = 0;
2020            standbyTime = systemTime() + kStandbyTimeInNsecs;
2021            //TODO: delay standby when effects have a tail
2022        } else {
2023            // If no tracks are ready, sleep once for the duration of an output
2024            // buffer size, then write 0s to the output
2025            if (sleepTime == 0) {
2026                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2027                    sleepTime = activeSleepTime;
2028                } else {
2029                    sleepTime = idleSleepTime;
2030                }
2031            } else if (mBytesWritten != 0 ||
2032                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
2033                memset (mMixBuffer, 0, mixBufferSize);
2034                sleepTime = 0;
2035                LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
2036            }
2037            // TODO add standby time extension fct of effect tail
2038        }
2039
2040        if (mSuspended) {
2041            sleepTime = suspendSleepTimeUs();
2042        }
2043        // sleepTime == 0 means we must write to audio hardware
2044        if (sleepTime == 0) {
2045             for (size_t i = 0; i < effectChains.size(); i ++) {
2046                 effectChains[i]->process_l();
2047             }
2048             // enable changes in effect chain
2049             unlockEffectChains(effectChains);
2050            mLastWriteTime = systemTime();
2051            mInWrite = true;
2052            mBytesWritten += mixBufferSize;
2053
2054            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2055            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2056            mNumWrites++;
2057            mInWrite = false;
2058            nsecs_t now = systemTime();
2059            nsecs_t delta = now - mLastWriteTime;
2060            if (delta > maxPeriod) {
2061                mNumDelayedWrites++;
2062                if ((now - lastWarning) > kWarningThrottle) {
2063                    LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2064                            ns2ms(delta), mNumDelayedWrites, this);
2065                    lastWarning = now;
2066                }
2067                if (mStandby) {
2068                    longStandbyExit = true;
2069                }
2070            }
2071            mStandby = false;
2072        } else {
2073            // enable changes in effect chain
2074            unlockEffectChains(effectChains);
2075            usleep(sleepTime);
2076        }
2077
2078        // finally let go of all our tracks, without the lock held
2079        // since we can't guarantee the destructors won't acquire that
2080        // same lock.
2081        tracksToRemove.clear();
2082
2083        // Effect chains will be actually deleted here if they were removed from
2084        // mEffectChains list during mixing or effects processing
2085        effectChains.clear();
2086    }
2087
2088    if (!mStandby) {
2089        mOutput->stream->common.standby(&mOutput->stream->common);
2090    }
2091
2092    releaseWakeLock();
2093
2094    LOGV("MixerThread %p exiting", this);
2095    return false;
2096}
2097
2098// prepareTracks_l() must be called with ThreadBase::mLock held
2099uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2100{
2101
2102    uint32_t mixerStatus = MIXER_IDLE;
2103    // find out which tracks need to be processed
2104    size_t count = activeTracks.size();
2105    size_t mixedTracks = 0;
2106    size_t tracksWithEffect = 0;
2107
2108    float masterVolume = mMasterVolume;
2109    bool  masterMute = mMasterMute;
2110
2111    if (masterMute) {
2112        masterVolume = 0;
2113    }
2114    // Delegate master volume control to effect in output mix effect chain if needed
2115    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2116    if (chain != 0) {
2117        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2118        chain->setVolume_l(&v, &v);
2119        masterVolume = (float)((v + (1 << 23)) >> 24);
2120        chain.clear();
2121    }
2122
2123    for (size_t i=0 ; i<count ; i++) {
2124        sp<Track> t = activeTracks[i].promote();
2125        if (t == 0) continue;
2126
2127        Track* const track = t.get();
2128        audio_track_cblk_t* cblk = track->cblk();
2129
2130        // The first time a track is added we wait
2131        // for all its buffers to be filled before processing it
2132        mAudioMixer->setActiveTrack(track->name());
2133        if (track->framesReady() && track->isReady() &&
2134                !track->isPaused() && !track->isTerminated())
2135        {
2136            //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
2137
2138            mixedTracks++;
2139
2140            // track->mainBuffer() != mMixBuffer means there is an effect chain
2141            // connected to the track
2142            chain.clear();
2143            if (track->mainBuffer() != mMixBuffer) {
2144                chain = getEffectChain_l(track->sessionId());
2145                // Delegate volume control to effect in track effect chain if needed
2146                if (chain != 0) {
2147                    tracksWithEffect++;
2148                } else {
2149                    LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
2150                            track->name(), track->sessionId());
2151                }
2152            }
2153
2154
2155            int param = AudioMixer::VOLUME;
2156            if (track->mFillingUpStatus == Track::FS_FILLED) {
2157                // no ramp for the first volume setting
2158                track->mFillingUpStatus = Track::FS_ACTIVE;
2159                if (track->mState == TrackBase::RESUMING) {
2160                    track->mState = TrackBase::ACTIVE;
2161                    param = AudioMixer::RAMP_VOLUME;
2162                }
2163                mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2164            } else if (cblk->server != 0) {
2165                // If the track is stopped before the first frame was mixed,
2166                // do not apply ramp
2167                param = AudioMixer::RAMP_VOLUME;
2168            }
2169
2170            // compute volume for this track
2171            uint32_t vl, vr, va;
2172            if (track->isMuted() || track->isPausing() ||
2173                mStreamTypes[track->type()].mute) {
2174                vl = vr = va = 0;
2175                if (track->isPausing()) {
2176                    track->setPaused();
2177                }
2178            } else {
2179
2180                // read original volumes with volume control
2181                float typeVolume = mStreamTypes[track->type()].volume;
2182                float v = masterVolume * typeVolume;
2183                vl = (uint32_t)(v * cblk->volume[0]) << 12;
2184                vr = (uint32_t)(v * cblk->volume[1]) << 12;
2185
2186                va = (uint32_t)(v * cblk->sendLevel);
2187            }
2188            // Delegate volume control to effect in track effect chain if needed
2189            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2190                // Do not ramp volume if volume is controlled by effect
2191                param = AudioMixer::VOLUME;
2192                track->mHasVolumeController = true;
2193            } else {
2194                // force no volume ramp when volume controller was just disabled or removed
2195                // from effect chain to avoid volume spike
2196                if (track->mHasVolumeController) {
2197                    param = AudioMixer::VOLUME;
2198                }
2199                track->mHasVolumeController = false;
2200            }
2201
2202            // Convert volumes from 8.24 to 4.12 format
2203            int16_t left, right, aux;
2204            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2205            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2206            left = int16_t(v_clamped);
2207            v_clamped = (vr + (1 << 11)) >> 12;
2208            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2209            right = int16_t(v_clamped);
2210
2211            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2212            aux = int16_t(va);
2213
2214            // XXX: these things DON'T need to be done each time
2215            mAudioMixer->setBufferProvider(track);
2216            mAudioMixer->enable(AudioMixer::MIXING);
2217
2218            mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
2219            mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
2220            mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
2221            mAudioMixer->setParameter(
2222                AudioMixer::TRACK,
2223                AudioMixer::FORMAT, (void *)track->format());
2224            mAudioMixer->setParameter(
2225                AudioMixer::TRACK,
2226                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2227            mAudioMixer->setParameter(
2228                AudioMixer::RESAMPLE,
2229                AudioMixer::SAMPLE_RATE,
2230                (void *)(cblk->sampleRate));
2231            mAudioMixer->setParameter(
2232                AudioMixer::TRACK,
2233                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2234            mAudioMixer->setParameter(
2235                AudioMixer::TRACK,
2236                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2237
2238            // reset retry count
2239            track->mRetryCount = kMaxTrackRetries;
2240            mixerStatus = MIXER_TRACKS_READY;
2241        } else {
2242            //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
2243            if (track->isStopped()) {
2244                track->reset();
2245            }
2246            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2247                // We have consumed all the buffers of this track.
2248                // Remove it from the list of active tracks.
2249                tracksToRemove->add(track);
2250            } else {
2251                // No buffers for this track. Give it a few chances to
2252                // fill a buffer, then remove it from active list.
2253                if (--(track->mRetryCount) <= 0) {
2254                    LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
2255                    tracksToRemove->add(track);
2256                    // indicate to client process that the track was disabled because of underrun
2257                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2258                } else if (mixerStatus != MIXER_TRACKS_READY) {
2259                    mixerStatus = MIXER_TRACKS_ENABLED;
2260                }
2261            }
2262            mAudioMixer->disable(AudioMixer::MIXING);
2263        }
2264    }
2265
2266    // remove all the tracks that need to be...
2267    count = tracksToRemove->size();
2268    if (UNLIKELY(count)) {
2269        for (size_t i=0 ; i<count ; i++) {
2270            const sp<Track>& track = tracksToRemove->itemAt(i);
2271            mActiveTracks.remove(track);
2272            if (track->mainBuffer() != mMixBuffer) {
2273                chain = getEffectChain_l(track->sessionId());
2274                if (chain != 0) {
2275                    LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2276                    chain->decActiveTrackCnt();
2277                }
2278            }
2279            if (track->isTerminated()) {
2280                removeTrack_l(track);
2281            }
2282        }
2283    }
2284
2285    // mix buffer must be cleared if all tracks are connected to an
2286    // effect chain as in this case the mixer will not write to
2287    // mix buffer and track effects will accumulate into it
2288    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2289        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2290    }
2291
2292    return mixerStatus;
2293}
2294
2295void AudioFlinger::MixerThread::invalidateTracks(int streamType)
2296{
2297    LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2298            this,  streamType, mTracks.size());
2299    Mutex::Autolock _l(mLock);
2300
2301    size_t size = mTracks.size();
2302    for (size_t i = 0; i < size; i++) {
2303        sp<Track> t = mTracks[i];
2304        if (t->type() == streamType) {
2305            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2306            t->mCblk->cv.signal();
2307        }
2308    }
2309}
2310
2311
2312// getTrackName_l() must be called with ThreadBase::mLock held
2313int AudioFlinger::MixerThread::getTrackName_l()
2314{
2315    return mAudioMixer->getTrackName();
2316}
2317
2318// deleteTrackName_l() must be called with ThreadBase::mLock held
2319void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2320{
2321    LOGV("remove track (%d) and delete from mixer", name);
2322    mAudioMixer->deleteTrackName(name);
2323}
2324
2325// checkForNewParameters_l() must be called with ThreadBase::mLock held
2326bool AudioFlinger::MixerThread::checkForNewParameters_l()
2327{
2328    bool reconfig = false;
2329
2330    while (!mNewParameters.isEmpty()) {
2331        status_t status = NO_ERROR;
2332        String8 keyValuePair = mNewParameters[0];
2333        AudioParameter param = AudioParameter(keyValuePair);
2334        int value;
2335
2336        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2337            reconfig = true;
2338        }
2339        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2340            if (value != AUDIO_FORMAT_PCM_16_BIT) {
2341                status = BAD_VALUE;
2342            } else {
2343                reconfig = true;
2344            }
2345        }
2346        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2347            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2348                status = BAD_VALUE;
2349            } else {
2350                reconfig = true;
2351            }
2352        }
2353        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2354            // do not accept frame count changes if tracks are open as the track buffer
2355            // size depends on frame count and correct behavior would not be garantied
2356            // if frame count is changed after track creation
2357            if (!mTracks.isEmpty()) {
2358                status = INVALID_OPERATION;
2359            } else {
2360                reconfig = true;
2361            }
2362        }
2363        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2364            // when changing the audio output device, call addBatteryData to notify
2365            // the change
2366            if ((int)mDevice != value) {
2367                uint32_t params = 0;
2368                // check whether speaker is on
2369                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2370                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2371                }
2372
2373                int deviceWithoutSpeaker
2374                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2375                // check if any other device (except speaker) is on
2376                if (value & deviceWithoutSpeaker ) {
2377                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2378                }
2379
2380                if (params != 0) {
2381                    addBatteryData(params);
2382                }
2383            }
2384
2385            // forward device change to effects that have requested to be
2386            // aware of attached audio device.
2387            mDevice = (uint32_t)value;
2388            for (size_t i = 0; i < mEffectChains.size(); i++) {
2389                mEffectChains[i]->setDevice_l(mDevice);
2390            }
2391        }
2392
2393        if (status == NO_ERROR) {
2394            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2395                                                    keyValuePair.string());
2396            if (!mStandby && status == INVALID_OPERATION) {
2397               mOutput->stream->common.standby(&mOutput->stream->common);
2398               mStandby = true;
2399               mBytesWritten = 0;
2400               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2401                                                       keyValuePair.string());
2402            }
2403            if (status == NO_ERROR && reconfig) {
2404                delete mAudioMixer;
2405                readOutputParameters();
2406                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2407                for (size_t i = 0; i < mTracks.size() ; i++) {
2408                    int name = getTrackName_l();
2409                    if (name < 0) break;
2410                    mTracks[i]->mName = name;
2411                    // limit track sample rate to 2 x new output sample rate
2412                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2413                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2414                    }
2415                }
2416                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2417            }
2418        }
2419
2420        mNewParameters.removeAt(0);
2421
2422        mParamStatus = status;
2423        mParamCond.signal();
2424        mWaitWorkCV.wait(mLock);
2425    }
2426    return reconfig;
2427}
2428
2429status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2430{
2431    const size_t SIZE = 256;
2432    char buffer[SIZE];
2433    String8 result;
2434
2435    PlaybackThread::dumpInternals(fd, args);
2436
2437    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2438    result.append(buffer);
2439    write(fd, result.string(), result.size());
2440    return NO_ERROR;
2441}
2442
2443uint32_t AudioFlinger::MixerThread::activeSleepTimeUs()
2444{
2445    return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2446}
2447
2448uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2449{
2450    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2451}
2452
2453uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2454{
2455    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2456}
2457
2458// ----------------------------------------------------------------------------
2459AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2460    :   PlaybackThread(audioFlinger, output, id, device)
2461{
2462    mType = ThreadBase::DIRECT;
2463}
2464
2465AudioFlinger::DirectOutputThread::~DirectOutputThread()
2466{
2467}
2468
2469
2470static inline int16_t clamp16(int32_t sample)
2471{
2472    if ((sample>>15) ^ (sample>>31))
2473        sample = 0x7FFF ^ (sample>>31);
2474    return sample;
2475}
2476
2477static inline
2478int32_t mul(int16_t in, int16_t v)
2479{
2480#if defined(__arm__) && !defined(__thumb__)
2481    int32_t out;
2482    asm( "smulbb %[out], %[in], %[v] \n"
2483         : [out]"=r"(out)
2484         : [in]"%r"(in), [v]"r"(v)
2485         : );
2486    return out;
2487#else
2488    return in * int32_t(v);
2489#endif
2490}
2491
2492void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2493{
2494    // Do not apply volume on compressed audio
2495    if (!audio_is_linear_pcm(mFormat)) {
2496        return;
2497    }
2498
2499    // convert to signed 16 bit before volume calculation
2500    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2501        size_t count = mFrameCount * mChannelCount;
2502        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2503        int16_t *dst = mMixBuffer + count-1;
2504        while(count--) {
2505            *dst-- = (int16_t)(*src--^0x80) << 8;
2506        }
2507    }
2508
2509    size_t frameCount = mFrameCount;
2510    int16_t *out = mMixBuffer;
2511    if (ramp) {
2512        if (mChannelCount == 1) {
2513            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2514            int32_t vlInc = d / (int32_t)frameCount;
2515            int32_t vl = ((int32_t)mLeftVolShort << 16);
2516            do {
2517                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2518                out++;
2519                vl += vlInc;
2520            } while (--frameCount);
2521
2522        } else {
2523            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2524            int32_t vlInc = d / (int32_t)frameCount;
2525            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2526            int32_t vrInc = d / (int32_t)frameCount;
2527            int32_t vl = ((int32_t)mLeftVolShort << 16);
2528            int32_t vr = ((int32_t)mRightVolShort << 16);
2529            do {
2530                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2531                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2532                out += 2;
2533                vl += vlInc;
2534                vr += vrInc;
2535            } while (--frameCount);
2536        }
2537    } else {
2538        if (mChannelCount == 1) {
2539            do {
2540                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2541                out++;
2542            } while (--frameCount);
2543        } else {
2544            do {
2545                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2546                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2547                out += 2;
2548            } while (--frameCount);
2549        }
2550    }
2551
2552    // convert back to unsigned 8 bit after volume calculation
2553    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2554        size_t count = mFrameCount * mChannelCount;
2555        int16_t *src = mMixBuffer;
2556        uint8_t *dst = (uint8_t *)mMixBuffer;
2557        while(count--) {
2558            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2559        }
2560    }
2561
2562    mLeftVolShort = leftVol;
2563    mRightVolShort = rightVol;
2564}
2565
2566bool AudioFlinger::DirectOutputThread::threadLoop()
2567{
2568    uint32_t mixerStatus = MIXER_IDLE;
2569    sp<Track> trackToRemove;
2570    sp<Track> activeTrack;
2571    nsecs_t standbyTime = systemTime();
2572    int8_t *curBuf;
2573    size_t mixBufferSize = mFrameCount*mFrameSize;
2574    uint32_t activeSleepTime = activeSleepTimeUs();
2575    uint32_t idleSleepTime = idleSleepTimeUs();
2576    uint32_t sleepTime = idleSleepTime;
2577    // use shorter standby delay as on normal output to release
2578    // hardware resources as soon as possible
2579    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2580
2581    acquireWakeLock();
2582
2583    while (!exitPending())
2584    {
2585        bool rampVolume;
2586        uint16_t leftVol;
2587        uint16_t rightVol;
2588        Vector< sp<EffectChain> > effectChains;
2589
2590        processConfigEvents();
2591
2592        mixerStatus = MIXER_IDLE;
2593
2594        { // scope for the mLock
2595
2596            Mutex::Autolock _l(mLock);
2597
2598            if (checkForNewParameters_l()) {
2599                mixBufferSize = mFrameCount*mFrameSize;
2600                activeSleepTime = activeSleepTimeUs();
2601                idleSleepTime = idleSleepTimeUs();
2602                standbyDelay = microseconds(activeSleepTime*2);
2603            }
2604
2605            // put audio hardware into standby after short delay
2606            if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2607                        mSuspended) {
2608                // wait until we have something to do...
2609                if (!mStandby) {
2610                    LOGV("Audio hardware entering standby, mixer %p\n", this);
2611                    mOutput->stream->common.standby(&mOutput->stream->common);
2612                    mStandby = true;
2613                    mBytesWritten = 0;
2614                }
2615
2616                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2617                    // we're about to wait, flush the binder command buffer
2618                    IPCThreadState::self()->flushCommands();
2619
2620                    if (exitPending()) break;
2621
2622                    releaseWakeLock_l();
2623                    LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2624                    mWaitWorkCV.wait(mLock);
2625                    LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2626                    acquireWakeLock_l();
2627
2628                    if (mMasterMute == false) {
2629                        char value[PROPERTY_VALUE_MAX];
2630                        property_get("ro.audio.silent", value, "0");
2631                        if (atoi(value)) {
2632                            LOGD("Silence is golden");
2633                            setMasterMute(true);
2634                        }
2635                    }
2636
2637                    standbyTime = systemTime() + standbyDelay;
2638                    sleepTime = idleSleepTime;
2639                    continue;
2640                }
2641            }
2642
2643            effectChains = mEffectChains;
2644
2645            // find out which tracks need to be processed
2646            if (mActiveTracks.size() != 0) {
2647                sp<Track> t = mActiveTracks[0].promote();
2648                if (t == 0) continue;
2649
2650                Track* const track = t.get();
2651                audio_track_cblk_t* cblk = track->cblk();
2652
2653                // The first time a track is added we wait
2654                // for all its buffers to be filled before processing it
2655                if (cblk->framesReady() && track->isReady() &&
2656                        !track->isPaused() && !track->isTerminated())
2657                {
2658                    //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2659
2660                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2661                        track->mFillingUpStatus = Track::FS_ACTIVE;
2662                        mLeftVolFloat = mRightVolFloat = 0;
2663                        mLeftVolShort = mRightVolShort = 0;
2664                        if (track->mState == TrackBase::RESUMING) {
2665                            track->mState = TrackBase::ACTIVE;
2666                            rampVolume = true;
2667                        }
2668                    } else if (cblk->server != 0) {
2669                        // If the track is stopped before the first frame was mixed,
2670                        // do not apply ramp
2671                        rampVolume = true;
2672                    }
2673                    // compute volume for this track
2674                    float left, right;
2675                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2676                        mStreamTypes[track->type()].mute) {
2677                        left = right = 0;
2678                        if (track->isPausing()) {
2679                            track->setPaused();
2680                        }
2681                    } else {
2682                        float typeVolume = mStreamTypes[track->type()].volume;
2683                        float v = mMasterVolume * typeVolume;
2684                        float v_clamped = v * cblk->volume[0];
2685                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2686                        left = v_clamped/MAX_GAIN;
2687                        v_clamped = v * cblk->volume[1];
2688                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2689                        right = v_clamped/MAX_GAIN;
2690                    }
2691
2692                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2693                        mLeftVolFloat = left;
2694                        mRightVolFloat = right;
2695
2696                        // If audio HAL implements volume control,
2697                        // force software volume to nominal value
2698                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2699                            left = 1.0f;
2700                            right = 1.0f;
2701                        }
2702
2703                        // Convert volumes from float to 8.24
2704                        uint32_t vl = (uint32_t)(left * (1 << 24));
2705                        uint32_t vr = (uint32_t)(right * (1 << 24));
2706
2707                        // Delegate volume control to effect in track effect chain if needed
2708                        // only one effect chain can be present on DirectOutputThread, so if
2709                        // there is one, the track is connected to it
2710                        if (!effectChains.isEmpty()) {
2711                            // Do not ramp volume if volume is controlled by effect
2712                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2713                                rampVolume = false;
2714                            }
2715                        }
2716
2717                        // Convert volumes from 8.24 to 4.12 format
2718                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2719                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2720                        leftVol = (uint16_t)v_clamped;
2721                        v_clamped = (vr + (1 << 11)) >> 12;
2722                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2723                        rightVol = (uint16_t)v_clamped;
2724                    } else {
2725                        leftVol = mLeftVolShort;
2726                        rightVol = mRightVolShort;
2727                        rampVolume = false;
2728                    }
2729
2730                    // reset retry count
2731                    track->mRetryCount = kMaxTrackRetriesDirect;
2732                    activeTrack = t;
2733                    mixerStatus = MIXER_TRACKS_READY;
2734                } else {
2735                    //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2736                    if (track->isStopped()) {
2737                        track->reset();
2738                    }
2739                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2740                        // We have consumed all the buffers of this track.
2741                        // Remove it from the list of active tracks.
2742                        trackToRemove = track;
2743                    } else {
2744                        // No buffers for this track. Give it a few chances to
2745                        // fill a buffer, then remove it from active list.
2746                        if (--(track->mRetryCount) <= 0) {
2747                            LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2748                            trackToRemove = track;
2749                        } else {
2750                            mixerStatus = MIXER_TRACKS_ENABLED;
2751                        }
2752                    }
2753                }
2754            }
2755
2756            // remove all the tracks that need to be...
2757            if (UNLIKELY(trackToRemove != 0)) {
2758                mActiveTracks.remove(trackToRemove);
2759                if (!effectChains.isEmpty()) {
2760                    LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2761                            trackToRemove->sessionId());
2762                    effectChains[0]->decActiveTrackCnt();
2763                }
2764                if (trackToRemove->isTerminated()) {
2765                    removeTrack_l(trackToRemove);
2766                }
2767            }
2768
2769            lockEffectChains_l(effectChains);
2770       }
2771
2772        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2773            AudioBufferProvider::Buffer buffer;
2774            size_t frameCount = mFrameCount;
2775            curBuf = (int8_t *)mMixBuffer;
2776            // output audio to hardware
2777            while (frameCount) {
2778                buffer.frameCount = frameCount;
2779                activeTrack->getNextBuffer(&buffer,
2780                                           AudioBufferProvider::kInvalidPTS);
2781                if (UNLIKELY(buffer.raw == 0)) {
2782                    memset(curBuf, 0, frameCount * mFrameSize);
2783                    break;
2784                }
2785                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2786                frameCount -= buffer.frameCount;
2787                curBuf += buffer.frameCount * mFrameSize;
2788                activeTrack->releaseBuffer(&buffer);
2789            }
2790            sleepTime = 0;
2791            standbyTime = systemTime() + standbyDelay;
2792        } else {
2793            if (sleepTime == 0) {
2794                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2795                    sleepTime = activeSleepTime;
2796                } else {
2797                    sleepTime = idleSleepTime;
2798                }
2799            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2800                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2801                sleepTime = 0;
2802            }
2803        }
2804
2805        if (mSuspended) {
2806            sleepTime = suspendSleepTimeUs();
2807        }
2808        // sleepTime == 0 means we must write to audio hardware
2809        if (sleepTime == 0) {
2810            if (mixerStatus == MIXER_TRACKS_READY) {
2811                applyVolume(leftVol, rightVol, rampVolume);
2812            }
2813            for (size_t i = 0; i < effectChains.size(); i ++) {
2814                effectChains[i]->process_l();
2815            }
2816            unlockEffectChains(effectChains);
2817
2818            mLastWriteTime = systemTime();
2819            mInWrite = true;
2820            mBytesWritten += mixBufferSize;
2821            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2822            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2823            mNumWrites++;
2824            mInWrite = false;
2825            mStandby = false;
2826        } else {
2827            unlockEffectChains(effectChains);
2828            usleep(sleepTime);
2829        }
2830
2831        // finally let go of removed track, without the lock held
2832        // since we can't guarantee the destructors won't acquire that
2833        // same lock.
2834        trackToRemove.clear();
2835        activeTrack.clear();
2836
2837        // Effect chains will be actually deleted here if they were removed from
2838        // mEffectChains list during mixing or effects processing
2839        effectChains.clear();
2840    }
2841
2842    if (!mStandby) {
2843        mOutput->stream->common.standby(&mOutput->stream->common);
2844    }
2845
2846    releaseWakeLock();
2847
2848    LOGV("DirectOutputThread %p exiting", this);
2849    return false;
2850}
2851
2852// getTrackName_l() must be called with ThreadBase::mLock held
2853int AudioFlinger::DirectOutputThread::getTrackName_l()
2854{
2855    return 0;
2856}
2857
2858// deleteTrackName_l() must be called with ThreadBase::mLock held
2859void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2860{
2861}
2862
2863// checkForNewParameters_l() must be called with ThreadBase::mLock held
2864bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2865{
2866    bool reconfig = false;
2867
2868    while (!mNewParameters.isEmpty()) {
2869        status_t status = NO_ERROR;
2870        String8 keyValuePair = mNewParameters[0];
2871        AudioParameter param = AudioParameter(keyValuePair);
2872        int value;
2873
2874        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2875            // do not accept frame count changes if tracks are open as the track buffer
2876            // size depends on frame count and correct behavior would not be garantied
2877            // if frame count is changed after track creation
2878            if (!mTracks.isEmpty()) {
2879                status = INVALID_OPERATION;
2880            } else {
2881                reconfig = true;
2882            }
2883        }
2884        if (status == NO_ERROR) {
2885            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2886                                                    keyValuePair.string());
2887            if (!mStandby && status == INVALID_OPERATION) {
2888               mOutput->stream->common.standby(&mOutput->stream->common);
2889               mStandby = true;
2890               mBytesWritten = 0;
2891               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2892                                                       keyValuePair.string());
2893            }
2894            if (status == NO_ERROR && reconfig) {
2895                readOutputParameters();
2896                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2897            }
2898        }
2899
2900        mNewParameters.removeAt(0);
2901
2902        mParamStatus = status;
2903        mParamCond.signal();
2904        mWaitWorkCV.wait(mLock);
2905    }
2906    return reconfig;
2907}
2908
2909uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2910{
2911    uint32_t time;
2912    if (audio_is_linear_pcm(mFormat)) {
2913        time = (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2914    } else {
2915        time = 10000;
2916    }
2917    return time;
2918}
2919
2920uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2921{
2922    uint32_t time;
2923    if (audio_is_linear_pcm(mFormat)) {
2924        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2925    } else {
2926        time = 10000;
2927    }
2928    return time;
2929}
2930
2931uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2932{
2933    uint32_t time;
2934    if (audio_is_linear_pcm(mFormat)) {
2935        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2936    } else {
2937        time = 10000;
2938    }
2939    return time;
2940}
2941
2942
2943// ----------------------------------------------------------------------------
2944
2945AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2946    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2947{
2948    mType = ThreadBase::DUPLICATING;
2949    addOutputTrack(mainThread);
2950}
2951
2952AudioFlinger::DuplicatingThread::~DuplicatingThread()
2953{
2954    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2955        mOutputTracks[i]->destroy();
2956    }
2957    mOutputTracks.clear();
2958}
2959
2960bool AudioFlinger::DuplicatingThread::threadLoop()
2961{
2962    Vector< sp<Track> > tracksToRemove;
2963    uint32_t mixerStatus = MIXER_IDLE;
2964    nsecs_t standbyTime = systemTime();
2965    size_t mixBufferSize = mFrameCount*mFrameSize;
2966    SortedVector< sp<OutputTrack> > outputTracks;
2967    uint32_t writeFrames = 0;
2968    uint32_t activeSleepTime = activeSleepTimeUs();
2969    uint32_t idleSleepTime = idleSleepTimeUs();
2970    uint32_t sleepTime = idleSleepTime;
2971    Vector< sp<EffectChain> > effectChains;
2972
2973    acquireWakeLock();
2974
2975    while (!exitPending())
2976    {
2977        processConfigEvents();
2978
2979        mixerStatus = MIXER_IDLE;
2980        { // scope for the mLock
2981
2982            Mutex::Autolock _l(mLock);
2983
2984            if (checkForNewParameters_l()) {
2985                mixBufferSize = mFrameCount*mFrameSize;
2986                updateWaitTime();
2987                activeSleepTime = activeSleepTimeUs();
2988                idleSleepTime = idleSleepTimeUs();
2989            }
2990
2991            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
2992
2993            for (size_t i = 0; i < mOutputTracks.size(); i++) {
2994                outputTracks.add(mOutputTracks[i]);
2995            }
2996
2997            // put audio hardware into standby after short delay
2998            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
2999                         mSuspended) {
3000                if (!mStandby) {
3001                    for (size_t i = 0; i < outputTracks.size(); i++) {
3002                        outputTracks[i]->stop();
3003                    }
3004                    mStandby = true;
3005                    mBytesWritten = 0;
3006                }
3007
3008                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
3009                    // we're about to wait, flush the binder command buffer
3010                    IPCThreadState::self()->flushCommands();
3011                    outputTracks.clear();
3012
3013                    if (exitPending()) break;
3014
3015                    releaseWakeLock_l();
3016                    LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3017                    mWaitWorkCV.wait(mLock);
3018                    LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3019                    acquireWakeLock_l();
3020
3021                    if (mMasterMute == false) {
3022                        char value[PROPERTY_VALUE_MAX];
3023                        property_get("ro.audio.silent", value, "0");
3024                        if (atoi(value)) {
3025                            LOGD("Silence is golden");
3026                            setMasterMute(true);
3027                        }
3028                    }
3029
3030                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3031                    sleepTime = idleSleepTime;
3032                    continue;
3033                }
3034            }
3035
3036            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3037
3038            // prevent any changes in effect chain list and in each effect chain
3039            // during mixing and effect process as the audio buffers could be deleted
3040            // or modified if an effect is created or deleted
3041            lockEffectChains_l(effectChains);
3042        }
3043
3044        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3045            // mix buffers...
3046            if (outputsReady(outputTracks)) {
3047                mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
3048            } else {
3049                memset(mMixBuffer, 0, mixBufferSize);
3050            }
3051            sleepTime = 0;
3052            writeFrames = mFrameCount;
3053        } else {
3054            if (sleepTime == 0) {
3055                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3056                    sleepTime = activeSleepTime;
3057                } else {
3058                    sleepTime = idleSleepTime;
3059                }
3060            } else if (mBytesWritten != 0) {
3061                // flush remaining overflow buffers in output tracks
3062                for (size_t i = 0; i < outputTracks.size(); i++) {
3063                    if (outputTracks[i]->isActive()) {
3064                        sleepTime = 0;
3065                        writeFrames = 0;
3066                        memset(mMixBuffer, 0, mixBufferSize);
3067                        break;
3068                    }
3069                }
3070            }
3071        }
3072
3073        if (mSuspended) {
3074            sleepTime = suspendSleepTimeUs();
3075        }
3076        // sleepTime == 0 means we must write to audio hardware
3077        if (sleepTime == 0) {
3078            for (size_t i = 0; i < effectChains.size(); i ++) {
3079                effectChains[i]->process_l();
3080            }
3081            // enable changes in effect chain
3082            unlockEffectChains(effectChains);
3083
3084            standbyTime = systemTime() + kStandbyTimeInNsecs;
3085            for (size_t i = 0; i < outputTracks.size(); i++) {
3086                outputTracks[i]->write(mMixBuffer, writeFrames);
3087            }
3088            mStandby = false;
3089            mBytesWritten += mixBufferSize;
3090        } else {
3091            // enable changes in effect chain
3092            unlockEffectChains(effectChains);
3093            usleep(sleepTime);
3094        }
3095
3096        // finally let go of all our tracks, without the lock held
3097        // since we can't guarantee the destructors won't acquire that
3098        // same lock.
3099        tracksToRemove.clear();
3100        outputTracks.clear();
3101
3102        // Effect chains will be actually deleted here if they were removed from
3103        // mEffectChains list during mixing or effects processing
3104        effectChains.clear();
3105    }
3106
3107    releaseWakeLock();
3108
3109    return false;
3110}
3111
3112void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3113{
3114    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3115    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3116                                            this,
3117                                            mSampleRate,
3118                                            mFormat,
3119                                            mChannelMask,
3120                                            frameCount);
3121    if (outputTrack->cblk() != NULL) {
3122        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3123        mOutputTracks.add(outputTrack);
3124        LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3125        updateWaitTime();
3126    }
3127}
3128
3129void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3130{
3131    Mutex::Autolock _l(mLock);
3132    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3133        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3134            mOutputTracks[i]->destroy();
3135            mOutputTracks.removeAt(i);
3136            updateWaitTime();
3137            return;
3138        }
3139    }
3140    LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3141}
3142
3143void AudioFlinger::DuplicatingThread::updateWaitTime()
3144{
3145    mWaitTimeMs = UINT_MAX;
3146    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3147        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3148        if (strong != NULL) {
3149            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3150            if (waitTimeMs < mWaitTimeMs) {
3151                mWaitTimeMs = waitTimeMs;
3152            }
3153        }
3154    }
3155}
3156
3157
3158bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3159{
3160    for (size_t i = 0; i < outputTracks.size(); i++) {
3161        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3162        if (thread == 0) {
3163            LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3164            return false;
3165        }
3166        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3167        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3168            LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3169            return false;
3170        }
3171    }
3172    return true;
3173}
3174
3175uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3176{
3177    return (mWaitTimeMs * 1000) / 2;
3178}
3179
3180// ----------------------------------------------------------------------------
3181
3182// TrackBase constructor must be called with AudioFlinger::mLock held
3183AudioFlinger::ThreadBase::TrackBase::TrackBase(
3184            const wp<ThreadBase>& thread,
3185            const sp<Client>& client,
3186            uint32_t sampleRate,
3187            uint32_t format,
3188            uint32_t channelMask,
3189            int frameCount,
3190            uint32_t flags,
3191            const sp<IMemory>& sharedBuffer,
3192            int sessionId)
3193    :   RefBase(),
3194        mThread(thread),
3195        mClient(client),
3196        mCblk(0),
3197        mFrameCount(0),
3198        mState(IDLE),
3199        mClientTid(-1),
3200        mFormat(format),
3201        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3202        mSessionId(sessionId)
3203{
3204    LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3205
3206    // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3207   size_t size = sizeof(audio_track_cblk_t);
3208   uint8_t channelCount = popcount(channelMask);
3209   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3210   if (sharedBuffer == 0) {
3211       size += bufferSize;
3212   }
3213
3214   if (client != NULL) {
3215        mCblkMemory = client->heap()->allocate(size);
3216        if (mCblkMemory != 0) {
3217            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3218            if (mCblk) { // construct the shared structure in-place.
3219                new(mCblk) audio_track_cblk_t();
3220                // clear all buffers
3221                mCblk->frameCount = frameCount;
3222                mCblk->sampleRate = sampleRate;
3223                mChannelCount = channelCount;
3224                mChannelMask = channelMask;
3225                if (sharedBuffer == 0) {
3226                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3227                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3228                    // Force underrun condition to avoid false underrun callback until first data is
3229                    // written to buffer (other flags are cleared)
3230                    mCblk->flags = CBLK_UNDERRUN_ON;
3231                } else {
3232                    mBuffer = sharedBuffer->pointer();
3233                }
3234                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3235            }
3236        } else {
3237            LOGE("not enough memory for AudioTrack size=%u", size);
3238            client->heap()->dump("AudioTrack");
3239            return;
3240        }
3241   } else {
3242       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3243       if (mCblk) { // construct the shared structure in-place.
3244           new(mCblk) audio_track_cblk_t();
3245           // clear all buffers
3246           mCblk->frameCount = frameCount;
3247           mCblk->sampleRate = sampleRate;
3248           mChannelCount = channelCount;
3249           mChannelMask = channelMask;
3250           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3251           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3252           // Force underrun condition to avoid false underrun callback until first data is
3253           // written to buffer (other flags are cleared)
3254           mCblk->flags = CBLK_UNDERRUN_ON;
3255           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3256       }
3257   }
3258}
3259
3260AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3261{
3262    if (mCblk) {
3263        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3264        if (mClient == NULL) {
3265            delete mCblk;
3266        }
3267    }
3268    mCblkMemory.clear();            // and free the shared memory
3269    if (mClient != NULL) {
3270        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3271        mClient.clear();
3272    }
3273}
3274
3275void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3276{
3277    buffer->raw = 0;
3278    mFrameCount = buffer->frameCount;
3279    step();
3280    buffer->frameCount = 0;
3281}
3282
3283bool AudioFlinger::ThreadBase::TrackBase::step() {
3284    bool result;
3285    audio_track_cblk_t* cblk = this->cblk();
3286
3287    result = cblk->stepServer(mFrameCount);
3288    if (!result) {
3289        LOGV("stepServer failed acquiring cblk mutex");
3290        mFlags |= STEPSERVER_FAILED;
3291    }
3292    return result;
3293}
3294
3295void AudioFlinger::ThreadBase::TrackBase::reset() {
3296    audio_track_cblk_t* cblk = this->cblk();
3297
3298    cblk->user = 0;
3299    cblk->server = 0;
3300    cblk->userBase = 0;
3301    cblk->serverBase = 0;
3302    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3303    LOGV("TrackBase::reset");
3304}
3305
3306sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3307{
3308    return mCblkMemory;
3309}
3310
3311int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3312    return (int)mCblk->sampleRate;
3313}
3314
3315int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3316    return (const int)mChannelCount;
3317}
3318
3319uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3320    return mChannelMask;
3321}
3322
3323void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3324    audio_track_cblk_t* cblk = this->cblk();
3325    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
3326    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
3327
3328    // Check validity of returned pointer in case the track control block would have been corrupted.
3329    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3330        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
3331        LOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3332                server %d, serverBase %d, user %d, userBase %d",
3333                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3334                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3335        return 0;
3336    }
3337
3338    return bufferStart;
3339}
3340
3341// ----------------------------------------------------------------------------
3342
3343// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3344AudioFlinger::PlaybackThread::Track::Track(
3345            const wp<ThreadBase>& thread,
3346            const sp<Client>& client,
3347            int streamType,
3348            uint32_t sampleRate,
3349            uint32_t format,
3350            uint32_t channelMask,
3351            int frameCount,
3352            const sp<IMemory>& sharedBuffer,
3353            int sessionId)
3354    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3355    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3356    mAuxEffectId(0), mHasVolumeController(false)
3357{
3358    if (mCblk != NULL) {
3359        sp<ThreadBase> baseThread = thread.promote();
3360        if (baseThread != 0) {
3361            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3362            mName = playbackThread->getTrackName_l();
3363            mMainBuffer = playbackThread->mixBuffer();
3364        }
3365        LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3366        if (mName < 0) {
3367            LOGE("no more track names available");
3368        }
3369        mVolume[0] = 1.0f;
3370        mVolume[1] = 1.0f;
3371        mStreamType = streamType;
3372        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3373        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3374        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3375    }
3376}
3377
3378AudioFlinger::PlaybackThread::Track::~Track()
3379{
3380    LOGV("PlaybackThread::Track destructor");
3381    sp<ThreadBase> thread = mThread.promote();
3382    if (thread != 0) {
3383        Mutex::Autolock _l(thread->mLock);
3384        mState = TERMINATED;
3385    }
3386}
3387
3388void AudioFlinger::PlaybackThread::Track::destroy()
3389{
3390    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3391    // by removing it from mTracks vector, so there is a risk that this Tracks's
3392    // desctructor is called. As the destructor needs to lock mLock,
3393    // we must acquire a strong reference on this Track before locking mLock
3394    // here so that the destructor is called only when exiting this function.
3395    // On the other hand, as long as Track::destroy() is only called by
3396    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3397    // this Track with its member mTrack.
3398    sp<Track> keep(this);
3399    { // scope for mLock
3400        sp<ThreadBase> thread = mThread.promote();
3401        if (thread != 0) {
3402            if (!isOutputTrack()) {
3403                if (mState == ACTIVE || mState == RESUMING) {
3404                    AudioSystem::stopOutput(thread->id(),
3405                                            (audio_stream_type_t)mStreamType,
3406                                            mSessionId);
3407
3408                    // to track the speaker usage
3409                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3410                }
3411                AudioSystem::releaseOutput(thread->id());
3412            }
3413            Mutex::Autolock _l(thread->mLock);
3414            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3415            playbackThread->destroyTrack_l(this);
3416        }
3417    }
3418}
3419
3420void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3421{
3422    snprintf(buffer, size, "   %05d %05d %03u %03u 0x%08x %05u   %04u %1d %1d %1d %05u %05u %05u  0x%08x 0x%08x 0x%08x 0x%08x\n",
3423            mName - AudioMixer::TRACK0,
3424            (mClient == NULL) ? getpid() : mClient->pid(),
3425            mStreamType,
3426            mFormat,
3427            mChannelMask,
3428            mSessionId,
3429            mFrameCount,
3430            mState,
3431            mMute,
3432            mFillingUpStatus,
3433            mCblk->sampleRate,
3434            mCblk->volume[0],
3435            mCblk->volume[1],
3436            mCblk->server,
3437            mCblk->user,
3438            (int)mMainBuffer,
3439            (int)mAuxBuffer);
3440}
3441
3442status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
3443    AudioBufferProvider::Buffer* buffer, int64_t pts)
3444{
3445     audio_track_cblk_t* cblk = this->cblk();
3446     uint32_t framesReady;
3447     uint32_t framesReq = buffer->frameCount;
3448
3449     // Check if last stepServer failed, try to step now
3450     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3451         if (!step())  goto getNextBuffer_exit;
3452         LOGV("stepServer recovered");
3453         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3454     }
3455
3456     framesReady = cblk->framesReady();
3457
3458     if (LIKELY(framesReady)) {
3459        uint32_t s = cblk->server;
3460        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3461
3462        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3463        if (framesReq > framesReady) {
3464            framesReq = framesReady;
3465        }
3466        if (s + framesReq > bufferEnd) {
3467            framesReq = bufferEnd - s;
3468        }
3469
3470         buffer->raw = getBuffer(s, framesReq);
3471         if (buffer->raw == 0) goto getNextBuffer_exit;
3472
3473         buffer->frameCount = framesReq;
3474        return NO_ERROR;
3475     }
3476
3477getNextBuffer_exit:
3478     buffer->raw = 0;
3479     buffer->frameCount = 0;
3480     LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3481     return NOT_ENOUGH_DATA;
3482}
3483
3484uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{
3485    return mCblk->framesReady();
3486}
3487
3488bool AudioFlinger::PlaybackThread::Track::isReady() const {
3489    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3490
3491    if (framesReady() >= mCblk->frameCount ||
3492            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3493        mFillingUpStatus = FS_FILLED;
3494        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3495        return true;
3496    }
3497    return false;
3498}
3499
3500status_t AudioFlinger::PlaybackThread::Track::start()
3501{
3502    status_t status = NO_ERROR;
3503    LOGV("start(%d), calling thread %d session %d",
3504            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3505    sp<ThreadBase> thread = mThread.promote();
3506    if (thread != 0) {
3507        Mutex::Autolock _l(thread->mLock);
3508        int state = mState;
3509        // here the track could be either new, or restarted
3510        // in both cases "unstop" the track
3511        if (mState == PAUSED) {
3512            mState = TrackBase::RESUMING;
3513            LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3514        } else {
3515            mState = TrackBase::ACTIVE;
3516            LOGV("? => ACTIVE (%d) on thread %p", mName, this);
3517        }
3518
3519        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3520            thread->mLock.unlock();
3521            status = AudioSystem::startOutput(thread->id(),
3522                                              (audio_stream_type_t)mStreamType,
3523                                              mSessionId);
3524            thread->mLock.lock();
3525
3526            // to track the speaker usage
3527            if (status == NO_ERROR) {
3528                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3529            }
3530        }
3531        if (status == NO_ERROR) {
3532            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3533            playbackThread->addTrack_l(this);
3534        } else {
3535            mState = state;
3536        }
3537    } else {
3538        status = BAD_VALUE;
3539    }
3540    return status;
3541}
3542
3543void AudioFlinger::PlaybackThread::Track::stop()
3544{
3545    LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3546    sp<ThreadBase> thread = mThread.promote();
3547    if (thread != 0) {
3548        Mutex::Autolock _l(thread->mLock);
3549        int state = mState;
3550        if (mState > STOPPED) {
3551            mState = STOPPED;
3552            // If the track is not active (PAUSED and buffers full), flush buffers
3553            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3554            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3555                reset();
3556            }
3557            LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3558        }
3559        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3560            thread->mLock.unlock();
3561            AudioSystem::stopOutput(thread->id(),
3562                                    (audio_stream_type_t)mStreamType,
3563                                    mSessionId);
3564            thread->mLock.lock();
3565
3566            // to track the speaker usage
3567            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3568        }
3569    }
3570}
3571
3572void AudioFlinger::PlaybackThread::Track::pause()
3573{
3574    LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3575    sp<ThreadBase> thread = mThread.promote();
3576    if (thread != 0) {
3577        Mutex::Autolock _l(thread->mLock);
3578        if (mState == ACTIVE || mState == RESUMING) {
3579            mState = PAUSING;
3580            LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3581            if (!isOutputTrack()) {
3582                thread->mLock.unlock();
3583                AudioSystem::stopOutput(thread->id(),
3584                                        (audio_stream_type_t)mStreamType,
3585                                        mSessionId);
3586                thread->mLock.lock();
3587
3588                // to track the speaker usage
3589                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3590            }
3591        }
3592    }
3593}
3594
3595void AudioFlinger::PlaybackThread::Track::flush()
3596{
3597    LOGV("flush(%d)", mName);
3598    sp<ThreadBase> thread = mThread.promote();
3599    if (thread != 0) {
3600        Mutex::Autolock _l(thread->mLock);
3601        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3602            return;
3603        }
3604        // No point remaining in PAUSED state after a flush => go to
3605        // STOPPED state
3606        mState = STOPPED;
3607
3608        // do not reset the track if it is still in the process of being stopped or paused.
3609        // this will be done by prepareTracks_l() when the track is stopped.
3610        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3611        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3612            reset();
3613        }
3614    }
3615}
3616
3617void AudioFlinger::PlaybackThread::Track::reset()
3618{
3619    // Do not reset twice to avoid discarding data written just after a flush and before
3620    // the audioflinger thread detects the track is stopped.
3621    if (!mResetDone) {
3622        TrackBase::reset();
3623        // Force underrun condition to avoid false underrun callback until first data is
3624        // written to buffer
3625        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3626        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3627        mFillingUpStatus = FS_FILLING;
3628        mResetDone = true;
3629    }
3630}
3631
3632void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3633{
3634    mMute = muted;
3635}
3636
3637void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3638{
3639    mVolume[0] = left;
3640    mVolume[1] = right;
3641}
3642
3643status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3644{
3645    status_t status = DEAD_OBJECT;
3646    sp<ThreadBase> thread = mThread.promote();
3647    if (thread != 0) {
3648       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3649       status = playbackThread->attachAuxEffect(this, EffectId);
3650    }
3651    return status;
3652}
3653
3654void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3655{
3656    mAuxEffectId = EffectId;
3657    mAuxBuffer = buffer;
3658}
3659
3660// timed audio tracks
3661
3662sp<AudioFlinger::PlaybackThread::TimedTrack>
3663AudioFlinger::PlaybackThread::TimedTrack::create(
3664            const wp<ThreadBase>& thread,
3665            const sp<Client>& client,
3666            int streamType,
3667            uint32_t sampleRate,
3668            uint32_t format,
3669            uint32_t channelMask,
3670            int frameCount,
3671            const sp<IMemory>& sharedBuffer,
3672            int sessionId) {
3673    if (!client->reserveTimedTrack())
3674        return NULL;
3675
3676    sp<TimedTrack> track = new TimedTrack(
3677        thread, client, streamType, sampleRate, format, channelMask, frameCount,
3678        sharedBuffer, sessionId);
3679
3680    if (track == NULL) {
3681        client->releaseTimedTrack();
3682        return NULL;
3683    }
3684
3685    return track;
3686}
3687
3688AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
3689            const wp<ThreadBase>& thread,
3690            const sp<Client>& client,
3691            int streamType,
3692            uint32_t sampleRate,
3693            uint32_t format,
3694            uint32_t channelMask,
3695            int frameCount,
3696            const sp<IMemory>& sharedBuffer,
3697            int sessionId)
3698    : Track(thread, client, streamType, sampleRate, format, channelMask,
3699            frameCount, sharedBuffer, sessionId),
3700      mTimedSilenceBuffer(NULL),
3701      mTimedSilenceBufferSize(0),
3702      mTimedAudioOutputOnTime(false),
3703      mMediaTimeTransformValid(false)
3704{
3705    LocalClock lc;
3706    mLocalTimeFreq = lc.getLocalFreq();
3707
3708    mLocalTimeToSampleTransform.a_zero = 0;
3709    mLocalTimeToSampleTransform.b_zero = 0;
3710    mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
3711    mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
3712    LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
3713                            &mLocalTimeToSampleTransform.a_to_b_denom);
3714}
3715
3716AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
3717    mClient->releaseTimedTrack();
3718    delete [] mTimedSilenceBuffer;
3719}
3720
3721status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
3722    size_t size, sp<IMemory>* buffer) {
3723
3724    Mutex::Autolock _l(mTimedBufferQueueLock);
3725
3726    trimTimedBufferQueue_l();
3727
3728    // lazily initialize the shared memory heap for timed buffers
3729    if (mTimedMemoryDealer == NULL) {
3730        const int kTimedBufferHeapSize = 512 << 10;
3731
3732        mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
3733                                              "AudioFlingerTimed");
3734        if (mTimedMemoryDealer == NULL)
3735            return NO_MEMORY;
3736    }
3737
3738    sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
3739    if (newBuffer == NULL) {
3740        newBuffer = mTimedMemoryDealer->allocate(size);
3741        if (newBuffer == NULL)
3742            return NO_MEMORY;
3743    }
3744
3745    *buffer = newBuffer;
3746    return NO_ERROR;
3747}
3748
3749// caller must hold mTimedBufferQueueLock
3750void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
3751    int64_t mediaTimeNow;
3752    {
3753        Mutex::Autolock mttLock(mMediaTimeTransformLock);
3754        if (!mMediaTimeTransformValid)
3755            return;
3756
3757        int64_t targetTimeNow;
3758        status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
3759            ? CCHelper::getCommonTime(&targetTimeNow)
3760            : CCHelper::getLocalTime(&targetTimeNow);
3761
3762        if (OK != res)
3763            return;
3764
3765        if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
3766                                                    &mediaTimeNow)) {
3767            return;
3768        }
3769    }
3770
3771    size_t trimIndex;
3772    for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) {
3773        if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow)
3774            break;
3775    }
3776
3777    if (trimIndex) {
3778        mTimedBufferQueue.removeItemsAt(0, trimIndex);
3779    }
3780}
3781
3782status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
3783    const sp<IMemory>& buffer, int64_t pts) {
3784
3785    {
3786        Mutex::Autolock mttLock(mMediaTimeTransformLock);
3787        if (!mMediaTimeTransformValid)
3788            return INVALID_OPERATION;
3789    }
3790
3791    Mutex::Autolock _l(mTimedBufferQueueLock);
3792
3793    mTimedBufferQueue.add(TimedBuffer(buffer, pts));
3794
3795    return NO_ERROR;
3796}
3797
3798status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
3799    const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
3800
3801    LOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__,
3802         xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
3803         target);
3804
3805    if (!(target == TimedAudioTrack::LOCAL_TIME ||
3806          target == TimedAudioTrack::COMMON_TIME)) {
3807        return BAD_VALUE;
3808    }
3809
3810    Mutex::Autolock lock(mMediaTimeTransformLock);
3811    mMediaTimeTransform = xform;
3812    mMediaTimeTransformTarget = target;
3813    mMediaTimeTransformValid = true;
3814
3815    return NO_ERROR;
3816}
3817
3818#define min(a, b) ((a) < (b) ? (a) : (b))
3819
3820// implementation of getNextBuffer for tracks whose buffers have timestamps
3821status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
3822    AudioBufferProvider::Buffer* buffer, int64_t pts)
3823{
3824    if (pts == AudioBufferProvider::kInvalidPTS) {
3825        buffer->raw = 0;
3826        buffer->frameCount = 0;
3827        return INVALID_OPERATION;
3828    }
3829
3830    // get ahold of the output stream that these samples will be written to
3831    sp<ThreadBase> thread = mThread.promote();
3832    if (thread == NULL) {
3833        buffer->raw = 0;
3834        buffer->frameCount = 0;
3835        return INVALID_OPERATION;
3836    }
3837    PlaybackThread* playbackThread = static_cast<PlaybackThread*>(thread.get());
3838
3839    Mutex::Autolock _l(mTimedBufferQueueLock);
3840
3841    while (true) {
3842
3843        // if we have no timed buffers, then fail
3844        if (mTimedBufferQueue.isEmpty()) {
3845            buffer->raw = 0;
3846            buffer->frameCount = 0;
3847            return NOT_ENOUGH_DATA;
3848        }
3849
3850        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
3851
3852        // calculate the PTS of the head of the timed buffer queue expressed in
3853        // local time
3854        int64_t headLocalPTS;
3855        {
3856            Mutex::Autolock mttLock(mMediaTimeTransformLock);
3857
3858            assert(mMediaTimeTransformValid);
3859
3860            if (mMediaTimeTransform.a_to_b_denom == 0) {
3861                // the transform represents a pause, so yield silence
3862                timedYieldSilence(buffer->frameCount, buffer);
3863                return NO_ERROR;
3864            }
3865
3866            int64_t transformedPTS;
3867            if (!mMediaTimeTransform.doForwardTransform(head.pts(),
3868                                                        &transformedPTS)) {
3869                // the transform failed.  this shouldn't happen, but if it does
3870                // then just drop this buffer
3871                LOGW("timedGetNextBuffer transform failed");
3872                buffer->raw = 0;
3873                buffer->frameCount = 0;
3874                mTimedBufferQueue.removeAt(0);
3875                return NO_ERROR;
3876            }
3877
3878            if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
3879                if (OK != CCHelper::commonTimeToLocalTime(transformedPTS,
3880                                                          &headLocalPTS)) {
3881                    buffer->raw = 0;
3882                    buffer->frameCount = 0;
3883                    return INVALID_OPERATION;
3884                }
3885            } else {
3886                headLocalPTS = transformedPTS;
3887            }
3888        }
3889
3890        // adjust the head buffer's PTS to reflect the portion of the head buffer
3891        // that has already been consumed
3892        int64_t effectivePTS = headLocalPTS +
3893                ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate());
3894
3895        // Calculate the delta in samples between the head of the input buffer
3896        // queue and the start of the next output buffer that will be written.
3897        // If the transformation fails because of over or underflow, it means
3898        // that the sample's position in the output stream is so far out of
3899        // whack that it should just be dropped.
3900        int64_t sampleDelta;
3901        if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
3902            LOGV("*** head buffer is too far from PTS: dropped buffer");
3903            mTimedBufferQueue.removeAt(0);
3904            continue;
3905        }
3906        if (!mLocalTimeToSampleTransform.doForwardTransform(
3907                (effectivePTS - pts) << 32, &sampleDelta)) {
3908            LOGV("*** too late during sample rate transform: dropped buffer");
3909            mTimedBufferQueue.removeAt(0);
3910            continue;
3911        }
3912
3913        LOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]",
3914             __PRETTY_FUNCTION__, head.pts(), head.position(), pts,
3915             static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)),
3916             static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
3917
3918        // if the delta between the ideal placement for the next input sample and
3919        // the current output position is within this threshold, then we will
3920        // concatenate the next input samples to the previous output
3921        const int64_t kSampleContinuityThreshold =
3922                (static_cast<int64_t>(sampleRate()) << 32) / 10;
3923
3924        // if this is the first buffer of audio that we're emitting from this track
3925        // then it should be almost exactly on time.
3926        const int64_t kSampleStartupThreshold = 1LL << 32;
3927
3928        if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
3929            (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
3930            // the next input is close enough to being on time, so concatenate it
3931            // with the last output
3932            timedYieldSamples(buffer);
3933
3934            LOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
3935            return NO_ERROR;
3936        } else if (sampleDelta > 0) {
3937            // the gap between the current output position and the proper start of
3938            // the next input sample is too big, so fill it with silence
3939            uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
3940
3941            timedYieldSilence(framesUntilNextInput, buffer);
3942            LOGV("*** silence: frameCount=%u", buffer->frameCount);
3943            return NO_ERROR;
3944        } else {
3945            // the next input sample is late
3946            uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
3947            size_t onTimeSamplePosition =
3948                    head.position() + lateFrames * mCblk->frameSize;
3949
3950            if (onTimeSamplePosition > head.buffer()->size()) {
3951                // all the remaining samples in the head are too late, so
3952                // drop it and move on
3953                LOGV("*** too late: dropped buffer");
3954                mTimedBufferQueue.removeAt(0);
3955                continue;
3956            } else {
3957                // skip over the late samples
3958                head.setPosition(onTimeSamplePosition);
3959
3960                // yield the available samples
3961                timedYieldSamples(buffer);
3962
3963                LOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
3964                return NO_ERROR;
3965            }
3966        }
3967    }
3968}
3969
3970// Yield samples from the timed buffer queue head up to the given output
3971// buffer's capacity.
3972//
3973// Caller must hold mTimedBufferQueueLock
3974void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples(
3975    AudioBufferProvider::Buffer* buffer) {
3976
3977    const TimedBuffer& head = mTimedBufferQueue[0];
3978
3979    buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
3980                   head.position());
3981
3982    uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
3983                                 mCblk->frameSize);
3984    size_t framesRequested = buffer->frameCount;
3985    buffer->frameCount = min(framesLeftInHead, framesRequested);
3986
3987    mTimedAudioOutputOnTime = true;
3988}
3989
3990// Yield samples of silence up to the given output buffer's capacity
3991//
3992// Caller must hold mTimedBufferQueueLock
3993void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence(
3994    uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
3995
3996    // lazily allocate a buffer filled with silence
3997    if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) {
3998        delete [] mTimedSilenceBuffer;
3999        mTimedSilenceBufferSize = numFrames * mCblk->frameSize;
4000        mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
4001        memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
4002    }
4003
4004    buffer->raw = mTimedSilenceBuffer;
4005    size_t framesRequested = buffer->frameCount;
4006    buffer->frameCount = min(numFrames, framesRequested);
4007
4008    mTimedAudioOutputOnTime = false;
4009}
4010
4011void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
4012    AudioBufferProvider::Buffer* buffer) {
4013
4014    Mutex::Autolock _l(mTimedBufferQueueLock);
4015
4016    if (buffer->raw != mTimedSilenceBuffer) {
4017        TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
4018        head.setPosition(head.position() + buffer->frameCount * mCblk->frameSize);
4019        if (static_cast<size_t>(head.position()) >= head.buffer()->size()) {
4020            mTimedBufferQueue.removeAt(0);
4021        }
4022    }
4023
4024    buffer->raw = 0;
4025    buffer->frameCount = 0;
4026}
4027
4028uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
4029    Mutex::Autolock _l(mTimedBufferQueueLock);
4030
4031    uint32_t frames = 0;
4032    for (size_t i = 0; i < mTimedBufferQueue.size(); i++) {
4033        const TimedBuffer& tb = mTimedBufferQueue[i];
4034        frames += (tb.buffer()->size() - tb.position())  / mCblk->frameSize;
4035    }
4036
4037    return frames;
4038}
4039
4040AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
4041        : mPTS(0), mPosition(0) {}
4042
4043AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
4044    const sp<IMemory>& buffer, int64_t pts)
4045        : mBuffer(buffer), mPTS(pts), mPosition(0) {}
4046
4047// ----------------------------------------------------------------------------
4048
4049// RecordTrack constructor must be called with AudioFlinger::mLock held
4050AudioFlinger::RecordThread::RecordTrack::RecordTrack(
4051            const wp<ThreadBase>& thread,
4052            const sp<Client>& client,
4053            uint32_t sampleRate,
4054            uint32_t format,
4055            uint32_t channelMask,
4056            int frameCount,
4057            uint32_t flags,
4058            int sessionId)
4059    :   TrackBase(thread, client, sampleRate, format,
4060                  channelMask, frameCount, flags, 0, sessionId),
4061        mOverflow(false)
4062{
4063    if (mCblk != NULL) {
4064       LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
4065       if (format == AUDIO_FORMAT_PCM_16_BIT) {
4066           mCblk->frameSize = mChannelCount * sizeof(int16_t);
4067       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
4068           mCblk->frameSize = mChannelCount * sizeof(int8_t);
4069       } else {
4070           mCblk->frameSize = sizeof(int8_t);
4071       }
4072    }
4073}
4074
4075AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
4076{
4077    sp<ThreadBase> thread = mThread.promote();
4078    if (thread != 0) {
4079        AudioSystem::releaseInput(thread->id());
4080    }
4081}
4082
4083status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
4084{
4085    audio_track_cblk_t* cblk = this->cblk();
4086    uint32_t framesAvail;
4087    uint32_t framesReq = buffer->frameCount;
4088
4089     // Check if last stepServer failed, try to step now
4090    if (mFlags & TrackBase::STEPSERVER_FAILED) {
4091        if (!step()) goto getNextBuffer_exit;
4092        LOGV("stepServer recovered");
4093        mFlags &= ~TrackBase::STEPSERVER_FAILED;
4094    }
4095
4096    framesAvail = cblk->framesAvailable_l();
4097
4098    if (LIKELY(framesAvail)) {
4099        uint32_t s = cblk->server;
4100        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
4101
4102        if (framesReq > framesAvail) {
4103            framesReq = framesAvail;
4104        }
4105        if (s + framesReq > bufferEnd) {
4106            framesReq = bufferEnd - s;
4107        }
4108
4109        buffer->raw = getBuffer(s, framesReq);
4110        if (buffer->raw == 0) goto getNextBuffer_exit;
4111
4112        buffer->frameCount = framesReq;
4113        return NO_ERROR;
4114    }
4115
4116getNextBuffer_exit:
4117    buffer->raw = 0;
4118    buffer->frameCount = 0;
4119    return NOT_ENOUGH_DATA;
4120}
4121
4122status_t AudioFlinger::RecordThread::RecordTrack::start()
4123{
4124    sp<ThreadBase> thread = mThread.promote();
4125    if (thread != 0) {
4126        RecordThread *recordThread = (RecordThread *)thread.get();
4127        return recordThread->start(this);
4128    } else {
4129        return BAD_VALUE;
4130    }
4131}
4132
4133void AudioFlinger::RecordThread::RecordTrack::stop()
4134{
4135    sp<ThreadBase> thread = mThread.promote();
4136    if (thread != 0) {
4137        RecordThread *recordThread = (RecordThread *)thread.get();
4138        recordThread->stop(this);
4139        TrackBase::reset();
4140        // Force overerrun condition to avoid false overrun callback until first data is
4141        // read from buffer
4142        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
4143    }
4144}
4145
4146void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
4147{
4148    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
4149            (mClient == NULL) ? getpid() : mClient->pid(),
4150            mFormat,
4151            mChannelMask,
4152            mSessionId,
4153            mFrameCount,
4154            mState,
4155            mCblk->sampleRate,
4156            mCblk->server,
4157            mCblk->user);
4158}
4159
4160
4161// ----------------------------------------------------------------------------
4162
4163AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
4164            const wp<ThreadBase>& thread,
4165            DuplicatingThread *sourceThread,
4166            uint32_t sampleRate,
4167            uint32_t format,
4168            uint32_t channelMask,
4169            int frameCount)
4170    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
4171    mActive(false), mSourceThread(sourceThread)
4172{
4173
4174    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
4175    if (mCblk != NULL) {
4176        mCblk->flags |= CBLK_DIRECTION_OUT;
4177        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
4178        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
4179        mOutBuffer.frameCount = 0;
4180        playbackThread->mTracks.add(this);
4181        LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
4182                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
4183                mCblk, mBuffer, mCblk->buffers,
4184                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
4185    } else {
4186        LOGW("Error creating output track on thread %p", playbackThread);
4187    }
4188}
4189
4190AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
4191{
4192    clearBufferQueue();
4193}
4194
4195status_t AudioFlinger::PlaybackThread::OutputTrack::start()
4196{
4197    status_t status = Track::start();
4198    if (status != NO_ERROR) {
4199        return status;
4200    }
4201
4202    mActive = true;
4203    mRetryCount = 127;
4204    return status;
4205}
4206
4207void AudioFlinger::PlaybackThread::OutputTrack::stop()
4208{
4209    Track::stop();
4210    clearBufferQueue();
4211    mOutBuffer.frameCount = 0;
4212    mActive = false;
4213}
4214
4215bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
4216{
4217    Buffer *pInBuffer;
4218    Buffer inBuffer;
4219    uint32_t channelCount = mChannelCount;
4220    bool outputBufferFull = false;
4221    inBuffer.frameCount = frames;
4222    inBuffer.i16 = data;
4223
4224    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
4225
4226    if (!mActive && frames != 0) {
4227        start();
4228        sp<ThreadBase> thread = mThread.promote();
4229        if (thread != 0) {
4230            MixerThread *mixerThread = (MixerThread *)thread.get();
4231            if (mCblk->frameCount > frames){
4232                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
4233                    uint32_t startFrames = (mCblk->frameCount - frames);
4234                    pInBuffer = new Buffer;
4235                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
4236                    pInBuffer->frameCount = startFrames;
4237                    pInBuffer->i16 = pInBuffer->mBuffer;
4238                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
4239                    mBufferQueue.add(pInBuffer);
4240                } else {
4241                    LOGW ("OutputTrack::write() %p no more buffers in queue", this);
4242                }
4243            }
4244        }
4245    }
4246
4247    while (waitTimeLeftMs) {
4248        // First write pending buffers, then new data
4249        if (mBufferQueue.size()) {
4250            pInBuffer = mBufferQueue.itemAt(0);
4251        } else {
4252            pInBuffer = &inBuffer;
4253        }
4254
4255        if (pInBuffer->frameCount == 0) {
4256            break;
4257        }
4258
4259        if (mOutBuffer.frameCount == 0) {
4260            mOutBuffer.frameCount = pInBuffer->frameCount;
4261            nsecs_t startTime = systemTime();
4262            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
4263                LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
4264                outputBufferFull = true;
4265                break;
4266            }
4267            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
4268            if (waitTimeLeftMs >= waitTimeMs) {
4269                waitTimeLeftMs -= waitTimeMs;
4270            } else {
4271                waitTimeLeftMs = 0;
4272            }
4273        }
4274
4275        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
4276        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
4277        mCblk->stepUser(outFrames);
4278        pInBuffer->frameCount -= outFrames;
4279        pInBuffer->i16 += outFrames * channelCount;
4280        mOutBuffer.frameCount -= outFrames;
4281        mOutBuffer.i16 += outFrames * channelCount;
4282
4283        if (pInBuffer->frameCount == 0) {
4284            if (mBufferQueue.size()) {
4285                mBufferQueue.removeAt(0);
4286                delete [] pInBuffer->mBuffer;
4287                delete pInBuffer;
4288                LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
4289            } else {
4290                break;
4291            }
4292        }
4293    }
4294
4295    // If we could not write all frames, allocate a buffer and queue it for next time.
4296    if (inBuffer.frameCount) {
4297        sp<ThreadBase> thread = mThread.promote();
4298        if (thread != 0 && !thread->standby()) {
4299            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
4300                pInBuffer = new Buffer;
4301                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
4302                pInBuffer->frameCount = inBuffer.frameCount;
4303                pInBuffer->i16 = pInBuffer->mBuffer;
4304                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
4305                mBufferQueue.add(pInBuffer);
4306                LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
4307            } else {
4308                LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
4309            }
4310        }
4311    }
4312
4313    // Calling write() with a 0 length buffer, means that no more data will be written:
4314    // If no more buffers are pending, fill output track buffer to make sure it is started
4315    // by output mixer.
4316    if (frames == 0 && mBufferQueue.size() == 0) {
4317        if (mCblk->user < mCblk->frameCount) {
4318            frames = mCblk->frameCount - mCblk->user;
4319            pInBuffer = new Buffer;
4320            pInBuffer->mBuffer = new int16_t[frames * channelCount];
4321            pInBuffer->frameCount = frames;
4322            pInBuffer->i16 = pInBuffer->mBuffer;
4323            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
4324            mBufferQueue.add(pInBuffer);
4325        } else if (mActive) {
4326            stop();
4327        }
4328    }
4329
4330    return outputBufferFull;
4331}
4332
4333status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
4334{
4335    int active;
4336    status_t result;
4337    audio_track_cblk_t* cblk = mCblk;
4338    uint32_t framesReq = buffer->frameCount;
4339
4340//    LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
4341    buffer->frameCount  = 0;
4342
4343    uint32_t framesAvail = cblk->framesAvailable();
4344
4345
4346    if (framesAvail == 0) {
4347        Mutex::Autolock _l(cblk->lock);
4348        goto start_loop_here;
4349        while (framesAvail == 0) {
4350            active = mActive;
4351            if (UNLIKELY(!active)) {
4352                LOGV("Not active and NO_MORE_BUFFERS");
4353                return AudioTrack::NO_MORE_BUFFERS;
4354            }
4355            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
4356            if (result != NO_ERROR) {
4357                return AudioTrack::NO_MORE_BUFFERS;
4358            }
4359            // read the server count again
4360        start_loop_here:
4361            framesAvail = cblk->framesAvailable_l();
4362        }
4363    }
4364
4365//    if (framesAvail < framesReq) {
4366//        return AudioTrack::NO_MORE_BUFFERS;
4367//    }
4368
4369    if (framesReq > framesAvail) {
4370        framesReq = framesAvail;
4371    }
4372
4373    uint32_t u = cblk->user;
4374    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
4375
4376    if (u + framesReq > bufferEnd) {
4377        framesReq = bufferEnd - u;
4378    }
4379
4380    buffer->frameCount  = framesReq;
4381    buffer->raw         = (void *)cblk->buffer(u);
4382    return NO_ERROR;
4383}
4384
4385
4386void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
4387{
4388    size_t size = mBufferQueue.size();
4389    Buffer *pBuffer;
4390
4391    for (size_t i = 0; i < size; i++) {
4392        pBuffer = mBufferQueue.itemAt(i);
4393        delete [] pBuffer->mBuffer;
4394        delete pBuffer;
4395    }
4396    mBufferQueue.clear();
4397}
4398
4399// ----------------------------------------------------------------------------
4400
4401AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
4402    :   RefBase(),
4403        mAudioFlinger(audioFlinger),
4404        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
4405        mPid(pid),
4406        mTimedTrackCount(0)
4407{
4408    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
4409}
4410
4411// Client destructor must be called with AudioFlinger::mLock held
4412AudioFlinger::Client::~Client()
4413{
4414    mAudioFlinger->removeClient_l(mPid);
4415}
4416
4417const sp<MemoryDealer>& AudioFlinger::Client::heap() const
4418{
4419    return mMemoryDealer;
4420}
4421
4422// Reserve one of the limited slots for a timed audio track associated
4423// with this client
4424bool AudioFlinger::Client::reserveTimedTrack()
4425{
4426    const int kMaxTimedTracksPerClient = 4;
4427
4428    Mutex::Autolock _l(mTimedTrackLock);
4429
4430    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
4431        LOGW("can not create timed track - pid %d has exceeded the limit",
4432             mPid);
4433        return false;
4434    }
4435
4436    mTimedTrackCount++;
4437    return true;
4438}
4439
4440// Release a slot for a timed audio track
4441void AudioFlinger::Client::releaseTimedTrack()
4442{
4443    Mutex::Autolock _l(mTimedTrackLock);
4444    mTimedTrackCount--;
4445}
4446
4447// ----------------------------------------------------------------------------
4448
4449AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4450                                                     const sp<IAudioFlingerClient>& client,
4451                                                     pid_t pid)
4452    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
4453{
4454}
4455
4456AudioFlinger::NotificationClient::~NotificationClient()
4457{
4458    mClient.clear();
4459}
4460
4461void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4462{
4463    sp<NotificationClient> keep(this);
4464    {
4465        mAudioFlinger->removeNotificationClient(mPid);
4466    }
4467}
4468
4469// ----------------------------------------------------------------------------
4470
4471AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4472    : BnAudioTrack(),
4473      mTrack(track)
4474{
4475}
4476
4477AudioFlinger::TrackHandle::~TrackHandle() {
4478    // just stop the track on deletion, associated resources
4479    // will be freed from the main thread once all pending buffers have
4480    // been played. Unless it's not in the active track list, in which
4481    // case we free everything now...
4482    mTrack->destroy();
4483}
4484
4485status_t AudioFlinger::TrackHandle::start() {
4486    return mTrack->start();
4487}
4488
4489void AudioFlinger::TrackHandle::stop() {
4490    mTrack->stop();
4491}
4492
4493void AudioFlinger::TrackHandle::flush() {
4494    mTrack->flush();
4495}
4496
4497void AudioFlinger::TrackHandle::mute(bool e) {
4498    mTrack->mute(e);
4499}
4500
4501void AudioFlinger::TrackHandle::pause() {
4502    mTrack->pause();
4503}
4504
4505void AudioFlinger::TrackHandle::setVolume(float left, float right) {
4506    mTrack->setVolume(left, right);
4507}
4508
4509sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4510    return mTrack->getCblk();
4511}
4512
4513status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4514{
4515    return mTrack->attachAuxEffect(EffectId);
4516}
4517
4518status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
4519                                                         sp<IMemory>* buffer) {
4520    if (!mTrack->isTimedTrack())
4521        return INVALID_OPERATION;
4522
4523    PlaybackThread::TimedTrack* tt =
4524            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
4525    return tt->allocateTimedBuffer(size, buffer);
4526}
4527
4528status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
4529                                                     int64_t pts) {
4530    if (!mTrack->isTimedTrack())
4531        return INVALID_OPERATION;
4532
4533    PlaybackThread::TimedTrack* tt =
4534            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
4535    return tt->queueTimedBuffer(buffer, pts);
4536}
4537
4538status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
4539    const LinearTransform& xform, int target) {
4540
4541    if (!mTrack->isTimedTrack())
4542        return INVALID_OPERATION;
4543
4544    PlaybackThread::TimedTrack* tt =
4545            reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
4546    return tt->setMediaTimeTransform(
4547        xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
4548}
4549
4550status_t AudioFlinger::TrackHandle::onTransact(
4551    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4552{
4553    return BnAudioTrack::onTransact(code, data, reply, flags);
4554}
4555
4556// ----------------------------------------------------------------------------
4557
4558sp<IAudioRecord> AudioFlinger::openRecord(
4559        pid_t pid,
4560        int input,
4561        uint32_t sampleRate,
4562        uint32_t format,
4563        uint32_t channelMask,
4564        int frameCount,
4565        uint32_t flags,
4566        int *sessionId,
4567        status_t *status)
4568{
4569    sp<RecordThread::RecordTrack> recordTrack;
4570    sp<RecordHandle> recordHandle;
4571    sp<Client> client;
4572    wp<Client> wclient;
4573    status_t lStatus;
4574    RecordThread *thread;
4575    size_t inFrameCount;
4576    int lSessionId;
4577
4578    // check calling permissions
4579    if (!recordingAllowed()) {
4580        lStatus = PERMISSION_DENIED;
4581        goto Exit;
4582    }
4583
4584    // add client to list
4585    { // scope for mLock
4586        Mutex::Autolock _l(mLock);
4587        thread = checkRecordThread_l(input);
4588        if (thread == NULL) {
4589            lStatus = BAD_VALUE;
4590            goto Exit;
4591        }
4592
4593        wclient = mClients.valueFor(pid);
4594        if (wclient != NULL) {
4595            client = wclient.promote();
4596        } else {
4597            client = new Client(this, pid);
4598            mClients.add(pid, client);
4599        }
4600
4601        // If no audio session id is provided, create one here
4602        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4603            lSessionId = *sessionId;
4604        } else {
4605            lSessionId = nextUniqueId();
4606            if (sessionId != NULL) {
4607                *sessionId = lSessionId;
4608            }
4609        }
4610        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4611        recordTrack = thread->createRecordTrack_l(client,
4612                                                sampleRate,
4613                                                format,
4614                                                channelMask,
4615                                                frameCount,
4616                                                flags,
4617                                                lSessionId,
4618                                                &lStatus);
4619    }
4620    if (lStatus != NO_ERROR) {
4621        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4622        // destructor is called by the TrackBase destructor with mLock held
4623        client.clear();
4624        recordTrack.clear();
4625        goto Exit;
4626    }
4627
4628    // return to handle to client
4629    recordHandle = new RecordHandle(recordTrack);
4630    lStatus = NO_ERROR;
4631
4632Exit:
4633    if (status) {
4634        *status = lStatus;
4635    }
4636    return recordHandle;
4637}
4638
4639// ----------------------------------------------------------------------------
4640
4641AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4642    : BnAudioRecord(),
4643    mRecordTrack(recordTrack)
4644{
4645}
4646
4647AudioFlinger::RecordHandle::~RecordHandle() {
4648    stop();
4649}
4650
4651status_t AudioFlinger::RecordHandle::start() {
4652    LOGV("RecordHandle::start()");
4653    return mRecordTrack->start();
4654}
4655
4656void AudioFlinger::RecordHandle::stop() {
4657    LOGV("RecordHandle::stop()");
4658    mRecordTrack->stop();
4659}
4660
4661sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4662    return mRecordTrack->getCblk();
4663}
4664
4665status_t AudioFlinger::RecordHandle::onTransact(
4666    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4667{
4668    return BnAudioRecord::onTransact(code, data, reply, flags);
4669}
4670
4671// ----------------------------------------------------------------------------
4672
4673AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4674                                         AudioStreamIn *input,
4675                                         uint32_t sampleRate,
4676                                         uint32_t channels,
4677                                         int id,
4678                                         uint32_t device) :
4679    ThreadBase(audioFlinger, id, device),
4680    mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
4681{
4682    mType = ThreadBase::RECORD;
4683
4684    snprintf(mName, kNameLength, "AudioIn_%d", id);
4685
4686    mReqChannelCount = popcount(channels);
4687    mReqSampleRate = sampleRate;
4688    readInputParameters();
4689}
4690
4691
4692AudioFlinger::RecordThread::~RecordThread()
4693{
4694    delete[] mRsmpInBuffer;
4695    if (mResampler != 0) {
4696        delete mResampler;
4697        delete[] mRsmpOutBuffer;
4698    }
4699}
4700
4701void AudioFlinger::RecordThread::onFirstRef()
4702{
4703    run(mName, PRIORITY_URGENT_AUDIO);
4704}
4705
4706status_t AudioFlinger::RecordThread::readyToRun()
4707{
4708    status_t status = initCheck();
4709    LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4710    return status;
4711}
4712
4713bool AudioFlinger::RecordThread::threadLoop()
4714{
4715    AudioBufferProvider::Buffer buffer;
4716    sp<RecordTrack> activeTrack;
4717    Vector< sp<EffectChain> > effectChains;
4718
4719    nsecs_t lastWarning = 0;
4720
4721    acquireWakeLock();
4722
4723    // start recording
4724    while (!exitPending()) {
4725
4726        processConfigEvents();
4727
4728        { // scope for mLock
4729            Mutex::Autolock _l(mLock);
4730            checkForNewParameters_l();
4731            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4732                if (!mStandby) {
4733                    mInput->stream->common.standby(&mInput->stream->common);
4734                    mStandby = true;
4735                }
4736
4737                if (exitPending()) break;
4738
4739                releaseWakeLock_l();
4740                LOGV("RecordThread: loop stopping");
4741                // go to sleep
4742                mWaitWorkCV.wait(mLock);
4743                LOGV("RecordThread: loop starting");
4744                acquireWakeLock_l();
4745                continue;
4746            }
4747            if (mActiveTrack != 0) {
4748                if (mActiveTrack->mState == TrackBase::PAUSING) {
4749                    if (!mStandby) {
4750                        mInput->stream->common.standby(&mInput->stream->common);
4751                        mStandby = true;
4752                    }
4753                    mActiveTrack.clear();
4754                    mStartStopCond.broadcast();
4755                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4756                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4757                        mActiveTrack.clear();
4758                        mStartStopCond.broadcast();
4759                    } else if (mBytesRead != 0) {
4760                        // record start succeeds only if first read from audio input
4761                        // succeeds
4762                        if (mBytesRead > 0) {
4763                            mActiveTrack->mState = TrackBase::ACTIVE;
4764                        } else {
4765                            mActiveTrack.clear();
4766                        }
4767                        mStartStopCond.broadcast();
4768                    }
4769                    mStandby = false;
4770                }
4771            }
4772            lockEffectChains_l(effectChains);
4773        }
4774
4775        if (mActiveTrack != 0) {
4776            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4777                mActiveTrack->mState != TrackBase::RESUMING) {
4778                unlockEffectChains(effectChains);
4779                usleep(kRecordThreadSleepUs);
4780                continue;
4781            }
4782            for (size_t i = 0; i < effectChains.size(); i ++) {
4783                effectChains[i]->process_l();
4784            }
4785
4786            buffer.frameCount = mFrameCount;
4787            if (LIKELY(mActiveTrack->getNextBuffer(
4788                    &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) {
4789                size_t framesOut = buffer.frameCount;
4790                if (mResampler == 0) {
4791                    // no resampling
4792                    while (framesOut) {
4793                        size_t framesIn = mFrameCount - mRsmpInIndex;
4794                        if (framesIn) {
4795                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4796                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4797                            if (framesIn > framesOut)
4798                                framesIn = framesOut;
4799                            mRsmpInIndex += framesIn;
4800                            framesOut -= framesIn;
4801                            if ((int)mChannelCount == mReqChannelCount ||
4802                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4803                                memcpy(dst, src, framesIn * mFrameSize);
4804                            } else {
4805                                int16_t *src16 = (int16_t *)src;
4806                                int16_t *dst16 = (int16_t *)dst;
4807                                if (mChannelCount == 1) {
4808                                    while (framesIn--) {
4809                                        *dst16++ = *src16;
4810                                        *dst16++ = *src16++;
4811                                    }
4812                                } else {
4813                                    while (framesIn--) {
4814                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4815                                        src16 += 2;
4816                                    }
4817                                }
4818                            }
4819                        }
4820                        if (framesOut && mFrameCount == mRsmpInIndex) {
4821                            if (framesOut == mFrameCount &&
4822                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4823                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4824                                framesOut = 0;
4825                            } else {
4826                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4827                                mRsmpInIndex = 0;
4828                            }
4829                            if (mBytesRead < 0) {
4830                                LOGE("Error reading audio input");
4831                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4832                                    // Force input into standby so that it tries to
4833                                    // recover at next read attempt
4834                                    mInput->stream->common.standby(&mInput->stream->common);
4835                                    usleep(kRecordThreadSleepUs);
4836                                }
4837                                mRsmpInIndex = mFrameCount;
4838                                framesOut = 0;
4839                                buffer.frameCount = 0;
4840                            }
4841                        }
4842                    }
4843                } else {
4844                    // resampling
4845
4846                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4847                    // alter output frame count as if we were expecting stereo samples
4848                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4849                        framesOut >>= 1;
4850                    }
4851                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4852                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4853                    // are 32 bit aligned which should be always true.
4854                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4855                        AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4856                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4857                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4858                        int16_t *dst = buffer.i16;
4859                        while (framesOut--) {
4860                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4861                            src += 2;
4862                        }
4863                    } else {
4864                        AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4865                    }
4866
4867                }
4868                mActiveTrack->releaseBuffer(&buffer);
4869                mActiveTrack->overflow();
4870            }
4871            // client isn't retrieving buffers fast enough
4872            else {
4873                if (!mActiveTrack->setOverflow()) {
4874                    nsecs_t now = systemTime();
4875                    if ((now - lastWarning) > kWarningThrottle) {
4876                        LOGW("RecordThread: buffer overflow");
4877                        lastWarning = now;
4878                    }
4879                }
4880                // Release the processor for a while before asking for a new buffer.
4881                // This will give the application more chance to read from the buffer and
4882                // clear the overflow.
4883                usleep(kRecordThreadSleepUs);
4884            }
4885        }
4886        // enable changes in effect chain
4887        unlockEffectChains(effectChains);
4888        effectChains.clear();
4889    }
4890
4891    if (!mStandby) {
4892        mInput->stream->common.standby(&mInput->stream->common);
4893    }
4894    mActiveTrack.clear();
4895
4896    mStartStopCond.broadcast();
4897
4898    releaseWakeLock();
4899
4900    LOGV("RecordThread %p exiting", this);
4901    return false;
4902}
4903
4904
4905sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4906        const sp<AudioFlinger::Client>& client,
4907        uint32_t sampleRate,
4908        int format,
4909        int channelMask,
4910        int frameCount,
4911        uint32_t flags,
4912        int sessionId,
4913        status_t *status)
4914{
4915    sp<RecordTrack> track;
4916    status_t lStatus;
4917
4918    lStatus = initCheck();
4919    if (lStatus != NO_ERROR) {
4920        LOGE("Audio driver not initialized.");
4921        goto Exit;
4922    }
4923
4924    { // scope for mLock
4925        Mutex::Autolock _l(mLock);
4926
4927        track = new RecordTrack(this, client, sampleRate,
4928                      format, channelMask, frameCount, flags, sessionId);
4929
4930        if (track->getCblk() == NULL) {
4931            lStatus = NO_MEMORY;
4932            goto Exit;
4933        }
4934
4935        mTrack = track.get();
4936        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4937        bool suspend = audio_is_bluetooth_sco_device(
4938                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrec();
4939        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4940        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4941    }
4942    lStatus = NO_ERROR;
4943
4944Exit:
4945    if (status) {
4946        *status = lStatus;
4947    }
4948    return track;
4949}
4950
4951status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4952{
4953    LOGV("RecordThread::start");
4954    sp <ThreadBase> strongMe = this;
4955    status_t status = NO_ERROR;
4956    {
4957        AutoMutex lock(&mLock);
4958        if (mActiveTrack != 0) {
4959            if (recordTrack != mActiveTrack.get()) {
4960                status = -EBUSY;
4961            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4962                mActiveTrack->mState = TrackBase::ACTIVE;
4963            }
4964            return status;
4965        }
4966
4967        recordTrack->mState = TrackBase::IDLE;
4968        mActiveTrack = recordTrack;
4969        mLock.unlock();
4970        status_t status = AudioSystem::startInput(mId);
4971        mLock.lock();
4972        if (status != NO_ERROR) {
4973            mActiveTrack.clear();
4974            return status;
4975        }
4976        mRsmpInIndex = mFrameCount;
4977        mBytesRead = 0;
4978        if (mResampler != NULL) {
4979            mResampler->reset();
4980        }
4981        mActiveTrack->mState = TrackBase::RESUMING;
4982        // signal thread to start
4983        LOGV("Signal record thread");
4984        mWaitWorkCV.signal();
4985        // do not wait for mStartStopCond if exiting
4986        if (mExiting) {
4987            mActiveTrack.clear();
4988            status = INVALID_OPERATION;
4989            goto startError;
4990        }
4991        mStartStopCond.wait(mLock);
4992        if (mActiveTrack == 0) {
4993            LOGV("Record failed to start");
4994            status = BAD_VALUE;
4995            goto startError;
4996        }
4997        LOGV("Record started OK");
4998        return status;
4999    }
5000startError:
5001    AudioSystem::stopInput(mId);
5002    return status;
5003}
5004
5005void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
5006    LOGV("RecordThread::stop");
5007    sp <ThreadBase> strongMe = this;
5008    {
5009        AutoMutex lock(&mLock);
5010        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
5011            mActiveTrack->mState = TrackBase::PAUSING;
5012            // do not wait for mStartStopCond if exiting
5013            if (mExiting) {
5014                return;
5015            }
5016            mStartStopCond.wait(mLock);
5017            // if we have been restarted, recordTrack == mActiveTrack.get() here
5018            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
5019                mLock.unlock();
5020                AudioSystem::stopInput(mId);
5021                mLock.lock();
5022                LOGV("Record stopped OK");
5023            }
5024        }
5025    }
5026}
5027
5028status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
5029{
5030    const size_t SIZE = 256;
5031    char buffer[SIZE];
5032    String8 result;
5033    pid_t pid = 0;
5034
5035    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
5036    result.append(buffer);
5037
5038    if (mActiveTrack != 0) {
5039        result.append("Active Track:\n");
5040        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
5041        mActiveTrack->dump(buffer, SIZE);
5042        result.append(buffer);
5043
5044        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
5045        result.append(buffer);
5046        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
5047        result.append(buffer);
5048        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
5049        result.append(buffer);
5050        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
5051        result.append(buffer);
5052        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
5053        result.append(buffer);
5054
5055
5056    } else {
5057        result.append("No record client\n");
5058    }
5059    write(fd, result.string(), result.size());
5060
5061    dumpBase(fd, args);
5062    dumpEffectChains(fd, args);
5063
5064    return NO_ERROR;
5065}
5066
5067status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
5068{
5069    size_t framesReq = buffer->frameCount;
5070    size_t framesReady = mFrameCount - mRsmpInIndex;
5071    int channelCount;
5072
5073    if (framesReady == 0) {
5074        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
5075        if (mBytesRead < 0) {
5076            LOGE("RecordThread::getNextBuffer() Error reading audio input");
5077            if (mActiveTrack->mState == TrackBase::ACTIVE) {
5078                // Force input into standby so that it tries to
5079                // recover at next read attempt
5080                mInput->stream->common.standby(&mInput->stream->common);
5081                usleep(kRecordThreadSleepUs);
5082            }
5083            buffer->raw = 0;
5084            buffer->frameCount = 0;
5085            return NOT_ENOUGH_DATA;
5086        }
5087        mRsmpInIndex = 0;
5088        framesReady = mFrameCount;
5089    }
5090
5091    if (framesReq > framesReady) {
5092        framesReq = framesReady;
5093    }
5094
5095    if (mChannelCount == 1 && mReqChannelCount == 2) {
5096        channelCount = 1;
5097    } else {
5098        channelCount = 2;
5099    }
5100    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
5101    buffer->frameCount = framesReq;
5102    return NO_ERROR;
5103}
5104
5105void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
5106{
5107    mRsmpInIndex += buffer->frameCount;
5108    buffer->frameCount = 0;
5109}
5110
5111bool AudioFlinger::RecordThread::checkForNewParameters_l()
5112{
5113    bool reconfig = false;
5114
5115    while (!mNewParameters.isEmpty()) {
5116        status_t status = NO_ERROR;
5117        String8 keyValuePair = mNewParameters[0];
5118        AudioParameter param = AudioParameter(keyValuePair);
5119        int value;
5120        int reqFormat = mFormat;
5121        int reqSamplingRate = mReqSampleRate;
5122        int reqChannelCount = mReqChannelCount;
5123
5124        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
5125            reqSamplingRate = value;
5126            reconfig = true;
5127        }
5128        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
5129            reqFormat = value;
5130            reconfig = true;
5131        }
5132        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
5133            reqChannelCount = popcount(value);
5134            reconfig = true;
5135        }
5136        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
5137            // do not accept frame count changes if tracks are open as the track buffer
5138            // size depends on frame count and correct behavior would not be garantied
5139            // if frame count is changed after track creation
5140            if (mActiveTrack != 0) {
5141                status = INVALID_OPERATION;
5142            } else {
5143                reconfig = true;
5144            }
5145        }
5146        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
5147            // forward device change to effects that have requested to be
5148            // aware of attached audio device.
5149            for (size_t i = 0; i < mEffectChains.size(); i++) {
5150                mEffectChains[i]->setDevice_l(value);
5151            }
5152            // store input device and output device but do not forward output device to audio HAL.
5153            // Note that status is ignored by the caller for output device
5154            // (see AudioFlinger::setParameters()
5155            if (value & AUDIO_DEVICE_OUT_ALL) {
5156                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
5157                status = BAD_VALUE;
5158            } else {
5159                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
5160                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
5161                if (mTrack != NULL) {
5162                    bool suspend = audio_is_bluetooth_sco_device(
5163                            (audio_devices_t)value) && mAudioFlinger->btNrec();
5164                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
5165                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
5166                }
5167            }
5168            mDevice |= (uint32_t)value;
5169        }
5170        if (status == NO_ERROR) {
5171            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
5172            if (status == INVALID_OPERATION) {
5173               mInput->stream->common.standby(&mInput->stream->common);
5174               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
5175            }
5176            if (reconfig) {
5177                if (status == BAD_VALUE &&
5178                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
5179                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
5180                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
5181                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
5182                    (reqChannelCount < 3)) {
5183                    status = NO_ERROR;
5184                }
5185                if (status == NO_ERROR) {
5186                    readInputParameters();
5187                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
5188                }
5189            }
5190        }
5191
5192        mNewParameters.removeAt(0);
5193
5194        mParamStatus = status;
5195        mParamCond.signal();
5196        mWaitWorkCV.wait(mLock);
5197    }
5198    return reconfig;
5199}
5200
5201String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
5202{
5203    char *s;
5204    String8 out_s8 = String8();
5205
5206    Mutex::Autolock _l(mLock);
5207    if (initCheck() != NO_ERROR) {
5208        return out_s8;
5209    }
5210
5211    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
5212    out_s8 = String8(s);
5213    free(s);
5214    return out_s8;
5215}
5216
5217void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
5218    AudioSystem::OutputDescriptor desc;
5219    void *param2 = 0;
5220
5221    switch (event) {
5222    case AudioSystem::INPUT_OPENED:
5223    case AudioSystem::INPUT_CONFIG_CHANGED:
5224        desc.channels = mChannelMask;
5225        desc.samplingRate = mSampleRate;
5226        desc.format = mFormat;
5227        desc.frameCount = mFrameCount;
5228        desc.latency = 0;
5229        param2 = &desc;
5230        break;
5231
5232    case AudioSystem::INPUT_CLOSED:
5233    default:
5234        break;
5235    }
5236    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
5237}
5238
5239void AudioFlinger::RecordThread::readInputParameters()
5240{
5241    if (mRsmpInBuffer) delete mRsmpInBuffer;
5242    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
5243    if (mResampler) delete mResampler;
5244    mResampler = 0;
5245
5246    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
5247    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
5248    mChannelCount = (uint16_t)popcount(mChannelMask);
5249    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
5250    mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
5251    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
5252    mFrameCount = mInputBytes / mFrameSize;
5253    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
5254
5255    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
5256    {
5257        int channelCount;
5258         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
5259         // stereo to mono post process as the resampler always outputs stereo.
5260        if (mChannelCount == 1 && mReqChannelCount == 2) {
5261            channelCount = 1;
5262        } else {
5263            channelCount = 2;
5264        }
5265        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
5266        mResampler->setSampleRate(mSampleRate);
5267        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
5268        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
5269
5270        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
5271        if (mChannelCount == 1 && mReqChannelCount == 1) {
5272            mFrameCount >>= 1;
5273        }
5274
5275    }
5276    mRsmpInIndex = mFrameCount;
5277}
5278
5279unsigned int AudioFlinger::RecordThread::getInputFramesLost()
5280{
5281    Mutex::Autolock _l(mLock);
5282    if (initCheck() != NO_ERROR) {
5283        return 0;
5284    }
5285
5286    return mInput->stream->get_input_frames_lost(mInput->stream);
5287}
5288
5289uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
5290{
5291    Mutex::Autolock _l(mLock);
5292    uint32_t result = 0;
5293    if (getEffectChain_l(sessionId) != 0) {
5294        result = EFFECT_SESSION;
5295    }
5296
5297    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
5298        result |= TRACK_SESSION;
5299    }
5300
5301    return result;
5302}
5303
5304AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
5305{
5306    Mutex::Autolock _l(mLock);
5307    return mTrack;
5308}
5309
5310AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
5311{
5312    Mutex::Autolock _l(mLock);
5313    return mInput;
5314}
5315
5316AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
5317{
5318    Mutex::Autolock _l(mLock);
5319    AudioStreamIn *input = mInput;
5320    mInput = NULL;
5321    return input;
5322}
5323
5324// this method must always be called either with ThreadBase mLock held or inside the thread loop
5325audio_stream_t* AudioFlinger::RecordThread::stream()
5326{
5327    if (mInput == NULL) {
5328        return NULL;
5329    }
5330    return &mInput->stream->common;
5331}
5332
5333
5334// ----------------------------------------------------------------------------
5335
5336int AudioFlinger::openOutput(uint32_t *pDevices,
5337                                uint32_t *pSamplingRate,
5338                                uint32_t *pFormat,
5339                                uint32_t *pChannels,
5340                                uint32_t *pLatencyMs,
5341                                uint32_t flags)
5342{
5343    status_t status;
5344    PlaybackThread *thread = NULL;
5345    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
5346    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5347    uint32_t format = pFormat ? *pFormat : 0;
5348    uint32_t channels = pChannels ? *pChannels : 0;
5349    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
5350    audio_stream_out_t *outStream;
5351    audio_hw_device_t *outHwDev;
5352
5353    LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
5354            pDevices ? *pDevices : 0,
5355            samplingRate,
5356            format,
5357            channels,
5358            flags);
5359
5360    if (pDevices == NULL || *pDevices == 0) {
5361        return 0;
5362    }
5363
5364    Mutex::Autolock _l(mLock);
5365
5366    outHwDev = findSuitableHwDev_l(*pDevices);
5367    if (outHwDev == NULL)
5368        return 0;
5369
5370    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
5371                                          &channels, &samplingRate, &outStream);
5372    LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
5373            outStream,
5374            samplingRate,
5375            format,
5376            channels,
5377            status);
5378
5379    mHardwareStatus = AUDIO_HW_IDLE;
5380    if (outStream != NULL) {
5381        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
5382        int id = nextUniqueId();
5383
5384        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
5385            (format != AUDIO_FORMAT_PCM_16_BIT) ||
5386            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
5387            thread = new DirectOutputThread(this, output, id, *pDevices);
5388            LOGV("openOutput() created direct output: ID %d thread %p", id, thread);
5389        } else {
5390            thread = new MixerThread(this, output, id, *pDevices);
5391            LOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
5392        }
5393        mPlaybackThreads.add(id, thread);
5394
5395        if (pSamplingRate) *pSamplingRate = samplingRate;
5396        if (pFormat) *pFormat = format;
5397        if (pChannels) *pChannels = channels;
5398        if (pLatencyMs) *pLatencyMs = thread->latency();
5399
5400        // notify client processes of the new output creation
5401        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
5402        return id;
5403    }
5404
5405    return 0;
5406}
5407
5408int AudioFlinger::openDuplicateOutput(int output1, int output2)
5409{
5410    Mutex::Autolock _l(mLock);
5411    MixerThread *thread1 = checkMixerThread_l(output1);
5412    MixerThread *thread2 = checkMixerThread_l(output2);
5413
5414    if (thread1 == NULL || thread2 == NULL) {
5415        LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
5416        return 0;
5417    }
5418
5419    int id = nextUniqueId();
5420    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
5421    thread->addOutputTrack(thread2);
5422    mPlaybackThreads.add(id, thread);
5423    // notify client processes of the new output creation
5424    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
5425    return id;
5426}
5427
5428status_t AudioFlinger::closeOutput(int output)
5429{
5430    // keep strong reference on the playback thread so that
5431    // it is not destroyed while exit() is executed
5432    sp <PlaybackThread> thread;
5433    {
5434        Mutex::Autolock _l(mLock);
5435        thread = checkPlaybackThread_l(output);
5436        if (thread == NULL) {
5437            return BAD_VALUE;
5438        }
5439
5440        LOGV("closeOutput() %d", output);
5441
5442        if (thread->type() == ThreadBase::MIXER) {
5443            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5444                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
5445                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
5446                    dupThread->removeOutputTrack((MixerThread *)thread.get());
5447                }
5448            }
5449        }
5450        void *param2 = 0;
5451        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
5452        mPlaybackThreads.removeItem(output);
5453    }
5454    thread->exit();
5455
5456    if (thread->type() != ThreadBase::DUPLICATING) {
5457        AudioStreamOut *out = thread->clearOutput();
5458        // from now on thread->mOutput is NULL
5459        out->hwDev->close_output_stream(out->hwDev, out->stream);
5460        delete out;
5461    }
5462    return NO_ERROR;
5463}
5464
5465status_t AudioFlinger::suspendOutput(int output)
5466{
5467    Mutex::Autolock _l(mLock);
5468    PlaybackThread *thread = checkPlaybackThread_l(output);
5469
5470    if (thread == NULL) {
5471        return BAD_VALUE;
5472    }
5473
5474    LOGV("suspendOutput() %d", output);
5475    thread->suspend();
5476
5477    return NO_ERROR;
5478}
5479
5480status_t AudioFlinger::restoreOutput(int output)
5481{
5482    Mutex::Autolock _l(mLock);
5483    PlaybackThread *thread = checkPlaybackThread_l(output);
5484
5485    if (thread == NULL) {
5486        return BAD_VALUE;
5487    }
5488
5489    LOGV("restoreOutput() %d", output);
5490
5491    thread->restore();
5492
5493    return NO_ERROR;
5494}
5495
5496int AudioFlinger::openInput(uint32_t *pDevices,
5497                                uint32_t *pSamplingRate,
5498                                uint32_t *pFormat,
5499                                uint32_t *pChannels,
5500                                uint32_t acoustics)
5501{
5502    status_t status;
5503    RecordThread *thread = NULL;
5504    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5505    uint32_t format = pFormat ? *pFormat : 0;
5506    uint32_t channels = pChannels ? *pChannels : 0;
5507    uint32_t reqSamplingRate = samplingRate;
5508    uint32_t reqFormat = format;
5509    uint32_t reqChannels = channels;
5510    audio_stream_in_t *inStream;
5511    audio_hw_device_t *inHwDev;
5512
5513    if (pDevices == NULL || *pDevices == 0) {
5514        return 0;
5515    }
5516
5517    Mutex::Autolock _l(mLock);
5518
5519    inHwDev = findSuitableHwDev_l(*pDevices);
5520    if (inHwDev == NULL)
5521        return 0;
5522
5523    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5524                                        &channels, &samplingRate,
5525                                        (audio_in_acoustics_t)acoustics,
5526                                        &inStream);
5527    LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5528            inStream,
5529            samplingRate,
5530            format,
5531            channels,
5532            acoustics,
5533            status);
5534
5535    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5536    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5537    // or stereo to mono conversions on 16 bit PCM inputs.
5538    if (inStream == NULL && status == BAD_VALUE &&
5539        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5540        (samplingRate <= 2 * reqSamplingRate) &&
5541        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5542        LOGV("openInput() reopening with proposed sampling rate and channels");
5543        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5544                                            &channels, &samplingRate,
5545                                            (audio_in_acoustics_t)acoustics,
5546                                            &inStream);
5547    }
5548
5549    if (inStream != NULL) {
5550        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5551
5552        int id = nextUniqueId();
5553        // Start record thread
5554        // RecorThread require both input and output device indication to forward to audio
5555        // pre processing modules
5556        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5557        thread = new RecordThread(this,
5558                                  input,
5559                                  reqSamplingRate,
5560                                  reqChannels,
5561                                  id,
5562                                  device);
5563        mRecordThreads.add(id, thread);
5564        LOGV("openInput() created record thread: ID %d thread %p", id, thread);
5565        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5566        if (pFormat) *pFormat = format;
5567        if (pChannels) *pChannels = reqChannels;
5568
5569        input->stream->common.standby(&input->stream->common);
5570
5571        // notify client processes of the new input creation
5572        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5573        return id;
5574    }
5575
5576    return 0;
5577}
5578
5579status_t AudioFlinger::closeInput(int input)
5580{
5581    // keep strong reference on the record thread so that
5582    // it is not destroyed while exit() is executed
5583    sp <RecordThread> thread;
5584    {
5585        Mutex::Autolock _l(mLock);
5586        thread = checkRecordThread_l(input);
5587        if (thread == NULL) {
5588            return BAD_VALUE;
5589        }
5590
5591        LOGV("closeInput() %d", input);
5592        void *param2 = 0;
5593        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5594        mRecordThreads.removeItem(input);
5595    }
5596    thread->exit();
5597
5598    AudioStreamIn *in = thread->clearInput();
5599    // from now on thread->mInput is NULL
5600    in->hwDev->close_input_stream(in->hwDev, in->stream);
5601    delete in;
5602
5603    return NO_ERROR;
5604}
5605
5606status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
5607{
5608    Mutex::Autolock _l(mLock);
5609    MixerThread *dstThread = checkMixerThread_l(output);
5610    if (dstThread == NULL) {
5611        LOGW("setStreamOutput() bad output id %d", output);
5612        return BAD_VALUE;
5613    }
5614
5615    LOGV("setStreamOutput() stream %d to output %d", stream, output);
5616    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5617
5618    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5619        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5620        if (thread != dstThread &&
5621            thread->type() != ThreadBase::DIRECT) {
5622            MixerThread *srcThread = (MixerThread *)thread;
5623            srcThread->invalidateTracks(stream);
5624        }
5625    }
5626
5627    return NO_ERROR;
5628}
5629
5630
5631int AudioFlinger::newAudioSessionId()
5632{
5633    return nextUniqueId();
5634}
5635
5636void AudioFlinger::acquireAudioSessionId(int audioSession)
5637{
5638    Mutex::Autolock _l(mLock);
5639    int caller = IPCThreadState::self()->getCallingPid();
5640    LOGV("acquiring %d from %d", audioSession, caller);
5641    int num = mAudioSessionRefs.size();
5642    for (int i = 0; i< num; i++) {
5643        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5644        if (ref->sessionid == audioSession && ref->pid == caller) {
5645            ref->cnt++;
5646            LOGV(" incremented refcount to %d", ref->cnt);
5647            return;
5648        }
5649    }
5650    AudioSessionRef *ref = new AudioSessionRef();
5651    ref->sessionid = audioSession;
5652    ref->pid = caller;
5653    ref->cnt = 1;
5654    mAudioSessionRefs.push(ref);
5655    LOGV(" added new entry for %d", ref->sessionid);
5656}
5657
5658void AudioFlinger::releaseAudioSessionId(int audioSession)
5659{
5660    Mutex::Autolock _l(mLock);
5661    int caller = IPCThreadState::self()->getCallingPid();
5662    LOGV("releasing %d from %d", audioSession, caller);
5663    int num = mAudioSessionRefs.size();
5664    for (int i = 0; i< num; i++) {
5665        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5666        if (ref->sessionid == audioSession && ref->pid == caller) {
5667            ref->cnt--;
5668            LOGV(" decremented refcount to %d", ref->cnt);
5669            if (ref->cnt == 0) {
5670                mAudioSessionRefs.removeAt(i);
5671                delete ref;
5672                purgeStaleEffects_l();
5673            }
5674            return;
5675        }
5676    }
5677    LOGW("session id %d not found for pid %d", audioSession, caller);
5678}
5679
5680void AudioFlinger::purgeStaleEffects_l() {
5681
5682    LOGV("purging stale effects");
5683
5684    Vector< sp<EffectChain> > chains;
5685
5686    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5687        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5688        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5689            sp<EffectChain> ec = t->mEffectChains[j];
5690            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5691                chains.push(ec);
5692            }
5693        }
5694    }
5695    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5696        sp<RecordThread> t = mRecordThreads.valueAt(i);
5697        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5698            sp<EffectChain> ec = t->mEffectChains[j];
5699            chains.push(ec);
5700        }
5701    }
5702
5703    for (size_t i = 0; i < chains.size(); i++) {
5704        sp<EffectChain> ec = chains[i];
5705        int sessionid = ec->sessionId();
5706        sp<ThreadBase> t = ec->mThread.promote();
5707        if (t == 0) {
5708            continue;
5709        }
5710        size_t numsessionrefs = mAudioSessionRefs.size();
5711        bool found = false;
5712        for (size_t k = 0; k < numsessionrefs; k++) {
5713            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5714            if (ref->sessionid == sessionid) {
5715                LOGV(" session %d still exists for %d with %d refs",
5716                     sessionid, ref->pid, ref->cnt);
5717                found = true;
5718                break;
5719            }
5720        }
5721        if (!found) {
5722            // remove all effects from the chain
5723            while (ec->mEffects.size()) {
5724                sp<EffectModule> effect = ec->mEffects[0];
5725                effect->unPin();
5726                Mutex::Autolock _l (t->mLock);
5727                t->removeEffect_l(effect);
5728                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5729                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5730                    if (handle != 0) {
5731                        handle->mEffect.clear();
5732                    }
5733                }
5734                AudioSystem::unregisterEffect(effect->id());
5735            }
5736        }
5737    }
5738    return;
5739}
5740
5741// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5742AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5743{
5744    PlaybackThread *thread = NULL;
5745    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5746        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5747    }
5748    return thread;
5749}
5750
5751// checkMixerThread_l() must be called with AudioFlinger::mLock held
5752AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5753{
5754    PlaybackThread *thread = checkPlaybackThread_l(output);
5755    if (thread != NULL) {
5756        if (thread->type() == ThreadBase::DIRECT) {
5757            thread = NULL;
5758        }
5759    }
5760    return (MixerThread *)thread;
5761}
5762
5763// checkRecordThread_l() must be called with AudioFlinger::mLock held
5764AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5765{
5766    RecordThread *thread = NULL;
5767    if (mRecordThreads.indexOfKey(input) >= 0) {
5768        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5769    }
5770    return thread;
5771}
5772
5773uint32_t AudioFlinger::nextUniqueId()
5774{
5775    return android_atomic_inc(&mNextUniqueId);
5776}
5777
5778AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5779{
5780    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5781        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5782        AudioStreamOut *output = thread->getOutput();
5783        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5784            return thread;
5785        }
5786    }
5787    return NULL;
5788}
5789
5790uint32_t AudioFlinger::primaryOutputDevice_l()
5791{
5792    PlaybackThread *thread = primaryPlaybackThread_l();
5793
5794    if (thread == NULL) {
5795        return 0;
5796    }
5797
5798    return thread->device();
5799}
5800
5801
5802// ----------------------------------------------------------------------------
5803//  Effect management
5804// ----------------------------------------------------------------------------
5805
5806
5807status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5808{
5809    Mutex::Autolock _l(mLock);
5810    return EffectQueryNumberEffects(numEffects);
5811}
5812
5813status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5814{
5815    Mutex::Autolock _l(mLock);
5816    return EffectQueryEffect(index, descriptor);
5817}
5818
5819status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5820{
5821    Mutex::Autolock _l(mLock);
5822    return EffectGetDescriptor(pUuid, descriptor);
5823}
5824
5825
5826sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5827        effect_descriptor_t *pDesc,
5828        const sp<IEffectClient>& effectClient,
5829        int32_t priority,
5830        int io,
5831        int sessionId,
5832        status_t *status,
5833        int *id,
5834        int *enabled)
5835{
5836    status_t lStatus = NO_ERROR;
5837    sp<EffectHandle> handle;
5838    effect_descriptor_t desc;
5839    sp<Client> client;
5840    wp<Client> wclient;
5841
5842    LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5843            pid, effectClient.get(), priority, sessionId, io);
5844
5845    if (pDesc == NULL) {
5846        lStatus = BAD_VALUE;
5847        goto Exit;
5848    }
5849
5850    // check audio settings permission for global effects
5851    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5852        lStatus = PERMISSION_DENIED;
5853        goto Exit;
5854    }
5855
5856    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5857    // that can only be created by audio policy manager (running in same process)
5858    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5859        lStatus = PERMISSION_DENIED;
5860        goto Exit;
5861    }
5862
5863    if (io == 0) {
5864        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5865            // output must be specified by AudioPolicyManager when using session
5866            // AUDIO_SESSION_OUTPUT_STAGE
5867            lStatus = BAD_VALUE;
5868            goto Exit;
5869        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5870            // if the output returned by getOutputForEffect() is removed before we lock the
5871            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5872            // and we will exit safely
5873            io = AudioSystem::getOutputForEffect(&desc);
5874        }
5875    }
5876
5877    {
5878        Mutex::Autolock _l(mLock);
5879
5880
5881        if (!EffectIsNullUuid(&pDesc->uuid)) {
5882            // if uuid is specified, request effect descriptor
5883            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5884            if (lStatus < 0) {
5885                LOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5886                goto Exit;
5887            }
5888        } else {
5889            // if uuid is not specified, look for an available implementation
5890            // of the required type in effect factory
5891            if (EffectIsNullUuid(&pDesc->type)) {
5892                LOGW("createEffect() no effect type");
5893                lStatus = BAD_VALUE;
5894                goto Exit;
5895            }
5896            uint32_t numEffects = 0;
5897            effect_descriptor_t d;
5898            d.flags = 0; // prevent compiler warning
5899            bool found = false;
5900
5901            lStatus = EffectQueryNumberEffects(&numEffects);
5902            if (lStatus < 0) {
5903                LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5904                goto Exit;
5905            }
5906            for (uint32_t i = 0; i < numEffects; i++) {
5907                lStatus = EffectQueryEffect(i, &desc);
5908                if (lStatus < 0) {
5909                    LOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5910                    continue;
5911                }
5912                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5913                    // If matching type found save effect descriptor. If the session is
5914                    // 0 and the effect is not auxiliary, continue enumeration in case
5915                    // an auxiliary version of this effect type is available
5916                    found = true;
5917                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5918                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5919                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5920                        break;
5921                    }
5922                }
5923            }
5924            if (!found) {
5925                lStatus = BAD_VALUE;
5926                LOGW("createEffect() effect not found");
5927                goto Exit;
5928            }
5929            // For same effect type, chose auxiliary version over insert version if
5930            // connect to output mix (Compliance to OpenSL ES)
5931            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5932                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5933                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5934            }
5935        }
5936
5937        // Do not allow auxiliary effects on a session different from 0 (output mix)
5938        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5939             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5940            lStatus = INVALID_OPERATION;
5941            goto Exit;
5942        }
5943
5944        // check recording permission for visualizer
5945        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5946            !recordingAllowed()) {
5947            lStatus = PERMISSION_DENIED;
5948            goto Exit;
5949        }
5950
5951        // return effect descriptor
5952        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5953
5954        // If output is not specified try to find a matching audio session ID in one of the
5955        // output threads.
5956        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5957        // because of code checking output when entering the function.
5958        // Note: io is never 0 when creating an effect on an input
5959        if (io == 0) {
5960             // look for the thread where the specified audio session is present
5961            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5962                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5963                    io = mPlaybackThreads.keyAt(i);
5964                    break;
5965                }
5966            }
5967            if (io == 0) {
5968               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5969                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5970                       io = mRecordThreads.keyAt(i);
5971                       break;
5972                   }
5973               }
5974            }
5975            // If no output thread contains the requested session ID, default to
5976            // first output. The effect chain will be moved to the correct output
5977            // thread when a track with the same session ID is created
5978            if (io == 0 && mPlaybackThreads.size()) {
5979                io = mPlaybackThreads.keyAt(0);
5980            }
5981            LOGV("createEffect() got io %d for effect %s", io, desc.name);
5982        }
5983        ThreadBase *thread = checkRecordThread_l(io);
5984        if (thread == NULL) {
5985            thread = checkPlaybackThread_l(io);
5986            if (thread == NULL) {
5987                LOGE("createEffect() unknown output thread");
5988                lStatus = BAD_VALUE;
5989                goto Exit;
5990            }
5991        }
5992
5993        wclient = mClients.valueFor(pid);
5994
5995        if (wclient != NULL) {
5996            client = wclient.promote();
5997        } else {
5998            client = new Client(this, pid);
5999            mClients.add(pid, client);
6000        }
6001
6002        // create effect on selected output thread
6003        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
6004                &desc, enabled, &lStatus);
6005        if (handle != 0 && id != NULL) {
6006            *id = handle->id();
6007        }
6008    }
6009
6010Exit:
6011    if(status) {
6012        *status = lStatus;
6013    }
6014    return handle;
6015}
6016
6017status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
6018{
6019    LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
6020            sessionId, srcOutput, dstOutput);
6021    Mutex::Autolock _l(mLock);
6022    if (srcOutput == dstOutput) {
6023        LOGW("moveEffects() same dst and src outputs %d", dstOutput);
6024        return NO_ERROR;
6025    }
6026    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
6027    if (srcThread == NULL) {
6028        LOGW("moveEffects() bad srcOutput %d", srcOutput);
6029        return BAD_VALUE;
6030    }
6031    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
6032    if (dstThread == NULL) {
6033        LOGW("moveEffects() bad dstOutput %d", dstOutput);
6034        return BAD_VALUE;
6035    }
6036
6037    Mutex::Autolock _dl(dstThread->mLock);
6038    Mutex::Autolock _sl(srcThread->mLock);
6039    moveEffectChain_l(sessionId, srcThread, dstThread, false);
6040
6041    return NO_ERROR;
6042}
6043
6044// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
6045status_t AudioFlinger::moveEffectChain_l(int sessionId,
6046                                   AudioFlinger::PlaybackThread *srcThread,
6047                                   AudioFlinger::PlaybackThread *dstThread,
6048                                   bool reRegister)
6049{
6050    LOGV("moveEffectChain_l() session %d from thread %p to thread %p",
6051            sessionId, srcThread, dstThread);
6052
6053    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
6054    if (chain == 0) {
6055        LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
6056                sessionId, srcThread);
6057        return INVALID_OPERATION;
6058    }
6059
6060    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
6061    // so that a new chain is created with correct parameters when first effect is added. This is
6062    // otherwise unecessary as removeEffect_l() will remove the chain when last effect is
6063    // removed.
6064    srcThread->removeEffectChain_l(chain);
6065
6066    // transfer all effects one by one so that new effect chain is created on new thread with
6067    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
6068    int dstOutput = dstThread->id();
6069    sp<EffectChain> dstChain;
6070    uint32_t strategy = 0; // prevent compiler warning
6071    sp<EffectModule> effect = chain->getEffectFromId_l(0);
6072    while (effect != 0) {
6073        srcThread->removeEffect_l(effect);
6074        dstThread->addEffect_l(effect);
6075        // if the move request is not received from audio policy manager, the effect must be
6076        // re-registered with the new strategy and output
6077        if (dstChain == 0) {
6078            dstChain = effect->chain().promote();
6079            if (dstChain == 0) {
6080                LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
6081                srcThread->addEffect_l(effect);
6082                return NO_INIT;
6083            }
6084            strategy = dstChain->strategy();
6085        }
6086        if (reRegister) {
6087            AudioSystem::unregisterEffect(effect->id());
6088            AudioSystem::registerEffect(&effect->desc(),
6089                                        dstOutput,
6090                                        strategy,
6091                                        sessionId,
6092                                        effect->id());
6093        }
6094        effect = chain->getEffectFromId_l(0);
6095    }
6096
6097    return NO_ERROR;
6098}
6099
6100
6101// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
6102sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
6103        const sp<AudioFlinger::Client>& client,
6104        const sp<IEffectClient>& effectClient,
6105        int32_t priority,
6106        int sessionId,
6107        effect_descriptor_t *desc,
6108        int *enabled,
6109        status_t *status
6110        )
6111{
6112    sp<EffectModule> effect;
6113    sp<EffectHandle> handle;
6114    status_t lStatus;
6115    sp<EffectChain> chain;
6116    bool chainCreated = false;
6117    bool effectCreated = false;
6118    bool effectRegistered = false;
6119
6120    lStatus = initCheck();
6121    if (lStatus != NO_ERROR) {
6122        LOGW("createEffect_l() Audio driver not initialized.");
6123        goto Exit;
6124    }
6125
6126    // Do not allow effects with session ID 0 on direct output or duplicating threads
6127    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
6128    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
6129        LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
6130                desc->name, sessionId);
6131        lStatus = BAD_VALUE;
6132        goto Exit;
6133    }
6134    // Only Pre processor effects are allowed on input threads and only on input threads
6135    if ((mType == RECORD &&
6136            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
6137            (mType != RECORD &&
6138                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
6139        LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
6140                desc->name, desc->flags, mType);
6141        lStatus = BAD_VALUE;
6142        goto Exit;
6143    }
6144
6145    LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
6146
6147    { // scope for mLock
6148        Mutex::Autolock _l(mLock);
6149
6150        // check for existing effect chain with the requested audio session
6151        chain = getEffectChain_l(sessionId);
6152        if (chain == 0) {
6153            // create a new chain for this session
6154            LOGV("createEffect_l() new effect chain for session %d", sessionId);
6155            chain = new EffectChain(this, sessionId);
6156            addEffectChain_l(chain);
6157            chain->setStrategy(getStrategyForSession_l(sessionId));
6158            chainCreated = true;
6159        } else {
6160            effect = chain->getEffectFromDesc_l(desc);
6161        }
6162
6163        LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
6164
6165        if (effect == 0) {
6166            int id = mAudioFlinger->nextUniqueId();
6167            // Check CPU and memory usage
6168            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
6169            if (lStatus != NO_ERROR) {
6170                goto Exit;
6171            }
6172            effectRegistered = true;
6173            // create a new effect module if none present in the chain
6174            effect = new EffectModule(this, chain, desc, id, sessionId);
6175            lStatus = effect->status();
6176            if (lStatus != NO_ERROR) {
6177                goto Exit;
6178            }
6179            lStatus = chain->addEffect_l(effect);
6180            if (lStatus != NO_ERROR) {
6181                goto Exit;
6182            }
6183            effectCreated = true;
6184
6185            effect->setDevice(mDevice);
6186            effect->setMode(mAudioFlinger->getMode());
6187        }
6188        // create effect handle and connect it to effect module
6189        handle = new EffectHandle(effect, client, effectClient, priority);
6190        lStatus = effect->addHandle(handle);
6191        if (enabled) {
6192            *enabled = (int)effect->isEnabled();
6193        }
6194    }
6195
6196Exit:
6197    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
6198        Mutex::Autolock _l(mLock);
6199        if (effectCreated) {
6200            chain->removeEffect_l(effect);
6201        }
6202        if (effectRegistered) {
6203            AudioSystem::unregisterEffect(effect->id());
6204        }
6205        if (chainCreated) {
6206            removeEffectChain_l(chain);
6207        }
6208        handle.clear();
6209    }
6210
6211    if(status) {
6212        *status = lStatus;
6213    }
6214    return handle;
6215}
6216
6217sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
6218{
6219    sp<EffectModule> effect;
6220
6221    sp<EffectChain> chain = getEffectChain_l(sessionId);
6222    if (chain != 0) {
6223        effect = chain->getEffectFromId_l(effectId);
6224    }
6225    return effect;
6226}
6227
6228// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
6229// PlaybackThread::mLock held
6230status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
6231{
6232    // check for existing effect chain with the requested audio session
6233    int sessionId = effect->sessionId();
6234    sp<EffectChain> chain = getEffectChain_l(sessionId);
6235    bool chainCreated = false;
6236
6237    if (chain == 0) {
6238        // create a new chain for this session
6239        LOGV("addEffect_l() new effect chain for session %d", sessionId);
6240        chain = new EffectChain(this, sessionId);
6241        addEffectChain_l(chain);
6242        chain->setStrategy(getStrategyForSession_l(sessionId));
6243        chainCreated = true;
6244    }
6245    LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
6246
6247    if (chain->getEffectFromId_l(effect->id()) != 0) {
6248        LOGW("addEffect_l() %p effect %s already present in chain %p",
6249                this, effect->desc().name, chain.get());
6250        return BAD_VALUE;
6251    }
6252
6253    status_t status = chain->addEffect_l(effect);
6254    if (status != NO_ERROR) {
6255        if (chainCreated) {
6256            removeEffectChain_l(chain);
6257        }
6258        return status;
6259    }
6260
6261    effect->setDevice(mDevice);
6262    effect->setMode(mAudioFlinger->getMode());
6263    return NO_ERROR;
6264}
6265
6266void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
6267
6268    LOGV("removeEffect_l() %p effect %p", this, effect.get());
6269    effect_descriptor_t desc = effect->desc();
6270    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6271        detachAuxEffect_l(effect->id());
6272    }
6273
6274    sp<EffectChain> chain = effect->chain().promote();
6275    if (chain != 0) {
6276        // remove effect chain if removing last effect
6277        if (chain->removeEffect_l(effect) == 0) {
6278            removeEffectChain_l(chain);
6279        }
6280    } else {
6281        LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
6282    }
6283}
6284
6285void AudioFlinger::ThreadBase::lockEffectChains_l(
6286        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
6287{
6288    effectChains = mEffectChains;
6289    for (size_t i = 0; i < mEffectChains.size(); i++) {
6290        mEffectChains[i]->lock();
6291    }
6292}
6293
6294void AudioFlinger::ThreadBase::unlockEffectChains(
6295        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
6296{
6297    for (size_t i = 0; i < effectChains.size(); i++) {
6298        effectChains[i]->unlock();
6299    }
6300}
6301
6302sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
6303{
6304    Mutex::Autolock _l(mLock);
6305    return getEffectChain_l(sessionId);
6306}
6307
6308sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
6309{
6310    sp<EffectChain> chain;
6311
6312    size_t size = mEffectChains.size();
6313    for (size_t i = 0; i < size; i++) {
6314        if (mEffectChains[i]->sessionId() == sessionId) {
6315            chain = mEffectChains[i];
6316            break;
6317        }
6318    }
6319    return chain;
6320}
6321
6322void AudioFlinger::ThreadBase::setMode(uint32_t mode)
6323{
6324    Mutex::Autolock _l(mLock);
6325    size_t size = mEffectChains.size();
6326    for (size_t i = 0; i < size; i++) {
6327        mEffectChains[i]->setMode_l(mode);
6328    }
6329}
6330
6331void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
6332                                                    const wp<EffectHandle>& handle,
6333                                                    bool unpiniflast) {
6334
6335    Mutex::Autolock _l(mLock);
6336    LOGV("disconnectEffect() %p effect %p", this, effect.get());
6337    // delete the effect module if removing last handle on it
6338    if (effect->removeHandle(handle) == 0) {
6339        if (!effect->isPinned() || unpiniflast) {
6340            removeEffect_l(effect);
6341            AudioSystem::unregisterEffect(effect->id());
6342        }
6343    }
6344}
6345
6346status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
6347{
6348    int session = chain->sessionId();
6349    int16_t *buffer = mMixBuffer;
6350    bool ownsBuffer = false;
6351
6352    LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
6353    if (session > 0) {
6354        // Only one effect chain can be present in direct output thread and it uses
6355        // the mix buffer as input
6356        if (mType != DIRECT) {
6357            size_t numSamples = mFrameCount * mChannelCount;
6358            buffer = new int16_t[numSamples];
6359            memset(buffer, 0, numSamples * sizeof(int16_t));
6360            LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
6361            ownsBuffer = true;
6362        }
6363
6364        // Attach all tracks with same session ID to this chain.
6365        for (size_t i = 0; i < mTracks.size(); ++i) {
6366            sp<Track> track = mTracks[i];
6367            if (session == track->sessionId()) {
6368                LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
6369                track->setMainBuffer(buffer);
6370                chain->incTrackCnt();
6371            }
6372        }
6373
6374        // indicate all active tracks in the chain
6375        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
6376            sp<Track> track = mActiveTracks[i].promote();
6377            if (track == 0) continue;
6378            if (session == track->sessionId()) {
6379                LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
6380                chain->incActiveTrackCnt();
6381            }
6382        }
6383    }
6384
6385    chain->setInBuffer(buffer, ownsBuffer);
6386    chain->setOutBuffer(mMixBuffer);
6387    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
6388    // chains list in order to be processed last as it contains output stage effects
6389    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
6390    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
6391    // after track specific effects and before output stage
6392    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
6393    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
6394    // Effect chain for other sessions are inserted at beginning of effect
6395    // chains list to be processed before output mix effects. Relative order between other
6396    // sessions is not important
6397    size_t size = mEffectChains.size();
6398    size_t i = 0;
6399    for (i = 0; i < size; i++) {
6400        if (mEffectChains[i]->sessionId() < session) break;
6401    }
6402    mEffectChains.insertAt(chain, i);
6403    checkSuspendOnAddEffectChain_l(chain);
6404
6405    return NO_ERROR;
6406}
6407
6408size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
6409{
6410    int session = chain->sessionId();
6411
6412    LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
6413
6414    for (size_t i = 0; i < mEffectChains.size(); i++) {
6415        if (chain == mEffectChains[i]) {
6416            mEffectChains.removeAt(i);
6417            // detach all active tracks from the chain
6418            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
6419                sp<Track> track = mActiveTracks[i].promote();
6420                if (track == 0) continue;
6421                if (session == track->sessionId()) {
6422                    LOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
6423                            chain.get(), session);
6424                    chain->decActiveTrackCnt();
6425                }
6426            }
6427
6428            // detach all tracks with same session ID from this chain
6429            for (size_t i = 0; i < mTracks.size(); ++i) {
6430                sp<Track> track = mTracks[i];
6431                if (session == track->sessionId()) {
6432                    track->setMainBuffer(mMixBuffer);
6433                    chain->decTrackCnt();
6434                }
6435            }
6436            break;
6437        }
6438    }
6439    return mEffectChains.size();
6440}
6441
6442status_t AudioFlinger::PlaybackThread::attachAuxEffect(
6443        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6444{
6445    Mutex::Autolock _l(mLock);
6446    return attachAuxEffect_l(track, EffectId);
6447}
6448
6449status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6450        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6451{
6452    status_t status = NO_ERROR;
6453
6454    if (EffectId == 0) {
6455        track->setAuxBuffer(0, NULL);
6456    } else {
6457        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6458        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6459        if (effect != 0) {
6460            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6461                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6462            } else {
6463                status = INVALID_OPERATION;
6464            }
6465        } else {
6466            status = BAD_VALUE;
6467        }
6468    }
6469    return status;
6470}
6471
6472void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6473{
6474     for (size_t i = 0; i < mTracks.size(); ++i) {
6475        sp<Track> track = mTracks[i];
6476        if (track->auxEffectId() == effectId) {
6477            attachAuxEffect_l(track, 0);
6478        }
6479    }
6480}
6481
6482status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6483{
6484    // only one chain per input thread
6485    if (mEffectChains.size() != 0) {
6486        return INVALID_OPERATION;
6487    }
6488    LOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6489
6490    chain->setInBuffer(NULL);
6491    chain->setOutBuffer(NULL);
6492
6493    checkSuspendOnAddEffectChain_l(chain);
6494
6495    mEffectChains.add(chain);
6496
6497    return NO_ERROR;
6498}
6499
6500size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6501{
6502    LOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6503    LOGW_IF(mEffectChains.size() != 1,
6504            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6505            chain.get(), mEffectChains.size(), this);
6506    if (mEffectChains.size() == 1) {
6507        mEffectChains.removeAt(0);
6508    }
6509    return 0;
6510}
6511
6512// ----------------------------------------------------------------------------
6513//  EffectModule implementation
6514// ----------------------------------------------------------------------------
6515
6516#undef LOG_TAG
6517#define LOG_TAG "AudioFlinger::EffectModule"
6518
6519AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6520                                        const wp<AudioFlinger::EffectChain>& chain,
6521                                        effect_descriptor_t *desc,
6522                                        int id,
6523                                        int sessionId)
6524    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6525      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6526{
6527    LOGV("Constructor %p", this);
6528    int lStatus;
6529    sp<ThreadBase> thread = mThread.promote();
6530    if (thread == 0) {
6531        return;
6532    }
6533
6534    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6535
6536    // create effect engine from effect factory
6537    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6538
6539    if (mStatus != NO_ERROR) {
6540        return;
6541    }
6542    lStatus = init();
6543    if (lStatus < 0) {
6544        mStatus = lStatus;
6545        goto Error;
6546    }
6547
6548    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6549        mPinned = true;
6550    }
6551    LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6552    return;
6553Error:
6554    EffectRelease(mEffectInterface);
6555    mEffectInterface = NULL;
6556    LOGV("Constructor Error %d", mStatus);
6557}
6558
6559AudioFlinger::EffectModule::~EffectModule()
6560{
6561    LOGV("Destructor %p", this);
6562    if (mEffectInterface != NULL) {
6563        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6564                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6565            sp<ThreadBase> thread = mThread.promote();
6566            if (thread != 0) {
6567                audio_stream_t *stream = thread->stream();
6568                if (stream != NULL) {
6569                    stream->remove_audio_effect(stream, mEffectInterface);
6570                }
6571            }
6572        }
6573        // release effect engine
6574        EffectRelease(mEffectInterface);
6575    }
6576}
6577
6578status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6579{
6580    status_t status;
6581
6582    Mutex::Autolock _l(mLock);
6583    // First handle in mHandles has highest priority and controls the effect module
6584    int priority = handle->priority();
6585    size_t size = mHandles.size();
6586    sp<EffectHandle> h;
6587    size_t i;
6588    for (i = 0; i < size; i++) {
6589        h = mHandles[i].promote();
6590        if (h == 0) continue;
6591        if (h->priority() <= priority) break;
6592    }
6593    // if inserted in first place, move effect control from previous owner to this handle
6594    if (i == 0) {
6595        bool enabled = false;
6596        if (h != 0) {
6597            enabled = h->enabled();
6598            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6599        }
6600        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6601        status = NO_ERROR;
6602    } else {
6603        status = ALREADY_EXISTS;
6604    }
6605    LOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6606    mHandles.insertAt(handle, i);
6607    return status;
6608}
6609
6610size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6611{
6612    Mutex::Autolock _l(mLock);
6613    size_t size = mHandles.size();
6614    size_t i;
6615    for (i = 0; i < size; i++) {
6616        if (mHandles[i] == handle) break;
6617    }
6618    if (i == size) {
6619        return size;
6620    }
6621    LOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6622
6623    bool enabled = false;
6624    EffectHandle *hdl = handle.unsafe_get();
6625    if (hdl) {
6626        LOGV("removeHandle() unsafe_get OK");
6627        enabled = hdl->enabled();
6628    }
6629    mHandles.removeAt(i);
6630    size = mHandles.size();
6631    // if removed from first place, move effect control from this handle to next in line
6632    if (i == 0 && size != 0) {
6633        sp<EffectHandle> h = mHandles[0].promote();
6634        if (h != 0) {
6635            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6636        }
6637    }
6638
6639    // Prevent calls to process() and other functions on effect interface from now on.
6640    // The effect engine will be released by the destructor when the last strong reference on
6641    // this object is released which can happen after next process is called.
6642    if (size == 0 && !mPinned) {
6643        mState = DESTROYED;
6644    }
6645
6646    return size;
6647}
6648
6649sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6650{
6651    Mutex::Autolock _l(mLock);
6652    sp<EffectHandle> handle;
6653    if (mHandles.size() != 0) {
6654        handle = mHandles[0].promote();
6655    }
6656    return handle;
6657}
6658
6659void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6660{
6661    LOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6662    // keep a strong reference on this EffectModule to avoid calling the
6663    // destructor before we exit
6664    sp<EffectModule> keep(this);
6665    {
6666        sp<ThreadBase> thread = mThread.promote();
6667        if (thread != 0) {
6668            thread->disconnectEffect(keep, handle, unpiniflast);
6669        }
6670    }
6671}
6672
6673void AudioFlinger::EffectModule::updateState() {
6674    Mutex::Autolock _l(mLock);
6675
6676    switch (mState) {
6677    case RESTART:
6678        reset_l();
6679        // FALL THROUGH
6680
6681    case STARTING:
6682        // clear auxiliary effect input buffer for next accumulation
6683        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6684            memset(mConfig.inputCfg.buffer.raw,
6685                   0,
6686                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6687        }
6688        start_l();
6689        mState = ACTIVE;
6690        break;
6691    case STOPPING:
6692        stop_l();
6693        mDisableWaitCnt = mMaxDisableWaitCnt;
6694        mState = STOPPED;
6695        break;
6696    case STOPPED:
6697        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6698        // turn off sequence.
6699        if (--mDisableWaitCnt == 0) {
6700            reset_l();
6701            mState = IDLE;
6702        }
6703        break;
6704    default: //IDLE , ACTIVE, DESTROYED
6705        break;
6706    }
6707}
6708
6709void AudioFlinger::EffectModule::process()
6710{
6711    Mutex::Autolock _l(mLock);
6712
6713    if (mState == DESTROYED || mEffectInterface == NULL ||
6714            mConfig.inputCfg.buffer.raw == NULL ||
6715            mConfig.outputCfg.buffer.raw == NULL) {
6716        return;
6717    }
6718
6719    if (isProcessEnabled()) {
6720        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6721        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6722            AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
6723                                        mConfig.inputCfg.buffer.s32,
6724                                        mConfig.inputCfg.buffer.frameCount/2);
6725        }
6726
6727        // do the actual processing in the effect engine
6728        int ret = (*mEffectInterface)->process(mEffectInterface,
6729                                               &mConfig.inputCfg.buffer,
6730                                               &mConfig.outputCfg.buffer);
6731
6732        // force transition to IDLE state when engine is ready
6733        if (mState == STOPPED && ret == -ENODATA) {
6734            mDisableWaitCnt = 1;
6735        }
6736
6737        // clear auxiliary effect input buffer for next accumulation
6738        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6739            memset(mConfig.inputCfg.buffer.raw, 0,
6740                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6741        }
6742    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6743                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6744        // If an insert effect is idle and input buffer is different from output buffer,
6745        // accumulate input onto output
6746        sp<EffectChain> chain = mChain.promote();
6747        if (chain != 0 && chain->activeTrackCnt() != 0) {
6748            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6749            int16_t *in = mConfig.inputCfg.buffer.s16;
6750            int16_t *out = mConfig.outputCfg.buffer.s16;
6751            for (size_t i = 0; i < frameCnt; i++) {
6752                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6753            }
6754        }
6755    }
6756}
6757
6758void AudioFlinger::EffectModule::reset_l()
6759{
6760    if (mEffectInterface == NULL) {
6761        return;
6762    }
6763    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6764}
6765
6766status_t AudioFlinger::EffectModule::configure()
6767{
6768    uint32_t channels;
6769    if (mEffectInterface == NULL) {
6770        return NO_INIT;
6771    }
6772
6773    sp<ThreadBase> thread = mThread.promote();
6774    if (thread == 0) {
6775        return DEAD_OBJECT;
6776    }
6777
6778    // TODO: handle configuration of effects replacing track process
6779    if (thread->channelCount() == 1) {
6780        channels = AUDIO_CHANNEL_OUT_MONO;
6781    } else {
6782        channels = AUDIO_CHANNEL_OUT_STEREO;
6783    }
6784
6785    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6786        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6787    } else {
6788        mConfig.inputCfg.channels = channels;
6789    }
6790    mConfig.outputCfg.channels = channels;
6791    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6792    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6793    mConfig.inputCfg.samplingRate = thread->sampleRate();
6794    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6795    mConfig.inputCfg.bufferProvider.cookie = NULL;
6796    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6797    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6798    mConfig.outputCfg.bufferProvider.cookie = NULL;
6799    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6800    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6801    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6802    // Insert effect:
6803    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6804    // always overwrites output buffer: input buffer == output buffer
6805    // - in other sessions:
6806    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6807    //      other effect: overwrites output buffer: input buffer == output buffer
6808    // Auxiliary effect:
6809    //      accumulates in output buffer: input buffer != output buffer
6810    // Therefore: accumulate <=> input buffer != output buffer
6811    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6812        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6813    } else {
6814        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6815    }
6816    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6817    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6818    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6819    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6820
6821    LOGV("configure() %p thread %p buffer %p framecount %d",
6822            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6823
6824    status_t cmdStatus;
6825    uint32_t size = sizeof(int);
6826    status_t status = (*mEffectInterface)->command(mEffectInterface,
6827                                                   EFFECT_CMD_CONFIGURE,
6828                                                   sizeof(effect_config_t),
6829                                                   &mConfig,
6830                                                   &size,
6831                                                   &cmdStatus);
6832    if (status == 0) {
6833        status = cmdStatus;
6834    }
6835
6836    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6837            (1000 * mConfig.outputCfg.buffer.frameCount);
6838
6839    return status;
6840}
6841
6842status_t AudioFlinger::EffectModule::init()
6843{
6844    Mutex::Autolock _l(mLock);
6845    if (mEffectInterface == NULL) {
6846        return NO_INIT;
6847    }
6848    status_t cmdStatus;
6849    uint32_t size = sizeof(status_t);
6850    status_t status = (*mEffectInterface)->command(mEffectInterface,
6851                                                   EFFECT_CMD_INIT,
6852                                                   0,
6853                                                   NULL,
6854                                                   &size,
6855                                                   &cmdStatus);
6856    if (status == 0) {
6857        status = cmdStatus;
6858    }
6859    return status;
6860}
6861
6862status_t AudioFlinger::EffectModule::start_l()
6863{
6864    if (mEffectInterface == NULL) {
6865        return NO_INIT;
6866    }
6867    status_t cmdStatus;
6868    uint32_t size = sizeof(status_t);
6869    status_t status = (*mEffectInterface)->command(mEffectInterface,
6870                                                   EFFECT_CMD_ENABLE,
6871                                                   0,
6872                                                   NULL,
6873                                                   &size,
6874                                                   &cmdStatus);
6875    if (status == 0) {
6876        status = cmdStatus;
6877    }
6878    if (status == 0 &&
6879            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6880             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6881        sp<ThreadBase> thread = mThread.promote();
6882        if (thread != 0) {
6883            audio_stream_t *stream = thread->stream();
6884            if (stream != NULL) {
6885                stream->add_audio_effect(stream, mEffectInterface);
6886            }
6887        }
6888    }
6889    return status;
6890}
6891
6892status_t AudioFlinger::EffectModule::stop()
6893{
6894    Mutex::Autolock _l(mLock);
6895    return stop_l();
6896}
6897
6898status_t AudioFlinger::EffectModule::stop_l()
6899{
6900    if (mEffectInterface == NULL) {
6901        return NO_INIT;
6902    }
6903    status_t cmdStatus;
6904    uint32_t size = sizeof(status_t);
6905    status_t status = (*mEffectInterface)->command(mEffectInterface,
6906                                                   EFFECT_CMD_DISABLE,
6907                                                   0,
6908                                                   NULL,
6909                                                   &size,
6910                                                   &cmdStatus);
6911    if (status == 0) {
6912        status = cmdStatus;
6913    }
6914    if (status == 0 &&
6915            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6916             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6917        sp<ThreadBase> thread = mThread.promote();
6918        if (thread != 0) {
6919            audio_stream_t *stream = thread->stream();
6920            if (stream != NULL) {
6921                stream->remove_audio_effect(stream, mEffectInterface);
6922            }
6923        }
6924    }
6925    return status;
6926}
6927
6928status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6929                                             uint32_t cmdSize,
6930                                             void *pCmdData,
6931                                             uint32_t *replySize,
6932                                             void *pReplyData)
6933{
6934    Mutex::Autolock _l(mLock);
6935//    LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6936
6937    if (mState == DESTROYED || mEffectInterface == NULL) {
6938        return NO_INIT;
6939    }
6940    status_t status = (*mEffectInterface)->command(mEffectInterface,
6941                                                   cmdCode,
6942                                                   cmdSize,
6943                                                   pCmdData,
6944                                                   replySize,
6945                                                   pReplyData);
6946    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6947        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6948        for (size_t i = 1; i < mHandles.size(); i++) {
6949            sp<EffectHandle> h = mHandles[i].promote();
6950            if (h != 0) {
6951                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6952            }
6953        }
6954    }
6955    return status;
6956}
6957
6958status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6959{
6960
6961    Mutex::Autolock _l(mLock);
6962    LOGV("setEnabled %p enabled %d", this, enabled);
6963
6964    if (enabled != isEnabled()) {
6965        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6966        if (enabled && status != NO_ERROR) {
6967            return status;
6968        }
6969
6970        switch (mState) {
6971        // going from disabled to enabled
6972        case IDLE:
6973            mState = STARTING;
6974            break;
6975        case STOPPED:
6976            mState = RESTART;
6977            break;
6978        case STOPPING:
6979            mState = ACTIVE;
6980            break;
6981
6982        // going from enabled to disabled
6983        case RESTART:
6984            mState = STOPPED;
6985            break;
6986        case STARTING:
6987            mState = IDLE;
6988            break;
6989        case ACTIVE:
6990            mState = STOPPING;
6991            break;
6992        case DESTROYED:
6993            return NO_ERROR; // simply ignore as we are being destroyed
6994        }
6995        for (size_t i = 1; i < mHandles.size(); i++) {
6996            sp<EffectHandle> h = mHandles[i].promote();
6997            if (h != 0) {
6998                h->setEnabled(enabled);
6999            }
7000        }
7001    }
7002    return NO_ERROR;
7003}
7004
7005bool AudioFlinger::EffectModule::isEnabled()
7006{
7007    switch (mState) {
7008    case RESTART:
7009    case STARTING:
7010    case ACTIVE:
7011        return true;
7012    case IDLE:
7013    case STOPPING:
7014    case STOPPED:
7015    case DESTROYED:
7016    default:
7017        return false;
7018    }
7019}
7020
7021bool AudioFlinger::EffectModule::isProcessEnabled()
7022{
7023    switch (mState) {
7024    case RESTART:
7025    case ACTIVE:
7026    case STOPPING:
7027    case STOPPED:
7028        return true;
7029    case IDLE:
7030    case STARTING:
7031    case DESTROYED:
7032    default:
7033        return false;
7034    }
7035}
7036
7037status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
7038{
7039    Mutex::Autolock _l(mLock);
7040    status_t status = NO_ERROR;
7041
7042    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
7043    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
7044    if (isProcessEnabled() &&
7045            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
7046            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
7047        status_t cmdStatus;
7048        uint32_t volume[2];
7049        uint32_t *pVolume = NULL;
7050        uint32_t size = sizeof(volume);
7051        volume[0] = *left;
7052        volume[1] = *right;
7053        if (controller) {
7054            pVolume = volume;
7055        }
7056        status = (*mEffectInterface)->command(mEffectInterface,
7057                                              EFFECT_CMD_SET_VOLUME,
7058                                              size,
7059                                              volume,
7060                                              &size,
7061                                              pVolume);
7062        if (controller && status == NO_ERROR && size == sizeof(volume)) {
7063            *left = volume[0];
7064            *right = volume[1];
7065        }
7066    }
7067    return status;
7068}
7069
7070status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
7071{
7072    Mutex::Autolock _l(mLock);
7073    status_t status = NO_ERROR;
7074    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
7075        // audio pre processing modules on RecordThread can receive both output and
7076        // input device indication in the same call
7077        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
7078        if (dev) {
7079            status_t cmdStatus;
7080            uint32_t size = sizeof(status_t);
7081
7082            status = (*mEffectInterface)->command(mEffectInterface,
7083                                                  EFFECT_CMD_SET_DEVICE,
7084                                                  sizeof(uint32_t),
7085                                                  &dev,
7086                                                  &size,
7087                                                  &cmdStatus);
7088            if (status == NO_ERROR) {
7089                status = cmdStatus;
7090            }
7091        }
7092        dev = device & AUDIO_DEVICE_IN_ALL;
7093        if (dev) {
7094            status_t cmdStatus;
7095            uint32_t size = sizeof(status_t);
7096
7097            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
7098                                                  EFFECT_CMD_SET_INPUT_DEVICE,
7099                                                  sizeof(uint32_t),
7100                                                  &dev,
7101                                                  &size,
7102                                                  &cmdStatus);
7103            if (status2 == NO_ERROR) {
7104                status2 = cmdStatus;
7105            }
7106            if (status == NO_ERROR) {
7107                status = status2;
7108            }
7109        }
7110    }
7111    return status;
7112}
7113
7114status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
7115{
7116    Mutex::Autolock _l(mLock);
7117    status_t status = NO_ERROR;
7118    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
7119        status_t cmdStatus;
7120        uint32_t size = sizeof(status_t);
7121        status = (*mEffectInterface)->command(mEffectInterface,
7122                                              EFFECT_CMD_SET_AUDIO_MODE,
7123                                              sizeof(int),
7124                                              &mode,
7125                                              &size,
7126                                              &cmdStatus);
7127        if (status == NO_ERROR) {
7128            status = cmdStatus;
7129        }
7130    }
7131    return status;
7132}
7133
7134void AudioFlinger::EffectModule::setSuspended(bool suspended)
7135{
7136    Mutex::Autolock _l(mLock);
7137    mSuspended = suspended;
7138}
7139bool AudioFlinger::EffectModule::suspended()
7140{
7141    Mutex::Autolock _l(mLock);
7142    return mSuspended;
7143}
7144
7145status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
7146{
7147    const size_t SIZE = 256;
7148    char buffer[SIZE];
7149    String8 result;
7150
7151    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
7152    result.append(buffer);
7153
7154    bool locked = tryLock(mLock);
7155    // failed to lock - AudioFlinger is probably deadlocked
7156    if (!locked) {
7157        result.append("\t\tCould not lock Fx mutex:\n");
7158    }
7159
7160    result.append("\t\tSession Status State Engine:\n");
7161    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
7162            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
7163    result.append(buffer);
7164
7165    result.append("\t\tDescriptor:\n");
7166    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
7167            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
7168            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
7169            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
7170    result.append(buffer);
7171    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
7172                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
7173                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
7174                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
7175    result.append(buffer);
7176    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
7177            mDescriptor.apiVersion,
7178            mDescriptor.flags);
7179    result.append(buffer);
7180    snprintf(buffer, SIZE, "\t\t- name: %s\n",
7181            mDescriptor.name);
7182    result.append(buffer);
7183    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
7184            mDescriptor.implementor);
7185    result.append(buffer);
7186
7187    result.append("\t\t- Input configuration:\n");
7188    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
7189    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
7190            (uint32_t)mConfig.inputCfg.buffer.raw,
7191            mConfig.inputCfg.buffer.frameCount,
7192            mConfig.inputCfg.samplingRate,
7193            mConfig.inputCfg.channels,
7194            mConfig.inputCfg.format);
7195    result.append(buffer);
7196
7197    result.append("\t\t- Output configuration:\n");
7198    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
7199    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
7200            (uint32_t)mConfig.outputCfg.buffer.raw,
7201            mConfig.outputCfg.buffer.frameCount,
7202            mConfig.outputCfg.samplingRate,
7203            mConfig.outputCfg.channels,
7204            mConfig.outputCfg.format);
7205    result.append(buffer);
7206
7207    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
7208    result.append(buffer);
7209    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
7210    for (size_t i = 0; i < mHandles.size(); ++i) {
7211        sp<EffectHandle> handle = mHandles[i].promote();
7212        if (handle != 0) {
7213            handle->dump(buffer, SIZE);
7214            result.append(buffer);
7215        }
7216    }
7217
7218    result.append("\n");
7219
7220    write(fd, result.string(), result.length());
7221
7222    if (locked) {
7223        mLock.unlock();
7224    }
7225
7226    return NO_ERROR;
7227}
7228
7229// ----------------------------------------------------------------------------
7230//  EffectHandle implementation
7231// ----------------------------------------------------------------------------
7232
7233#undef LOG_TAG
7234#define LOG_TAG "AudioFlinger::EffectHandle"
7235
7236AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
7237                                        const sp<AudioFlinger::Client>& client,
7238                                        const sp<IEffectClient>& effectClient,
7239                                        int32_t priority)
7240    : BnEffect(),
7241    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
7242    mPriority(priority), mHasControl(false), mEnabled(false)
7243{
7244    LOGV("constructor %p", this);
7245
7246    if (client == 0) {
7247        return;
7248    }
7249    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
7250    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
7251    if (mCblkMemory != 0) {
7252        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
7253
7254        if (mCblk) {
7255            new(mCblk) effect_param_cblk_t();
7256            mBuffer = (uint8_t *)mCblk + bufOffset;
7257         }
7258    } else {
7259        LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
7260        return;
7261    }
7262}
7263
7264AudioFlinger::EffectHandle::~EffectHandle()
7265{
7266    LOGV("Destructor %p", this);
7267    disconnect(false);
7268    LOGV("Destructor DONE %p", this);
7269}
7270
7271status_t AudioFlinger::EffectHandle::enable()
7272{
7273    LOGV("enable %p", this);
7274    if (!mHasControl) return INVALID_OPERATION;
7275    if (mEffect == 0) return DEAD_OBJECT;
7276
7277    if (mEnabled) {
7278        return NO_ERROR;
7279    }
7280
7281    mEnabled = true;
7282
7283    sp<ThreadBase> thread = mEffect->thread().promote();
7284    if (thread != 0) {
7285        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
7286    }
7287
7288    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
7289    if (mEffect->suspended()) {
7290        return NO_ERROR;
7291    }
7292
7293    status_t status = mEffect->setEnabled(true);
7294    if (status != NO_ERROR) {
7295        if (thread != 0) {
7296            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
7297        }
7298        mEnabled = false;
7299    }
7300    return status;
7301}
7302
7303status_t AudioFlinger::EffectHandle::disable()
7304{
7305    LOGV("disable %p", this);
7306    if (!mHasControl) return INVALID_OPERATION;
7307    if (mEffect == 0) return DEAD_OBJECT;
7308
7309    if (!mEnabled) {
7310        return NO_ERROR;
7311    }
7312    mEnabled = false;
7313
7314    if (mEffect->suspended()) {
7315        return NO_ERROR;
7316    }
7317
7318    status_t status = mEffect->setEnabled(false);
7319
7320    sp<ThreadBase> thread = mEffect->thread().promote();
7321    if (thread != 0) {
7322        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
7323    }
7324
7325    return status;
7326}
7327
7328void AudioFlinger::EffectHandle::disconnect()
7329{
7330    disconnect(true);
7331}
7332
7333void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
7334{
7335    LOGV("disconnect(%s)", unpiniflast ? "true" : "false");
7336    if (mEffect == 0) {
7337        return;
7338    }
7339    mEffect->disconnect(this, unpiniflast);
7340
7341    if (mEnabled) {
7342        sp<ThreadBase> thread = mEffect->thread().promote();
7343        if (thread != 0) {
7344            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
7345        }
7346    }
7347
7348    // release sp on module => module destructor can be called now
7349    mEffect.clear();
7350    if (mClient != 0) {
7351        if (mCblk) {
7352            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
7353        }
7354        mCblkMemory.clear();            // and free the shared memory
7355        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
7356        mClient.clear();
7357    }
7358}
7359
7360status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
7361                                             uint32_t cmdSize,
7362                                             void *pCmdData,
7363                                             uint32_t *replySize,
7364                                             void *pReplyData)
7365{
7366//    LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
7367//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
7368
7369    // only get parameter command is permitted for applications not controlling the effect
7370    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
7371        return INVALID_OPERATION;
7372    }
7373    if (mEffect == 0) return DEAD_OBJECT;
7374    if (mClient == 0) return INVALID_OPERATION;
7375
7376    // handle commands that are not forwarded transparently to effect engine
7377    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
7378        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
7379        // no risk to block the whole media server process or mixer threads is we are stuck here
7380        Mutex::Autolock _l(mCblk->lock);
7381        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
7382            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
7383            mCblk->serverIndex = 0;
7384            mCblk->clientIndex = 0;
7385            return BAD_VALUE;
7386        }
7387        status_t status = NO_ERROR;
7388        while (mCblk->serverIndex < mCblk->clientIndex) {
7389            int reply;
7390            uint32_t rsize = sizeof(int);
7391            int *p = (int *)(mBuffer + mCblk->serverIndex);
7392            int size = *p++;
7393            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
7394                LOGW("command(): invalid parameter block size");
7395                break;
7396            }
7397            effect_param_t *param = (effect_param_t *)p;
7398            if (param->psize == 0 || param->vsize == 0) {
7399                LOGW("command(): null parameter or value size");
7400                mCblk->serverIndex += size;
7401                continue;
7402            }
7403            uint32_t psize = sizeof(effect_param_t) +
7404                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
7405                             param->vsize;
7406            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
7407                                            psize,
7408                                            p,
7409                                            &rsize,
7410                                            &reply);
7411            // stop at first error encountered
7412            if (ret != NO_ERROR) {
7413                status = ret;
7414                *(int *)pReplyData = reply;
7415                break;
7416            } else if (reply != NO_ERROR) {
7417                *(int *)pReplyData = reply;
7418                break;
7419            }
7420            mCblk->serverIndex += size;
7421        }
7422        mCblk->serverIndex = 0;
7423        mCblk->clientIndex = 0;
7424        return status;
7425    } else if (cmdCode == EFFECT_CMD_ENABLE) {
7426        *(int *)pReplyData = NO_ERROR;
7427        return enable();
7428    } else if (cmdCode == EFFECT_CMD_DISABLE) {
7429        *(int *)pReplyData = NO_ERROR;
7430        return disable();
7431    }
7432
7433    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7434}
7435
7436sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
7437    return mCblkMemory;
7438}
7439
7440void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
7441{
7442    LOGV("setControl %p control %d", this, hasControl);
7443
7444    mHasControl = hasControl;
7445    mEnabled = enabled;
7446
7447    if (signal && mEffectClient != 0) {
7448        mEffectClient->controlStatusChanged(hasControl);
7449    }
7450}
7451
7452void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7453                                                 uint32_t cmdSize,
7454                                                 void *pCmdData,
7455                                                 uint32_t replySize,
7456                                                 void *pReplyData)
7457{
7458    if (mEffectClient != 0) {
7459        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7460    }
7461}
7462
7463
7464
7465void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7466{
7467    if (mEffectClient != 0) {
7468        mEffectClient->enableStatusChanged(enabled);
7469    }
7470}
7471
7472status_t AudioFlinger::EffectHandle::onTransact(
7473    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7474{
7475    return BnEffect::onTransact(code, data, reply, flags);
7476}
7477
7478
7479void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7480{
7481    bool locked = mCblk ? tryLock(mCblk->lock) : false;
7482
7483    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7484            (mClient == NULL) ? getpid() : mClient->pid(),
7485            mPriority,
7486            mHasControl,
7487            !locked,
7488            mCblk ? mCblk->clientIndex : 0,
7489            mCblk ? mCblk->serverIndex : 0
7490            );
7491
7492    if (locked) {
7493        mCblk->lock.unlock();
7494    }
7495}
7496
7497#undef LOG_TAG
7498#define LOG_TAG "AudioFlinger::EffectChain"
7499
7500AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7501                                        int sessionId)
7502    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0),
7503      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7504      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7505{
7506    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7507}
7508
7509AudioFlinger::EffectChain::~EffectChain()
7510{
7511    if (mOwnInBuffer) {
7512        delete mInBuffer;
7513    }
7514
7515}
7516
7517// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7518sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7519{
7520    sp<EffectModule> effect;
7521    size_t size = mEffects.size();
7522
7523    for (size_t i = 0; i < size; i++) {
7524        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7525            effect = mEffects[i];
7526            break;
7527        }
7528    }
7529    return effect;
7530}
7531
7532// getEffectFromId_l() must be called with ThreadBase::mLock held
7533sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7534{
7535    sp<EffectModule> effect;
7536    size_t size = mEffects.size();
7537
7538    for (size_t i = 0; i < size; i++) {
7539        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7540        if (id == 0 || mEffects[i]->id() == id) {
7541            effect = mEffects[i];
7542            break;
7543        }
7544    }
7545    return effect;
7546}
7547
7548// getEffectFromType_l() must be called with ThreadBase::mLock held
7549sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7550        const effect_uuid_t *type)
7551{
7552    sp<EffectModule> effect;
7553    size_t size = mEffects.size();
7554
7555    for (size_t i = 0; i < size; i++) {
7556        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7557            effect = mEffects[i];
7558            break;
7559        }
7560    }
7561    return effect;
7562}
7563
7564// Must be called with EffectChain::mLock locked
7565void AudioFlinger::EffectChain::process_l()
7566{
7567    sp<ThreadBase> thread = mThread.promote();
7568    if (thread == 0) {
7569        LOGW("process_l(): cannot promote mixer thread");
7570        return;
7571    }
7572    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7573            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7574    bool tracksOnSession = false;
7575    if (!isGlobalSession) {
7576        tracksOnSession = (trackCnt() != 0);
7577    }
7578
7579    // if no track is active, input buffer must be cleared here as the mixer process
7580    // will not do it
7581    if (tracksOnSession &&
7582            activeTrackCnt() == 0) {
7583        size_t numSamples = thread->frameCount() * thread->channelCount();
7584        memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7585    }
7586
7587    size_t size = mEffects.size();
7588    // do not process effect if no track is present in same audio session
7589    if (isGlobalSession || tracksOnSession) {
7590        for (size_t i = 0; i < size; i++) {
7591            mEffects[i]->process();
7592        }
7593    }
7594    for (size_t i = 0; i < size; i++) {
7595        mEffects[i]->updateState();
7596    }
7597}
7598
7599// addEffect_l() must be called with PlaybackThread::mLock held
7600status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7601{
7602    effect_descriptor_t desc = effect->desc();
7603    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7604
7605    Mutex::Autolock _l(mLock);
7606    effect->setChain(this);
7607    sp<ThreadBase> thread = mThread.promote();
7608    if (thread == 0) {
7609        return NO_INIT;
7610    }
7611    effect->setThread(thread);
7612
7613    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7614        // Auxiliary effects are inserted at the beginning of mEffects vector as
7615        // they are processed first and accumulated in chain input buffer
7616        mEffects.insertAt(effect, 0);
7617
7618        // the input buffer for auxiliary effect contains mono samples in
7619        // 32 bit format. This is to avoid saturation in AudoMixer
7620        // accumulation stage. Saturation is done in EffectModule::process() before
7621        // calling the process in effect engine
7622        size_t numSamples = thread->frameCount();
7623        int32_t *buffer = new int32_t[numSamples];
7624        memset(buffer, 0, numSamples * sizeof(int32_t));
7625        effect->setInBuffer((int16_t *)buffer);
7626        // auxiliary effects output samples to chain input buffer for further processing
7627        // by insert effects
7628        effect->setOutBuffer(mInBuffer);
7629    } else {
7630        // Insert effects are inserted at the end of mEffects vector as they are processed
7631        //  after track and auxiliary effects.
7632        // Insert effect order as a function of indicated preference:
7633        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7634        //  another effect is present
7635        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7636        //  last effect claiming first position
7637        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7638        //  first effect claiming last position
7639        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7640        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7641        // already present
7642
7643        int size = (int)mEffects.size();
7644        int idx_insert = size;
7645        int idx_insert_first = -1;
7646        int idx_insert_last = -1;
7647
7648        for (int i = 0; i < size; i++) {
7649            effect_descriptor_t d = mEffects[i]->desc();
7650            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7651            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7652            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7653                // check invalid effect chaining combinations
7654                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7655                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7656                    LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7657                    return INVALID_OPERATION;
7658                }
7659                // remember position of first insert effect and by default
7660                // select this as insert position for new effect
7661                if (idx_insert == size) {
7662                    idx_insert = i;
7663                }
7664                // remember position of last insert effect claiming
7665                // first position
7666                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7667                    idx_insert_first = i;
7668                }
7669                // remember position of first insert effect claiming
7670                // last position
7671                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7672                    idx_insert_last == -1) {
7673                    idx_insert_last = i;
7674                }
7675            }
7676        }
7677
7678        // modify idx_insert from first position if needed
7679        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7680            if (idx_insert_last != -1) {
7681                idx_insert = idx_insert_last;
7682            } else {
7683                idx_insert = size;
7684            }
7685        } else {
7686            if (idx_insert_first != -1) {
7687                idx_insert = idx_insert_first + 1;
7688            }
7689        }
7690
7691        // always read samples from chain input buffer
7692        effect->setInBuffer(mInBuffer);
7693
7694        // if last effect in the chain, output samples to chain
7695        // output buffer, otherwise to chain input buffer
7696        if (idx_insert == size) {
7697            if (idx_insert != 0) {
7698                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7699                mEffects[idx_insert-1]->configure();
7700            }
7701            effect->setOutBuffer(mOutBuffer);
7702        } else {
7703            effect->setOutBuffer(mInBuffer);
7704        }
7705        mEffects.insertAt(effect, idx_insert);
7706
7707        LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7708    }
7709    effect->configure();
7710    return NO_ERROR;
7711}
7712
7713// removeEffect_l() must be called with PlaybackThread::mLock held
7714size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7715{
7716    Mutex::Autolock _l(mLock);
7717    int size = (int)mEffects.size();
7718    int i;
7719    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7720
7721    for (i = 0; i < size; i++) {
7722        if (effect == mEffects[i]) {
7723            // calling stop here will remove pre-processing effect from the audio HAL.
7724            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7725            // the middle of a read from audio HAL
7726            mEffects[i]->stop();
7727            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7728                delete[] effect->inBuffer();
7729            } else {
7730                if (i == size - 1 && i != 0) {
7731                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7732                    mEffects[i - 1]->configure();
7733                }
7734            }
7735            mEffects.removeAt(i);
7736            LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7737            break;
7738        }
7739    }
7740
7741    return mEffects.size();
7742}
7743
7744// setDevice_l() must be called with PlaybackThread::mLock held
7745void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7746{
7747    size_t size = mEffects.size();
7748    for (size_t i = 0; i < size; i++) {
7749        mEffects[i]->setDevice(device);
7750    }
7751}
7752
7753// setMode_l() must be called with PlaybackThread::mLock held
7754void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
7755{
7756    size_t size = mEffects.size();
7757    for (size_t i = 0; i < size; i++) {
7758        mEffects[i]->setMode(mode);
7759    }
7760}
7761
7762// setVolume_l() must be called with PlaybackThread::mLock held
7763bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7764{
7765    uint32_t newLeft = *left;
7766    uint32_t newRight = *right;
7767    bool hasControl = false;
7768    int ctrlIdx = -1;
7769    size_t size = mEffects.size();
7770
7771    // first update volume controller
7772    for (size_t i = size; i > 0; i--) {
7773        if (mEffects[i - 1]->isProcessEnabled() &&
7774            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7775            ctrlIdx = i - 1;
7776            hasControl = true;
7777            break;
7778        }
7779    }
7780
7781    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7782        if (hasControl) {
7783            *left = mNewLeftVolume;
7784            *right = mNewRightVolume;
7785        }
7786        return hasControl;
7787    }
7788
7789    mVolumeCtrlIdx = ctrlIdx;
7790    mLeftVolume = newLeft;
7791    mRightVolume = newRight;
7792
7793    // second get volume update from volume controller
7794    if (ctrlIdx >= 0) {
7795        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7796        mNewLeftVolume = newLeft;
7797        mNewRightVolume = newRight;
7798    }
7799    // then indicate volume to all other effects in chain.
7800    // Pass altered volume to effects before volume controller
7801    // and requested volume to effects after controller
7802    uint32_t lVol = newLeft;
7803    uint32_t rVol = newRight;
7804
7805    for (size_t i = 0; i < size; i++) {
7806        if ((int)i == ctrlIdx) continue;
7807        // this also works for ctrlIdx == -1 when there is no volume controller
7808        if ((int)i > ctrlIdx) {
7809            lVol = *left;
7810            rVol = *right;
7811        }
7812        mEffects[i]->setVolume(&lVol, &rVol, false);
7813    }
7814    *left = newLeft;
7815    *right = newRight;
7816
7817    return hasControl;
7818}
7819
7820status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7821{
7822    const size_t SIZE = 256;
7823    char buffer[SIZE];
7824    String8 result;
7825
7826    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7827    result.append(buffer);
7828
7829    bool locked = tryLock(mLock);
7830    // failed to lock - AudioFlinger is probably deadlocked
7831    if (!locked) {
7832        result.append("\tCould not lock mutex:\n");
7833    }
7834
7835    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7836    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7837            mEffects.size(),
7838            (uint32_t)mInBuffer,
7839            (uint32_t)mOutBuffer,
7840            mActiveTrackCnt);
7841    result.append(buffer);
7842    write(fd, result.string(), result.size());
7843
7844    for (size_t i = 0; i < mEffects.size(); ++i) {
7845        sp<EffectModule> effect = mEffects[i];
7846        if (effect != 0) {
7847            effect->dump(fd, args);
7848        }
7849    }
7850
7851    if (locked) {
7852        mLock.unlock();
7853    }
7854
7855    return NO_ERROR;
7856}
7857
7858// must be called with ThreadBase::mLock held
7859void AudioFlinger::EffectChain::setEffectSuspended_l(
7860        const effect_uuid_t *type, bool suspend)
7861{
7862    sp<SuspendedEffectDesc> desc;
7863    // use effect type UUID timelow as key as there is no real risk of identical
7864    // timeLow fields among effect type UUIDs.
7865    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7866    if (suspend) {
7867        if (index >= 0) {
7868            desc = mSuspendedEffects.valueAt(index);
7869        } else {
7870            desc = new SuspendedEffectDesc();
7871            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7872            mSuspendedEffects.add(type->timeLow, desc);
7873            LOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7874        }
7875        if (desc->mRefCount++ == 0) {
7876            sp<EffectModule> effect = getEffectIfEnabled(type);
7877            if (effect != 0) {
7878                desc->mEffect = effect;
7879                effect->setSuspended(true);
7880                effect->setEnabled(false);
7881            }
7882        }
7883    } else {
7884        if (index < 0) {
7885            return;
7886        }
7887        desc = mSuspendedEffects.valueAt(index);
7888        if (desc->mRefCount <= 0) {
7889            LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7890            desc->mRefCount = 1;
7891        }
7892        if (--desc->mRefCount == 0) {
7893            LOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7894            if (desc->mEffect != 0) {
7895                sp<EffectModule> effect = desc->mEffect.promote();
7896                if (effect != 0) {
7897                    effect->setSuspended(false);
7898                    sp<EffectHandle> handle = effect->controlHandle();
7899                    if (handle != 0) {
7900                        effect->setEnabled(handle->enabled());
7901                    }
7902                }
7903                desc->mEffect.clear();
7904            }
7905            mSuspendedEffects.removeItemsAt(index);
7906        }
7907    }
7908}
7909
7910// must be called with ThreadBase::mLock held
7911void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7912{
7913    sp<SuspendedEffectDesc> desc;
7914
7915    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7916    if (suspend) {
7917        if (index >= 0) {
7918            desc = mSuspendedEffects.valueAt(index);
7919        } else {
7920            desc = new SuspendedEffectDesc();
7921            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7922            LOGV("setEffectSuspendedAll_l() add entry for 0");
7923        }
7924        if (desc->mRefCount++ == 0) {
7925            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7926            for (size_t i = 0; i < effects.size(); i++) {
7927                setEffectSuspended_l(&effects[i]->desc().type, true);
7928            }
7929        }
7930    } else {
7931        if (index < 0) {
7932            return;
7933        }
7934        desc = mSuspendedEffects.valueAt(index);
7935        if (desc->mRefCount <= 0) {
7936            LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7937            desc->mRefCount = 1;
7938        }
7939        if (--desc->mRefCount == 0) {
7940            Vector<const effect_uuid_t *> types;
7941            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7942                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7943                    continue;
7944                }
7945                types.add(&mSuspendedEffects.valueAt(i)->mType);
7946            }
7947            for (size_t i = 0; i < types.size(); i++) {
7948                setEffectSuspended_l(types[i], false);
7949            }
7950            LOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7951            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7952        }
7953    }
7954}
7955
7956bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7957{
7958    // auxiliary effects and visualizer are never suspended on output mix
7959    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7960        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7961         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0))) {
7962        return false;
7963    }
7964    return true;
7965}
7966
7967Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7968{
7969    Vector< sp<EffectModule> > effects;
7970    for (size_t i = 0; i < mEffects.size(); i++) {
7971        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7972            continue;
7973        }
7974        effects.add(mEffects[i]);
7975    }
7976    return effects;
7977}
7978
7979sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7980                                                            const effect_uuid_t *type)
7981{
7982    sp<EffectModule> effect;
7983    effect = getEffectFromType_l(type);
7984    if (effect != 0 && !effect->isEnabled()) {
7985        effect.clear();
7986    }
7987    return effect;
7988}
7989
7990void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7991                                                            bool enabled)
7992{
7993    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7994    if (enabled) {
7995        if (index < 0) {
7996            // if the effect is not suspend check if all effects are suspended
7997            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7998            if (index < 0) {
7999                return;
8000            }
8001            if (!isEffectEligibleForSuspend(effect->desc())) {
8002                return;
8003            }
8004            setEffectSuspended_l(&effect->desc().type, enabled);
8005            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
8006            if (index < 0) {
8007                LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
8008                return;
8009            }
8010        }
8011        LOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
8012             effect->desc().type.timeLow);
8013        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
8014        // if effect is requested to suspended but was not yet enabled, supend it now.
8015        if (desc->mEffect == 0) {
8016            desc->mEffect = effect;
8017            effect->setEnabled(false);
8018            effect->setSuspended(true);
8019        }
8020    } else {
8021        if (index < 0) {
8022            return;
8023        }
8024        LOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
8025             effect->desc().type.timeLow);
8026        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
8027        desc->mEffect.clear();
8028        effect->setSuspended(false);
8029    }
8030}
8031
8032#undef LOG_TAG
8033#define LOG_TAG "AudioFlinger"
8034
8035// ----------------------------------------------------------------------------
8036
8037status_t AudioFlinger::onTransact(
8038        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
8039{
8040    return BnAudioFlinger::onTransact(code, data, reply, flags);
8041}
8042
8043}; // namespace android
8044