AudioFlinger.cpp revision f78aee70d15daf4690de7e7b4983ee68b0d1381d 
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#include <cutils/compiler.h>
39
40#include <media/AudioTrack.h>
41#include <media/AudioRecord.h>
42#include <media/IMediaPlayerService.h>
43#include <media/IMediaDeathNotifier.h>
44
45#include <private/media/AudioTrackShared.h>
46#include <private/media/AudioEffectShared.h>
47
48#include <system/audio.h>
49#include <hardware/audio.h>
50
51#include "AudioMixer.h"
52#include "AudioFlinger.h"
53
54#include <media/EffectsFactoryApi.h>
55#include <audio_effects/effect_visualizer.h>
56#include <audio_effects/effect_ns.h>
57#include <audio_effects/effect_aec.h>
58
59#include <audio_utils/primitives.h>
60
61#include <cpustats/ThreadCpuUsage.h>
62#include <powermanager/PowerManager.h>
63// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
64
65// ----------------------------------------------------------------------------
66
67
68namespace android {
69
70static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
71static const char kHardwareLockedString[] = "Hardware lock is taken\n";
72
73//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
74static const float MAX_GAIN = 4096.0f;
75static const float MAX_GAIN_INT = 0x1000;
76
77// retry counts for buffer fill timeout
78// 50 * ~20msecs = 1 second
79static const int8_t kMaxTrackRetries = 50;
80static const int8_t kMaxTrackStartupRetries = 50;
81// allow less retry attempts on direct output thread.
82// direct outputs can be a scarce resource in audio hardware and should
83// be released as quickly as possible.
84static const int8_t kMaxTrackRetriesDirect = 2;
85
86static const int kDumpLockRetries = 50;
87static const int kDumpLockSleepUs = 20000;
88
89// don't warn about blocked writes or record buffer overflows more often than this
90static const nsecs_t kWarningThrottleNs = seconds(5);
91
92// RecordThread loop sleep time upon application overrun or audio HAL read error
93static const int kRecordThreadSleepUs = 5000;
94
95// maximum time to wait for setParameters to complete
96static const nsecs_t kSetParametersTimeoutNs = seconds(2);
97
98// minimum sleep time for the mixer thread loop when tracks are active but in underrun
99static const uint32_t kMinThreadSleepTimeUs = 5000;
100// maximum divider applied to the active sleep time in the mixer thread loop
101static const uint32_t kMaxThreadSleepTimeShift = 2;
102
103
104// ----------------------------------------------------------------------------
105
106static bool recordingAllowed() {
107    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
108    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
109    if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
110    return ok;
111}
112
113static bool settingsAllowed() {
114    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
115    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
116    if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
117    return ok;
118}
119
120// To collect the amplifier usage
121static void addBatteryData(uint32_t params) {
122    sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
123    if (service == NULL) {
124        // it already logged
125        return;
126    }
127
128    service->addBatteryData(params);
129}
130
131static int load_audio_interface(const char *if_name, const hw_module_t **mod,
132                                audio_hw_device_t **dev)
133{
134    int rc;
135
136    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
137    if (rc)
138        goto out;
139
140    rc = audio_hw_device_open(*mod, dev);
141    ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
142            AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
143    if (rc)
144        goto out;
145
146    return 0;
147
148out:
149    *mod = NULL;
150    *dev = NULL;
151    return rc;
152}
153
154static const char * const audio_interfaces[] = {
155    "primary",
156    "a2dp",
157    "usb",
158};
159#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
160
161// ----------------------------------------------------------------------------
162
163AudioFlinger::AudioFlinger()
164    : BnAudioFlinger(),
165        mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
166        mBtNrecIsOff(false)
167{
168}
169
170void AudioFlinger::onFirstRef()
171{
172    int rc = 0;
173
174    Mutex::Autolock _l(mLock);
175
176    /* TODO: move all this work into an Init() function */
177    mHardwareStatus = AUDIO_HW_IDLE;
178
179    for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
180        const hw_module_t *mod;
181        audio_hw_device_t *dev;
182
183        rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
184        if (rc)
185            continue;
186
187        ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
188             mod->name, mod->id);
189        mAudioHwDevs.push(dev);
190
191        if (!mPrimaryHardwareDev) {
192            mPrimaryHardwareDev = dev;
193            ALOGI("Using '%s' (%s.%s) as the primary audio interface",
194                 mod->name, mod->id, audio_interfaces[i]);
195        }
196    }
197
198    mHardwareStatus = AUDIO_HW_INIT;
199
200    if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
201        ALOGE("Primary audio interface not found");
202        return;
203    }
204
205    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
206        audio_hw_device_t *dev = mAudioHwDevs[i];
207
208        mHardwareStatus = AUDIO_HW_INIT;
209        rc = dev->init_check(dev);
210        if (rc == 0) {
211            AutoMutex lock(mHardwareLock);
212
213            mMode = AUDIO_MODE_NORMAL;
214            mHardwareStatus = AUDIO_HW_SET_MODE;
215            dev->set_mode(dev, mMode);
216            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
217            dev->set_master_volume(dev, 1.0f);
218            mHardwareStatus = AUDIO_HW_IDLE;
219        }
220    }
221}
222
223status_t AudioFlinger::initCheck() const
224{
225    Mutex::Autolock _l(mLock);
226    if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
227        return NO_INIT;
228    return NO_ERROR;
229}
230
231AudioFlinger::~AudioFlinger()
232{
233    int num_devs = mAudioHwDevs.size();
234
235    while (!mRecordThreads.isEmpty()) {
236        // closeInput() will remove first entry from mRecordThreads
237        closeInput(mRecordThreads.keyAt(0));
238    }
239    while (!mPlaybackThreads.isEmpty()) {
240        // closeOutput() will remove first entry from mPlaybackThreads
241        closeOutput(mPlaybackThreads.keyAt(0));
242    }
243
244    for (int i = 0; i < num_devs; i++) {
245        audio_hw_device_t *dev = mAudioHwDevs[i];
246        audio_hw_device_close(dev);
247    }
248    mAudioHwDevs.clear();
249}
250
251audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
252{
253    /* first matching HW device is returned */
254    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
255        audio_hw_device_t *dev = mAudioHwDevs[i];
256        if ((dev->get_supported_devices(dev) & devices) == devices)
257            return dev;
258    }
259    return NULL;
260}
261
262status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
263{
264    const size_t SIZE = 256;
265    char buffer[SIZE];
266    String8 result;
267
268    result.append("Clients:\n");
269    for (size_t i = 0; i < mClients.size(); ++i) {
270        wp<Client> wClient = mClients.valueAt(i);
271        if (wClient != 0) {
272            sp<Client> client = wClient.promote();
273            if (client != 0) {
274                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
275                result.append(buffer);
276            }
277        }
278    }
279
280    result.append("Global session refs:\n");
281    result.append(" session pid cnt\n");
282    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
283        AudioSessionRef *r = mAudioSessionRefs[i];
284        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
285        result.append(buffer);
286    }
287    write(fd, result.string(), result.size());
288    return NO_ERROR;
289}
290
291
292status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
293{
294    const size_t SIZE = 256;
295    char buffer[SIZE];
296    String8 result;
297    hardware_call_state hardwareStatus = mHardwareStatus;
298
299    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
300    result.append(buffer);
301    write(fd, result.string(), result.size());
302    return NO_ERROR;
303}
304
305status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
306{
307    const size_t SIZE = 256;
308    char buffer[SIZE];
309    String8 result;
310    snprintf(buffer, SIZE, "Permission Denial: "
311            "can't dump AudioFlinger from pid=%d, uid=%d\n",
312            IPCThreadState::self()->getCallingPid(),
313            IPCThreadState::self()->getCallingUid());
314    result.append(buffer);
315    write(fd, result.string(), result.size());
316    return NO_ERROR;
317}
318
319static bool tryLock(Mutex& mutex)
320{
321    bool locked = false;
322    for (int i = 0; i < kDumpLockRetries; ++i) {
323        if (mutex.tryLock() == NO_ERROR) {
324            locked = true;
325            break;
326        }
327        usleep(kDumpLockSleepUs);
328    }
329    return locked;
330}
331
332status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
333{
334    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
335        dumpPermissionDenial(fd, args);
336    } else {
337        // get state of hardware lock
338        bool hardwareLocked = tryLock(mHardwareLock);
339        if (!hardwareLocked) {
340            String8 result(kHardwareLockedString);
341            write(fd, result.string(), result.size());
342        } else {
343            mHardwareLock.unlock();
344        }
345
346        bool locked = tryLock(mLock);
347
348        // failed to lock - AudioFlinger is probably deadlocked
349        if (!locked) {
350            String8 result(kDeadlockedString);
351            write(fd, result.string(), result.size());
352        }
353
354        dumpClients(fd, args);
355        dumpInternals(fd, args);
356
357        // dump playback threads
358        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
359            mPlaybackThreads.valueAt(i)->dump(fd, args);
360        }
361
362        // dump record threads
363        for (size_t i = 0; i < mRecordThreads.size(); i++) {
364            mRecordThreads.valueAt(i)->dump(fd, args);
365        }
366
367        // dump all hardware devs
368        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
369            audio_hw_device_t *dev = mAudioHwDevs[i];
370            dev->dump(dev, fd);
371        }
372        if (locked) mLock.unlock();
373    }
374    return NO_ERROR;
375}
376
377
378// IAudioFlinger interface
379
380
381sp<IAudioTrack> AudioFlinger::createTrack(
382        pid_t pid,
383        int streamType,
384        uint32_t sampleRate,
385        uint32_t format,
386        uint32_t channelMask,
387        int frameCount,
388        uint32_t flags,
389        const sp<IMemory>& sharedBuffer,
390        int output,
391        int *sessionId,
392        status_t *status)
393{
394    sp<PlaybackThread::Track> track;
395    sp<TrackHandle> trackHandle;
396    sp<Client> client;
397    wp<Client> wclient;
398    status_t lStatus;
399    int lSessionId;
400
401    if (streamType >= AUDIO_STREAM_CNT) {
402        ALOGE("createTrack() invalid stream type %d", streamType);
403        lStatus = BAD_VALUE;
404        goto Exit;
405    }
406
407    {
408        Mutex::Autolock _l(mLock);
409        PlaybackThread *thread = checkPlaybackThread_l(output);
410        PlaybackThread *effectThread = NULL;
411        if (thread == NULL) {
412            ALOGE("unknown output thread");
413            lStatus = BAD_VALUE;
414            goto Exit;
415        }
416
417        wclient = mClients.valueFor(pid);
418
419        if (wclient != NULL) {
420            client = wclient.promote();
421        } else {
422            client = new Client(this, pid);
423            mClients.add(pid, client);
424        }
425
426        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
427        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
428            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
429                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
430                if (mPlaybackThreads.keyAt(i) != output) {
431                    // prevent same audio session on different output threads
432                    uint32_t sessions = t->hasAudioSession(*sessionId);
433                    if (sessions & PlaybackThread::TRACK_SESSION) {
434                        ALOGE("createTrack() session ID %d already in use", *sessionId);
435                        lStatus = BAD_VALUE;
436                        goto Exit;
437                    }
438                    // check if an effect with same session ID is waiting for a track to be created
439                    if (sessions & PlaybackThread::EFFECT_SESSION) {
440                        effectThread = t.get();
441                    }
442                }
443            }
444            lSessionId = *sessionId;
445        } else {
446            // if no audio session id is provided, create one here
447            lSessionId = nextUniqueId();
448            if (sessionId != NULL) {
449                *sessionId = lSessionId;
450            }
451        }
452        ALOGV("createTrack() lSessionId: %d", lSessionId);
453
454        track = thread->createTrack_l(client, streamType, sampleRate, format,
455                channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
456
457        // move effect chain to this output thread if an effect on same session was waiting
458        // for a track to be created
459        if (lStatus == NO_ERROR && effectThread != NULL) {
460            Mutex::Autolock _dl(thread->mLock);
461            Mutex::Autolock _sl(effectThread->mLock);
462            moveEffectChain_l(lSessionId, effectThread, thread, true);
463        }
464    }
465    if (lStatus == NO_ERROR) {
466        trackHandle = new TrackHandle(track);
467    } else {
468        // remove local strong reference to Client before deleting the Track so that the Client
469        // destructor is called by the TrackBase destructor with mLock held
470        client.clear();
471        track.clear();
472    }
473
474Exit:
475    if(status) {
476        *status = lStatus;
477    }
478    return trackHandle;
479}
480
481uint32_t AudioFlinger::sampleRate(int output) const
482{
483    Mutex::Autolock _l(mLock);
484    PlaybackThread *thread = checkPlaybackThread_l(output);
485    if (thread == NULL) {
486        ALOGW("sampleRate() unknown thread %d", output);
487        return 0;
488    }
489    return thread->sampleRate();
490}
491
492int AudioFlinger::channelCount(int output) const
493{
494    Mutex::Autolock _l(mLock);
495    PlaybackThread *thread = checkPlaybackThread_l(output);
496    if (thread == NULL) {
497        ALOGW("channelCount() unknown thread %d", output);
498        return 0;
499    }
500    return thread->channelCount();
501}
502
503uint32_t AudioFlinger::format(int output) const
504{
505    Mutex::Autolock _l(mLock);
506    PlaybackThread *thread = checkPlaybackThread_l(output);
507    if (thread == NULL) {
508        ALOGW("format() unknown thread %d", output);
509        return 0;
510    }
511    return thread->format();
512}
513
514size_t AudioFlinger::frameCount(int output) const
515{
516    Mutex::Autolock _l(mLock);
517    PlaybackThread *thread = checkPlaybackThread_l(output);
518    if (thread == NULL) {
519        ALOGW("frameCount() unknown thread %d", output);
520        return 0;
521    }
522    return thread->frameCount();
523}
524
525uint32_t AudioFlinger::latency(int output) const
526{
527    Mutex::Autolock _l(mLock);
528    PlaybackThread *thread = checkPlaybackThread_l(output);
529    if (thread == NULL) {
530        ALOGW("latency() unknown thread %d", output);
531        return 0;
532    }
533    return thread->latency();
534}
535
536status_t AudioFlinger::setMasterVolume(float value)
537{
538    status_t ret = initCheck();
539    if (ret != NO_ERROR) {
540        return ret;
541    }
542
543    // check calling permissions
544    if (!settingsAllowed()) {
545        return PERMISSION_DENIED;
546    }
547
548    // when hw supports master volume, don't scale in sw mixer
549    { // scope for the lock
550        AutoMutex lock(mHardwareLock);
551        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
552        if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
553            value = 1.0f;
554        }
555        mHardwareStatus = AUDIO_HW_IDLE;
556    }
557
558    Mutex::Autolock _l(mLock);
559    mMasterVolume = value;
560    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
561       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
562
563    return NO_ERROR;
564}
565
566status_t AudioFlinger::setMode(audio_mode_t mode)
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    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
578        ALOGW("Illegal value: setMode(%d)", mode);
579        return BAD_VALUE;
580    }
581
582    { // scope for the lock
583        AutoMutex lock(mHardwareLock);
584        mHardwareStatus = AUDIO_HW_SET_MODE;
585        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
586        mHardwareStatus = AUDIO_HW_IDLE;
587    }
588
589    if (NO_ERROR == ret) {
590        Mutex::Autolock _l(mLock);
591        mMode = mode;
592        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
593           mPlaybackThreads.valueAt(i)->setMode(mode);
594    }
595
596    return ret;
597}
598
599status_t AudioFlinger::setMicMute(bool state)
600{
601    status_t ret = initCheck();
602    if (ret != NO_ERROR) {
603        return ret;
604    }
605
606    // check calling permissions
607    if (!settingsAllowed()) {
608        return PERMISSION_DENIED;
609    }
610
611    AutoMutex lock(mHardwareLock);
612    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
613    ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
614    mHardwareStatus = AUDIO_HW_IDLE;
615    return ret;
616}
617
618bool AudioFlinger::getMicMute() const
619{
620    status_t ret = initCheck();
621    if (ret != NO_ERROR) {
622        return false;
623    }
624
625    bool state = AUDIO_MODE_INVALID;
626    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
627    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
628    mHardwareStatus = AUDIO_HW_IDLE;
629    return state;
630}
631
632status_t AudioFlinger::setMasterMute(bool muted)
633{
634    // check calling permissions
635    if (!settingsAllowed()) {
636        return PERMISSION_DENIED;
637    }
638
639    Mutex::Autolock _l(mLock);
640    mMasterMute = muted;
641    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
642       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
643
644    return NO_ERROR;
645}
646
647float AudioFlinger::masterVolume() const
648{
649    return mMasterVolume;
650}
651
652bool AudioFlinger::masterMute() const
653{
654    return mMasterMute;
655}
656
657status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
658{
659    // check calling permissions
660    if (!settingsAllowed()) {
661        return PERMISSION_DENIED;
662    }
663
664    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
665        ALOGE("setStreamVolume() invalid stream %d", stream);
666        return BAD_VALUE;
667    }
668
669    AutoMutex lock(mLock);
670    PlaybackThread *thread = NULL;
671    if (output) {
672        thread = checkPlaybackThread_l(output);
673        if (thread == NULL) {
674            return BAD_VALUE;
675        }
676    }
677
678    mStreamTypes[stream].volume = value;
679
680    if (thread == NULL) {
681        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
682           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
683        }
684    } else {
685        thread->setStreamVolume(stream, value);
686    }
687
688    return NO_ERROR;
689}
690
691status_t AudioFlinger::setStreamMute(int stream, bool muted)
692{
693    // check calling permissions
694    if (!settingsAllowed()) {
695        return PERMISSION_DENIED;
696    }
697
698    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
699        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
700        ALOGE("setStreamMute() invalid stream %d", stream);
701        return BAD_VALUE;
702    }
703
704    AutoMutex lock(mLock);
705    mStreamTypes[stream].mute = muted;
706    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
707       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
708
709    return NO_ERROR;
710}
711
712float AudioFlinger::streamVolume(int stream, int output) const
713{
714    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
715        return 0.0f;
716    }
717
718    AutoMutex lock(mLock);
719    float volume;
720    if (output) {
721        PlaybackThread *thread = checkPlaybackThread_l(output);
722        if (thread == NULL) {
723            return 0.0f;
724        }
725        volume = thread->streamVolume(stream);
726    } else {
727        volume = mStreamTypes[stream].volume;
728    }
729
730    return volume;
731}
732
733bool AudioFlinger::streamMute(int stream) const
734{
735    if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
736        return true;
737    }
738
739    return mStreamTypes[stream].mute;
740}
741
742status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
743{
744    status_t result;
745
746    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
747            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
748    // check calling permissions
749    if (!settingsAllowed()) {
750        return PERMISSION_DENIED;
751    }
752
753    // ioHandle == 0 means the parameters are global to the audio hardware interface
754    if (ioHandle == 0) {
755        AutoMutex lock(mHardwareLock);
756        mHardwareStatus = AUDIO_SET_PARAMETER;
757        status_t final_result = NO_ERROR;
758        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
759            audio_hw_device_t *dev = mAudioHwDevs[i];
760            result = dev->set_parameters(dev, keyValuePairs.string());
761            final_result = result ?: final_result;
762        }
763        mHardwareStatus = AUDIO_HW_IDLE;
764        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
765        AudioParameter param = AudioParameter(keyValuePairs);
766        String8 value;
767        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
768            Mutex::Autolock _l(mLock);
769            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
770            if (mBtNrecIsOff != btNrecIsOff) {
771                for (size_t i = 0; i < mRecordThreads.size(); i++) {
772                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
773                    RecordThread::RecordTrack *track = thread->track();
774                    if (track != NULL) {
775                        audio_devices_t device = (audio_devices_t)(
776                                thread->device() & AUDIO_DEVICE_IN_ALL);
777                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
778                        thread->setEffectSuspended(FX_IID_AEC,
779                                                   suspend,
780                                                   track->sessionId());
781                        thread->setEffectSuspended(FX_IID_NS,
782                                                   suspend,
783                                                   track->sessionId());
784                    }
785                }
786                mBtNrecIsOff = btNrecIsOff;
787            }
788        }
789        return final_result;
790    }
791
792    // hold a strong ref on thread in case closeOutput() or closeInput() is called
793    // and the thread is exited once the lock is released
794    sp<ThreadBase> thread;
795    {
796        Mutex::Autolock _l(mLock);
797        thread = checkPlaybackThread_l(ioHandle);
798        if (thread == NULL) {
799            thread = checkRecordThread_l(ioHandle);
800        } else if (thread.get() == primaryPlaybackThread_l()) {
801            // indicate output device change to all input threads for pre processing
802            AudioParameter param = AudioParameter(keyValuePairs);
803            int value;
804            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
805                for (size_t i = 0; i < mRecordThreads.size(); i++) {
806                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
807                }
808            }
809        }
810    }
811    if (thread != NULL) {
812        result = thread->setParameters(keyValuePairs);
813        return result;
814    }
815    return BAD_VALUE;
816}
817
818String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
819{
820//    ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
821//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
822
823    if (ioHandle == 0) {
824        String8 out_s8;
825
826        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
827            audio_hw_device_t *dev = mAudioHwDevs[i];
828            char *s = dev->get_parameters(dev, keys.string());
829            out_s8 += String8(s);
830            free(s);
831        }
832        return out_s8;
833    }
834
835    Mutex::Autolock _l(mLock);
836
837    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
838    if (playbackThread != NULL) {
839        return playbackThread->getParameters(keys);
840    }
841    RecordThread *recordThread = checkRecordThread_l(ioHandle);
842    if (recordThread != NULL) {
843        return recordThread->getParameters(keys);
844    }
845    return String8("");
846}
847
848size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
849{
850    status_t ret = initCheck();
851    if (ret != NO_ERROR) {
852        return 0;
853    }
854
855    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
856}
857
858unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
859{
860    if (ioHandle == 0) {
861        return 0;
862    }
863
864    Mutex::Autolock _l(mLock);
865
866    RecordThread *recordThread = checkRecordThread_l(ioHandle);
867    if (recordThread != NULL) {
868        return recordThread->getInputFramesLost();
869    }
870    return 0;
871}
872
873status_t AudioFlinger::setVoiceVolume(float value)
874{
875    status_t ret = initCheck();
876    if (ret != NO_ERROR) {
877        return ret;
878    }
879
880    // check calling permissions
881    if (!settingsAllowed()) {
882        return PERMISSION_DENIED;
883    }
884
885    AutoMutex lock(mHardwareLock);
886    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
887    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
888    mHardwareStatus = AUDIO_HW_IDLE;
889
890    return ret;
891}
892
893status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
894{
895    status_t status;
896
897    Mutex::Autolock _l(mLock);
898
899    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
900    if (playbackThread != NULL) {
901        return playbackThread->getRenderPosition(halFrames, dspFrames);
902    }
903
904    return BAD_VALUE;
905}
906
907void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
908{
909
910    Mutex::Autolock _l(mLock);
911
912    int pid = IPCThreadState::self()->getCallingPid();
913    if (mNotificationClients.indexOfKey(pid) < 0) {
914        sp<NotificationClient> notificationClient = new NotificationClient(this,
915                                                                            client,
916                                                                            pid);
917        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
918
919        mNotificationClients.add(pid, notificationClient);
920
921        sp<IBinder> binder = client->asBinder();
922        binder->linkToDeath(notificationClient);
923
924        // the config change is always sent from playback or record threads to avoid deadlock
925        // with AudioSystem::gLock
926        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
927            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
928        }
929
930        for (size_t i = 0; i < mRecordThreads.size(); i++) {
931            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
932        }
933    }
934}
935
936void AudioFlinger::removeNotificationClient(pid_t pid)
937{
938    Mutex::Autolock _l(mLock);
939
940    int index = mNotificationClients.indexOfKey(pid);
941    if (index >= 0) {
942        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
943        ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
944        mNotificationClients.removeItem(pid);
945    }
946
947    ALOGV("%d died, releasing its sessions", pid);
948    int num = mAudioSessionRefs.size();
949    bool removed = false;
950    for (int i = 0; i< num; i++) {
951        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
952        ALOGV(" pid %d @ %d", ref->pid, i);
953        if (ref->pid == pid) {
954            ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
955            mAudioSessionRefs.removeAt(i);
956            delete ref;
957            removed = true;
958            i--;
959            num--;
960        }
961    }
962    if (removed) {
963        purgeStaleEffects_l();
964    }
965}
966
967// audioConfigChanged_l() must be called with AudioFlinger::mLock held
968void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
969{
970    size_t size = mNotificationClients.size();
971    for (size_t i = 0; i < size; i++) {
972        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
973    }
974}
975
976// removeClient_l() must be called with AudioFlinger::mLock held
977void AudioFlinger::removeClient_l(pid_t pid)
978{
979    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
980    mClients.removeItem(pid);
981}
982
983
984// ----------------------------------------------------------------------------
985
986AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
987    :   Thread(false),
988        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
989        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
990        mDevice(device)
991{
992    mDeathRecipient = new PMDeathRecipient(this);
993}
994
995AudioFlinger::ThreadBase::~ThreadBase()
996{
997    mParamCond.broadcast();
998    // do not lock the mutex in destructor
999    releaseWakeLock_l();
1000    if (mPowerManager != 0) {
1001        sp<IBinder> binder = mPowerManager->asBinder();
1002        binder->unlinkToDeath(mDeathRecipient);
1003    }
1004}
1005
1006void AudioFlinger::ThreadBase::exit()
1007{
1008    // keep a strong ref on ourself so that we won't get
1009    // destroyed in the middle of requestExitAndWait()
1010    sp <ThreadBase> strongMe = this;
1011
1012    ALOGV("ThreadBase::exit");
1013    {
1014        AutoMutex lock(mLock);
1015        mExiting = true;
1016        requestExit();
1017        mWaitWorkCV.signal();
1018    }
1019    requestExitAndWait();
1020}
1021
1022uint32_t AudioFlinger::ThreadBase::sampleRate() const
1023{
1024    return mSampleRate;
1025}
1026
1027int AudioFlinger::ThreadBase::channelCount() const
1028{
1029    return (int)mChannelCount;
1030}
1031
1032uint32_t AudioFlinger::ThreadBase::format() const
1033{
1034    return mFormat;
1035}
1036
1037size_t AudioFlinger::ThreadBase::frameCount() const
1038{
1039    return mFrameCount;
1040}
1041
1042status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1043{
1044    status_t status;
1045
1046    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1047    Mutex::Autolock _l(mLock);
1048
1049    mNewParameters.add(keyValuePairs);
1050    mWaitWorkCV.signal();
1051    // wait condition with timeout in case the thread loop has exited
1052    // before the request could be processed
1053    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
1054        status = mParamStatus;
1055        mWaitWorkCV.signal();
1056    } else {
1057        status = TIMED_OUT;
1058    }
1059    return status;
1060}
1061
1062void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1063{
1064    Mutex::Autolock _l(mLock);
1065    sendConfigEvent_l(event, param);
1066}
1067
1068// sendConfigEvent_l() must be called with ThreadBase::mLock held
1069void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1070{
1071    ConfigEvent configEvent;
1072    configEvent.mEvent = event;
1073    configEvent.mParam = param;
1074    mConfigEvents.add(configEvent);
1075    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1076    mWaitWorkCV.signal();
1077}
1078
1079void AudioFlinger::ThreadBase::processConfigEvents()
1080{
1081    mLock.lock();
1082    while(!mConfigEvents.isEmpty()) {
1083        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1084        ConfigEvent configEvent = mConfigEvents[0];
1085        mConfigEvents.removeAt(0);
1086        // release mLock before locking AudioFlinger mLock: lock order is always
1087        // AudioFlinger then ThreadBase to avoid cross deadlock
1088        mLock.unlock();
1089        mAudioFlinger->mLock.lock();
1090        audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
1091        mAudioFlinger->mLock.unlock();
1092        mLock.lock();
1093    }
1094    mLock.unlock();
1095}
1096
1097status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1098{
1099    const size_t SIZE = 256;
1100    char buffer[SIZE];
1101    String8 result;
1102
1103    bool locked = tryLock(mLock);
1104    if (!locked) {
1105        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1106        write(fd, buffer, strlen(buffer));
1107    }
1108
1109    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1110    result.append(buffer);
1111    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1112    result.append(buffer);
1113    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1114    result.append(buffer);
1115    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1116    result.append(buffer);
1117    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1118    result.append(buffer);
1119    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1120    result.append(buffer);
1121    snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
1122    result.append(buffer);
1123
1124    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1125    result.append(buffer);
1126    result.append(" Index Command");
1127    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1128        snprintf(buffer, SIZE, "\n %02d    ", i);
1129        result.append(buffer);
1130        result.append(mNewParameters[i]);
1131    }
1132
1133    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1134    result.append(buffer);
1135    snprintf(buffer, SIZE, " Index event param\n");
1136    result.append(buffer);
1137    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1138        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
1139        result.append(buffer);
1140    }
1141    result.append("\n");
1142
1143    write(fd, result.string(), result.size());
1144
1145    if (locked) {
1146        mLock.unlock();
1147    }
1148    return NO_ERROR;
1149}
1150
1151status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1152{
1153    const size_t SIZE = 256;
1154    char buffer[SIZE];
1155    String8 result;
1156
1157    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1158    write(fd, buffer, strlen(buffer));
1159
1160    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1161        sp<EffectChain> chain = mEffectChains[i];
1162        if (chain != 0) {
1163            chain->dump(fd, args);
1164        }
1165    }
1166    return NO_ERROR;
1167}
1168
1169void AudioFlinger::ThreadBase::acquireWakeLock()
1170{
1171    Mutex::Autolock _l(mLock);
1172    acquireWakeLock_l();
1173}
1174
1175void AudioFlinger::ThreadBase::acquireWakeLock_l()
1176{
1177    if (mPowerManager == 0) {
1178        // use checkService() to avoid blocking if power service is not up yet
1179        sp<IBinder> binder =
1180            defaultServiceManager()->checkService(String16("power"));
1181        if (binder == 0) {
1182            ALOGW("Thread %s cannot connect to the power manager service", mName);
1183        } else {
1184            mPowerManager = interface_cast<IPowerManager>(binder);
1185            binder->linkToDeath(mDeathRecipient);
1186        }
1187    }
1188    if (mPowerManager != 0) {
1189        sp<IBinder> binder = new BBinder();
1190        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
1191                                                         binder,
1192                                                         String16(mName));
1193        if (status == NO_ERROR) {
1194            mWakeLockToken = binder;
1195        }
1196        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
1197    }
1198}
1199
1200void AudioFlinger::ThreadBase::releaseWakeLock()
1201{
1202    Mutex::Autolock _l(mLock);
1203    releaseWakeLock_l();
1204}
1205
1206void AudioFlinger::ThreadBase::releaseWakeLock_l()
1207{
1208    if (mWakeLockToken != 0) {
1209        ALOGV("releaseWakeLock_l() %s", mName);
1210        if (mPowerManager != 0) {
1211            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
1212        }
1213        mWakeLockToken.clear();
1214    }
1215}
1216
1217void AudioFlinger::ThreadBase::clearPowerManager()
1218{
1219    Mutex::Autolock _l(mLock);
1220    releaseWakeLock_l();
1221    mPowerManager.clear();
1222}
1223
1224void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
1225{
1226    sp<ThreadBase> thread = mThread.promote();
1227    if (thread != 0) {
1228        thread->clearPowerManager();
1229    }
1230    ALOGW("power manager service died !!!");
1231}
1232
1233void AudioFlinger::ThreadBase::setEffectSuspended(
1234        const effect_uuid_t *type, bool suspend, int sessionId)
1235{
1236    Mutex::Autolock _l(mLock);
1237    setEffectSuspended_l(type, suspend, sessionId);
1238}
1239
1240void AudioFlinger::ThreadBase::setEffectSuspended_l(
1241        const effect_uuid_t *type, bool suspend, int sessionId)
1242{
1243    sp<EffectChain> chain;
1244    chain = getEffectChain_l(sessionId);
1245    if (chain != 0) {
1246        if (type != NULL) {
1247            chain->setEffectSuspended_l(type, suspend);
1248        } else {
1249            chain->setEffectSuspendedAll_l(suspend);
1250        }
1251    }
1252
1253    updateSuspendedSessions_l(type, suspend, sessionId);
1254}
1255
1256void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1257{
1258    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1259    if (index < 0) {
1260        return;
1261    }
1262
1263    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1264            mSuspendedSessions.editValueAt(index);
1265
1266    for (size_t i = 0; i < sessionEffects.size(); i++) {
1267        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1268        for (int j = 0; j < desc->mRefCount; j++) {
1269            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1270                chain->setEffectSuspendedAll_l(true);
1271            } else {
1272                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1273                     desc->mType.timeLow);
1274                chain->setEffectSuspended_l(&desc->mType, true);
1275            }
1276        }
1277    }
1278}
1279
1280void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1281                                                         bool suspend,
1282                                                         int sessionId)
1283{
1284    int index = mSuspendedSessions.indexOfKey(sessionId);
1285
1286    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1287
1288    if (suspend) {
1289        if (index >= 0) {
1290            sessionEffects = mSuspendedSessions.editValueAt(index);
1291        } else {
1292            mSuspendedSessions.add(sessionId, sessionEffects);
1293        }
1294    } else {
1295        if (index < 0) {
1296            return;
1297        }
1298        sessionEffects = mSuspendedSessions.editValueAt(index);
1299    }
1300
1301
1302    int key = EffectChain::kKeyForSuspendAll;
1303    if (type != NULL) {
1304        key = type->timeLow;
1305    }
1306    index = sessionEffects.indexOfKey(key);
1307
1308    sp <SuspendedSessionDesc> desc;
1309    if (suspend) {
1310        if (index >= 0) {
1311            desc = sessionEffects.valueAt(index);
1312        } else {
1313            desc = new SuspendedSessionDesc();
1314            if (type != NULL) {
1315                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1316            }
1317            sessionEffects.add(key, desc);
1318            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1319        }
1320        desc->mRefCount++;
1321    } else {
1322        if (index < 0) {
1323            return;
1324        }
1325        desc = sessionEffects.valueAt(index);
1326        if (--desc->mRefCount == 0) {
1327            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1328            sessionEffects.removeItemsAt(index);
1329            if (sessionEffects.isEmpty()) {
1330                ALOGV("updateSuspendedSessions_l() restore removing session %d",
1331                                 sessionId);
1332                mSuspendedSessions.removeItem(sessionId);
1333            }
1334        }
1335    }
1336    if (!sessionEffects.isEmpty()) {
1337        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1338    }
1339}
1340
1341void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1342                                                            bool enabled,
1343                                                            int sessionId)
1344{
1345    Mutex::Autolock _l(mLock);
1346    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1347}
1348
1349void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1350                                                            bool enabled,
1351                                                            int sessionId)
1352{
1353    if (mType != RECORD) {
1354        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1355        // another session. This gives the priority to well behaved effect control panels
1356        // and applications not using global effects.
1357        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1358            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1359        }
1360    }
1361
1362    sp<EffectChain> chain = getEffectChain_l(sessionId);
1363    if (chain != 0) {
1364        chain->checkSuspendOnEffectEnabled(effect, enabled);
1365    }
1366}
1367
1368// ----------------------------------------------------------------------------
1369
1370AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1371                                             AudioStreamOut* output,
1372                                             int id,
1373                                             uint32_t device)
1374    :   ThreadBase(audioFlinger, id, device),
1375        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output),
1376        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1377{
1378    snprintf(mName, kNameLength, "AudioOut_%d", id);
1379
1380    readOutputParameters();
1381
1382    mMasterVolume = mAudioFlinger->masterVolume();
1383    mMasterMute = mAudioFlinger->masterMute();
1384
1385    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
1386        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1387        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1388        mStreamTypes[stream].valid = true;
1389    }
1390}
1391
1392AudioFlinger::PlaybackThread::~PlaybackThread()
1393{
1394    delete [] mMixBuffer;
1395}
1396
1397status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1398{
1399    dumpInternals(fd, args);
1400    dumpTracks(fd, args);
1401    dumpEffectChains(fd, args);
1402    return NO_ERROR;
1403}
1404
1405status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1406{
1407    const size_t SIZE = 256;
1408    char buffer[SIZE];
1409    String8 result;
1410
1411    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1412    result.append(buffer);
1413    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1414    for (size_t i = 0; i < mTracks.size(); ++i) {
1415        sp<Track> track = mTracks[i];
1416        if (track != 0) {
1417            track->dump(buffer, SIZE);
1418            result.append(buffer);
1419        }
1420    }
1421
1422    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1423    result.append(buffer);
1424    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1425    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1426        wp<Track> wTrack = mActiveTracks[i];
1427        if (wTrack != 0) {
1428            sp<Track> track = wTrack.promote();
1429            if (track != 0) {
1430                track->dump(buffer, SIZE);
1431                result.append(buffer);
1432            }
1433        }
1434    }
1435    write(fd, result.string(), result.size());
1436    return NO_ERROR;
1437}
1438
1439status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1440{
1441    const size_t SIZE = 256;
1442    char buffer[SIZE];
1443    String8 result;
1444
1445    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1446    result.append(buffer);
1447    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1448    result.append(buffer);
1449    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1450    result.append(buffer);
1451    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1452    result.append(buffer);
1453    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1454    result.append(buffer);
1455    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1456    result.append(buffer);
1457    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1458    result.append(buffer);
1459    write(fd, result.string(), result.size());
1460
1461    dumpBase(fd, args);
1462
1463    return NO_ERROR;
1464}
1465
1466// Thread virtuals
1467status_t AudioFlinger::PlaybackThread::readyToRun()
1468{
1469    status_t status = initCheck();
1470    if (status == NO_ERROR) {
1471        ALOGI("AudioFlinger's thread %p ready to run", this);
1472    } else {
1473        ALOGE("No working audio driver found.");
1474    }
1475    return status;
1476}
1477
1478void AudioFlinger::PlaybackThread::onFirstRef()
1479{
1480    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1481}
1482
1483// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1484sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1485        const sp<AudioFlinger::Client>& client,
1486        int streamType,
1487        uint32_t sampleRate,
1488        uint32_t format,
1489        uint32_t channelMask,
1490        int frameCount,
1491        const sp<IMemory>& sharedBuffer,
1492        int sessionId,
1493        status_t *status)
1494{
1495    sp<Track> track;
1496    status_t lStatus;
1497
1498    if (mType == DIRECT) {
1499        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1500            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1501                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1502                        "for output %p with format %d",
1503                        sampleRate, format, channelMask, mOutput, mFormat);
1504                lStatus = BAD_VALUE;
1505                goto Exit;
1506            }
1507        }
1508    } else {
1509        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1510        if (sampleRate > mSampleRate*2) {
1511            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1512            lStatus = BAD_VALUE;
1513            goto Exit;
1514        }
1515    }
1516
1517    lStatus = initCheck();
1518    if (lStatus != NO_ERROR) {
1519        ALOGE("Audio driver not initialized.");
1520        goto Exit;
1521    }
1522
1523    { // scope for mLock
1524        Mutex::Autolock _l(mLock);
1525
1526        // all tracks in same audio session must share the same routing strategy otherwise
1527        // conflicts will happen when tracks are moved from one output to another by audio policy
1528        // manager
1529        uint32_t strategy =
1530                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1531        for (size_t i = 0; i < mTracks.size(); ++i) {
1532            sp<Track> t = mTracks[i];
1533            if (t != 0) {
1534                uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
1535                if (sessionId == t->sessionId() && strategy != actual) {
1536                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1537                            strategy, actual);
1538                    lStatus = BAD_VALUE;
1539                    goto Exit;
1540                }
1541            }
1542        }
1543
1544        track = new Track(this, client, streamType, sampleRate, format,
1545                channelMask, frameCount, sharedBuffer, sessionId);
1546        if (track->getCblk() == NULL || track->name() < 0) {
1547            lStatus = NO_MEMORY;
1548            goto Exit;
1549        }
1550        mTracks.add(track);
1551
1552        sp<EffectChain> chain = getEffectChain_l(sessionId);
1553        if (chain != 0) {
1554            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1555            track->setMainBuffer(chain->inBuffer());
1556            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1557            chain->incTrackCnt();
1558        }
1559
1560        // invalidate track immediately if the stream type was moved to another thread since
1561        // createTrack() was called by the client process.
1562        if (!mStreamTypes[streamType].valid) {
1563            ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1564                 this, streamType);
1565            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1566        }
1567    }
1568    lStatus = NO_ERROR;
1569
1570Exit:
1571    if(status) {
1572        *status = lStatus;
1573    }
1574    return track;
1575}
1576
1577uint32_t AudioFlinger::PlaybackThread::latency() const
1578{
1579    Mutex::Autolock _l(mLock);
1580    if (initCheck() == NO_ERROR) {
1581        return mOutput->stream->get_latency(mOutput->stream);
1582    } else {
1583        return 0;
1584    }
1585}
1586
1587status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1588{
1589    mMasterVolume = value;
1590    return NO_ERROR;
1591}
1592
1593status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1594{
1595    mMasterMute = muted;
1596    return NO_ERROR;
1597}
1598
1599float AudioFlinger::PlaybackThread::masterVolume() const
1600{
1601    return mMasterVolume;
1602}
1603
1604bool AudioFlinger::PlaybackThread::masterMute() const
1605{
1606    return mMasterMute;
1607}
1608
1609status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
1610{
1611    mStreamTypes[stream].volume = value;
1612    return NO_ERROR;
1613}
1614
1615status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
1616{
1617    mStreamTypes[stream].mute = muted;
1618    return NO_ERROR;
1619}
1620
1621float AudioFlinger::PlaybackThread::streamVolume(int stream) const
1622{
1623    return mStreamTypes[stream].volume;
1624}
1625
1626bool AudioFlinger::PlaybackThread::streamMute(int stream) const
1627{
1628    return mStreamTypes[stream].mute;
1629}
1630
1631// addTrack_l() must be called with ThreadBase::mLock held
1632status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1633{
1634    status_t status = ALREADY_EXISTS;
1635
1636    // set retry count for buffer fill
1637    track->mRetryCount = kMaxTrackStartupRetries;
1638    if (mActiveTracks.indexOf(track) < 0) {
1639        // the track is newly added, make sure it fills up all its
1640        // buffers before playing. This is to ensure the client will
1641        // effectively get the latency it requested.
1642        track->mFillingUpStatus = Track::FS_FILLING;
1643        track->mResetDone = false;
1644        mActiveTracks.add(track);
1645        if (track->mainBuffer() != mMixBuffer) {
1646            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1647            if (chain != 0) {
1648                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1649                chain->incActiveTrackCnt();
1650            }
1651        }
1652
1653        status = NO_ERROR;
1654    }
1655
1656    ALOGV("mWaitWorkCV.broadcast");
1657    mWaitWorkCV.broadcast();
1658
1659    return status;
1660}
1661
1662// destroyTrack_l() must be called with ThreadBase::mLock held
1663void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1664{
1665    track->mState = TrackBase::TERMINATED;
1666    if (mActiveTracks.indexOf(track) < 0) {
1667        removeTrack_l(track);
1668    }
1669}
1670
1671void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1672{
1673    mTracks.remove(track);
1674    deleteTrackName_l(track->name());
1675    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1676    if (chain != 0) {
1677        chain->decTrackCnt();
1678    }
1679}
1680
1681String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1682{
1683    String8 out_s8 = String8("");
1684    char *s;
1685
1686    Mutex::Autolock _l(mLock);
1687    if (initCheck() != NO_ERROR) {
1688        return out_s8;
1689    }
1690
1691    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1692    out_s8 = String8(s);
1693    free(s);
1694    return out_s8;
1695}
1696
1697// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1698void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1699    AudioSystem::OutputDescriptor desc;
1700    void *param2 = 0;
1701
1702    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1703
1704    switch (event) {
1705    case AudioSystem::OUTPUT_OPENED:
1706    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1707        desc.channels = mChannelMask;
1708        desc.samplingRate = mSampleRate;
1709        desc.format = mFormat;
1710        desc.frameCount = mFrameCount;
1711        desc.latency = latency();
1712        param2 = &desc;
1713        break;
1714
1715    case AudioSystem::STREAM_CONFIG_CHANGED:
1716        param2 = &param;
1717    case AudioSystem::OUTPUT_CLOSED:
1718    default:
1719        break;
1720    }
1721    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1722}
1723
1724void AudioFlinger::PlaybackThread::readOutputParameters()
1725{
1726    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1727    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1728    mChannelCount = (uint16_t)popcount(mChannelMask);
1729    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1730    mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
1731    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1732
1733    // FIXME - Current mixer implementation only supports stereo output: Always
1734    // Allocate a stereo buffer even if HW output is mono.
1735    if (mMixBuffer != NULL) delete[] mMixBuffer;
1736    mMixBuffer = new int16_t[mFrameCount * 2];
1737    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1738
1739    // force reconfiguration of effect chains and engines to take new buffer size and audio
1740    // parameters into account
1741    // Note that mLock is not held when readOutputParameters() is called from the constructor
1742    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1743    // matter.
1744    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1745    Vector< sp<EffectChain> > effectChains = mEffectChains;
1746    for (size_t i = 0; i < effectChains.size(); i ++) {
1747        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1748    }
1749}
1750
1751status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1752{
1753    if (halFrames == 0 || dspFrames == 0) {
1754        return BAD_VALUE;
1755    }
1756    Mutex::Autolock _l(mLock);
1757    if (initCheck() != NO_ERROR) {
1758        return INVALID_OPERATION;
1759    }
1760    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1761
1762    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1763}
1764
1765uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1766{
1767    Mutex::Autolock _l(mLock);
1768    uint32_t result = 0;
1769    if (getEffectChain_l(sessionId) != 0) {
1770        result = EFFECT_SESSION;
1771    }
1772
1773    for (size_t i = 0; i < mTracks.size(); ++i) {
1774        sp<Track> track = mTracks[i];
1775        if (sessionId == track->sessionId() &&
1776                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1777            result |= TRACK_SESSION;
1778            break;
1779        }
1780    }
1781
1782    return result;
1783}
1784
1785uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1786{
1787    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1788    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1789    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1790        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1791    }
1792    for (size_t i = 0; i < mTracks.size(); i++) {
1793        sp<Track> track = mTracks[i];
1794        if (sessionId == track->sessionId() &&
1795                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1796            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1797        }
1798    }
1799    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1800}
1801
1802
1803AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1804{
1805    Mutex::Autolock _l(mLock);
1806    return mOutput;
1807}
1808
1809AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1810{
1811    Mutex::Autolock _l(mLock);
1812    AudioStreamOut *output = mOutput;
1813    mOutput = NULL;
1814    return output;
1815}
1816
1817// this method must always be called either with ThreadBase mLock held or inside the thread loop
1818audio_stream_t* AudioFlinger::PlaybackThread::stream()
1819{
1820    if (mOutput == NULL) {
1821        return NULL;
1822    }
1823    return &mOutput->stream->common;
1824}
1825
1826uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
1827{
1828    // A2DP output latency is not due only to buffering capacity. It also reflects encoding,
1829    // decoding and transfer time. So sleeping for half of the latency would likely cause
1830    // underruns
1831    if (audio_is_a2dp_device((audio_devices_t)mDevice)) {
1832        return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000);
1833    } else {
1834        return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
1835    }
1836}
1837
1838// ----------------------------------------------------------------------------
1839
1840AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1841    :   PlaybackThread(audioFlinger, output, id, device),
1842        mAudioMixer(NULL)
1843{
1844    mType = ThreadBase::MIXER;
1845    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1846
1847    // FIXME - Current mixer implementation only supports stereo output
1848    if (mChannelCount == 1) {
1849        ALOGE("Invalid audio hardware channel count");
1850    }
1851}
1852
1853AudioFlinger::MixerThread::~MixerThread()
1854{
1855    delete mAudioMixer;
1856}
1857
1858bool AudioFlinger::MixerThread::threadLoop()
1859{
1860    Vector< sp<Track> > tracksToRemove;
1861    uint32_t mixerStatus = MIXER_IDLE;
1862    nsecs_t standbyTime = systemTime();
1863    size_t mixBufferSize = mFrameCount * mFrameSize;
1864    // FIXME: Relaxed timing because of a certain device that can't meet latency
1865    // Should be reduced to 2x after the vendor fixes the driver issue
1866    // increase threshold again due to low power audio mode. The way this warning threshold is
1867    // calculated and its usefulness should be reconsidered anyway.
1868    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1869    nsecs_t lastWarning = 0;
1870    bool longStandbyExit = false;
1871    uint32_t activeSleepTime = activeSleepTimeUs();
1872    uint32_t idleSleepTime = idleSleepTimeUs();
1873    uint32_t sleepTime = idleSleepTime;
1874    uint32_t sleepTimeShift = 0;
1875    Vector< sp<EffectChain> > effectChains;
1876#ifdef DEBUG_CPU_USAGE
1877    ThreadCpuUsage cpu;
1878    const CentralTendencyStatistics& stats = cpu.statistics();
1879#endif
1880
1881    acquireWakeLock();
1882
1883    while (!exitPending())
1884    {
1885#ifdef DEBUG_CPU_USAGE
1886        cpu.sampleAndEnable();
1887        unsigned n = stats.n();
1888        // cpu.elapsed() is expensive, so don't call it every loop
1889        if ((n & 127) == 1) {
1890            long long elapsed = cpu.elapsed();
1891            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1892                double perLoop = elapsed / (double) n;
1893                double perLoop100 = perLoop * 0.01;
1894                double mean = stats.mean();
1895                double stddev = stats.stddev();
1896                double minimum = stats.minimum();
1897                double maximum = stats.maximum();
1898                cpu.resetStatistics();
1899                ALOGI("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",
1900                        elapsed * .000000001, n, perLoop * .000001,
1901                        mean * .001,
1902                        stddev * .001,
1903                        minimum * .001,
1904                        maximum * .001,
1905                        mean / perLoop100,
1906                        stddev / perLoop100,
1907                        minimum / perLoop100,
1908                        maximum / perLoop100);
1909            }
1910        }
1911#endif
1912        processConfigEvents();
1913
1914        mixerStatus = MIXER_IDLE;
1915        { // scope for mLock
1916
1917            Mutex::Autolock _l(mLock);
1918
1919            if (checkForNewParameters_l()) {
1920                mixBufferSize = mFrameCount * mFrameSize;
1921                // FIXME: Relaxed timing because of a certain device that can't meet latency
1922                // Should be reduced to 2x after the vendor fixes the driver issue
1923                // increase threshold again due to low power audio mode. The way this warning
1924                // threshold is calculated and its usefulness should be reconsidered anyway.
1925                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1926                activeSleepTime = activeSleepTimeUs();
1927                idleSleepTime = idleSleepTimeUs();
1928            }
1929
1930            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1931
1932            // put audio hardware into standby after short delay
1933            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1934                        mSuspended)) {
1935                if (!mStandby) {
1936                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1937                    mOutput->stream->common.standby(&mOutput->stream->common);
1938                    mStandby = true;
1939                    mBytesWritten = 0;
1940                }
1941
1942                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1943                    // we're about to wait, flush the binder command buffer
1944                    IPCThreadState::self()->flushCommands();
1945
1946                    if (exitPending()) break;
1947
1948                    releaseWakeLock_l();
1949                    // wait until we have something to do...
1950                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1951                    mWaitWorkCV.wait(mLock);
1952                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1953                    acquireWakeLock_l();
1954
1955                    if (mMasterMute == false) {
1956                        char value[PROPERTY_VALUE_MAX];
1957                        property_get("ro.audio.silent", value, "0");
1958                        if (atoi(value)) {
1959                            ALOGD("Silence is golden");
1960                            setMasterMute(true);
1961                        }
1962                    }
1963
1964                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1965                    sleepTime = idleSleepTime;
1966                    sleepTimeShift = 0;
1967                    continue;
1968                }
1969            }
1970
1971            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1972
1973            // prevent any changes in effect chain list and in each effect chain
1974            // during mixing and effect process as the audio buffers could be deleted
1975            // or modified if an effect is created or deleted
1976            lockEffectChains_l(effectChains);
1977        }
1978
1979        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1980            // mix buffers...
1981            mAudioMixer->process();
1982            sleepTime = 0;
1983            // increase sleep time progressively when application underrun condition clears
1984            if (sleepTimeShift > 0) {
1985                sleepTimeShift--;
1986            }
1987            standbyTime = systemTime() + kStandbyTimeInNsecs;
1988            //TODO: delay standby when effects have a tail
1989        } else {
1990            // If no tracks are ready, sleep once for the duration of an output
1991            // buffer size, then write 0s to the output
1992            if (sleepTime == 0) {
1993                if (mixerStatus == MIXER_TRACKS_ENABLED) {
1994                    sleepTime = activeSleepTime >> sleepTimeShift;
1995                    if (sleepTime < kMinThreadSleepTimeUs) {
1996                        sleepTime = kMinThreadSleepTimeUs;
1997                    }
1998                    // reduce sleep time in case of consecutive application underruns to avoid
1999                    // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
2000                    // duration we would end up writing less data than needed by the audio HAL if
2001                    // the condition persists.
2002                    if (sleepTimeShift < kMaxThreadSleepTimeShift) {
2003                        sleepTimeShift++;
2004                    }
2005                } else {
2006                    sleepTime = idleSleepTime;
2007                }
2008            } else if (mBytesWritten != 0 ||
2009                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
2010                memset (mMixBuffer, 0, mixBufferSize);
2011                sleepTime = 0;
2012                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
2013            }
2014            // TODO add standby time extension fct of effect tail
2015        }
2016
2017        if (mSuspended) {
2018            sleepTime = suspendSleepTimeUs();
2019        }
2020        // sleepTime == 0 means we must write to audio hardware
2021        if (sleepTime == 0) {
2022            for (size_t i = 0; i < effectChains.size(); i ++) {
2023                effectChains[i]->process_l();
2024            }
2025            // enable changes in effect chain
2026            unlockEffectChains(effectChains);
2027            mLastWriteTime = systemTime();
2028            mInWrite = true;
2029            mBytesWritten += mixBufferSize;
2030
2031            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2032            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2033            mNumWrites++;
2034            mInWrite = false;
2035            nsecs_t now = systemTime();
2036            nsecs_t delta = now - mLastWriteTime;
2037            if (!mStandby && delta > maxPeriod) {
2038                mNumDelayedWrites++;
2039                if ((now - lastWarning) > kWarningThrottleNs) {
2040                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2041                            ns2ms(delta), mNumDelayedWrites, this);
2042                    lastWarning = now;
2043                }
2044                if (mStandby) {
2045                    longStandbyExit = true;
2046                }
2047            }
2048            mStandby = false;
2049        } else {
2050            // enable changes in effect chain
2051            unlockEffectChains(effectChains);
2052            usleep(sleepTime);
2053        }
2054
2055        // finally let go of all our tracks, without the lock held
2056        // since we can't guarantee the destructors won't acquire that
2057        // same lock.
2058        tracksToRemove.clear();
2059
2060        // Effect chains will be actually deleted here if they were removed from
2061        // mEffectChains list during mixing or effects processing
2062        effectChains.clear();
2063    }
2064
2065    if (!mStandby) {
2066        mOutput->stream->common.standby(&mOutput->stream->common);
2067    }
2068
2069    releaseWakeLock();
2070
2071    ALOGV("MixerThread %p exiting", this);
2072    return false;
2073}
2074
2075// prepareTracks_l() must be called with ThreadBase::mLock held
2076uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2077{
2078
2079    uint32_t mixerStatus = MIXER_IDLE;
2080    // find out which tracks need to be processed
2081    size_t count = activeTracks.size();
2082    size_t mixedTracks = 0;
2083    size_t tracksWithEffect = 0;
2084
2085    float masterVolume = mMasterVolume;
2086    bool  masterMute = mMasterMute;
2087
2088    if (masterMute) {
2089        masterVolume = 0;
2090    }
2091    // Delegate master volume control to effect in output mix effect chain if needed
2092    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2093    if (chain != 0) {
2094        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2095        chain->setVolume_l(&v, &v);
2096        masterVolume = (float)((v + (1 << 23)) >> 24);
2097        chain.clear();
2098    }
2099
2100    for (size_t i=0 ; i<count ; i++) {
2101        sp<Track> t = activeTracks[i].promote();
2102        if (t == 0) continue;
2103
2104        // this const just means the local variable doesn't change
2105        Track* const track = t.get();
2106        audio_track_cblk_t* cblk = track->cblk();
2107
2108        // The first time a track is added we wait
2109        // for all its buffers to be filled before processing it
2110        int name = track->name();
2111        // make sure that we have enough frames to mix one full buffer.
2112        // enforce this condition only once to enable draining the buffer in case the client
2113        // app does not call stop() and relies on underrun to stop:
2114        // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed
2115        // during last round
2116        uint32_t minFrames = 1;
2117        if (!track->isStopped() && !track->isPausing() &&
2118                (track->mRetryCount >= kMaxTrackRetries)) {
2119            if (t->sampleRate() == (int)mSampleRate) {
2120                minFrames = mFrameCount;
2121            } else {
2122                // +1 for rounding and +1 for additional sample needed for interpolation
2123                minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
2124                // add frames already consumed but not yet released by the resampler
2125                // because cblk->framesReady() will  include these frames
2126                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
2127                // the minimum track buffer size is normally twice the number of frames necessary
2128                // to fill one buffer and the resampler should not leave more than one buffer worth
2129                // of unreleased frames after each pass, but just in case...
2130                ALOG_ASSERT(minFrames <= cblk->frameCount);
2131            }
2132        }
2133        if ((cblk->framesReady() >= minFrames) && track->isReady() &&
2134                !track->isPaused() && !track->isTerminated())
2135        {
2136            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", 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                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
2150                            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(name, 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(name, track);
2216            mAudioMixer->enable(name);
2217
2218            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left);
2219            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right);
2220            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux);
2221            mAudioMixer->setParameter(
2222                name,
2223                AudioMixer::TRACK,
2224                AudioMixer::FORMAT, (void *)track->format());
2225            mAudioMixer->setParameter(
2226                name,
2227                AudioMixer::TRACK,
2228                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2229            mAudioMixer->setParameter(
2230                name,
2231                AudioMixer::RESAMPLE,
2232                AudioMixer::SAMPLE_RATE,
2233                (void *)(cblk->sampleRate));
2234            mAudioMixer->setParameter(
2235                name,
2236                AudioMixer::TRACK,
2237                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2238            mAudioMixer->setParameter(
2239                name,
2240                AudioMixer::TRACK,
2241                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2242
2243            // reset retry count
2244            track->mRetryCount = kMaxTrackRetries;
2245            mixerStatus = MIXER_TRACKS_READY;
2246        } else {
2247            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
2248            if (track->isStopped()) {
2249                track->reset();
2250            }
2251            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2252                // We have consumed all the buffers of this track.
2253                // Remove it from the list of active tracks.
2254                tracksToRemove->add(track);
2255            } else {
2256                // No buffers for this track. Give it a few chances to
2257                // fill a buffer, then remove it from active list.
2258                if (--(track->mRetryCount) <= 0) {
2259                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
2260                    tracksToRemove->add(track);
2261                    // indicate to client process that the track was disabled because of underrun
2262                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2263                } else if (mixerStatus != MIXER_TRACKS_READY) {
2264                    mixerStatus = MIXER_TRACKS_ENABLED;
2265                }
2266            }
2267            mAudioMixer->disable(name);
2268        }
2269    }
2270
2271    // remove all the tracks that need to be...
2272    count = tracksToRemove->size();
2273    if (CC_UNLIKELY(count)) {
2274        for (size_t i=0 ; i<count ; i++) {
2275            const sp<Track>& track = tracksToRemove->itemAt(i);
2276            mActiveTracks.remove(track);
2277            if (track->mainBuffer() != mMixBuffer) {
2278                chain = getEffectChain_l(track->sessionId());
2279                if (chain != 0) {
2280                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2281                    chain->decActiveTrackCnt();
2282                }
2283            }
2284            if (track->isTerminated()) {
2285                removeTrack_l(track);
2286            }
2287        }
2288    }
2289
2290    // mix buffer must be cleared if all tracks are connected to an
2291    // effect chain as in this case the mixer will not write to
2292    // mix buffer and track effects will accumulate into it
2293    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2294        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2295    }
2296
2297    return mixerStatus;
2298}
2299
2300void AudioFlinger::MixerThread::invalidateTracks(int streamType)
2301{
2302    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2303            this,  streamType, mTracks.size());
2304    Mutex::Autolock _l(mLock);
2305
2306    size_t size = mTracks.size();
2307    for (size_t i = 0; i < size; i++) {
2308        sp<Track> t = mTracks[i];
2309        if (t->type() == streamType) {
2310            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2311            t->mCblk->cv.signal();
2312        }
2313    }
2314}
2315
2316void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid)
2317{
2318    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2319            this,  streamType, valid);
2320    Mutex::Autolock _l(mLock);
2321
2322    mStreamTypes[streamType].valid = valid;
2323}
2324
2325// getTrackName_l() must be called with ThreadBase::mLock held
2326int AudioFlinger::MixerThread::getTrackName_l()
2327{
2328    return mAudioMixer->getTrackName();
2329}
2330
2331// deleteTrackName_l() must be called with ThreadBase::mLock held
2332void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2333{
2334    ALOGV("remove track (%d) and delete from mixer", name);
2335    mAudioMixer->deleteTrackName(name);
2336}
2337
2338// checkForNewParameters_l() must be called with ThreadBase::mLock held
2339bool AudioFlinger::MixerThread::checkForNewParameters_l()
2340{
2341    bool reconfig = false;
2342
2343    while (!mNewParameters.isEmpty()) {
2344        status_t status = NO_ERROR;
2345        String8 keyValuePair = mNewParameters[0];
2346        AudioParameter param = AudioParameter(keyValuePair);
2347        int value;
2348
2349        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2350            reconfig = true;
2351        }
2352        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2353            if (value != AUDIO_FORMAT_PCM_16_BIT) {
2354                status = BAD_VALUE;
2355            } else {
2356                reconfig = true;
2357            }
2358        }
2359        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2360            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2361                status = BAD_VALUE;
2362            } else {
2363                reconfig = true;
2364            }
2365        }
2366        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2367            // do not accept frame count changes if tracks are open as the track buffer
2368            // size depends on frame count and correct behavior would not be guaranteed
2369            // if frame count is changed after track creation
2370            if (!mTracks.isEmpty()) {
2371                status = INVALID_OPERATION;
2372            } else {
2373                reconfig = true;
2374            }
2375        }
2376        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2377            // when changing the audio output device, call addBatteryData to notify
2378            // the change
2379            if ((int)mDevice != value) {
2380                uint32_t params = 0;
2381                // check whether speaker is on
2382                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2383                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2384                }
2385
2386                int deviceWithoutSpeaker
2387                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2388                // check if any other device (except speaker) is on
2389                if (value & deviceWithoutSpeaker ) {
2390                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2391                }
2392
2393                if (params != 0) {
2394                    addBatteryData(params);
2395                }
2396            }
2397
2398            // forward device change to effects that have requested to be
2399            // aware of attached audio device.
2400            mDevice = (uint32_t)value;
2401            for (size_t i = 0; i < mEffectChains.size(); i++) {
2402                mEffectChains[i]->setDevice_l(mDevice);
2403            }
2404        }
2405
2406        if (status == NO_ERROR) {
2407            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2408                                                    keyValuePair.string());
2409            if (!mStandby && status == INVALID_OPERATION) {
2410               mOutput->stream->common.standby(&mOutput->stream->common);
2411               mStandby = true;
2412               mBytesWritten = 0;
2413               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2414                                                       keyValuePair.string());
2415            }
2416            if (status == NO_ERROR && reconfig) {
2417                delete mAudioMixer;
2418                readOutputParameters();
2419                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2420                for (size_t i = 0; i < mTracks.size() ; i++) {
2421                    int name = getTrackName_l();
2422                    if (name < 0) break;
2423                    mTracks[i]->mName = name;
2424                    // limit track sample rate to 2 x new output sample rate
2425                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2426                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2427                    }
2428                }
2429                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2430            }
2431        }
2432
2433        mNewParameters.removeAt(0);
2434
2435        mParamStatus = status;
2436        mParamCond.signal();
2437        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2438        // already timed out waiting for the status and will never signal the condition.
2439        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2440    }
2441    return reconfig;
2442}
2443
2444status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2445{
2446    const size_t SIZE = 256;
2447    char buffer[SIZE];
2448    String8 result;
2449
2450    PlaybackThread::dumpInternals(fd, args);
2451
2452    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2453    result.append(buffer);
2454    write(fd, result.string(), result.size());
2455    return NO_ERROR;
2456}
2457
2458uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2459{
2460    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2461}
2462
2463uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2464{
2465    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2466}
2467
2468// ----------------------------------------------------------------------------
2469AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2470    :   PlaybackThread(audioFlinger, output, id, device)
2471{
2472    mType = ThreadBase::DIRECT;
2473}
2474
2475AudioFlinger::DirectOutputThread::~DirectOutputThread()
2476{
2477}
2478
2479static inline
2480int32_t mul(int16_t in, int16_t v)
2481{
2482#if defined(__arm__) && !defined(__thumb__)
2483    int32_t out;
2484    asm( "smulbb %[out], %[in], %[v] \n"
2485         : [out]"=r"(out)
2486         : [in]"%r"(in), [v]"r"(v)
2487         : );
2488    return out;
2489#else
2490    return in * int32_t(v);
2491#endif
2492}
2493
2494void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2495{
2496    // Do not apply volume on compressed audio
2497    if (!audio_is_linear_pcm(mFormat)) {
2498        return;
2499    }
2500
2501    // convert to signed 16 bit before volume calculation
2502    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2503        size_t count = mFrameCount * mChannelCount;
2504        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2505        int16_t *dst = mMixBuffer + count-1;
2506        while(count--) {
2507            *dst-- = (int16_t)(*src--^0x80) << 8;
2508        }
2509    }
2510
2511    size_t frameCount = mFrameCount;
2512    int16_t *out = mMixBuffer;
2513    if (ramp) {
2514        if (mChannelCount == 1) {
2515            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2516            int32_t vlInc = d / (int32_t)frameCount;
2517            int32_t vl = ((int32_t)mLeftVolShort << 16);
2518            do {
2519                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2520                out++;
2521                vl += vlInc;
2522            } while (--frameCount);
2523
2524        } else {
2525            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2526            int32_t vlInc = d / (int32_t)frameCount;
2527            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2528            int32_t vrInc = d / (int32_t)frameCount;
2529            int32_t vl = ((int32_t)mLeftVolShort << 16);
2530            int32_t vr = ((int32_t)mRightVolShort << 16);
2531            do {
2532                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2533                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2534                out += 2;
2535                vl += vlInc;
2536                vr += vrInc;
2537            } while (--frameCount);
2538        }
2539    } else {
2540        if (mChannelCount == 1) {
2541            do {
2542                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2543                out++;
2544            } while (--frameCount);
2545        } else {
2546            do {
2547                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2548                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2549                out += 2;
2550            } while (--frameCount);
2551        }
2552    }
2553
2554    // convert back to unsigned 8 bit after volume calculation
2555    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2556        size_t count = mFrameCount * mChannelCount;
2557        int16_t *src = mMixBuffer;
2558        uint8_t *dst = (uint8_t *)mMixBuffer;
2559        while(count--) {
2560            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2561        }
2562    }
2563
2564    mLeftVolShort = leftVol;
2565    mRightVolShort = rightVol;
2566}
2567
2568bool AudioFlinger::DirectOutputThread::threadLoop()
2569{
2570    uint32_t mixerStatus = MIXER_IDLE;
2571    sp<Track> trackToRemove;
2572    sp<Track> activeTrack;
2573    nsecs_t standbyTime = systemTime();
2574    int8_t *curBuf;
2575    size_t mixBufferSize = mFrameCount*mFrameSize;
2576    uint32_t activeSleepTime = activeSleepTimeUs();
2577    uint32_t idleSleepTime = idleSleepTimeUs();
2578    uint32_t sleepTime = idleSleepTime;
2579    // use shorter standby delay as on normal output to release
2580    // hardware resources as soon as possible
2581    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2582
2583    acquireWakeLock();
2584
2585    while (!exitPending())
2586    {
2587        bool rampVolume;
2588        uint16_t leftVol;
2589        uint16_t rightVol;
2590        Vector< sp<EffectChain> > effectChains;
2591
2592        processConfigEvents();
2593
2594        mixerStatus = MIXER_IDLE;
2595
2596        { // scope for the mLock
2597
2598            Mutex::Autolock _l(mLock);
2599
2600            if (checkForNewParameters_l()) {
2601                mixBufferSize = mFrameCount*mFrameSize;
2602                activeSleepTime = activeSleepTimeUs();
2603                idleSleepTime = idleSleepTimeUs();
2604                standbyDelay = microseconds(activeSleepTime*2);
2605            }
2606
2607            // put audio hardware into standby after short delay
2608            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2609                        mSuspended)) {
2610                // wait until we have something to do...
2611                if (!mStandby) {
2612                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2613                    mOutput->stream->common.standby(&mOutput->stream->common);
2614                    mStandby = true;
2615                    mBytesWritten = 0;
2616                }
2617
2618                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2619                    // we're about to wait, flush the binder command buffer
2620                    IPCThreadState::self()->flushCommands();
2621
2622                    if (exitPending()) break;
2623
2624                    releaseWakeLock_l();
2625                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2626                    mWaitWorkCV.wait(mLock);
2627                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2628                    acquireWakeLock_l();
2629
2630                    if (mMasterMute == false) {
2631                        char value[PROPERTY_VALUE_MAX];
2632                        property_get("ro.audio.silent", value, "0");
2633                        if (atoi(value)) {
2634                            ALOGD("Silence is golden");
2635                            setMasterMute(true);
2636                        }
2637                    }
2638
2639                    standbyTime = systemTime() + standbyDelay;
2640                    sleepTime = idleSleepTime;
2641                    continue;
2642                }
2643            }
2644
2645            effectChains = mEffectChains;
2646
2647            // find out which tracks need to be processed
2648            if (mActiveTracks.size() != 0) {
2649                sp<Track> t = mActiveTracks[0].promote();
2650                if (t == 0) continue;
2651
2652                Track* const track = t.get();
2653                audio_track_cblk_t* cblk = track->cblk();
2654
2655                // The first time a track is added we wait
2656                // for all its buffers to be filled before processing it
2657                if (cblk->framesReady() && track->isReady() &&
2658                        !track->isPaused() && !track->isTerminated())
2659                {
2660                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2661
2662                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2663                        track->mFillingUpStatus = Track::FS_ACTIVE;
2664                        mLeftVolFloat = mRightVolFloat = 0;
2665                        mLeftVolShort = mRightVolShort = 0;
2666                        if (track->mState == TrackBase::RESUMING) {
2667                            track->mState = TrackBase::ACTIVE;
2668                            rampVolume = true;
2669                        }
2670                    } else if (cblk->server != 0) {
2671                        // If the track is stopped before the first frame was mixed,
2672                        // do not apply ramp
2673                        rampVolume = true;
2674                    }
2675                    // compute volume for this track
2676                    float left, right;
2677                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2678                        mStreamTypes[track->type()].mute) {
2679                        left = right = 0;
2680                        if (track->isPausing()) {
2681                            track->setPaused();
2682                        }
2683                    } else {
2684                        float typeVolume = mStreamTypes[track->type()].volume;
2685                        float v = mMasterVolume * typeVolume;
2686                        float v_clamped = v * cblk->volume[0];
2687                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2688                        left = v_clamped/MAX_GAIN;
2689                        v_clamped = v * cblk->volume[1];
2690                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2691                        right = v_clamped/MAX_GAIN;
2692                    }
2693
2694                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2695                        mLeftVolFloat = left;
2696                        mRightVolFloat = right;
2697
2698                        // If audio HAL implements volume control,
2699                        // force software volume to nominal value
2700                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2701                            left = 1.0f;
2702                            right = 1.0f;
2703                        }
2704
2705                        // Convert volumes from float to 8.24
2706                        uint32_t vl = (uint32_t)(left * (1 << 24));
2707                        uint32_t vr = (uint32_t)(right * (1 << 24));
2708
2709                        // Delegate volume control to effect in track effect chain if needed
2710                        // only one effect chain can be present on DirectOutputThread, so if
2711                        // there is one, the track is connected to it
2712                        if (!effectChains.isEmpty()) {
2713                            // Do not ramp volume if volume is controlled by effect
2714                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2715                                rampVolume = false;
2716                            }
2717                        }
2718
2719                        // Convert volumes from 8.24 to 4.12 format
2720                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2721                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2722                        leftVol = (uint16_t)v_clamped;
2723                        v_clamped = (vr + (1 << 11)) >> 12;
2724                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2725                        rightVol = (uint16_t)v_clamped;
2726                    } else {
2727                        leftVol = mLeftVolShort;
2728                        rightVol = mRightVolShort;
2729                        rampVolume = false;
2730                    }
2731
2732                    // reset retry count
2733                    track->mRetryCount = kMaxTrackRetriesDirect;
2734                    activeTrack = t;
2735                    mixerStatus = MIXER_TRACKS_READY;
2736                } else {
2737                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2738                    if (track->isStopped()) {
2739                        track->reset();
2740                    }
2741                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2742                        // We have consumed all the buffers of this track.
2743                        // Remove it from the list of active tracks.
2744                        trackToRemove = track;
2745                    } else {
2746                        // No buffers for this track. Give it a few chances to
2747                        // fill a buffer, then remove it from active list.
2748                        if (--(track->mRetryCount) <= 0) {
2749                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2750                            trackToRemove = track;
2751                        } else {
2752                            mixerStatus = MIXER_TRACKS_ENABLED;
2753                        }
2754                    }
2755                }
2756            }
2757
2758            // remove all the tracks that need to be...
2759            if (CC_UNLIKELY(trackToRemove != 0)) {
2760                mActiveTracks.remove(trackToRemove);
2761                if (!effectChains.isEmpty()) {
2762                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2763                            trackToRemove->sessionId());
2764                    effectChains[0]->decActiveTrackCnt();
2765                }
2766                if (trackToRemove->isTerminated()) {
2767                    removeTrack_l(trackToRemove);
2768                }
2769            }
2770
2771            lockEffectChains_l(effectChains);
2772       }
2773
2774        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2775            AudioBufferProvider::Buffer buffer;
2776            size_t frameCount = mFrameCount;
2777            curBuf = (int8_t *)mMixBuffer;
2778            // output audio to hardware
2779            while (frameCount) {
2780                buffer.frameCount = frameCount;
2781                activeTrack->getNextBuffer(&buffer);
2782                if (CC_UNLIKELY(buffer.raw == NULL)) {
2783                    memset(curBuf, 0, frameCount * mFrameSize);
2784                    break;
2785                }
2786                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2787                frameCount -= buffer.frameCount;
2788                curBuf += buffer.frameCount * mFrameSize;
2789                activeTrack->releaseBuffer(&buffer);
2790            }
2791            sleepTime = 0;
2792            standbyTime = systemTime() + standbyDelay;
2793        } else {
2794            if (sleepTime == 0) {
2795                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2796                    sleepTime = activeSleepTime;
2797                } else {
2798                    sleepTime = idleSleepTime;
2799                }
2800            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2801                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2802                sleepTime = 0;
2803            }
2804        }
2805
2806        if (mSuspended) {
2807            sleepTime = suspendSleepTimeUs();
2808        }
2809        // sleepTime == 0 means we must write to audio hardware
2810        if (sleepTime == 0) {
2811            if (mixerStatus == MIXER_TRACKS_READY) {
2812                applyVolume(leftVol, rightVol, rampVolume);
2813            }
2814            for (size_t i = 0; i < effectChains.size(); i ++) {
2815                effectChains[i]->process_l();
2816            }
2817            unlockEffectChains(effectChains);
2818
2819            mLastWriteTime = systemTime();
2820            mInWrite = true;
2821            mBytesWritten += mixBufferSize;
2822            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2823            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2824            mNumWrites++;
2825            mInWrite = false;
2826            mStandby = false;
2827        } else {
2828            unlockEffectChains(effectChains);
2829            usleep(sleepTime);
2830        }
2831
2832        // finally let go of removed track, without the lock held
2833        // since we can't guarantee the destructors won't acquire that
2834        // same lock.
2835        trackToRemove.clear();
2836        activeTrack.clear();
2837
2838        // Effect chains will be actually deleted here if they were removed from
2839        // mEffectChains list during mixing or effects processing
2840        effectChains.clear();
2841    }
2842
2843    if (!mStandby) {
2844        mOutput->stream->common.standby(&mOutput->stream->common);
2845    }
2846
2847    releaseWakeLock();
2848
2849    ALOGV("DirectOutputThread %p exiting", this);
2850    return false;
2851}
2852
2853// getTrackName_l() must be called with ThreadBase::mLock held
2854int AudioFlinger::DirectOutputThread::getTrackName_l()
2855{
2856    return 0;
2857}
2858
2859// deleteTrackName_l() must be called with ThreadBase::mLock held
2860void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2861{
2862}
2863
2864// checkForNewParameters_l() must be called with ThreadBase::mLock held
2865bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2866{
2867    bool reconfig = false;
2868
2869    while (!mNewParameters.isEmpty()) {
2870        status_t status = NO_ERROR;
2871        String8 keyValuePair = mNewParameters[0];
2872        AudioParameter param = AudioParameter(keyValuePair);
2873        int value;
2874
2875        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2876            // do not accept frame count changes if tracks are open as the track buffer
2877            // size depends on frame count and correct behavior would not be garantied
2878            // if frame count is changed after track creation
2879            if (!mTracks.isEmpty()) {
2880                status = INVALID_OPERATION;
2881            } else {
2882                reconfig = true;
2883            }
2884        }
2885        if (status == NO_ERROR) {
2886            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2887                                                    keyValuePair.string());
2888            if (!mStandby && status == INVALID_OPERATION) {
2889               mOutput->stream->common.standby(&mOutput->stream->common);
2890               mStandby = true;
2891               mBytesWritten = 0;
2892               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2893                                                       keyValuePair.string());
2894            }
2895            if (status == NO_ERROR && reconfig) {
2896                readOutputParameters();
2897                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2898            }
2899        }
2900
2901        mNewParameters.removeAt(0);
2902
2903        mParamStatus = status;
2904        mParamCond.signal();
2905        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2906        // already timed out waiting for the status and will never signal the condition.
2907        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2908    }
2909    return reconfig;
2910}
2911
2912uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2913{
2914    uint32_t time;
2915    if (audio_is_linear_pcm(mFormat)) {
2916        time = PlaybackThread::activeSleepTimeUs();
2917    } else {
2918        time = 10000;
2919    }
2920    return time;
2921}
2922
2923uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2924{
2925    uint32_t time;
2926    if (audio_is_linear_pcm(mFormat)) {
2927        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2928    } else {
2929        time = 10000;
2930    }
2931    return time;
2932}
2933
2934uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2935{
2936    uint32_t time;
2937    if (audio_is_linear_pcm(mFormat)) {
2938        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2939    } else {
2940        time = 10000;
2941    }
2942    return time;
2943}
2944
2945
2946// ----------------------------------------------------------------------------
2947
2948AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2949    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2950{
2951    mType = ThreadBase::DUPLICATING;
2952    addOutputTrack(mainThread);
2953}
2954
2955AudioFlinger::DuplicatingThread::~DuplicatingThread()
2956{
2957    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2958        mOutputTracks[i]->destroy();
2959    }
2960    mOutputTracks.clear();
2961}
2962
2963bool AudioFlinger::DuplicatingThread::threadLoop()
2964{
2965    Vector< sp<Track> > tracksToRemove;
2966    uint32_t mixerStatus = MIXER_IDLE;
2967    nsecs_t standbyTime = systemTime();
2968    size_t mixBufferSize = mFrameCount*mFrameSize;
2969    SortedVector< sp<OutputTrack> > outputTracks;
2970    uint32_t writeFrames = 0;
2971    uint32_t activeSleepTime = activeSleepTimeUs();
2972    uint32_t idleSleepTime = idleSleepTimeUs();
2973    uint32_t sleepTime = idleSleepTime;
2974    Vector< sp<EffectChain> > effectChains;
2975
2976    acquireWakeLock();
2977
2978    while (!exitPending())
2979    {
2980        processConfigEvents();
2981
2982        mixerStatus = MIXER_IDLE;
2983        { // scope for the mLock
2984
2985            Mutex::Autolock _l(mLock);
2986
2987            if (checkForNewParameters_l()) {
2988                mixBufferSize = mFrameCount*mFrameSize;
2989                updateWaitTime();
2990                activeSleepTime = activeSleepTimeUs();
2991                idleSleepTime = idleSleepTimeUs();
2992            }
2993
2994            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
2995
2996            for (size_t i = 0; i < mOutputTracks.size(); i++) {
2997                outputTracks.add(mOutputTracks[i]);
2998            }
2999
3000            // put audio hardware into standby after short delay
3001            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
3002                         mSuspended)) {
3003                if (!mStandby) {
3004                    for (size_t i = 0; i < outputTracks.size(); i++) {
3005                        outputTracks[i]->stop();
3006                    }
3007                    mStandby = true;
3008                    mBytesWritten = 0;
3009                }
3010
3011                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
3012                    // we're about to wait, flush the binder command buffer
3013                    IPCThreadState::self()->flushCommands();
3014                    outputTracks.clear();
3015
3016                    if (exitPending()) break;
3017
3018                    releaseWakeLock_l();
3019                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3020                    mWaitWorkCV.wait(mLock);
3021                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3022                    acquireWakeLock_l();
3023
3024                    if (mMasterMute == false) {
3025                        char value[PROPERTY_VALUE_MAX];
3026                        property_get("ro.audio.silent", value, "0");
3027                        if (atoi(value)) {
3028                            ALOGD("Silence is golden");
3029                            setMasterMute(true);
3030                        }
3031                    }
3032
3033                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3034                    sleepTime = idleSleepTime;
3035                    continue;
3036                }
3037            }
3038
3039            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3040
3041            // prevent any changes in effect chain list and in each effect chain
3042            // during mixing and effect process as the audio buffers could be deleted
3043            // or modified if an effect is created or deleted
3044            lockEffectChains_l(effectChains);
3045        }
3046
3047        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3048            // mix buffers...
3049            if (outputsReady(outputTracks)) {
3050                mAudioMixer->process();
3051            } else {
3052                memset(mMixBuffer, 0, mixBufferSize);
3053            }
3054            sleepTime = 0;
3055            writeFrames = mFrameCount;
3056        } else {
3057            if (sleepTime == 0) {
3058                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3059                    sleepTime = activeSleepTime;
3060                } else {
3061                    sleepTime = idleSleepTime;
3062                }
3063            } else if (mBytesWritten != 0) {
3064                // flush remaining overflow buffers in output tracks
3065                for (size_t i = 0; i < outputTracks.size(); i++) {
3066                    if (outputTracks[i]->isActive()) {
3067                        sleepTime = 0;
3068                        writeFrames = 0;
3069                        memset(mMixBuffer, 0, mixBufferSize);
3070                        break;
3071                    }
3072                }
3073            }
3074        }
3075
3076        if (mSuspended) {
3077            sleepTime = suspendSleepTimeUs();
3078        }
3079        // sleepTime == 0 means we must write to audio hardware
3080        if (sleepTime == 0) {
3081            for (size_t i = 0; i < effectChains.size(); i ++) {
3082                effectChains[i]->process_l();
3083            }
3084            // enable changes in effect chain
3085            unlockEffectChains(effectChains);
3086
3087            standbyTime = systemTime() + kStandbyTimeInNsecs;
3088            for (size_t i = 0; i < outputTracks.size(); i++) {
3089                outputTracks[i]->write(mMixBuffer, writeFrames);
3090            }
3091            mStandby = false;
3092            mBytesWritten += mixBufferSize;
3093        } else {
3094            // enable changes in effect chain
3095            unlockEffectChains(effectChains);
3096            usleep(sleepTime);
3097        }
3098
3099        // finally let go of all our tracks, without the lock held
3100        // since we can't guarantee the destructors won't acquire that
3101        // same lock.
3102        tracksToRemove.clear();
3103        outputTracks.clear();
3104
3105        // Effect chains will be actually deleted here if they were removed from
3106        // mEffectChains list during mixing or effects processing
3107        effectChains.clear();
3108    }
3109
3110    releaseWakeLock();
3111
3112    return false;
3113}
3114
3115void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3116{
3117    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3118    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3119                                            this,
3120                                            mSampleRate,
3121                                            mFormat,
3122                                            mChannelMask,
3123                                            frameCount);
3124    if (outputTrack->cblk() != NULL) {
3125        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3126        mOutputTracks.add(outputTrack);
3127        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3128        updateWaitTime();
3129    }
3130}
3131
3132void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3133{
3134    Mutex::Autolock _l(mLock);
3135    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3136        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3137            mOutputTracks[i]->destroy();
3138            mOutputTracks.removeAt(i);
3139            updateWaitTime();
3140            return;
3141        }
3142    }
3143    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3144}
3145
3146void AudioFlinger::DuplicatingThread::updateWaitTime()
3147{
3148    mWaitTimeMs = UINT_MAX;
3149    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3150        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3151        if (strong != NULL) {
3152            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3153            if (waitTimeMs < mWaitTimeMs) {
3154                mWaitTimeMs = waitTimeMs;
3155            }
3156        }
3157    }
3158}
3159
3160
3161bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3162{
3163    for (size_t i = 0; i < outputTracks.size(); i++) {
3164        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3165        if (thread == 0) {
3166            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3167            return false;
3168        }
3169        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3170        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3171            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3172            return false;
3173        }
3174    }
3175    return true;
3176}
3177
3178uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3179{
3180    return (mWaitTimeMs * 1000) / 2;
3181}
3182
3183// ----------------------------------------------------------------------------
3184
3185// TrackBase constructor must be called with AudioFlinger::mLock held
3186AudioFlinger::ThreadBase::TrackBase::TrackBase(
3187            const wp<ThreadBase>& thread,
3188            const sp<Client>& client,
3189            uint32_t sampleRate,
3190            uint32_t format,
3191            uint32_t channelMask,
3192            int frameCount,
3193            uint32_t flags,
3194            const sp<IMemory>& sharedBuffer,
3195            int sessionId)
3196    :   RefBase(),
3197        mThread(thread),
3198        mClient(client),
3199        mCblk(0),
3200        mFrameCount(0),
3201        mState(IDLE),
3202        mClientTid(-1),
3203        mFormat(format),
3204        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3205        mSessionId(sessionId)
3206{
3207    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3208
3209    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3210   size_t size = sizeof(audio_track_cblk_t);
3211   uint8_t channelCount = popcount(channelMask);
3212   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3213   if (sharedBuffer == 0) {
3214       size += bufferSize;
3215   }
3216
3217   if (client != NULL) {
3218        mCblkMemory = client->heap()->allocate(size);
3219        if (mCblkMemory != 0) {
3220            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3221            if (mCblk) { // construct the shared structure in-place.
3222                new(mCblk) audio_track_cblk_t();
3223                // clear all buffers
3224                mCblk->frameCount = frameCount;
3225                mCblk->sampleRate = sampleRate;
3226                mChannelCount = channelCount;
3227                mChannelMask = channelMask;
3228                if (sharedBuffer == 0) {
3229                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3230                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3231                    // Force underrun condition to avoid false underrun callback until first data is
3232                    // written to buffer (other flags are cleared)
3233                    mCblk->flags = CBLK_UNDERRUN_ON;
3234                } else {
3235                    mBuffer = sharedBuffer->pointer();
3236                }
3237                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3238            }
3239        } else {
3240            ALOGE("not enough memory for AudioTrack size=%u", size);
3241            client->heap()->dump("AudioTrack");
3242            return;
3243        }
3244   } else {
3245       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3246           // construct the shared structure in-place.
3247           new(mCblk) audio_track_cblk_t();
3248           // clear all buffers
3249           mCblk->frameCount = frameCount;
3250           mCblk->sampleRate = sampleRate;
3251           mChannelCount = channelCount;
3252           mChannelMask = channelMask;
3253           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3254           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3255           // Force underrun condition to avoid false underrun callback until first data is
3256           // written to buffer (other flags are cleared)
3257           mCblk->flags = CBLK_UNDERRUN_ON;
3258           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3259   }
3260}
3261
3262AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3263{
3264    if (mCblk) {
3265        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3266        if (mClient == NULL) {
3267            delete mCblk;
3268        }
3269    }
3270    mCblkMemory.clear();            // and free the shared memory
3271    if (mClient != NULL) {
3272        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3273        mClient.clear();
3274    }
3275}
3276
3277void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3278{
3279    buffer->raw = NULL;
3280    mFrameCount = buffer->frameCount;
3281    step();
3282    buffer->frameCount = 0;
3283}
3284
3285bool AudioFlinger::ThreadBase::TrackBase::step() {
3286    bool result;
3287    audio_track_cblk_t* cblk = this->cblk();
3288
3289    result = cblk->stepServer(mFrameCount);
3290    if (!result) {
3291        ALOGV("stepServer failed acquiring cblk mutex");
3292        mFlags |= STEPSERVER_FAILED;
3293    }
3294    return result;
3295}
3296
3297void AudioFlinger::ThreadBase::TrackBase::reset() {
3298    audio_track_cblk_t* cblk = this->cblk();
3299
3300    cblk->user = 0;
3301    cblk->server = 0;
3302    cblk->userBase = 0;
3303    cblk->serverBase = 0;
3304    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3305    ALOGV("TrackBase::reset");
3306}
3307
3308sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3309{
3310    return mCblkMemory;
3311}
3312
3313int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3314    return (int)mCblk->sampleRate;
3315}
3316
3317int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3318    return (const int)mChannelCount;
3319}
3320
3321uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3322    return mChannelMask;
3323}
3324
3325void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3326    audio_track_cblk_t* cblk = this->cblk();
3327    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
3328    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
3329
3330    // Check validity of returned pointer in case the track control block would have been corrupted.
3331    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3332        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
3333        ALOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3334                server %d, serverBase %d, user %d, userBase %d",
3335                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3336                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3337        return 0;
3338    }
3339
3340    return bufferStart;
3341}
3342
3343// ----------------------------------------------------------------------------
3344
3345// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3346AudioFlinger::PlaybackThread::Track::Track(
3347            const wp<ThreadBase>& thread,
3348            const sp<Client>& client,
3349            int streamType,
3350            uint32_t sampleRate,
3351            uint32_t format,
3352            uint32_t channelMask,
3353            int frameCount,
3354            const sp<IMemory>& sharedBuffer,
3355            int sessionId)
3356    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3357    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3358    mAuxEffectId(0), mHasVolumeController(false)
3359{
3360    if (mCblk != NULL) {
3361        sp<ThreadBase> baseThread = thread.promote();
3362        if (baseThread != 0) {
3363            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3364            mName = playbackThread->getTrackName_l();
3365            mMainBuffer = playbackThread->mixBuffer();
3366        }
3367        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3368        if (mName < 0) {
3369            ALOGE("no more track names available");
3370        }
3371        mVolume[0] = 1.0f;
3372        mVolume[1] = 1.0f;
3373        mStreamType = streamType;
3374        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3375        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3376        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3377    }
3378}
3379
3380AudioFlinger::PlaybackThread::Track::~Track()
3381{
3382    ALOGV("PlaybackThread::Track destructor");
3383    sp<ThreadBase> thread = mThread.promote();
3384    if (thread != 0) {
3385        Mutex::Autolock _l(thread->mLock);
3386        mState = TERMINATED;
3387    }
3388}
3389
3390void AudioFlinger::PlaybackThread::Track::destroy()
3391{
3392    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3393    // by removing it from mTracks vector, so there is a risk that this Tracks's
3394    // desctructor is called. As the destructor needs to lock mLock,
3395    // we must acquire a strong reference on this Track before locking mLock
3396    // here so that the destructor is called only when exiting this function.
3397    // On the other hand, as long as Track::destroy() is only called by
3398    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3399    // this Track with its member mTrack.
3400    sp<Track> keep(this);
3401    { // scope for mLock
3402        sp<ThreadBase> thread = mThread.promote();
3403        if (thread != 0) {
3404            if (!isOutputTrack()) {
3405                if (mState == ACTIVE || mState == RESUMING) {
3406                    AudioSystem::stopOutput(thread->id(),
3407                                            (audio_stream_type_t)mStreamType,
3408                                            mSessionId);
3409
3410                    // to track the speaker usage
3411                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3412                }
3413                AudioSystem::releaseOutput(thread->id());
3414            }
3415            Mutex::Autolock _l(thread->mLock);
3416            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3417            playbackThread->destroyTrack_l(this);
3418        }
3419    }
3420}
3421
3422void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3423{
3424    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",
3425            mName - AudioMixer::TRACK0,
3426            (mClient == NULL) ? getpid() : mClient->pid(),
3427            mStreamType,
3428            mFormat,
3429            mChannelMask,
3430            mSessionId,
3431            mFrameCount,
3432            mState,
3433            mMute,
3434            mFillingUpStatus,
3435            mCblk->sampleRate,
3436            mCblk->volume[0],
3437            mCblk->volume[1],
3438            mCblk->server,
3439            mCblk->user,
3440            (int)mMainBuffer,
3441            (int)mAuxBuffer);
3442}
3443
3444status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3445{
3446     audio_track_cblk_t* cblk = this->cblk();
3447     uint32_t framesReady;
3448     uint32_t framesReq = buffer->frameCount;
3449
3450     // Check if last stepServer failed, try to step now
3451     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3452         if (!step())  goto getNextBuffer_exit;
3453         ALOGV("stepServer recovered");
3454         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3455     }
3456
3457     framesReady = cblk->framesReady();
3458
3459     if (CC_LIKELY(framesReady)) {
3460        uint32_t s = cblk->server;
3461        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3462
3463        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3464        if (framesReq > framesReady) {
3465            framesReq = framesReady;
3466        }
3467        if (s + framesReq > bufferEnd) {
3468            framesReq = bufferEnd - s;
3469        }
3470
3471         buffer->raw = getBuffer(s, framesReq);
3472         if (buffer->raw == NULL) goto getNextBuffer_exit;
3473
3474         buffer->frameCount = framesReq;
3475        return NO_ERROR;
3476     }
3477
3478getNextBuffer_exit:
3479     buffer->raw = NULL;
3480     buffer->frameCount = 0;
3481     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3482     return NOT_ENOUGH_DATA;
3483}
3484
3485bool AudioFlinger::PlaybackThread::Track::isReady() const {
3486    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3487
3488    if (mCblk->framesReady() >= mCblk->frameCount ||
3489            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3490        mFillingUpStatus = FS_FILLED;
3491        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3492        return true;
3493    }
3494    return false;
3495}
3496
3497status_t AudioFlinger::PlaybackThread::Track::start()
3498{
3499    status_t status = NO_ERROR;
3500    ALOGV("start(%d), calling thread %d session %d",
3501            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3502    sp<ThreadBase> thread = mThread.promote();
3503    if (thread != 0) {
3504        Mutex::Autolock _l(thread->mLock);
3505        int state = mState;
3506        // here the track could be either new, or restarted
3507        // in both cases "unstop" the track
3508        if (mState == PAUSED) {
3509            mState = TrackBase::RESUMING;
3510            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3511        } else {
3512            mState = TrackBase::ACTIVE;
3513            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3514        }
3515
3516        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3517            thread->mLock.unlock();
3518            status = AudioSystem::startOutput(thread->id(),
3519                                              (audio_stream_type_t)mStreamType,
3520                                              mSessionId);
3521            thread->mLock.lock();
3522
3523            // to track the speaker usage
3524            if (status == NO_ERROR) {
3525                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3526            }
3527        }
3528        if (status == NO_ERROR) {
3529            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3530            playbackThread->addTrack_l(this);
3531        } else {
3532            mState = state;
3533        }
3534    } else {
3535        status = BAD_VALUE;
3536    }
3537    return status;
3538}
3539
3540void AudioFlinger::PlaybackThread::Track::stop()
3541{
3542    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3543    sp<ThreadBase> thread = mThread.promote();
3544    if (thread != 0) {
3545        Mutex::Autolock _l(thread->mLock);
3546        int state = mState;
3547        if (mState > STOPPED) {
3548            mState = STOPPED;
3549            // If the track is not active (PAUSED and buffers full), flush buffers
3550            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3551            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3552                reset();
3553            }
3554            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3555        }
3556        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3557            thread->mLock.unlock();
3558            AudioSystem::stopOutput(thread->id(),
3559                                    (audio_stream_type_t)mStreamType,
3560                                    mSessionId);
3561            thread->mLock.lock();
3562
3563            // to track the speaker usage
3564            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3565        }
3566    }
3567}
3568
3569void AudioFlinger::PlaybackThread::Track::pause()
3570{
3571    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3572    sp<ThreadBase> thread = mThread.promote();
3573    if (thread != 0) {
3574        Mutex::Autolock _l(thread->mLock);
3575        if (mState == ACTIVE || mState == RESUMING) {
3576            mState = PAUSING;
3577            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3578            if (!isOutputTrack()) {
3579                thread->mLock.unlock();
3580                AudioSystem::stopOutput(thread->id(),
3581                                        (audio_stream_type_t)mStreamType,
3582                                        mSessionId);
3583                thread->mLock.lock();
3584
3585                // to track the speaker usage
3586                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3587            }
3588        }
3589    }
3590}
3591
3592void AudioFlinger::PlaybackThread::Track::flush()
3593{
3594    ALOGV("flush(%d)", mName);
3595    sp<ThreadBase> thread = mThread.promote();
3596    if (thread != 0) {
3597        Mutex::Autolock _l(thread->mLock);
3598        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3599            return;
3600        }
3601        // No point remaining in PAUSED state after a flush => go to
3602        // STOPPED state
3603        mState = STOPPED;
3604
3605        // do not reset the track if it is still in the process of being stopped or paused.
3606        // this will be done by prepareTracks_l() when the track is stopped.
3607        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3608        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3609            reset();
3610        }
3611    }
3612}
3613
3614void AudioFlinger::PlaybackThread::Track::reset()
3615{
3616    // Do not reset twice to avoid discarding data written just after a flush and before
3617    // the audioflinger thread detects the track is stopped.
3618    if (!mResetDone) {
3619        TrackBase::reset();
3620        // Force underrun condition to avoid false underrun callback until first data is
3621        // written to buffer
3622        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3623        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3624        mFillingUpStatus = FS_FILLING;
3625        mResetDone = true;
3626    }
3627}
3628
3629void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3630{
3631    mMute = muted;
3632}
3633
3634void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3635{
3636    mVolume[0] = left;
3637    mVolume[1] = right;
3638}
3639
3640status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3641{
3642    status_t status = DEAD_OBJECT;
3643    sp<ThreadBase> thread = mThread.promote();
3644    if (thread != 0) {
3645       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3646       status = playbackThread->attachAuxEffect(this, EffectId);
3647    }
3648    return status;
3649}
3650
3651void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3652{
3653    mAuxEffectId = EffectId;
3654    mAuxBuffer = buffer;
3655}
3656
3657// ----------------------------------------------------------------------------
3658
3659// RecordTrack constructor must be called with AudioFlinger::mLock held
3660AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3661            const wp<ThreadBase>& thread,
3662            const sp<Client>& client,
3663            uint32_t sampleRate,
3664            uint32_t format,
3665            uint32_t channelMask,
3666            int frameCount,
3667            uint32_t flags,
3668            int sessionId)
3669    :   TrackBase(thread, client, sampleRate, format,
3670                  channelMask, frameCount, flags, 0, sessionId),
3671        mOverflow(false)
3672{
3673    if (mCblk != NULL) {
3674       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3675       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3676           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3677       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3678           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3679       } else {
3680           mCblk->frameSize = sizeof(int8_t);
3681       }
3682    }
3683}
3684
3685AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3686{
3687    sp<ThreadBase> thread = mThread.promote();
3688    if (thread != 0) {
3689        AudioSystem::releaseInput(thread->id());
3690    }
3691}
3692
3693status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3694{
3695    audio_track_cblk_t* cblk = this->cblk();
3696    uint32_t framesAvail;
3697    uint32_t framesReq = buffer->frameCount;
3698
3699     // Check if last stepServer failed, try to step now
3700    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3701        if (!step()) goto getNextBuffer_exit;
3702        ALOGV("stepServer recovered");
3703        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3704    }
3705
3706    framesAvail = cblk->framesAvailable_l();
3707
3708    if (CC_LIKELY(framesAvail)) {
3709        uint32_t s = cblk->server;
3710        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3711
3712        if (framesReq > framesAvail) {
3713            framesReq = framesAvail;
3714        }
3715        if (s + framesReq > bufferEnd) {
3716            framesReq = bufferEnd - s;
3717        }
3718
3719        buffer->raw = getBuffer(s, framesReq);
3720        if (buffer->raw == NULL) goto getNextBuffer_exit;
3721
3722        buffer->frameCount = framesReq;
3723        return NO_ERROR;
3724    }
3725
3726getNextBuffer_exit:
3727    buffer->raw = NULL;
3728    buffer->frameCount = 0;
3729    return NOT_ENOUGH_DATA;
3730}
3731
3732status_t AudioFlinger::RecordThread::RecordTrack::start()
3733{
3734    sp<ThreadBase> thread = mThread.promote();
3735    if (thread != 0) {
3736        RecordThread *recordThread = (RecordThread *)thread.get();
3737        return recordThread->start(this);
3738    } else {
3739        return BAD_VALUE;
3740    }
3741}
3742
3743void AudioFlinger::RecordThread::RecordTrack::stop()
3744{
3745    sp<ThreadBase> thread = mThread.promote();
3746    if (thread != 0) {
3747        RecordThread *recordThread = (RecordThread *)thread.get();
3748        recordThread->stop(this);
3749        TrackBase::reset();
3750        // Force overerrun condition to avoid false overrun callback until first data is
3751        // read from buffer
3752        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3753    }
3754}
3755
3756void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3757{
3758    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3759            (mClient == NULL) ? getpid() : mClient->pid(),
3760            mFormat,
3761            mChannelMask,
3762            mSessionId,
3763            mFrameCount,
3764            mState,
3765            mCblk->sampleRate,
3766            mCblk->server,
3767            mCblk->user);
3768}
3769
3770
3771// ----------------------------------------------------------------------------
3772
3773AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3774            const wp<ThreadBase>& thread,
3775            DuplicatingThread *sourceThread,
3776            uint32_t sampleRate,
3777            uint32_t format,
3778            uint32_t channelMask,
3779            int frameCount)
3780    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3781    mActive(false), mSourceThread(sourceThread)
3782{
3783
3784    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3785    if (mCblk != NULL) {
3786        mCblk->flags |= CBLK_DIRECTION_OUT;
3787        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3788        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
3789        mOutBuffer.frameCount = 0;
3790        playbackThread->mTracks.add(this);
3791        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3792                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3793                mCblk, mBuffer, mCblk->buffers,
3794                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3795    } else {
3796        ALOGW("Error creating output track on thread %p", playbackThread);
3797    }
3798}
3799
3800AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3801{
3802    clearBufferQueue();
3803}
3804
3805status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3806{
3807    status_t status = Track::start();
3808    if (status != NO_ERROR) {
3809        return status;
3810    }
3811
3812    mActive = true;
3813    mRetryCount = 127;
3814    return status;
3815}
3816
3817void AudioFlinger::PlaybackThread::OutputTrack::stop()
3818{
3819    Track::stop();
3820    clearBufferQueue();
3821    mOutBuffer.frameCount = 0;
3822    mActive = false;
3823}
3824
3825bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3826{
3827    Buffer *pInBuffer;
3828    Buffer inBuffer;
3829    uint32_t channelCount = mChannelCount;
3830    bool outputBufferFull = false;
3831    inBuffer.frameCount = frames;
3832    inBuffer.i16 = data;
3833
3834    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3835
3836    if (!mActive && frames != 0) {
3837        start();
3838        sp<ThreadBase> thread = mThread.promote();
3839        if (thread != 0) {
3840            MixerThread *mixerThread = (MixerThread *)thread.get();
3841            if (mCblk->frameCount > frames){
3842                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3843                    uint32_t startFrames = (mCblk->frameCount - frames);
3844                    pInBuffer = new Buffer;
3845                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3846                    pInBuffer->frameCount = startFrames;
3847                    pInBuffer->i16 = pInBuffer->mBuffer;
3848                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3849                    mBufferQueue.add(pInBuffer);
3850                } else {
3851                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
3852                }
3853            }
3854        }
3855    }
3856
3857    while (waitTimeLeftMs) {
3858        // First write pending buffers, then new data
3859        if (mBufferQueue.size()) {
3860            pInBuffer = mBufferQueue.itemAt(0);
3861        } else {
3862            pInBuffer = &inBuffer;
3863        }
3864
3865        if (pInBuffer->frameCount == 0) {
3866            break;
3867        }
3868
3869        if (mOutBuffer.frameCount == 0) {
3870            mOutBuffer.frameCount = pInBuffer->frameCount;
3871            nsecs_t startTime = systemTime();
3872            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
3873                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3874                outputBufferFull = true;
3875                break;
3876            }
3877            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3878            if (waitTimeLeftMs >= waitTimeMs) {
3879                waitTimeLeftMs -= waitTimeMs;
3880            } else {
3881                waitTimeLeftMs = 0;
3882            }
3883        }
3884
3885        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3886        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3887        mCblk->stepUser(outFrames);
3888        pInBuffer->frameCount -= outFrames;
3889        pInBuffer->i16 += outFrames * channelCount;
3890        mOutBuffer.frameCount -= outFrames;
3891        mOutBuffer.i16 += outFrames * channelCount;
3892
3893        if (pInBuffer->frameCount == 0) {
3894            if (mBufferQueue.size()) {
3895                mBufferQueue.removeAt(0);
3896                delete [] pInBuffer->mBuffer;
3897                delete pInBuffer;
3898                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3899            } else {
3900                break;
3901            }
3902        }
3903    }
3904
3905    // If we could not write all frames, allocate a buffer and queue it for next time.
3906    if (inBuffer.frameCount) {
3907        sp<ThreadBase> thread = mThread.promote();
3908        if (thread != 0 && !thread->standby()) {
3909            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3910                pInBuffer = new Buffer;
3911                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3912                pInBuffer->frameCount = inBuffer.frameCount;
3913                pInBuffer->i16 = pInBuffer->mBuffer;
3914                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3915                mBufferQueue.add(pInBuffer);
3916                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3917            } else {
3918                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3919            }
3920        }
3921    }
3922
3923    // Calling write() with a 0 length buffer, means that no more data will be written:
3924    // If no more buffers are pending, fill output track buffer to make sure it is started
3925    // by output mixer.
3926    if (frames == 0 && mBufferQueue.size() == 0) {
3927        if (mCblk->user < mCblk->frameCount) {
3928            frames = mCblk->frameCount - mCblk->user;
3929            pInBuffer = new Buffer;
3930            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3931            pInBuffer->frameCount = frames;
3932            pInBuffer->i16 = pInBuffer->mBuffer;
3933            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3934            mBufferQueue.add(pInBuffer);
3935        } else if (mActive) {
3936            stop();
3937        }
3938    }
3939
3940    return outputBufferFull;
3941}
3942
3943status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3944{
3945    int active;
3946    status_t result;
3947    audio_track_cblk_t* cblk = mCblk;
3948    uint32_t framesReq = buffer->frameCount;
3949
3950//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3951    buffer->frameCount  = 0;
3952
3953    uint32_t framesAvail = cblk->framesAvailable();
3954
3955
3956    if (framesAvail == 0) {
3957        Mutex::Autolock _l(cblk->lock);
3958        goto start_loop_here;
3959        while (framesAvail == 0) {
3960            active = mActive;
3961            if (CC_UNLIKELY(!active)) {
3962                ALOGV("Not active and NO_MORE_BUFFERS");
3963                return AudioTrack::NO_MORE_BUFFERS;
3964            }
3965            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
3966            if (result != NO_ERROR) {
3967                return AudioTrack::NO_MORE_BUFFERS;
3968            }
3969            // read the server count again
3970        start_loop_here:
3971            framesAvail = cblk->framesAvailable_l();
3972        }
3973    }
3974
3975//    if (framesAvail < framesReq) {
3976//        return AudioTrack::NO_MORE_BUFFERS;
3977//    }
3978
3979    if (framesReq > framesAvail) {
3980        framesReq = framesAvail;
3981    }
3982
3983    uint32_t u = cblk->user;
3984    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
3985
3986    if (u + framesReq > bufferEnd) {
3987        framesReq = bufferEnd - u;
3988    }
3989
3990    buffer->frameCount  = framesReq;
3991    buffer->raw         = (void *)cblk->buffer(u);
3992    return NO_ERROR;
3993}
3994
3995
3996void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
3997{
3998    size_t size = mBufferQueue.size();
3999    Buffer *pBuffer;
4000
4001    for (size_t i = 0; i < size; i++) {
4002        pBuffer = mBufferQueue.itemAt(i);
4003        delete [] pBuffer->mBuffer;
4004        delete pBuffer;
4005    }
4006    mBufferQueue.clear();
4007}
4008
4009// ----------------------------------------------------------------------------
4010
4011AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
4012    :   RefBase(),
4013        mAudioFlinger(audioFlinger),
4014        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
4015        mPid(pid)
4016{
4017    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
4018}
4019
4020// Client destructor must be called with AudioFlinger::mLock held
4021AudioFlinger::Client::~Client()
4022{
4023    mAudioFlinger->removeClient_l(mPid);
4024}
4025
4026const sp<MemoryDealer>& AudioFlinger::Client::heap() const
4027{
4028    return mMemoryDealer;
4029}
4030
4031// ----------------------------------------------------------------------------
4032
4033AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4034                                                     const sp<IAudioFlingerClient>& client,
4035                                                     pid_t pid)
4036    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
4037{
4038}
4039
4040AudioFlinger::NotificationClient::~NotificationClient()
4041{
4042    mClient.clear();
4043}
4044
4045void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4046{
4047    sp<NotificationClient> keep(this);
4048    {
4049        mAudioFlinger->removeNotificationClient(mPid);
4050    }
4051}
4052
4053// ----------------------------------------------------------------------------
4054
4055AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4056    : BnAudioTrack(),
4057      mTrack(track)
4058{
4059}
4060
4061AudioFlinger::TrackHandle::~TrackHandle() {
4062    // just stop the track on deletion, associated resources
4063    // will be freed from the main thread once all pending buffers have
4064    // been played. Unless it's not in the active track list, in which
4065    // case we free everything now...
4066    mTrack->destroy();
4067}
4068
4069status_t AudioFlinger::TrackHandle::start() {
4070    return mTrack->start();
4071}
4072
4073void AudioFlinger::TrackHandle::stop() {
4074    mTrack->stop();
4075}
4076
4077void AudioFlinger::TrackHandle::flush() {
4078    mTrack->flush();
4079}
4080
4081void AudioFlinger::TrackHandle::mute(bool e) {
4082    mTrack->mute(e);
4083}
4084
4085void AudioFlinger::TrackHandle::pause() {
4086    mTrack->pause();
4087}
4088
4089void AudioFlinger::TrackHandle::setVolume(float left, float right) {
4090    mTrack->setVolume(left, right);
4091}
4092
4093sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4094    return mTrack->getCblk();
4095}
4096
4097status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4098{
4099    return mTrack->attachAuxEffect(EffectId);
4100}
4101
4102status_t AudioFlinger::TrackHandle::onTransact(
4103    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4104{
4105    return BnAudioTrack::onTransact(code, data, reply, flags);
4106}
4107
4108// ----------------------------------------------------------------------------
4109
4110sp<IAudioRecord> AudioFlinger::openRecord(
4111        pid_t pid,
4112        int input,
4113        uint32_t sampleRate,
4114        uint32_t format,
4115        uint32_t channelMask,
4116        int frameCount,
4117        uint32_t flags,
4118        int *sessionId,
4119        status_t *status)
4120{
4121    sp<RecordThread::RecordTrack> recordTrack;
4122    sp<RecordHandle> recordHandle;
4123    sp<Client> client;
4124    wp<Client> wclient;
4125    status_t lStatus;
4126    RecordThread *thread;
4127    size_t inFrameCount;
4128    int lSessionId;
4129
4130    // check calling permissions
4131    if (!recordingAllowed()) {
4132        lStatus = PERMISSION_DENIED;
4133        goto Exit;
4134    }
4135
4136    // add client to list
4137    { // scope for mLock
4138        Mutex::Autolock _l(mLock);
4139        thread = checkRecordThread_l(input);
4140        if (thread == NULL) {
4141            lStatus = BAD_VALUE;
4142            goto Exit;
4143        }
4144
4145        wclient = mClients.valueFor(pid);
4146        if (wclient != NULL) {
4147            client = wclient.promote();
4148        } else {
4149            client = new Client(this, pid);
4150            mClients.add(pid, client);
4151        }
4152
4153        // If no audio session id is provided, create one here
4154        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4155            lSessionId = *sessionId;
4156        } else {
4157            lSessionId = nextUniqueId();
4158            if (sessionId != NULL) {
4159                *sessionId = lSessionId;
4160            }
4161        }
4162        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4163        recordTrack = thread->createRecordTrack_l(client,
4164                                                sampleRate,
4165                                                format,
4166                                                channelMask,
4167                                                frameCount,
4168                                                flags,
4169                                                lSessionId,
4170                                                &lStatus);
4171    }
4172    if (lStatus != NO_ERROR) {
4173        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4174        // destructor is called by the TrackBase destructor with mLock held
4175        client.clear();
4176        recordTrack.clear();
4177        goto Exit;
4178    }
4179
4180    // return to handle to client
4181    recordHandle = new RecordHandle(recordTrack);
4182    lStatus = NO_ERROR;
4183
4184Exit:
4185    if (status) {
4186        *status = lStatus;
4187    }
4188    return recordHandle;
4189}
4190
4191// ----------------------------------------------------------------------------
4192
4193AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4194    : BnAudioRecord(),
4195    mRecordTrack(recordTrack)
4196{
4197}
4198
4199AudioFlinger::RecordHandle::~RecordHandle() {
4200    stop();
4201}
4202
4203status_t AudioFlinger::RecordHandle::start() {
4204    ALOGV("RecordHandle::start()");
4205    return mRecordTrack->start();
4206}
4207
4208void AudioFlinger::RecordHandle::stop() {
4209    ALOGV("RecordHandle::stop()");
4210    mRecordTrack->stop();
4211}
4212
4213sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4214    return mRecordTrack->getCblk();
4215}
4216
4217status_t AudioFlinger::RecordHandle::onTransact(
4218    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4219{
4220    return BnAudioRecord::onTransact(code, data, reply, flags);
4221}
4222
4223// ----------------------------------------------------------------------------
4224
4225AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4226                                         AudioStreamIn *input,
4227                                         uint32_t sampleRate,
4228                                         uint32_t channels,
4229                                         int id,
4230                                         uint32_t device) :
4231    ThreadBase(audioFlinger, id, device),
4232    mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
4233{
4234    mType = ThreadBase::RECORD;
4235
4236    snprintf(mName, kNameLength, "AudioIn_%d", id);
4237
4238    mReqChannelCount = popcount(channels);
4239    mReqSampleRate = sampleRate;
4240    readInputParameters();
4241}
4242
4243
4244AudioFlinger::RecordThread::~RecordThread()
4245{
4246    delete[] mRsmpInBuffer;
4247    if (mResampler != NULL) {
4248        delete mResampler;
4249        delete[] mRsmpOutBuffer;
4250    }
4251}
4252
4253void AudioFlinger::RecordThread::onFirstRef()
4254{
4255    run(mName, PRIORITY_URGENT_AUDIO);
4256}
4257
4258status_t AudioFlinger::RecordThread::readyToRun()
4259{
4260    status_t status = initCheck();
4261    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4262    return status;
4263}
4264
4265bool AudioFlinger::RecordThread::threadLoop()
4266{
4267    AudioBufferProvider::Buffer buffer;
4268    sp<RecordTrack> activeTrack;
4269    Vector< sp<EffectChain> > effectChains;
4270
4271    nsecs_t lastWarning = 0;
4272
4273    acquireWakeLock();
4274
4275    // start recording
4276    while (!exitPending()) {
4277
4278        processConfigEvents();
4279
4280        { // scope for mLock
4281            Mutex::Autolock _l(mLock);
4282            checkForNewParameters_l();
4283            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4284                if (!mStandby) {
4285                    mInput->stream->common.standby(&mInput->stream->common);
4286                    mStandby = true;
4287                }
4288
4289                if (exitPending()) break;
4290
4291                releaseWakeLock_l();
4292                ALOGV("RecordThread: loop stopping");
4293                // go to sleep
4294                mWaitWorkCV.wait(mLock);
4295                ALOGV("RecordThread: loop starting");
4296                acquireWakeLock_l();
4297                continue;
4298            }
4299            if (mActiveTrack != 0) {
4300                if (mActiveTrack->mState == TrackBase::PAUSING) {
4301                    if (!mStandby) {
4302                        mInput->stream->common.standby(&mInput->stream->common);
4303                        mStandby = true;
4304                    }
4305                    mActiveTrack.clear();
4306                    mStartStopCond.broadcast();
4307                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4308                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4309                        mActiveTrack.clear();
4310                        mStartStopCond.broadcast();
4311                    } else if (mBytesRead != 0) {
4312                        // record start succeeds only if first read from audio input
4313                        // succeeds
4314                        if (mBytesRead > 0) {
4315                            mActiveTrack->mState = TrackBase::ACTIVE;
4316                        } else {
4317                            mActiveTrack.clear();
4318                        }
4319                        mStartStopCond.broadcast();
4320                    }
4321                    mStandby = false;
4322                }
4323            }
4324            lockEffectChains_l(effectChains);
4325        }
4326
4327        if (mActiveTrack != 0) {
4328            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4329                mActiveTrack->mState != TrackBase::RESUMING) {
4330                unlockEffectChains(effectChains);
4331                usleep(kRecordThreadSleepUs);
4332                continue;
4333            }
4334            for (size_t i = 0; i < effectChains.size(); i ++) {
4335                effectChains[i]->process_l();
4336            }
4337
4338            buffer.frameCount = mFrameCount;
4339            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4340                size_t framesOut = buffer.frameCount;
4341                if (mResampler == NULL) {
4342                    // no resampling
4343                    while (framesOut) {
4344                        size_t framesIn = mFrameCount - mRsmpInIndex;
4345                        if (framesIn) {
4346                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4347                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4348                            if (framesIn > framesOut)
4349                                framesIn = framesOut;
4350                            mRsmpInIndex += framesIn;
4351                            framesOut -= framesIn;
4352                            if ((int)mChannelCount == mReqChannelCount ||
4353                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4354                                memcpy(dst, src, framesIn * mFrameSize);
4355                            } else {
4356                                int16_t *src16 = (int16_t *)src;
4357                                int16_t *dst16 = (int16_t *)dst;
4358                                if (mChannelCount == 1) {
4359                                    while (framesIn--) {
4360                                        *dst16++ = *src16;
4361                                        *dst16++ = *src16++;
4362                                    }
4363                                } else {
4364                                    while (framesIn--) {
4365                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4366                                        src16 += 2;
4367                                    }
4368                                }
4369                            }
4370                        }
4371                        if (framesOut && mFrameCount == mRsmpInIndex) {
4372                            if (framesOut == mFrameCount &&
4373                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4374                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4375                                framesOut = 0;
4376                            } else {
4377                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4378                                mRsmpInIndex = 0;
4379                            }
4380                            if (mBytesRead < 0) {
4381                                ALOGE("Error reading audio input");
4382                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4383                                    // Force input into standby so that it tries to
4384                                    // recover at next read attempt
4385                                    mInput->stream->common.standby(&mInput->stream->common);
4386                                    usleep(kRecordThreadSleepUs);
4387                                }
4388                                mRsmpInIndex = mFrameCount;
4389                                framesOut = 0;
4390                                buffer.frameCount = 0;
4391                            }
4392                        }
4393                    }
4394                } else {
4395                    // resampling
4396
4397                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4398                    // alter output frame count as if we were expecting stereo samples
4399                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4400                        framesOut >>= 1;
4401                    }
4402                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4403                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4404                    // are 32 bit aligned which should be always true.
4405                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4406                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4407                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4408                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4409                        int16_t *dst = buffer.i16;
4410                        while (framesOut--) {
4411                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4412                            src += 2;
4413                        }
4414                    } else {
4415                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4416                    }
4417
4418                }
4419                mActiveTrack->releaseBuffer(&buffer);
4420                mActiveTrack->overflow();
4421            }
4422            // client isn't retrieving buffers fast enough
4423            else {
4424                if (!mActiveTrack->setOverflow()) {
4425                    nsecs_t now = systemTime();
4426                    if ((now - lastWarning) > kWarningThrottleNs) {
4427                        ALOGW("RecordThread: buffer overflow");
4428                        lastWarning = now;
4429                    }
4430                }
4431                // Release the processor for a while before asking for a new buffer.
4432                // This will give the application more chance to read from the buffer and
4433                // clear the overflow.
4434                usleep(kRecordThreadSleepUs);
4435            }
4436        }
4437        // enable changes in effect chain
4438        unlockEffectChains(effectChains);
4439        effectChains.clear();
4440    }
4441
4442    if (!mStandby) {
4443        mInput->stream->common.standby(&mInput->stream->common);
4444    }
4445    mActiveTrack.clear();
4446
4447    mStartStopCond.broadcast();
4448
4449    releaseWakeLock();
4450
4451    ALOGV("RecordThread %p exiting", this);
4452    return false;
4453}
4454
4455
4456sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4457        const sp<AudioFlinger::Client>& client,
4458        uint32_t sampleRate,
4459        int format,
4460        int channelMask,
4461        int frameCount,
4462        uint32_t flags,
4463        int sessionId,
4464        status_t *status)
4465{
4466    sp<RecordTrack> track;
4467    status_t lStatus;
4468
4469    lStatus = initCheck();
4470    if (lStatus != NO_ERROR) {
4471        ALOGE("Audio driver not initialized.");
4472        goto Exit;
4473    }
4474
4475    { // scope for mLock
4476        Mutex::Autolock _l(mLock);
4477
4478        track = new RecordTrack(this, client, sampleRate,
4479                      format, channelMask, frameCount, flags, sessionId);
4480
4481        if (track->getCblk() == NULL) {
4482            lStatus = NO_MEMORY;
4483            goto Exit;
4484        }
4485
4486        mTrack = track.get();
4487        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4488        bool suspend = audio_is_bluetooth_sco_device(
4489                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4490        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4491        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4492    }
4493    lStatus = NO_ERROR;
4494
4495Exit:
4496    if (status) {
4497        *status = lStatus;
4498    }
4499    return track;
4500}
4501
4502status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4503{
4504    ALOGV("RecordThread::start");
4505    sp <ThreadBase> strongMe = this;
4506    status_t status = NO_ERROR;
4507    {
4508        AutoMutex lock(mLock);
4509        if (mActiveTrack != 0) {
4510            if (recordTrack != mActiveTrack.get()) {
4511                status = -EBUSY;
4512            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4513                mActiveTrack->mState = TrackBase::ACTIVE;
4514            }
4515            return status;
4516        }
4517
4518        recordTrack->mState = TrackBase::IDLE;
4519        mActiveTrack = recordTrack;
4520        mLock.unlock();
4521        status_t status = AudioSystem::startInput(mId);
4522        mLock.lock();
4523        if (status != NO_ERROR) {
4524            mActiveTrack.clear();
4525            return status;
4526        }
4527        mRsmpInIndex = mFrameCount;
4528        mBytesRead = 0;
4529        if (mResampler != NULL) {
4530            mResampler->reset();
4531        }
4532        mActiveTrack->mState = TrackBase::RESUMING;
4533        // signal thread to start
4534        ALOGV("Signal record thread");
4535        mWaitWorkCV.signal();
4536        // do not wait for mStartStopCond if exiting
4537        if (mExiting) {
4538            mActiveTrack.clear();
4539            status = INVALID_OPERATION;
4540            goto startError;
4541        }
4542        mStartStopCond.wait(mLock);
4543        if (mActiveTrack == 0) {
4544            ALOGV("Record failed to start");
4545            status = BAD_VALUE;
4546            goto startError;
4547        }
4548        ALOGV("Record started OK");
4549        return status;
4550    }
4551startError:
4552    AudioSystem::stopInput(mId);
4553    return status;
4554}
4555
4556void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4557    ALOGV("RecordThread::stop");
4558    sp <ThreadBase> strongMe = this;
4559    {
4560        AutoMutex lock(mLock);
4561        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4562            mActiveTrack->mState = TrackBase::PAUSING;
4563            // do not wait for mStartStopCond if exiting
4564            if (mExiting) {
4565                return;
4566            }
4567            mStartStopCond.wait(mLock);
4568            // if we have been restarted, recordTrack == mActiveTrack.get() here
4569            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4570                mLock.unlock();
4571                AudioSystem::stopInput(mId);
4572                mLock.lock();
4573                ALOGV("Record stopped OK");
4574            }
4575        }
4576    }
4577}
4578
4579status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4580{
4581    const size_t SIZE = 256;
4582    char buffer[SIZE];
4583    String8 result;
4584    pid_t pid = 0;
4585
4586    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4587    result.append(buffer);
4588
4589    if (mActiveTrack != 0) {
4590        result.append("Active Track:\n");
4591        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4592        mActiveTrack->dump(buffer, SIZE);
4593        result.append(buffer);
4594
4595        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4596        result.append(buffer);
4597        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4598        result.append(buffer);
4599        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4600        result.append(buffer);
4601        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4602        result.append(buffer);
4603        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4604        result.append(buffer);
4605
4606
4607    } else {
4608        result.append("No record client\n");
4609    }
4610    write(fd, result.string(), result.size());
4611
4612    dumpBase(fd, args);
4613    dumpEffectChains(fd, args);
4614
4615    return NO_ERROR;
4616}
4617
4618status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4619{
4620    size_t framesReq = buffer->frameCount;
4621    size_t framesReady = mFrameCount - mRsmpInIndex;
4622    int channelCount;
4623
4624    if (framesReady == 0) {
4625        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4626        if (mBytesRead < 0) {
4627            ALOGE("RecordThread::getNextBuffer() Error reading audio input");
4628            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4629                // Force input into standby so that it tries to
4630                // recover at next read attempt
4631                mInput->stream->common.standby(&mInput->stream->common);
4632                usleep(kRecordThreadSleepUs);
4633            }
4634            buffer->raw = NULL;
4635            buffer->frameCount = 0;
4636            return NOT_ENOUGH_DATA;
4637        }
4638        mRsmpInIndex = 0;
4639        framesReady = mFrameCount;
4640    }
4641
4642    if (framesReq > framesReady) {
4643        framesReq = framesReady;
4644    }
4645
4646    if (mChannelCount == 1 && mReqChannelCount == 2) {
4647        channelCount = 1;
4648    } else {
4649        channelCount = 2;
4650    }
4651    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4652    buffer->frameCount = framesReq;
4653    return NO_ERROR;
4654}
4655
4656void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4657{
4658    mRsmpInIndex += buffer->frameCount;
4659    buffer->frameCount = 0;
4660}
4661
4662bool AudioFlinger::RecordThread::checkForNewParameters_l()
4663{
4664    bool reconfig = false;
4665
4666    while (!mNewParameters.isEmpty()) {
4667        status_t status = NO_ERROR;
4668        String8 keyValuePair = mNewParameters[0];
4669        AudioParameter param = AudioParameter(keyValuePair);
4670        int value;
4671        int reqFormat = mFormat;
4672        int reqSamplingRate = mReqSampleRate;
4673        int reqChannelCount = mReqChannelCount;
4674
4675        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4676            reqSamplingRate = value;
4677            reconfig = true;
4678        }
4679        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4680            reqFormat = value;
4681            reconfig = true;
4682        }
4683        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4684            reqChannelCount = popcount(value);
4685            reconfig = true;
4686        }
4687        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4688            // do not accept frame count changes if tracks are open as the track buffer
4689            // size depends on frame count and correct behavior would not be garantied
4690            // if frame count is changed after track creation
4691            if (mActiveTrack != 0) {
4692                status = INVALID_OPERATION;
4693            } else {
4694                reconfig = true;
4695            }
4696        }
4697        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4698            // forward device change to effects that have requested to be
4699            // aware of attached audio device.
4700            for (size_t i = 0; i < mEffectChains.size(); i++) {
4701                mEffectChains[i]->setDevice_l(value);
4702            }
4703            // store input device and output device but do not forward output device to audio HAL.
4704            // Note that status is ignored by the caller for output device
4705            // (see AudioFlinger::setParameters()
4706            if (value & AUDIO_DEVICE_OUT_ALL) {
4707                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4708                status = BAD_VALUE;
4709            } else {
4710                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4711                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4712                if (mTrack != NULL) {
4713                    bool suspend = audio_is_bluetooth_sco_device(
4714                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4715                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4716                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4717                }
4718            }
4719            mDevice |= (uint32_t)value;
4720        }
4721        if (status == NO_ERROR) {
4722            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4723            if (status == INVALID_OPERATION) {
4724               mInput->stream->common.standby(&mInput->stream->common);
4725               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4726            }
4727            if (reconfig) {
4728                if (status == BAD_VALUE &&
4729                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4730                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4731                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4732                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4733                    (reqChannelCount < 3)) {
4734                    status = NO_ERROR;
4735                }
4736                if (status == NO_ERROR) {
4737                    readInputParameters();
4738                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4739                }
4740            }
4741        }
4742
4743        mNewParameters.removeAt(0);
4744
4745        mParamStatus = status;
4746        mParamCond.signal();
4747        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4748        // already timed out waiting for the status and will never signal the condition.
4749        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4750    }
4751    return reconfig;
4752}
4753
4754String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4755{
4756    char *s;
4757    String8 out_s8 = String8();
4758
4759    Mutex::Autolock _l(mLock);
4760    if (initCheck() != NO_ERROR) {
4761        return out_s8;
4762    }
4763
4764    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4765    out_s8 = String8(s);
4766    free(s);
4767    return out_s8;
4768}
4769
4770void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4771    AudioSystem::OutputDescriptor desc;
4772    void *param2 = 0;
4773
4774    switch (event) {
4775    case AudioSystem::INPUT_OPENED:
4776    case AudioSystem::INPUT_CONFIG_CHANGED:
4777        desc.channels = mChannelMask;
4778        desc.samplingRate = mSampleRate;
4779        desc.format = mFormat;
4780        desc.frameCount = mFrameCount;
4781        desc.latency = 0;
4782        param2 = &desc;
4783        break;
4784
4785    case AudioSystem::INPUT_CLOSED:
4786    default:
4787        break;
4788    }
4789    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4790}
4791
4792void AudioFlinger::RecordThread::readInputParameters()
4793{
4794    if (mRsmpInBuffer) delete mRsmpInBuffer;
4795    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4796    if (mResampler) delete mResampler;
4797    mResampler = NULL;
4798
4799    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4800    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4801    mChannelCount = (uint16_t)popcount(mChannelMask);
4802    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4803    mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
4804    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4805    mFrameCount = mInputBytes / mFrameSize;
4806    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4807
4808    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4809    {
4810        int channelCount;
4811         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4812         // stereo to mono post process as the resampler always outputs stereo.
4813        if (mChannelCount == 1 && mReqChannelCount == 2) {
4814            channelCount = 1;
4815        } else {
4816            channelCount = 2;
4817        }
4818        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4819        mResampler->setSampleRate(mSampleRate);
4820        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4821        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4822
4823        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4824        if (mChannelCount == 1 && mReqChannelCount == 1) {
4825            mFrameCount >>= 1;
4826        }
4827
4828    }
4829    mRsmpInIndex = mFrameCount;
4830}
4831
4832unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4833{
4834    Mutex::Autolock _l(mLock);
4835    if (initCheck() != NO_ERROR) {
4836        return 0;
4837    }
4838
4839    return mInput->stream->get_input_frames_lost(mInput->stream);
4840}
4841
4842uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4843{
4844    Mutex::Autolock _l(mLock);
4845    uint32_t result = 0;
4846    if (getEffectChain_l(sessionId) != 0) {
4847        result = EFFECT_SESSION;
4848    }
4849
4850    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4851        result |= TRACK_SESSION;
4852    }
4853
4854    return result;
4855}
4856
4857AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4858{
4859    Mutex::Autolock _l(mLock);
4860    return mTrack;
4861}
4862
4863AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
4864{
4865    Mutex::Autolock _l(mLock);
4866    return mInput;
4867}
4868
4869AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4870{
4871    Mutex::Autolock _l(mLock);
4872    AudioStreamIn *input = mInput;
4873    mInput = NULL;
4874    return input;
4875}
4876
4877// this method must always be called either with ThreadBase mLock held or inside the thread loop
4878audio_stream_t* AudioFlinger::RecordThread::stream()
4879{
4880    if (mInput == NULL) {
4881        return NULL;
4882    }
4883    return &mInput->stream->common;
4884}
4885
4886
4887// ----------------------------------------------------------------------------
4888
4889int AudioFlinger::openOutput(uint32_t *pDevices,
4890                                uint32_t *pSamplingRate,
4891                                uint32_t *pFormat,
4892                                uint32_t *pChannels,
4893                                uint32_t *pLatencyMs,
4894                                uint32_t flags)
4895{
4896    status_t status;
4897    PlaybackThread *thread = NULL;
4898    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4899    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4900    uint32_t format = pFormat ? *pFormat : 0;
4901    uint32_t channels = pChannels ? *pChannels : 0;
4902    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4903    audio_stream_out_t *outStream;
4904    audio_hw_device_t *outHwDev;
4905
4906    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4907            pDevices ? *pDevices : 0,
4908            samplingRate,
4909            format,
4910            channels,
4911            flags);
4912
4913    if (pDevices == NULL || *pDevices == 0) {
4914        return 0;
4915    }
4916
4917    Mutex::Autolock _l(mLock);
4918
4919    outHwDev = findSuitableHwDev_l(*pDevices);
4920    if (outHwDev == NULL)
4921        return 0;
4922
4923    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
4924                                          &channels, &samplingRate, &outStream);
4925    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4926            outStream,
4927            samplingRate,
4928            format,
4929            channels,
4930            status);
4931
4932    mHardwareStatus = AUDIO_HW_IDLE;
4933    if (outStream != NULL) {
4934        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4935        int id = nextUniqueId();
4936
4937        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4938            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4939            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4940            thread = new DirectOutputThread(this, output, id, *pDevices);
4941            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4942        } else {
4943            thread = new MixerThread(this, output, id, *pDevices);
4944            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4945        }
4946        mPlaybackThreads.add(id, thread);
4947
4948        if (pSamplingRate) *pSamplingRate = samplingRate;
4949        if (pFormat) *pFormat = format;
4950        if (pChannels) *pChannels = channels;
4951        if (pLatencyMs) *pLatencyMs = thread->latency();
4952
4953        // notify client processes of the new output creation
4954        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4955        return id;
4956    }
4957
4958    return 0;
4959}
4960
4961int AudioFlinger::openDuplicateOutput(int output1, int output2)
4962{
4963    Mutex::Autolock _l(mLock);
4964    MixerThread *thread1 = checkMixerThread_l(output1);
4965    MixerThread *thread2 = checkMixerThread_l(output2);
4966
4967    if (thread1 == NULL || thread2 == NULL) {
4968        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
4969        return 0;
4970    }
4971
4972    int id = nextUniqueId();
4973    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
4974    thread->addOutputTrack(thread2);
4975    mPlaybackThreads.add(id, thread);
4976    // notify client processes of the new output creation
4977    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4978    return id;
4979}
4980
4981status_t AudioFlinger::closeOutput(int output)
4982{
4983    // keep strong reference on the playback thread so that
4984    // it is not destroyed while exit() is executed
4985    sp <PlaybackThread> thread;
4986    {
4987        Mutex::Autolock _l(mLock);
4988        thread = checkPlaybackThread_l(output);
4989        if (thread == NULL) {
4990            return BAD_VALUE;
4991        }
4992
4993        ALOGV("closeOutput() %d", output);
4994
4995        if (thread->type() == ThreadBase::MIXER) {
4996            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4997                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
4998                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
4999                    dupThread->removeOutputTrack((MixerThread *)thread.get());
5000                }
5001            }
5002        }
5003        void *param2 = 0;
5004        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
5005        mPlaybackThreads.removeItem(output);
5006    }
5007    thread->exit();
5008
5009    if (thread->type() != ThreadBase::DUPLICATING) {
5010        AudioStreamOut *out = thread->clearOutput();
5011        // from now on thread->mOutput is NULL
5012        out->hwDev->close_output_stream(out->hwDev, out->stream);
5013        delete out;
5014    }
5015    return NO_ERROR;
5016}
5017
5018status_t AudioFlinger::suspendOutput(int output)
5019{
5020    Mutex::Autolock _l(mLock);
5021    PlaybackThread *thread = checkPlaybackThread_l(output);
5022
5023    if (thread == NULL) {
5024        return BAD_VALUE;
5025    }
5026
5027    ALOGV("suspendOutput() %d", output);
5028    thread->suspend();
5029
5030    return NO_ERROR;
5031}
5032
5033status_t AudioFlinger::restoreOutput(int output)
5034{
5035    Mutex::Autolock _l(mLock);
5036    PlaybackThread *thread = checkPlaybackThread_l(output);
5037
5038    if (thread == NULL) {
5039        return BAD_VALUE;
5040    }
5041
5042    ALOGV("restoreOutput() %d", output);
5043
5044    thread->restore();
5045
5046    return NO_ERROR;
5047}
5048
5049int AudioFlinger::openInput(uint32_t *pDevices,
5050                                uint32_t *pSamplingRate,
5051                                uint32_t *pFormat,
5052                                uint32_t *pChannels,
5053                                uint32_t acoustics)
5054{
5055    status_t status;
5056    RecordThread *thread = NULL;
5057    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5058    uint32_t format = pFormat ? *pFormat : 0;
5059    uint32_t channels = pChannels ? *pChannels : 0;
5060    uint32_t reqSamplingRate = samplingRate;
5061    uint32_t reqFormat = format;
5062    uint32_t reqChannels = channels;
5063    audio_stream_in_t *inStream;
5064    audio_hw_device_t *inHwDev;
5065
5066    if (pDevices == NULL || *pDevices == 0) {
5067        return 0;
5068    }
5069
5070    Mutex::Autolock _l(mLock);
5071
5072    inHwDev = findSuitableHwDev_l(*pDevices);
5073    if (inHwDev == NULL)
5074        return 0;
5075
5076    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5077                                        &channels, &samplingRate,
5078                                        (audio_in_acoustics_t)acoustics,
5079                                        &inStream);
5080    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5081            inStream,
5082            samplingRate,
5083            format,
5084            channels,
5085            acoustics,
5086            status);
5087
5088    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5089    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5090    // or stereo to mono conversions on 16 bit PCM inputs.
5091    if (inStream == NULL && status == BAD_VALUE &&
5092        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5093        (samplingRate <= 2 * reqSamplingRate) &&
5094        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5095        ALOGV("openInput() reopening with proposed sampling rate and channels");
5096        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5097                                            &channels, &samplingRate,
5098                                            (audio_in_acoustics_t)acoustics,
5099                                            &inStream);
5100    }
5101
5102    if (inStream != NULL) {
5103        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5104
5105        int id = nextUniqueId();
5106        // Start record thread
5107        // RecorThread require both input and output device indication to forward to audio
5108        // pre processing modules
5109        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5110        thread = new RecordThread(this,
5111                                  input,
5112                                  reqSamplingRate,
5113                                  reqChannels,
5114                                  id,
5115                                  device);
5116        mRecordThreads.add(id, thread);
5117        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5118        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5119        if (pFormat) *pFormat = format;
5120        if (pChannels) *pChannels = reqChannels;
5121
5122        input->stream->common.standby(&input->stream->common);
5123
5124        // notify client processes of the new input creation
5125        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5126        return id;
5127    }
5128
5129    return 0;
5130}
5131
5132status_t AudioFlinger::closeInput(int input)
5133{
5134    // keep strong reference on the record thread so that
5135    // it is not destroyed while exit() is executed
5136    sp <RecordThread> thread;
5137    {
5138        Mutex::Autolock _l(mLock);
5139        thread = checkRecordThread_l(input);
5140        if (thread == NULL) {
5141            return BAD_VALUE;
5142        }
5143
5144        ALOGV("closeInput() %d", input);
5145        void *param2 = 0;
5146        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5147        mRecordThreads.removeItem(input);
5148    }
5149    thread->exit();
5150
5151    AudioStreamIn *in = thread->clearInput();
5152    // from now on thread->mInput is NULL
5153    in->hwDev->close_input_stream(in->hwDev, in->stream);
5154    delete in;
5155
5156    return NO_ERROR;
5157}
5158
5159status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
5160{
5161    Mutex::Autolock _l(mLock);
5162    MixerThread *dstThread = checkMixerThread_l(output);
5163    if (dstThread == NULL) {
5164        ALOGW("setStreamOutput() bad output id %d", output);
5165        return BAD_VALUE;
5166    }
5167
5168    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5169    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5170
5171    dstThread->setStreamValid(stream, true);
5172
5173    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5174        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5175        if (thread != dstThread &&
5176            thread->type() != ThreadBase::DIRECT) {
5177            MixerThread *srcThread = (MixerThread *)thread;
5178            srcThread->setStreamValid(stream, false);
5179            srcThread->invalidateTracks(stream);
5180        }
5181    }
5182
5183    return NO_ERROR;
5184}
5185
5186
5187int AudioFlinger::newAudioSessionId()
5188{
5189    return nextUniqueId();
5190}
5191
5192void AudioFlinger::acquireAudioSessionId(int audioSession)
5193{
5194    Mutex::Autolock _l(mLock);
5195    int caller = IPCThreadState::self()->getCallingPid();
5196    ALOGV("acquiring %d from %d", audioSession, caller);
5197    int num = mAudioSessionRefs.size();
5198    for (int i = 0; i< num; i++) {
5199        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5200        if (ref->sessionid == audioSession && ref->pid == caller) {
5201            ref->cnt++;
5202            ALOGV(" incremented refcount to %d", ref->cnt);
5203            return;
5204        }
5205    }
5206    AudioSessionRef *ref = new AudioSessionRef();
5207    ref->sessionid = audioSession;
5208    ref->pid = caller;
5209    ref->cnt = 1;
5210    mAudioSessionRefs.push(ref);
5211    ALOGV(" added new entry for %d", ref->sessionid);
5212}
5213
5214void AudioFlinger::releaseAudioSessionId(int audioSession)
5215{
5216    Mutex::Autolock _l(mLock);
5217    int caller = IPCThreadState::self()->getCallingPid();
5218    ALOGV("releasing %d from %d", audioSession, caller);
5219    int num = mAudioSessionRefs.size();
5220    for (int i = 0; i< num; i++) {
5221        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5222        if (ref->sessionid == audioSession && ref->pid == caller) {
5223            ref->cnt--;
5224            ALOGV(" decremented refcount to %d", ref->cnt);
5225            if (ref->cnt == 0) {
5226                mAudioSessionRefs.removeAt(i);
5227                delete ref;
5228                purgeStaleEffects_l();
5229            }
5230            return;
5231        }
5232    }
5233    ALOGW("session id %d not found for pid %d", audioSession, caller);
5234}
5235
5236void AudioFlinger::purgeStaleEffects_l() {
5237
5238    ALOGV("purging stale effects");
5239
5240    Vector< sp<EffectChain> > chains;
5241
5242    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5243        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5244        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5245            sp<EffectChain> ec = t->mEffectChains[j];
5246            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5247                chains.push(ec);
5248            }
5249        }
5250    }
5251    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5252        sp<RecordThread> t = mRecordThreads.valueAt(i);
5253        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5254            sp<EffectChain> ec = t->mEffectChains[j];
5255            chains.push(ec);
5256        }
5257    }
5258
5259    for (size_t i = 0; i < chains.size(); i++) {
5260        sp<EffectChain> ec = chains[i];
5261        int sessionid = ec->sessionId();
5262        sp<ThreadBase> t = ec->mThread.promote();
5263        if (t == 0) {
5264            continue;
5265        }
5266        size_t numsessionrefs = mAudioSessionRefs.size();
5267        bool found = false;
5268        for (size_t k = 0; k < numsessionrefs; k++) {
5269            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5270            if (ref->sessionid == sessionid) {
5271                ALOGV(" session %d still exists for %d with %d refs",
5272                     sessionid, ref->pid, ref->cnt);
5273                found = true;
5274                break;
5275            }
5276        }
5277        if (!found) {
5278            // remove all effects from the chain
5279            while (ec->mEffects.size()) {
5280                sp<EffectModule> effect = ec->mEffects[0];
5281                effect->unPin();
5282                Mutex::Autolock _l (t->mLock);
5283                t->removeEffect_l(effect);
5284                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5285                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5286                    if (handle != 0) {
5287                        handle->mEffect.clear();
5288                        if (handle->mHasControl && handle->mEnabled) {
5289                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5290                        }
5291                    }
5292                }
5293                AudioSystem::unregisterEffect(effect->id());
5294            }
5295        }
5296    }
5297    return;
5298}
5299
5300// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5301AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5302{
5303    PlaybackThread *thread = NULL;
5304    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5305        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5306    }
5307    return thread;
5308}
5309
5310// checkMixerThread_l() must be called with AudioFlinger::mLock held
5311AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5312{
5313    PlaybackThread *thread = checkPlaybackThread_l(output);
5314    if (thread != NULL) {
5315        if (thread->type() == ThreadBase::DIRECT) {
5316            thread = NULL;
5317        }
5318    }
5319    return (MixerThread *)thread;
5320}
5321
5322// checkRecordThread_l() must be called with AudioFlinger::mLock held
5323AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5324{
5325    RecordThread *thread = NULL;
5326    if (mRecordThreads.indexOfKey(input) >= 0) {
5327        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5328    }
5329    return thread;
5330}
5331
5332uint32_t AudioFlinger::nextUniqueId()
5333{
5334    return android_atomic_inc(&mNextUniqueId);
5335}
5336
5337AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5338{
5339    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5340        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5341        AudioStreamOut *output = thread->getOutput();
5342        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5343            return thread;
5344        }
5345    }
5346    return NULL;
5347}
5348
5349uint32_t AudioFlinger::primaryOutputDevice_l()
5350{
5351    PlaybackThread *thread = primaryPlaybackThread_l();
5352
5353    if (thread == NULL) {
5354        return 0;
5355    }
5356
5357    return thread->device();
5358}
5359
5360
5361// ----------------------------------------------------------------------------
5362//  Effect management
5363// ----------------------------------------------------------------------------
5364
5365
5366status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5367{
5368    Mutex::Autolock _l(mLock);
5369    return EffectQueryNumberEffects(numEffects);
5370}
5371
5372status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5373{
5374    Mutex::Autolock _l(mLock);
5375    return EffectQueryEffect(index, descriptor);
5376}
5377
5378status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5379{
5380    Mutex::Autolock _l(mLock);
5381    return EffectGetDescriptor(pUuid, descriptor);
5382}
5383
5384
5385sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5386        effect_descriptor_t *pDesc,
5387        const sp<IEffectClient>& effectClient,
5388        int32_t priority,
5389        int io,
5390        int sessionId,
5391        status_t *status,
5392        int *id,
5393        int *enabled)
5394{
5395    status_t lStatus = NO_ERROR;
5396    sp<EffectHandle> handle;
5397    effect_descriptor_t desc;
5398    sp<Client> client;
5399    wp<Client> wclient;
5400
5401    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5402            pid, effectClient.get(), priority, sessionId, io);
5403
5404    if (pDesc == NULL) {
5405        lStatus = BAD_VALUE;
5406        goto Exit;
5407    }
5408
5409    // check audio settings permission for global effects
5410    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5411        lStatus = PERMISSION_DENIED;
5412        goto Exit;
5413    }
5414
5415    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5416    // that can only be created by audio policy manager (running in same process)
5417    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5418        lStatus = PERMISSION_DENIED;
5419        goto Exit;
5420    }
5421
5422    if (io == 0) {
5423        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5424            // output must be specified by AudioPolicyManager when using session
5425            // AUDIO_SESSION_OUTPUT_STAGE
5426            lStatus = BAD_VALUE;
5427            goto Exit;
5428        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5429            // if the output returned by getOutputForEffect() is removed before we lock the
5430            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5431            // and we will exit safely
5432            io = AudioSystem::getOutputForEffect(&desc);
5433        }
5434    }
5435
5436    {
5437        Mutex::Autolock _l(mLock);
5438
5439
5440        if (!EffectIsNullUuid(&pDesc->uuid)) {
5441            // if uuid is specified, request effect descriptor
5442            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5443            if (lStatus < 0) {
5444                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5445                goto Exit;
5446            }
5447        } else {
5448            // if uuid is not specified, look for an available implementation
5449            // of the required type in effect factory
5450            if (EffectIsNullUuid(&pDesc->type)) {
5451                ALOGW("createEffect() no effect type");
5452                lStatus = BAD_VALUE;
5453                goto Exit;
5454            }
5455            uint32_t numEffects = 0;
5456            effect_descriptor_t d;
5457            d.flags = 0; // prevent compiler warning
5458            bool found = false;
5459
5460            lStatus = EffectQueryNumberEffects(&numEffects);
5461            if (lStatus < 0) {
5462                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5463                goto Exit;
5464            }
5465            for (uint32_t i = 0; i < numEffects; i++) {
5466                lStatus = EffectQueryEffect(i, &desc);
5467                if (lStatus < 0) {
5468                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5469                    continue;
5470                }
5471                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5472                    // If matching type found save effect descriptor. If the session is
5473                    // 0 and the effect is not auxiliary, continue enumeration in case
5474                    // an auxiliary version of this effect type is available
5475                    found = true;
5476                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5477                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5478                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5479                        break;
5480                    }
5481                }
5482            }
5483            if (!found) {
5484                lStatus = BAD_VALUE;
5485                ALOGW("createEffect() effect not found");
5486                goto Exit;
5487            }
5488            // For same effect type, chose auxiliary version over insert version if
5489            // connect to output mix (Compliance to OpenSL ES)
5490            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5491                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5492                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5493            }
5494        }
5495
5496        // Do not allow auxiliary effects on a session different from 0 (output mix)
5497        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5498             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5499            lStatus = INVALID_OPERATION;
5500            goto Exit;
5501        }
5502
5503        // check recording permission for visualizer
5504        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5505            !recordingAllowed()) {
5506            lStatus = PERMISSION_DENIED;
5507            goto Exit;
5508        }
5509
5510        // return effect descriptor
5511        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5512
5513        // If output is not specified try to find a matching audio session ID in one of the
5514        // output threads.
5515        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5516        // because of code checking output when entering the function.
5517        // Note: io is never 0 when creating an effect on an input
5518        if (io == 0) {
5519             // look for the thread where the specified audio session is present
5520            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5521                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5522                    io = mPlaybackThreads.keyAt(i);
5523                    break;
5524                }
5525            }
5526            if (io == 0) {
5527               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5528                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5529                       io = mRecordThreads.keyAt(i);
5530                       break;
5531                   }
5532               }
5533            }
5534            // If no output thread contains the requested session ID, default to
5535            // first output. The effect chain will be moved to the correct output
5536            // thread when a track with the same session ID is created
5537            if (io == 0 && mPlaybackThreads.size()) {
5538                io = mPlaybackThreads.keyAt(0);
5539            }
5540            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5541        }
5542        ThreadBase *thread = checkRecordThread_l(io);
5543        if (thread == NULL) {
5544            thread = checkPlaybackThread_l(io);
5545            if (thread == NULL) {
5546                ALOGE("createEffect() unknown output thread");
5547                lStatus = BAD_VALUE;
5548                goto Exit;
5549            }
5550        }
5551
5552        wclient = mClients.valueFor(pid);
5553
5554        if (wclient != NULL) {
5555            client = wclient.promote();
5556        } else {
5557            client = new Client(this, pid);
5558            mClients.add(pid, client);
5559        }
5560
5561        // create effect on selected output thread
5562        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5563                &desc, enabled, &lStatus);
5564        if (handle != 0 && id != NULL) {
5565            *id = handle->id();
5566        }
5567    }
5568
5569Exit:
5570    if(status) {
5571        *status = lStatus;
5572    }
5573    return handle;
5574}
5575
5576status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5577{
5578    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5579            sessionId, srcOutput, dstOutput);
5580    Mutex::Autolock _l(mLock);
5581    if (srcOutput == dstOutput) {
5582        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
5583        return NO_ERROR;
5584    }
5585    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5586    if (srcThread == NULL) {
5587        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
5588        return BAD_VALUE;
5589    }
5590    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5591    if (dstThread == NULL) {
5592        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
5593        return BAD_VALUE;
5594    }
5595
5596    Mutex::Autolock _dl(dstThread->mLock);
5597    Mutex::Autolock _sl(srcThread->mLock);
5598    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5599
5600    return NO_ERROR;
5601}
5602
5603// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5604status_t AudioFlinger::moveEffectChain_l(int sessionId,
5605                                   AudioFlinger::PlaybackThread *srcThread,
5606                                   AudioFlinger::PlaybackThread *dstThread,
5607                                   bool reRegister)
5608{
5609    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5610            sessionId, srcThread, dstThread);
5611
5612    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5613    if (chain == 0) {
5614        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5615                sessionId, srcThread);
5616        return INVALID_OPERATION;
5617    }
5618
5619    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5620    // so that a new chain is created with correct parameters when first effect is added. This is
5621    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5622    // removed.
5623    srcThread->removeEffectChain_l(chain);
5624
5625    // transfer all effects one by one so that new effect chain is created on new thread with
5626    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5627    int dstOutput = dstThread->id();
5628    sp<EffectChain> dstChain;
5629    uint32_t strategy = 0; // prevent compiler warning
5630    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5631    while (effect != 0) {
5632        srcThread->removeEffect_l(effect);
5633        dstThread->addEffect_l(effect);
5634        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5635        if (effect->state() == EffectModule::ACTIVE ||
5636                effect->state() == EffectModule::STOPPING) {
5637            effect->start();
5638        }
5639        // if the move request is not received from audio policy manager, the effect must be
5640        // re-registered with the new strategy and output
5641        if (dstChain == 0) {
5642            dstChain = effect->chain().promote();
5643            if (dstChain == 0) {
5644                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5645                srcThread->addEffect_l(effect);
5646                return NO_INIT;
5647            }
5648            strategy = dstChain->strategy();
5649        }
5650        if (reRegister) {
5651            AudioSystem::unregisterEffect(effect->id());
5652            AudioSystem::registerEffect(&effect->desc(),
5653                                        dstOutput,
5654                                        strategy,
5655                                        sessionId,
5656                                        effect->id());
5657        }
5658        effect = chain->getEffectFromId_l(0);
5659    }
5660
5661    return NO_ERROR;
5662}
5663
5664
5665// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5666sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5667        const sp<AudioFlinger::Client>& client,
5668        const sp<IEffectClient>& effectClient,
5669        int32_t priority,
5670        int sessionId,
5671        effect_descriptor_t *desc,
5672        int *enabled,
5673        status_t *status
5674        )
5675{
5676    sp<EffectModule> effect;
5677    sp<EffectHandle> handle;
5678    status_t lStatus;
5679    sp<EffectChain> chain;
5680    bool chainCreated = false;
5681    bool effectCreated = false;
5682    bool effectRegistered = false;
5683
5684    lStatus = initCheck();
5685    if (lStatus != NO_ERROR) {
5686        ALOGW("createEffect_l() Audio driver not initialized.");
5687        goto Exit;
5688    }
5689
5690    // Do not allow effects with session ID 0 on direct output or duplicating threads
5691    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5692    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5693        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5694                desc->name, sessionId);
5695        lStatus = BAD_VALUE;
5696        goto Exit;
5697    }
5698    // Only Pre processor effects are allowed on input threads and only on input threads
5699    if ((mType == RECORD &&
5700            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5701            (mType != RECORD &&
5702                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5703        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5704                desc->name, desc->flags, mType);
5705        lStatus = BAD_VALUE;
5706        goto Exit;
5707    }
5708
5709    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5710
5711    { // scope for mLock
5712        Mutex::Autolock _l(mLock);
5713
5714        // check for existing effect chain with the requested audio session
5715        chain = getEffectChain_l(sessionId);
5716        if (chain == 0) {
5717            // create a new chain for this session
5718            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5719            chain = new EffectChain(this, sessionId);
5720            addEffectChain_l(chain);
5721            chain->setStrategy(getStrategyForSession_l(sessionId));
5722            chainCreated = true;
5723        } else {
5724            effect = chain->getEffectFromDesc_l(desc);
5725        }
5726
5727        ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
5728
5729        if (effect == 0) {
5730            int id = mAudioFlinger->nextUniqueId();
5731            // Check CPU and memory usage
5732            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5733            if (lStatus != NO_ERROR) {
5734                goto Exit;
5735            }
5736            effectRegistered = true;
5737            // create a new effect module if none present in the chain
5738            effect = new EffectModule(this, chain, desc, id, sessionId);
5739            lStatus = effect->status();
5740            if (lStatus != NO_ERROR) {
5741                goto Exit;
5742            }
5743            lStatus = chain->addEffect_l(effect);
5744            if (lStatus != NO_ERROR) {
5745                goto Exit;
5746            }
5747            effectCreated = true;
5748
5749            effect->setDevice(mDevice);
5750            effect->setMode(mAudioFlinger->getMode());
5751        }
5752        // create effect handle and connect it to effect module
5753        handle = new EffectHandle(effect, client, effectClient, priority);
5754        lStatus = effect->addHandle(handle);
5755        if (enabled) {
5756            *enabled = (int)effect->isEnabled();
5757        }
5758    }
5759
5760Exit:
5761    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5762        Mutex::Autolock _l(mLock);
5763        if (effectCreated) {
5764            chain->removeEffect_l(effect);
5765        }
5766        if (effectRegistered) {
5767            AudioSystem::unregisterEffect(effect->id());
5768        }
5769        if (chainCreated) {
5770            removeEffectChain_l(chain);
5771        }
5772        handle.clear();
5773    }
5774
5775    if(status) {
5776        *status = lStatus;
5777    }
5778    return handle;
5779}
5780
5781sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5782{
5783    sp<EffectModule> effect;
5784
5785    sp<EffectChain> chain = getEffectChain_l(sessionId);
5786    if (chain != 0) {
5787        effect = chain->getEffectFromId_l(effectId);
5788    }
5789    return effect;
5790}
5791
5792// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5793// PlaybackThread::mLock held
5794status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5795{
5796    // check for existing effect chain with the requested audio session
5797    int sessionId = effect->sessionId();
5798    sp<EffectChain> chain = getEffectChain_l(sessionId);
5799    bool chainCreated = false;
5800
5801    if (chain == 0) {
5802        // create a new chain for this session
5803        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5804        chain = new EffectChain(this, sessionId);
5805        addEffectChain_l(chain);
5806        chain->setStrategy(getStrategyForSession_l(sessionId));
5807        chainCreated = true;
5808    }
5809    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5810
5811    if (chain->getEffectFromId_l(effect->id()) != 0) {
5812        ALOGW("addEffect_l() %p effect %s already present in chain %p",
5813                this, effect->desc().name, chain.get());
5814        return BAD_VALUE;
5815    }
5816
5817    status_t status = chain->addEffect_l(effect);
5818    if (status != NO_ERROR) {
5819        if (chainCreated) {
5820            removeEffectChain_l(chain);
5821        }
5822        return status;
5823    }
5824
5825    effect->setDevice(mDevice);
5826    effect->setMode(mAudioFlinger->getMode());
5827    return NO_ERROR;
5828}
5829
5830void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5831
5832    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5833    effect_descriptor_t desc = effect->desc();
5834    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5835        detachAuxEffect_l(effect->id());
5836    }
5837
5838    sp<EffectChain> chain = effect->chain().promote();
5839    if (chain != 0) {
5840        // remove effect chain if removing last effect
5841        if (chain->removeEffect_l(effect) == 0) {
5842            removeEffectChain_l(chain);
5843        }
5844    } else {
5845        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5846    }
5847}
5848
5849void AudioFlinger::ThreadBase::lockEffectChains_l(
5850        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5851{
5852    effectChains = mEffectChains;
5853    for (size_t i = 0; i < mEffectChains.size(); i++) {
5854        mEffectChains[i]->lock();
5855    }
5856}
5857
5858void AudioFlinger::ThreadBase::unlockEffectChains(
5859        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5860{
5861    for (size_t i = 0; i < effectChains.size(); i++) {
5862        effectChains[i]->unlock();
5863    }
5864}
5865
5866sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5867{
5868    Mutex::Autolock _l(mLock);
5869    return getEffectChain_l(sessionId);
5870}
5871
5872sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5873{
5874    sp<EffectChain> chain;
5875
5876    size_t size = mEffectChains.size();
5877    for (size_t i = 0; i < size; i++) {
5878        if (mEffectChains[i]->sessionId() == sessionId) {
5879            chain = mEffectChains[i];
5880            break;
5881        }
5882    }
5883    return chain;
5884}
5885
5886void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
5887{
5888    Mutex::Autolock _l(mLock);
5889    size_t size = mEffectChains.size();
5890    for (size_t i = 0; i < size; i++) {
5891        mEffectChains[i]->setMode_l(mode);
5892    }
5893}
5894
5895void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5896                                                    const wp<EffectHandle>& handle,
5897                                                    bool unpiniflast) {
5898
5899    Mutex::Autolock _l(mLock);
5900    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5901    // delete the effect module if removing last handle on it
5902    if (effect->removeHandle(handle) == 0) {
5903        if (!effect->isPinned() || unpiniflast) {
5904            removeEffect_l(effect);
5905            AudioSystem::unregisterEffect(effect->id());
5906        }
5907    }
5908}
5909
5910status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5911{
5912    int session = chain->sessionId();
5913    int16_t *buffer = mMixBuffer;
5914    bool ownsBuffer = false;
5915
5916    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5917    if (session > 0) {
5918        // Only one effect chain can be present in direct output thread and it uses
5919        // the mix buffer as input
5920        if (mType != DIRECT) {
5921            size_t numSamples = mFrameCount * mChannelCount;
5922            buffer = new int16_t[numSamples];
5923            memset(buffer, 0, numSamples * sizeof(int16_t));
5924            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5925            ownsBuffer = true;
5926        }
5927
5928        // Attach all tracks with same session ID to this chain.
5929        for (size_t i = 0; i < mTracks.size(); ++i) {
5930            sp<Track> track = mTracks[i];
5931            if (session == track->sessionId()) {
5932                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5933                track->setMainBuffer(buffer);
5934                chain->incTrackCnt();
5935            }
5936        }
5937
5938        // indicate all active tracks in the chain
5939        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5940            sp<Track> track = mActiveTracks[i].promote();
5941            if (track == 0) continue;
5942            if (session == track->sessionId()) {
5943                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5944                chain->incActiveTrackCnt();
5945            }
5946        }
5947    }
5948
5949    chain->setInBuffer(buffer, ownsBuffer);
5950    chain->setOutBuffer(mMixBuffer);
5951    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5952    // chains list in order to be processed last as it contains output stage effects
5953    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5954    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5955    // after track specific effects and before output stage
5956    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5957    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5958    // Effect chain for other sessions are inserted at beginning of effect
5959    // chains list to be processed before output mix effects. Relative order between other
5960    // sessions is not important
5961    size_t size = mEffectChains.size();
5962    size_t i = 0;
5963    for (i = 0; i < size; i++) {
5964        if (mEffectChains[i]->sessionId() < session) break;
5965    }
5966    mEffectChains.insertAt(chain, i);
5967    checkSuspendOnAddEffectChain_l(chain);
5968
5969    return NO_ERROR;
5970}
5971
5972size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
5973{
5974    int session = chain->sessionId();
5975
5976    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
5977
5978    for (size_t i = 0; i < mEffectChains.size(); i++) {
5979        if (chain == mEffectChains[i]) {
5980            mEffectChains.removeAt(i);
5981            // detach all active tracks from the chain
5982            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5983                sp<Track> track = mActiveTracks[i].promote();
5984                if (track == 0) continue;
5985                if (session == track->sessionId()) {
5986                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
5987                            chain.get(), session);
5988                    chain->decActiveTrackCnt();
5989                }
5990            }
5991
5992            // detach all tracks with same session ID from this chain
5993            for (size_t i = 0; i < mTracks.size(); ++i) {
5994                sp<Track> track = mTracks[i];
5995                if (session == track->sessionId()) {
5996                    track->setMainBuffer(mMixBuffer);
5997                    chain->decTrackCnt();
5998                }
5999            }
6000            break;
6001        }
6002    }
6003    return mEffectChains.size();
6004}
6005
6006status_t AudioFlinger::PlaybackThread::attachAuxEffect(
6007        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6008{
6009    Mutex::Autolock _l(mLock);
6010    return attachAuxEffect_l(track, EffectId);
6011}
6012
6013status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6014        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6015{
6016    status_t status = NO_ERROR;
6017
6018    if (EffectId == 0) {
6019        track->setAuxBuffer(0, NULL);
6020    } else {
6021        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6022        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6023        if (effect != 0) {
6024            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6025                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6026            } else {
6027                status = INVALID_OPERATION;
6028            }
6029        } else {
6030            status = BAD_VALUE;
6031        }
6032    }
6033    return status;
6034}
6035
6036void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6037{
6038     for (size_t i = 0; i < mTracks.size(); ++i) {
6039        sp<Track> track = mTracks[i];
6040        if (track->auxEffectId() == effectId) {
6041            attachAuxEffect_l(track, 0);
6042        }
6043    }
6044}
6045
6046status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6047{
6048    // only one chain per input thread
6049    if (mEffectChains.size() != 0) {
6050        return INVALID_OPERATION;
6051    }
6052    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6053
6054    chain->setInBuffer(NULL);
6055    chain->setOutBuffer(NULL);
6056
6057    checkSuspendOnAddEffectChain_l(chain);
6058
6059    mEffectChains.add(chain);
6060
6061    return NO_ERROR;
6062}
6063
6064size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6065{
6066    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6067    ALOGW_IF(mEffectChains.size() != 1,
6068            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6069            chain.get(), mEffectChains.size(), this);
6070    if (mEffectChains.size() == 1) {
6071        mEffectChains.removeAt(0);
6072    }
6073    return 0;
6074}
6075
6076// ----------------------------------------------------------------------------
6077//  EffectModule implementation
6078// ----------------------------------------------------------------------------
6079
6080#undef LOG_TAG
6081#define LOG_TAG "AudioFlinger::EffectModule"
6082
6083AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6084                                        const wp<AudioFlinger::EffectChain>& chain,
6085                                        effect_descriptor_t *desc,
6086                                        int id,
6087                                        int sessionId)
6088    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6089      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6090{
6091    ALOGV("Constructor %p", this);
6092    int lStatus;
6093    sp<ThreadBase> thread = mThread.promote();
6094    if (thread == 0) {
6095        return;
6096    }
6097
6098    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6099
6100    // create effect engine from effect factory
6101    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6102
6103    if (mStatus != NO_ERROR) {
6104        return;
6105    }
6106    lStatus = init();
6107    if (lStatus < 0) {
6108        mStatus = lStatus;
6109        goto Error;
6110    }
6111
6112    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6113        mPinned = true;
6114    }
6115    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6116    return;
6117Error:
6118    EffectRelease(mEffectInterface);
6119    mEffectInterface = NULL;
6120    ALOGV("Constructor Error %d", mStatus);
6121}
6122
6123AudioFlinger::EffectModule::~EffectModule()
6124{
6125    ALOGV("Destructor %p", this);
6126    if (mEffectInterface != NULL) {
6127        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6128                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6129            sp<ThreadBase> thread = mThread.promote();
6130            if (thread != 0) {
6131                audio_stream_t *stream = thread->stream();
6132                if (stream != NULL) {
6133                    stream->remove_audio_effect(stream, mEffectInterface);
6134                }
6135            }
6136        }
6137        // release effect engine
6138        EffectRelease(mEffectInterface);
6139    }
6140}
6141
6142status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6143{
6144    status_t status;
6145
6146    Mutex::Autolock _l(mLock);
6147    // First handle in mHandles has highest priority and controls the effect module
6148    int priority = handle->priority();
6149    size_t size = mHandles.size();
6150    sp<EffectHandle> h;
6151    size_t i;
6152    for (i = 0; i < size; i++) {
6153        h = mHandles[i].promote();
6154        if (h == 0) continue;
6155        if (h->priority() <= priority) break;
6156    }
6157    // if inserted in first place, move effect control from previous owner to this handle
6158    if (i == 0) {
6159        bool enabled = false;
6160        if (h != 0) {
6161            enabled = h->enabled();
6162            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6163        }
6164        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6165        status = NO_ERROR;
6166    } else {
6167        status = ALREADY_EXISTS;
6168    }
6169    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6170    mHandles.insertAt(handle, i);
6171    return status;
6172}
6173
6174size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6175{
6176    Mutex::Autolock _l(mLock);
6177    size_t size = mHandles.size();
6178    size_t i;
6179    for (i = 0; i < size; i++) {
6180        if (mHandles[i] == handle) break;
6181    }
6182    if (i == size) {
6183        return size;
6184    }
6185    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6186
6187    bool enabled = false;
6188    EffectHandle *hdl = handle.unsafe_get();
6189    if (hdl) {
6190        ALOGV("removeHandle() unsafe_get OK");
6191        enabled = hdl->enabled();
6192    }
6193    mHandles.removeAt(i);
6194    size = mHandles.size();
6195    // if removed from first place, move effect control from this handle to next in line
6196    if (i == 0 && size != 0) {
6197        sp<EffectHandle> h = mHandles[0].promote();
6198        if (h != 0) {
6199            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6200        }
6201    }
6202
6203    // Prevent calls to process() and other functions on effect interface from now on.
6204    // The effect engine will be released by the destructor when the last strong reference on
6205    // this object is released which can happen after next process is called.
6206    if (size == 0 && !mPinned) {
6207        mState = DESTROYED;
6208    }
6209
6210    return size;
6211}
6212
6213sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6214{
6215    Mutex::Autolock _l(mLock);
6216    sp<EffectHandle> handle;
6217    if (mHandles.size() != 0) {
6218        handle = mHandles[0].promote();
6219    }
6220    return handle;
6221}
6222
6223void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6224{
6225    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6226    // keep a strong reference on this EffectModule to avoid calling the
6227    // destructor before we exit
6228    sp<EffectModule> keep(this);
6229    {
6230        sp<ThreadBase> thread = mThread.promote();
6231        if (thread != 0) {
6232            thread->disconnectEffect(keep, handle, unpiniflast);
6233        }
6234    }
6235}
6236
6237void AudioFlinger::EffectModule::updateState() {
6238    Mutex::Autolock _l(mLock);
6239
6240    switch (mState) {
6241    case RESTART:
6242        reset_l();
6243        // FALL THROUGH
6244
6245    case STARTING:
6246        // clear auxiliary effect input buffer for next accumulation
6247        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6248            memset(mConfig.inputCfg.buffer.raw,
6249                   0,
6250                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6251        }
6252        start_l();
6253        mState = ACTIVE;
6254        break;
6255    case STOPPING:
6256        stop_l();
6257        mDisableWaitCnt = mMaxDisableWaitCnt;
6258        mState = STOPPED;
6259        break;
6260    case STOPPED:
6261        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6262        // turn off sequence.
6263        if (--mDisableWaitCnt == 0) {
6264            reset_l();
6265            mState = IDLE;
6266        }
6267        break;
6268    default: //IDLE , ACTIVE, DESTROYED
6269        break;
6270    }
6271}
6272
6273void AudioFlinger::EffectModule::process()
6274{
6275    Mutex::Autolock _l(mLock);
6276
6277    if (mState == DESTROYED || mEffectInterface == NULL ||
6278            mConfig.inputCfg.buffer.raw == NULL ||
6279            mConfig.outputCfg.buffer.raw == NULL) {
6280        return;
6281    }
6282
6283    if (isProcessEnabled()) {
6284        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6285        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6286            ditherAndClamp(mConfig.inputCfg.buffer.s32,
6287                                        mConfig.inputCfg.buffer.s32,
6288                                        mConfig.inputCfg.buffer.frameCount/2);
6289        }
6290
6291        // do the actual processing in the effect engine
6292        int ret = (*mEffectInterface)->process(mEffectInterface,
6293                                               &mConfig.inputCfg.buffer,
6294                                               &mConfig.outputCfg.buffer);
6295
6296        // force transition to IDLE state when engine is ready
6297        if (mState == STOPPED && ret == -ENODATA) {
6298            mDisableWaitCnt = 1;
6299        }
6300
6301        // clear auxiliary effect input buffer for next accumulation
6302        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6303            memset(mConfig.inputCfg.buffer.raw, 0,
6304                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6305        }
6306    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6307                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6308        // If an insert effect is idle and input buffer is different from output buffer,
6309        // accumulate input onto output
6310        sp<EffectChain> chain = mChain.promote();
6311        if (chain != 0 && chain->activeTrackCnt() != 0) {
6312            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6313            int16_t *in = mConfig.inputCfg.buffer.s16;
6314            int16_t *out = mConfig.outputCfg.buffer.s16;
6315            for (size_t i = 0; i < frameCnt; i++) {
6316                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6317            }
6318        }
6319    }
6320}
6321
6322void AudioFlinger::EffectModule::reset_l()
6323{
6324    if (mEffectInterface == NULL) {
6325        return;
6326    }
6327    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6328}
6329
6330status_t AudioFlinger::EffectModule::configure()
6331{
6332    uint32_t channels;
6333    if (mEffectInterface == NULL) {
6334        return NO_INIT;
6335    }
6336
6337    sp<ThreadBase> thread = mThread.promote();
6338    if (thread == 0) {
6339        return DEAD_OBJECT;
6340    }
6341
6342    // TODO: handle configuration of effects replacing track process
6343    if (thread->channelCount() == 1) {
6344        channels = AUDIO_CHANNEL_OUT_MONO;
6345    } else {
6346        channels = AUDIO_CHANNEL_OUT_STEREO;
6347    }
6348
6349    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6350        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6351    } else {
6352        mConfig.inputCfg.channels = channels;
6353    }
6354    mConfig.outputCfg.channels = channels;
6355    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6356    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6357    mConfig.inputCfg.samplingRate = thread->sampleRate();
6358    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6359    mConfig.inputCfg.bufferProvider.cookie = NULL;
6360    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6361    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6362    mConfig.outputCfg.bufferProvider.cookie = NULL;
6363    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6364    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6365    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6366    // Insert effect:
6367    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6368    // always overwrites output buffer: input buffer == output buffer
6369    // - in other sessions:
6370    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6371    //      other effect: overwrites output buffer: input buffer == output buffer
6372    // Auxiliary effect:
6373    //      accumulates in output buffer: input buffer != output buffer
6374    // Therefore: accumulate <=> input buffer != output buffer
6375    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6376        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6377    } else {
6378        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6379    }
6380    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6381    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6382    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6383    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6384
6385    ALOGV("configure() %p thread %p buffer %p framecount %d",
6386            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6387
6388    status_t cmdStatus;
6389    uint32_t size = sizeof(int);
6390    status_t status = (*mEffectInterface)->command(mEffectInterface,
6391                                                   EFFECT_CMD_SET_CONFIG,
6392                                                   sizeof(effect_config_t),
6393                                                   &mConfig,
6394                                                   &size,
6395                                                   &cmdStatus);
6396    if (status == 0) {
6397        status = cmdStatus;
6398    }
6399
6400    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6401            (1000 * mConfig.outputCfg.buffer.frameCount);
6402
6403    return status;
6404}
6405
6406status_t AudioFlinger::EffectModule::init()
6407{
6408    Mutex::Autolock _l(mLock);
6409    if (mEffectInterface == NULL) {
6410        return NO_INIT;
6411    }
6412    status_t cmdStatus;
6413    uint32_t size = sizeof(status_t);
6414    status_t status = (*mEffectInterface)->command(mEffectInterface,
6415                                                   EFFECT_CMD_INIT,
6416                                                   0,
6417                                                   NULL,
6418                                                   &size,
6419                                                   &cmdStatus);
6420    if (status == 0) {
6421        status = cmdStatus;
6422    }
6423    return status;
6424}
6425
6426status_t AudioFlinger::EffectModule::start()
6427{
6428    Mutex::Autolock _l(mLock);
6429    return start_l();
6430}
6431
6432status_t AudioFlinger::EffectModule::start_l()
6433{
6434    if (mEffectInterface == NULL) {
6435        return NO_INIT;
6436    }
6437    status_t cmdStatus;
6438    uint32_t size = sizeof(status_t);
6439    status_t status = (*mEffectInterface)->command(mEffectInterface,
6440                                                   EFFECT_CMD_ENABLE,
6441                                                   0,
6442                                                   NULL,
6443                                                   &size,
6444                                                   &cmdStatus);
6445    if (status == 0) {
6446        status = cmdStatus;
6447    }
6448    if (status == 0 &&
6449            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6450             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6451        sp<ThreadBase> thread = mThread.promote();
6452        if (thread != 0) {
6453            audio_stream_t *stream = thread->stream();
6454            if (stream != NULL) {
6455                stream->add_audio_effect(stream, mEffectInterface);
6456            }
6457        }
6458    }
6459    return status;
6460}
6461
6462status_t AudioFlinger::EffectModule::stop()
6463{
6464    Mutex::Autolock _l(mLock);
6465    return stop_l();
6466}
6467
6468status_t AudioFlinger::EffectModule::stop_l()
6469{
6470    if (mEffectInterface == NULL) {
6471        return NO_INIT;
6472    }
6473    status_t cmdStatus;
6474    uint32_t size = sizeof(status_t);
6475    status_t status = (*mEffectInterface)->command(mEffectInterface,
6476                                                   EFFECT_CMD_DISABLE,
6477                                                   0,
6478                                                   NULL,
6479                                                   &size,
6480                                                   &cmdStatus);
6481    if (status == 0) {
6482        status = cmdStatus;
6483    }
6484    if (status == 0 &&
6485            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6486             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6487        sp<ThreadBase> thread = mThread.promote();
6488        if (thread != 0) {
6489            audio_stream_t *stream = thread->stream();
6490            if (stream != NULL) {
6491                stream->remove_audio_effect(stream, mEffectInterface);
6492            }
6493        }
6494    }
6495    return status;
6496}
6497
6498status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6499                                             uint32_t cmdSize,
6500                                             void *pCmdData,
6501                                             uint32_t *replySize,
6502                                             void *pReplyData)
6503{
6504    Mutex::Autolock _l(mLock);
6505//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6506
6507    if (mState == DESTROYED || mEffectInterface == NULL) {
6508        return NO_INIT;
6509    }
6510    status_t status = (*mEffectInterface)->command(mEffectInterface,
6511                                                   cmdCode,
6512                                                   cmdSize,
6513                                                   pCmdData,
6514                                                   replySize,
6515                                                   pReplyData);
6516    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6517        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6518        for (size_t i = 1; i < mHandles.size(); i++) {
6519            sp<EffectHandle> h = mHandles[i].promote();
6520            if (h != 0) {
6521                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6522            }
6523        }
6524    }
6525    return status;
6526}
6527
6528status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6529{
6530
6531    Mutex::Autolock _l(mLock);
6532    ALOGV("setEnabled %p enabled %d", this, enabled);
6533
6534    if (enabled != isEnabled()) {
6535        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6536        if (enabled && status != NO_ERROR) {
6537            return status;
6538        }
6539
6540        switch (mState) {
6541        // going from disabled to enabled
6542        case IDLE:
6543            mState = STARTING;
6544            break;
6545        case STOPPED:
6546            mState = RESTART;
6547            break;
6548        case STOPPING:
6549            mState = ACTIVE;
6550            break;
6551
6552        // going from enabled to disabled
6553        case RESTART:
6554            mState = STOPPED;
6555            break;
6556        case STARTING:
6557            mState = IDLE;
6558            break;
6559        case ACTIVE:
6560            mState = STOPPING;
6561            break;
6562        case DESTROYED:
6563            return NO_ERROR; // simply ignore as we are being destroyed
6564        }
6565        for (size_t i = 1; i < mHandles.size(); i++) {
6566            sp<EffectHandle> h = mHandles[i].promote();
6567            if (h != 0) {
6568                h->setEnabled(enabled);
6569            }
6570        }
6571    }
6572    return NO_ERROR;
6573}
6574
6575bool AudioFlinger::EffectModule::isEnabled()
6576{
6577    switch (mState) {
6578    case RESTART:
6579    case STARTING:
6580    case ACTIVE:
6581        return true;
6582    case IDLE:
6583    case STOPPING:
6584    case STOPPED:
6585    case DESTROYED:
6586    default:
6587        return false;
6588    }
6589}
6590
6591bool AudioFlinger::EffectModule::isProcessEnabled()
6592{
6593    switch (mState) {
6594    case RESTART:
6595    case ACTIVE:
6596    case STOPPING:
6597    case STOPPED:
6598        return true;
6599    case IDLE:
6600    case STARTING:
6601    case DESTROYED:
6602    default:
6603        return false;
6604    }
6605}
6606
6607status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6608{
6609    Mutex::Autolock _l(mLock);
6610    status_t status = NO_ERROR;
6611
6612    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6613    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6614    if (isProcessEnabled() &&
6615            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6616            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6617        status_t cmdStatus;
6618        uint32_t volume[2];
6619        uint32_t *pVolume = NULL;
6620        uint32_t size = sizeof(volume);
6621        volume[0] = *left;
6622        volume[1] = *right;
6623        if (controller) {
6624            pVolume = volume;
6625        }
6626        status = (*mEffectInterface)->command(mEffectInterface,
6627                                              EFFECT_CMD_SET_VOLUME,
6628                                              size,
6629                                              volume,
6630                                              &size,
6631                                              pVolume);
6632        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6633            *left = volume[0];
6634            *right = volume[1];
6635        }
6636    }
6637    return status;
6638}
6639
6640status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6641{
6642    Mutex::Autolock _l(mLock);
6643    status_t status = NO_ERROR;
6644    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6645        // audio pre processing modules on RecordThread can receive both output and
6646        // input device indication in the same call
6647        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6648        if (dev) {
6649            status_t cmdStatus;
6650            uint32_t size = sizeof(status_t);
6651
6652            status = (*mEffectInterface)->command(mEffectInterface,
6653                                                  EFFECT_CMD_SET_DEVICE,
6654                                                  sizeof(uint32_t),
6655                                                  &dev,
6656                                                  &size,
6657                                                  &cmdStatus);
6658            if (status == NO_ERROR) {
6659                status = cmdStatus;
6660            }
6661        }
6662        dev = device & AUDIO_DEVICE_IN_ALL;
6663        if (dev) {
6664            status_t cmdStatus;
6665            uint32_t size = sizeof(status_t);
6666
6667            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6668                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6669                                                  sizeof(uint32_t),
6670                                                  &dev,
6671                                                  &size,
6672                                                  &cmdStatus);
6673            if (status2 == NO_ERROR) {
6674                status2 = cmdStatus;
6675            }
6676            if (status == NO_ERROR) {
6677                status = status2;
6678            }
6679        }
6680    }
6681    return status;
6682}
6683
6684status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
6685{
6686    Mutex::Autolock _l(mLock);
6687    status_t status = NO_ERROR;
6688    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6689        status_t cmdStatus;
6690        uint32_t size = sizeof(status_t);
6691        status = (*mEffectInterface)->command(mEffectInterface,
6692                                              EFFECT_CMD_SET_AUDIO_MODE,
6693                                              sizeof(audio_mode_t),
6694                                              &mode,
6695                                              &size,
6696                                              &cmdStatus);
6697        if (status == NO_ERROR) {
6698            status = cmdStatus;
6699        }
6700    }
6701    return status;
6702}
6703
6704void AudioFlinger::EffectModule::setSuspended(bool suspended)
6705{
6706    Mutex::Autolock _l(mLock);
6707    mSuspended = suspended;
6708}
6709
6710bool AudioFlinger::EffectModule::suspended() const
6711{
6712    Mutex::Autolock _l(mLock);
6713    return mSuspended;
6714}
6715
6716status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6717{
6718    const size_t SIZE = 256;
6719    char buffer[SIZE];
6720    String8 result;
6721
6722    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6723    result.append(buffer);
6724
6725    bool locked = tryLock(mLock);
6726    // failed to lock - AudioFlinger is probably deadlocked
6727    if (!locked) {
6728        result.append("\t\tCould not lock Fx mutex:\n");
6729    }
6730
6731    result.append("\t\tSession Status State Engine:\n");
6732    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6733            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6734    result.append(buffer);
6735
6736    result.append("\t\tDescriptor:\n");
6737    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6738            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6739            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6740            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6741    result.append(buffer);
6742    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6743                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6744                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6745                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6746    result.append(buffer);
6747    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6748            mDescriptor.apiVersion,
6749            mDescriptor.flags);
6750    result.append(buffer);
6751    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6752            mDescriptor.name);
6753    result.append(buffer);
6754    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6755            mDescriptor.implementor);
6756    result.append(buffer);
6757
6758    result.append("\t\t- Input configuration:\n");
6759    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6760    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6761            (uint32_t)mConfig.inputCfg.buffer.raw,
6762            mConfig.inputCfg.buffer.frameCount,
6763            mConfig.inputCfg.samplingRate,
6764            mConfig.inputCfg.channels,
6765            mConfig.inputCfg.format);
6766    result.append(buffer);
6767
6768    result.append("\t\t- Output configuration:\n");
6769    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6770    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6771            (uint32_t)mConfig.outputCfg.buffer.raw,
6772            mConfig.outputCfg.buffer.frameCount,
6773            mConfig.outputCfg.samplingRate,
6774            mConfig.outputCfg.channels,
6775            mConfig.outputCfg.format);
6776    result.append(buffer);
6777
6778    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6779    result.append(buffer);
6780    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6781    for (size_t i = 0; i < mHandles.size(); ++i) {
6782        sp<EffectHandle> handle = mHandles[i].promote();
6783        if (handle != 0) {
6784            handle->dump(buffer, SIZE);
6785            result.append(buffer);
6786        }
6787    }
6788
6789    result.append("\n");
6790
6791    write(fd, result.string(), result.length());
6792
6793    if (locked) {
6794        mLock.unlock();
6795    }
6796
6797    return NO_ERROR;
6798}
6799
6800// ----------------------------------------------------------------------------
6801//  EffectHandle implementation
6802// ----------------------------------------------------------------------------
6803
6804#undef LOG_TAG
6805#define LOG_TAG "AudioFlinger::EffectHandle"
6806
6807AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6808                                        const sp<AudioFlinger::Client>& client,
6809                                        const sp<IEffectClient>& effectClient,
6810                                        int32_t priority)
6811    : BnEffect(),
6812    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6813    mPriority(priority), mHasControl(false), mEnabled(false)
6814{
6815    ALOGV("constructor %p", this);
6816
6817    if (client == 0) {
6818        return;
6819    }
6820    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6821    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6822    if (mCblkMemory != 0) {
6823        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6824
6825        if (mCblk) {
6826            new(mCblk) effect_param_cblk_t();
6827            mBuffer = (uint8_t *)mCblk + bufOffset;
6828         }
6829    } else {
6830        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6831        return;
6832    }
6833}
6834
6835AudioFlinger::EffectHandle::~EffectHandle()
6836{
6837    ALOGV("Destructor %p", this);
6838    disconnect(false);
6839    ALOGV("Destructor DONE %p", this);
6840}
6841
6842status_t AudioFlinger::EffectHandle::enable()
6843{
6844    ALOGV("enable %p", this);
6845    if (!mHasControl) return INVALID_OPERATION;
6846    if (mEffect == 0) return DEAD_OBJECT;
6847
6848    if (mEnabled) {
6849        return NO_ERROR;
6850    }
6851
6852    mEnabled = true;
6853
6854    sp<ThreadBase> thread = mEffect->thread().promote();
6855    if (thread != 0) {
6856        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6857    }
6858
6859    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6860    if (mEffect->suspended()) {
6861        return NO_ERROR;
6862    }
6863
6864    status_t status = mEffect->setEnabled(true);
6865    if (status != NO_ERROR) {
6866        if (thread != 0) {
6867            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6868        }
6869        mEnabled = false;
6870    }
6871    return status;
6872}
6873
6874status_t AudioFlinger::EffectHandle::disable()
6875{
6876    ALOGV("disable %p", this);
6877    if (!mHasControl) return INVALID_OPERATION;
6878    if (mEffect == 0) return DEAD_OBJECT;
6879
6880    if (!mEnabled) {
6881        return NO_ERROR;
6882    }
6883    mEnabled = false;
6884
6885    if (mEffect->suspended()) {
6886        return NO_ERROR;
6887    }
6888
6889    status_t status = mEffect->setEnabled(false);
6890
6891    sp<ThreadBase> thread = mEffect->thread().promote();
6892    if (thread != 0) {
6893        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6894    }
6895
6896    return status;
6897}
6898
6899void AudioFlinger::EffectHandle::disconnect()
6900{
6901    disconnect(true);
6902}
6903
6904void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6905{
6906    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6907    if (mEffect == 0) {
6908        return;
6909    }
6910    mEffect->disconnect(this, unpiniflast);
6911
6912    if (mHasControl && mEnabled) {
6913        sp<ThreadBase> thread = mEffect->thread().promote();
6914        if (thread != 0) {
6915            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6916        }
6917    }
6918
6919    // release sp on module => module destructor can be called now
6920    mEffect.clear();
6921    if (mClient != 0) {
6922        if (mCblk) {
6923            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6924        }
6925        mCblkMemory.clear();            // and free the shared memory
6926        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6927        mClient.clear();
6928    }
6929}
6930
6931status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6932                                             uint32_t cmdSize,
6933                                             void *pCmdData,
6934                                             uint32_t *replySize,
6935                                             void *pReplyData)
6936{
6937//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6938//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6939
6940    // only get parameter command is permitted for applications not controlling the effect
6941    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6942        return INVALID_OPERATION;
6943    }
6944    if (mEffect == 0) return DEAD_OBJECT;
6945    if (mClient == 0) return INVALID_OPERATION;
6946
6947    // handle commands that are not forwarded transparently to effect engine
6948    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6949        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6950        // no risk to block the whole media server process or mixer threads is we are stuck here
6951        Mutex::Autolock _l(mCblk->lock);
6952        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6953            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6954            mCblk->serverIndex = 0;
6955            mCblk->clientIndex = 0;
6956            return BAD_VALUE;
6957        }
6958        status_t status = NO_ERROR;
6959        while (mCblk->serverIndex < mCblk->clientIndex) {
6960            int reply;
6961            uint32_t rsize = sizeof(int);
6962            int *p = (int *)(mBuffer + mCblk->serverIndex);
6963            int size = *p++;
6964            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
6965                ALOGW("command(): invalid parameter block size");
6966                break;
6967            }
6968            effect_param_t *param = (effect_param_t *)p;
6969            if (param->psize == 0 || param->vsize == 0) {
6970                ALOGW("command(): null parameter or value size");
6971                mCblk->serverIndex += size;
6972                continue;
6973            }
6974            uint32_t psize = sizeof(effect_param_t) +
6975                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
6976                             param->vsize;
6977            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
6978                                            psize,
6979                                            p,
6980                                            &rsize,
6981                                            &reply);
6982            // stop at first error encountered
6983            if (ret != NO_ERROR) {
6984                status = ret;
6985                *(int *)pReplyData = reply;
6986                break;
6987            } else if (reply != NO_ERROR) {
6988                *(int *)pReplyData = reply;
6989                break;
6990            }
6991            mCblk->serverIndex += size;
6992        }
6993        mCblk->serverIndex = 0;
6994        mCblk->clientIndex = 0;
6995        return status;
6996    } else if (cmdCode == EFFECT_CMD_ENABLE) {
6997        *(int *)pReplyData = NO_ERROR;
6998        return enable();
6999    } else if (cmdCode == EFFECT_CMD_DISABLE) {
7000        *(int *)pReplyData = NO_ERROR;
7001        return disable();
7002    }
7003
7004    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7005}
7006
7007sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
7008    return mCblkMemory;
7009}
7010
7011void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
7012{
7013    ALOGV("setControl %p control %d", this, hasControl);
7014
7015    mHasControl = hasControl;
7016    mEnabled = enabled;
7017
7018    if (signal && mEffectClient != 0) {
7019        mEffectClient->controlStatusChanged(hasControl);
7020    }
7021}
7022
7023void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7024                                                 uint32_t cmdSize,
7025                                                 void *pCmdData,
7026                                                 uint32_t replySize,
7027                                                 void *pReplyData)
7028{
7029    if (mEffectClient != 0) {
7030        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7031    }
7032}
7033
7034
7035
7036void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7037{
7038    if (mEffectClient != 0) {
7039        mEffectClient->enableStatusChanged(enabled);
7040    }
7041}
7042
7043status_t AudioFlinger::EffectHandle::onTransact(
7044    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7045{
7046    return BnEffect::onTransact(code, data, reply, flags);
7047}
7048
7049
7050void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7051{
7052    bool locked = mCblk ? tryLock(mCblk->lock) : false;
7053
7054    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7055            (mClient == NULL) ? getpid() : mClient->pid(),
7056            mPriority,
7057            mHasControl,
7058            !locked,
7059            mCblk ? mCblk->clientIndex : 0,
7060            mCblk ? mCblk->serverIndex : 0
7061            );
7062
7063    if (locked) {
7064        mCblk->lock.unlock();
7065    }
7066}
7067
7068#undef LOG_TAG
7069#define LOG_TAG "AudioFlinger::EffectChain"
7070
7071AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7072                                        int sessionId)
7073    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7074      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7075      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7076{
7077    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7078    sp<ThreadBase> thread = mThread.promote();
7079    if (thread == 0) {
7080        return;
7081    }
7082    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7083                                    thread->frameCount();
7084}
7085
7086AudioFlinger::EffectChain::~EffectChain()
7087{
7088    if (mOwnInBuffer) {
7089        delete mInBuffer;
7090    }
7091
7092}
7093
7094// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7095sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7096{
7097    sp<EffectModule> effect;
7098    size_t size = mEffects.size();
7099
7100    for (size_t i = 0; i < size; i++) {
7101        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7102            effect = mEffects[i];
7103            break;
7104        }
7105    }
7106    return effect;
7107}
7108
7109// getEffectFromId_l() must be called with ThreadBase::mLock held
7110sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7111{
7112    sp<EffectModule> effect;
7113    size_t size = mEffects.size();
7114
7115    for (size_t i = 0; i < size; i++) {
7116        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7117        if (id == 0 || mEffects[i]->id() == id) {
7118            effect = mEffects[i];
7119            break;
7120        }
7121    }
7122    return effect;
7123}
7124
7125// getEffectFromType_l() must be called with ThreadBase::mLock held
7126sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7127        const effect_uuid_t *type)
7128{
7129    sp<EffectModule> effect;
7130    size_t size = mEffects.size();
7131
7132    for (size_t i = 0; i < size; i++) {
7133        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7134            effect = mEffects[i];
7135            break;
7136        }
7137    }
7138    return effect;
7139}
7140
7141// Must be called with EffectChain::mLock locked
7142void AudioFlinger::EffectChain::process_l()
7143{
7144    sp<ThreadBase> thread = mThread.promote();
7145    if (thread == 0) {
7146        ALOGW("process_l(): cannot promote mixer thread");
7147        return;
7148    }
7149    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7150            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7151    // always process effects unless no more tracks are on the session and the effect tail
7152    // has been rendered
7153    bool doProcess = true;
7154    if (!isGlobalSession) {
7155        bool tracksOnSession = (trackCnt() != 0);
7156
7157        if (!tracksOnSession && mTailBufferCount == 0) {
7158            doProcess = false;
7159        }
7160
7161        if (activeTrackCnt() == 0) {
7162            // if no track is active and the effect tail has not been rendered,
7163            // the input buffer must be cleared here as the mixer process will not do it
7164            if (tracksOnSession || mTailBufferCount > 0) {
7165                size_t numSamples = thread->frameCount() * thread->channelCount();
7166                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7167                if (mTailBufferCount > 0) {
7168                    mTailBufferCount--;
7169                }
7170            }
7171        }
7172    }
7173
7174    size_t size = mEffects.size();
7175    if (doProcess) {
7176        for (size_t i = 0; i < size; i++) {
7177            mEffects[i]->process();
7178        }
7179    }
7180    for (size_t i = 0; i < size; i++) {
7181        mEffects[i]->updateState();
7182    }
7183}
7184
7185// addEffect_l() must be called with PlaybackThread::mLock held
7186status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7187{
7188    effect_descriptor_t desc = effect->desc();
7189    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7190
7191    Mutex::Autolock _l(mLock);
7192    effect->setChain(this);
7193    sp<ThreadBase> thread = mThread.promote();
7194    if (thread == 0) {
7195        return NO_INIT;
7196    }
7197    effect->setThread(thread);
7198
7199    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7200        // Auxiliary effects are inserted at the beginning of mEffects vector as
7201        // they are processed first and accumulated in chain input buffer
7202        mEffects.insertAt(effect, 0);
7203
7204        // the input buffer for auxiliary effect contains mono samples in
7205        // 32 bit format. This is to avoid saturation in AudoMixer
7206        // accumulation stage. Saturation is done in EffectModule::process() before
7207        // calling the process in effect engine
7208        size_t numSamples = thread->frameCount();
7209        int32_t *buffer = new int32_t[numSamples];
7210        memset(buffer, 0, numSamples * sizeof(int32_t));
7211        effect->setInBuffer((int16_t *)buffer);
7212        // auxiliary effects output samples to chain input buffer for further processing
7213        // by insert effects
7214        effect->setOutBuffer(mInBuffer);
7215    } else {
7216        // Insert effects are inserted at the end of mEffects vector as they are processed
7217        //  after track and auxiliary effects.
7218        // Insert effect order as a function of indicated preference:
7219        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7220        //  another effect is present
7221        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7222        //  last effect claiming first position
7223        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7224        //  first effect claiming last position
7225        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7226        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7227        // already present
7228
7229        int size = (int)mEffects.size();
7230        int idx_insert = size;
7231        int idx_insert_first = -1;
7232        int idx_insert_last = -1;
7233
7234        for (int i = 0; i < size; i++) {
7235            effect_descriptor_t d = mEffects[i]->desc();
7236            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7237            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7238            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7239                // check invalid effect chaining combinations
7240                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7241                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7242                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7243                    return INVALID_OPERATION;
7244                }
7245                // remember position of first insert effect and by default
7246                // select this as insert position for new effect
7247                if (idx_insert == size) {
7248                    idx_insert = i;
7249                }
7250                // remember position of last insert effect claiming
7251                // first position
7252                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7253                    idx_insert_first = i;
7254                }
7255                // remember position of first insert effect claiming
7256                // last position
7257                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7258                    idx_insert_last == -1) {
7259                    idx_insert_last = i;
7260                }
7261            }
7262        }
7263
7264        // modify idx_insert from first position if needed
7265        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7266            if (idx_insert_last != -1) {
7267                idx_insert = idx_insert_last;
7268            } else {
7269                idx_insert = size;
7270            }
7271        } else {
7272            if (idx_insert_first != -1) {
7273                idx_insert = idx_insert_first + 1;
7274            }
7275        }
7276
7277        // always read samples from chain input buffer
7278        effect->setInBuffer(mInBuffer);
7279
7280        // if last effect in the chain, output samples to chain
7281        // output buffer, otherwise to chain input buffer
7282        if (idx_insert == size) {
7283            if (idx_insert != 0) {
7284                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7285                mEffects[idx_insert-1]->configure();
7286            }
7287            effect->setOutBuffer(mOutBuffer);
7288        } else {
7289            effect->setOutBuffer(mInBuffer);
7290        }
7291        mEffects.insertAt(effect, idx_insert);
7292
7293        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7294    }
7295    effect->configure();
7296    return NO_ERROR;
7297}
7298
7299// removeEffect_l() must be called with PlaybackThread::mLock held
7300size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7301{
7302    Mutex::Autolock _l(mLock);
7303    int size = (int)mEffects.size();
7304    int i;
7305    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7306
7307    for (i = 0; i < size; i++) {
7308        if (effect == mEffects[i]) {
7309            // calling stop here will remove pre-processing effect from the audio HAL.
7310            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7311            // the middle of a read from audio HAL
7312            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7313                    mEffects[i]->state() == EffectModule::STOPPING) {
7314                mEffects[i]->stop();
7315            }
7316            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7317                delete[] effect->inBuffer();
7318            } else {
7319                if (i == size - 1 && i != 0) {
7320                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7321                    mEffects[i - 1]->configure();
7322                }
7323            }
7324            mEffects.removeAt(i);
7325            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7326            break;
7327        }
7328    }
7329
7330    return mEffects.size();
7331}
7332
7333// setDevice_l() must be called with PlaybackThread::mLock held
7334void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7335{
7336    size_t size = mEffects.size();
7337    for (size_t i = 0; i < size; i++) {
7338        mEffects[i]->setDevice(device);
7339    }
7340}
7341
7342// setMode_l() must be called with PlaybackThread::mLock held
7343void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode)
7344{
7345    size_t size = mEffects.size();
7346    for (size_t i = 0; i < size; i++) {
7347        mEffects[i]->setMode(mode);
7348    }
7349}
7350
7351// setVolume_l() must be called with PlaybackThread::mLock held
7352bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7353{
7354    uint32_t newLeft = *left;
7355    uint32_t newRight = *right;
7356    bool hasControl = false;
7357    int ctrlIdx = -1;
7358    size_t size = mEffects.size();
7359
7360    // first update volume controller
7361    for (size_t i = size; i > 0; i--) {
7362        if (mEffects[i - 1]->isProcessEnabled() &&
7363            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7364            ctrlIdx = i - 1;
7365            hasControl = true;
7366            break;
7367        }
7368    }
7369
7370    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7371        if (hasControl) {
7372            *left = mNewLeftVolume;
7373            *right = mNewRightVolume;
7374        }
7375        return hasControl;
7376    }
7377
7378    mVolumeCtrlIdx = ctrlIdx;
7379    mLeftVolume = newLeft;
7380    mRightVolume = newRight;
7381
7382    // second get volume update from volume controller
7383    if (ctrlIdx >= 0) {
7384        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7385        mNewLeftVolume = newLeft;
7386        mNewRightVolume = newRight;
7387    }
7388    // then indicate volume to all other effects in chain.
7389    // Pass altered volume to effects before volume controller
7390    // and requested volume to effects after controller
7391    uint32_t lVol = newLeft;
7392    uint32_t rVol = newRight;
7393
7394    for (size_t i = 0; i < size; i++) {
7395        if ((int)i == ctrlIdx) continue;
7396        // this also works for ctrlIdx == -1 when there is no volume controller
7397        if ((int)i > ctrlIdx) {
7398            lVol = *left;
7399            rVol = *right;
7400        }
7401        mEffects[i]->setVolume(&lVol, &rVol, false);
7402    }
7403    *left = newLeft;
7404    *right = newRight;
7405
7406    return hasControl;
7407}
7408
7409status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7410{
7411    const size_t SIZE = 256;
7412    char buffer[SIZE];
7413    String8 result;
7414
7415    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7416    result.append(buffer);
7417
7418    bool locked = tryLock(mLock);
7419    // failed to lock - AudioFlinger is probably deadlocked
7420    if (!locked) {
7421        result.append("\tCould not lock mutex:\n");
7422    }
7423
7424    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7425    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7426            mEffects.size(),
7427            (uint32_t)mInBuffer,
7428            (uint32_t)mOutBuffer,
7429            mActiveTrackCnt);
7430    result.append(buffer);
7431    write(fd, result.string(), result.size());
7432
7433    for (size_t i = 0; i < mEffects.size(); ++i) {
7434        sp<EffectModule> effect = mEffects[i];
7435        if (effect != 0) {
7436            effect->dump(fd, args);
7437        }
7438    }
7439
7440    if (locked) {
7441        mLock.unlock();
7442    }
7443
7444    return NO_ERROR;
7445}
7446
7447// must be called with ThreadBase::mLock held
7448void AudioFlinger::EffectChain::setEffectSuspended_l(
7449        const effect_uuid_t *type, bool suspend)
7450{
7451    sp<SuspendedEffectDesc> desc;
7452    // use effect type UUID timelow as key as there is no real risk of identical
7453    // timeLow fields among effect type UUIDs.
7454    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7455    if (suspend) {
7456        if (index >= 0) {
7457            desc = mSuspendedEffects.valueAt(index);
7458        } else {
7459            desc = new SuspendedEffectDesc();
7460            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7461            mSuspendedEffects.add(type->timeLow, desc);
7462            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7463        }
7464        if (desc->mRefCount++ == 0) {
7465            sp<EffectModule> effect = getEffectIfEnabled(type);
7466            if (effect != 0) {
7467                desc->mEffect = effect;
7468                effect->setSuspended(true);
7469                effect->setEnabled(false);
7470            }
7471        }
7472    } else {
7473        if (index < 0) {
7474            return;
7475        }
7476        desc = mSuspendedEffects.valueAt(index);
7477        if (desc->mRefCount <= 0) {
7478            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7479            desc->mRefCount = 1;
7480        }
7481        if (--desc->mRefCount == 0) {
7482            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7483            if (desc->mEffect != 0) {
7484                sp<EffectModule> effect = desc->mEffect.promote();
7485                if (effect != 0) {
7486                    effect->setSuspended(false);
7487                    sp<EffectHandle> handle = effect->controlHandle();
7488                    if (handle != 0) {
7489                        effect->setEnabled(handle->enabled());
7490                    }
7491                }
7492                desc->mEffect.clear();
7493            }
7494            mSuspendedEffects.removeItemsAt(index);
7495        }
7496    }
7497}
7498
7499// must be called with ThreadBase::mLock held
7500void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7501{
7502    sp<SuspendedEffectDesc> desc;
7503
7504    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7505    if (suspend) {
7506        if (index >= 0) {
7507            desc = mSuspendedEffects.valueAt(index);
7508        } else {
7509            desc = new SuspendedEffectDesc();
7510            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7511            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7512        }
7513        if (desc->mRefCount++ == 0) {
7514            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7515            for (size_t i = 0; i < effects.size(); i++) {
7516                setEffectSuspended_l(&effects[i]->desc().type, true);
7517            }
7518        }
7519    } else {
7520        if (index < 0) {
7521            return;
7522        }
7523        desc = mSuspendedEffects.valueAt(index);
7524        if (desc->mRefCount <= 0) {
7525            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7526            desc->mRefCount = 1;
7527        }
7528        if (--desc->mRefCount == 0) {
7529            Vector<const effect_uuid_t *> types;
7530            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7531                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7532                    continue;
7533                }
7534                types.add(&mSuspendedEffects.valueAt(i)->mType);
7535            }
7536            for (size_t i = 0; i < types.size(); i++) {
7537                setEffectSuspended_l(types[i], false);
7538            }
7539            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7540            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7541        }
7542    }
7543}
7544
7545
7546// The volume effect is used for automated tests only
7547#ifndef OPENSL_ES_H_
7548static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7549                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7550const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7551#endif //OPENSL_ES_H_
7552
7553bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7554{
7555    // auxiliary effects and visualizer are never suspended on output mix
7556    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7557        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7558         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7559         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7560        return false;
7561    }
7562    return true;
7563}
7564
7565Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7566{
7567    Vector< sp<EffectModule> > effects;
7568    for (size_t i = 0; i < mEffects.size(); i++) {
7569        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7570            continue;
7571        }
7572        effects.add(mEffects[i]);
7573    }
7574    return effects;
7575}
7576
7577sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7578                                                            const effect_uuid_t *type)
7579{
7580    sp<EffectModule> effect;
7581    effect = getEffectFromType_l(type);
7582    if (effect != 0 && !effect->isEnabled()) {
7583        effect.clear();
7584    }
7585    return effect;
7586}
7587
7588void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7589                                                            bool enabled)
7590{
7591    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7592    if (enabled) {
7593        if (index < 0) {
7594            // if the effect is not suspend check if all effects are suspended
7595            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7596            if (index < 0) {
7597                return;
7598            }
7599            if (!isEffectEligibleForSuspend(effect->desc())) {
7600                return;
7601            }
7602            setEffectSuspended_l(&effect->desc().type, enabled);
7603            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7604            if (index < 0) {
7605                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7606                return;
7607            }
7608        }
7609        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7610             effect->desc().type.timeLow);
7611        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7612        // if effect is requested to suspended but was not yet enabled, supend it now.
7613        if (desc->mEffect == 0) {
7614            desc->mEffect = effect;
7615            effect->setEnabled(false);
7616            effect->setSuspended(true);
7617        }
7618    } else {
7619        if (index < 0) {
7620            return;
7621        }
7622        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7623             effect->desc().type.timeLow);
7624        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7625        desc->mEffect.clear();
7626        effect->setSuspended(false);
7627    }
7628}
7629
7630#undef LOG_TAG
7631#define LOG_TAG "AudioFlinger"
7632
7633// ----------------------------------------------------------------------------
7634
7635status_t AudioFlinger::onTransact(
7636        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7637{
7638    return BnAudioFlinger::onTransact(code, data, reply, flags);
7639}
7640
7641}; // namespace android
7642