AudioFlinger.cpp revision fff6d715a8db0daf08a50634f242c40268de3d49 
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        audio_stream_type_t 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(int 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(audio_stream_type_t 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(audio_stream_type_t 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(audio_stream_type_t 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(audio_stream_type_t 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    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
1386    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
1387            stream = (audio_stream_type_t) (stream + 1)) {
1388        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1389        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1390        mStreamTypes[stream].valid = true;
1391    }
1392}
1393
1394AudioFlinger::PlaybackThread::~PlaybackThread()
1395{
1396    delete [] mMixBuffer;
1397}
1398
1399status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1400{
1401    dumpInternals(fd, args);
1402    dumpTracks(fd, args);
1403    dumpEffectChains(fd, args);
1404    return NO_ERROR;
1405}
1406
1407status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1408{
1409    const size_t SIZE = 256;
1410    char buffer[SIZE];
1411    String8 result;
1412
1413    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1414    result.append(buffer);
1415    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1416    for (size_t i = 0; i < mTracks.size(); ++i) {
1417        sp<Track> track = mTracks[i];
1418        if (track != 0) {
1419            track->dump(buffer, SIZE);
1420            result.append(buffer);
1421        }
1422    }
1423
1424    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1425    result.append(buffer);
1426    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1427    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1428        wp<Track> wTrack = mActiveTracks[i];
1429        if (wTrack != 0) {
1430            sp<Track> track = wTrack.promote();
1431            if (track != 0) {
1432                track->dump(buffer, SIZE);
1433                result.append(buffer);
1434            }
1435        }
1436    }
1437    write(fd, result.string(), result.size());
1438    return NO_ERROR;
1439}
1440
1441status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1442{
1443    const size_t SIZE = 256;
1444    char buffer[SIZE];
1445    String8 result;
1446
1447    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1448    result.append(buffer);
1449    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1450    result.append(buffer);
1451    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1452    result.append(buffer);
1453    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1454    result.append(buffer);
1455    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1456    result.append(buffer);
1457    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1458    result.append(buffer);
1459    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1460    result.append(buffer);
1461    write(fd, result.string(), result.size());
1462
1463    dumpBase(fd, args);
1464
1465    return NO_ERROR;
1466}
1467
1468// Thread virtuals
1469status_t AudioFlinger::PlaybackThread::readyToRun()
1470{
1471    status_t status = initCheck();
1472    if (status == NO_ERROR) {
1473        ALOGI("AudioFlinger's thread %p ready to run", this);
1474    } else {
1475        ALOGE("No working audio driver found.");
1476    }
1477    return status;
1478}
1479
1480void AudioFlinger::PlaybackThread::onFirstRef()
1481{
1482    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1483}
1484
1485// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1486sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1487        const sp<AudioFlinger::Client>& client,
1488        audio_stream_type_t streamType,
1489        uint32_t sampleRate,
1490        uint32_t format,
1491        uint32_t channelMask,
1492        int frameCount,
1493        const sp<IMemory>& sharedBuffer,
1494        int sessionId,
1495        status_t *status)
1496{
1497    sp<Track> track;
1498    status_t lStatus;
1499
1500    if (mType == DIRECT) {
1501        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1502            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1503                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1504                        "for output %p with format %d",
1505                        sampleRate, format, channelMask, mOutput, mFormat);
1506                lStatus = BAD_VALUE;
1507                goto Exit;
1508            }
1509        }
1510    } else {
1511        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1512        if (sampleRate > mSampleRate*2) {
1513            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1514            lStatus = BAD_VALUE;
1515            goto Exit;
1516        }
1517    }
1518
1519    lStatus = initCheck();
1520    if (lStatus != NO_ERROR) {
1521        ALOGE("Audio driver not initialized.");
1522        goto Exit;
1523    }
1524
1525    { // scope for mLock
1526        Mutex::Autolock _l(mLock);
1527
1528        // all tracks in same audio session must share the same routing strategy otherwise
1529        // conflicts will happen when tracks are moved from one output to another by audio policy
1530        // manager
1531        uint32_t strategy =
1532                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1533        for (size_t i = 0; i < mTracks.size(); ++i) {
1534            sp<Track> t = mTracks[i];
1535            if (t != 0) {
1536                uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
1537                if (sessionId == t->sessionId() && strategy != actual) {
1538                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1539                            strategy, actual);
1540                    lStatus = BAD_VALUE;
1541                    goto Exit;
1542                }
1543            }
1544        }
1545
1546        track = new Track(this, client, streamType, sampleRate, format,
1547                channelMask, frameCount, sharedBuffer, sessionId);
1548        if (track->getCblk() == NULL || track->name() < 0) {
1549            lStatus = NO_MEMORY;
1550            goto Exit;
1551        }
1552        mTracks.add(track);
1553
1554        sp<EffectChain> chain = getEffectChain_l(sessionId);
1555        if (chain != 0) {
1556            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1557            track->setMainBuffer(chain->inBuffer());
1558            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1559            chain->incTrackCnt();
1560        }
1561
1562        // invalidate track immediately if the stream type was moved to another thread since
1563        // createTrack() was called by the client process.
1564        if (!mStreamTypes[streamType].valid) {
1565            ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1566                 this, streamType);
1567            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1568        }
1569    }
1570    lStatus = NO_ERROR;
1571
1572Exit:
1573    if(status) {
1574        *status = lStatus;
1575    }
1576    return track;
1577}
1578
1579uint32_t AudioFlinger::PlaybackThread::latency() const
1580{
1581    Mutex::Autolock _l(mLock);
1582    if (initCheck() == NO_ERROR) {
1583        return mOutput->stream->get_latency(mOutput->stream);
1584    } else {
1585        return 0;
1586    }
1587}
1588
1589status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1590{
1591    mMasterVolume = value;
1592    return NO_ERROR;
1593}
1594
1595status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1596{
1597    mMasterMute = muted;
1598    return NO_ERROR;
1599}
1600
1601float AudioFlinger::PlaybackThread::masterVolume() const
1602{
1603    return mMasterVolume;
1604}
1605
1606bool AudioFlinger::PlaybackThread::masterMute() const
1607{
1608    return mMasterMute;
1609}
1610
1611status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
1612{
1613    mStreamTypes[stream].volume = value;
1614    return NO_ERROR;
1615}
1616
1617status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
1618{
1619    mStreamTypes[stream].mute = muted;
1620    return NO_ERROR;
1621}
1622
1623float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
1624{
1625    return mStreamTypes[stream].volume;
1626}
1627
1628bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const
1629{
1630    return mStreamTypes[stream].mute;
1631}
1632
1633// addTrack_l() must be called with ThreadBase::mLock held
1634status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1635{
1636    status_t status = ALREADY_EXISTS;
1637
1638    // set retry count for buffer fill
1639    track->mRetryCount = kMaxTrackStartupRetries;
1640    if (mActiveTracks.indexOf(track) < 0) {
1641        // the track is newly added, make sure it fills up all its
1642        // buffers before playing. This is to ensure the client will
1643        // effectively get the latency it requested.
1644        track->mFillingUpStatus = Track::FS_FILLING;
1645        track->mResetDone = false;
1646        mActiveTracks.add(track);
1647        if (track->mainBuffer() != mMixBuffer) {
1648            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1649            if (chain != 0) {
1650                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1651                chain->incActiveTrackCnt();
1652            }
1653        }
1654
1655        status = NO_ERROR;
1656    }
1657
1658    ALOGV("mWaitWorkCV.broadcast");
1659    mWaitWorkCV.broadcast();
1660
1661    return status;
1662}
1663
1664// destroyTrack_l() must be called with ThreadBase::mLock held
1665void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1666{
1667    track->mState = TrackBase::TERMINATED;
1668    if (mActiveTracks.indexOf(track) < 0) {
1669        removeTrack_l(track);
1670    }
1671}
1672
1673void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1674{
1675    mTracks.remove(track);
1676    deleteTrackName_l(track->name());
1677    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1678    if (chain != 0) {
1679        chain->decTrackCnt();
1680    }
1681}
1682
1683String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1684{
1685    String8 out_s8 = String8("");
1686    char *s;
1687
1688    Mutex::Autolock _l(mLock);
1689    if (initCheck() != NO_ERROR) {
1690        return out_s8;
1691    }
1692
1693    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1694    out_s8 = String8(s);
1695    free(s);
1696    return out_s8;
1697}
1698
1699// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1700void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1701    AudioSystem::OutputDescriptor desc;
1702    void *param2 = 0;
1703
1704    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1705
1706    switch (event) {
1707    case AudioSystem::OUTPUT_OPENED:
1708    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1709        desc.channels = mChannelMask;
1710        desc.samplingRate = mSampleRate;
1711        desc.format = mFormat;
1712        desc.frameCount = mFrameCount;
1713        desc.latency = latency();
1714        param2 = &desc;
1715        break;
1716
1717    case AudioSystem::STREAM_CONFIG_CHANGED:
1718        param2 = &param;
1719    case AudioSystem::OUTPUT_CLOSED:
1720    default:
1721        break;
1722    }
1723    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1724}
1725
1726void AudioFlinger::PlaybackThread::readOutputParameters()
1727{
1728    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1729    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1730    mChannelCount = (uint16_t)popcount(mChannelMask);
1731    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1732    mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
1733    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1734
1735    // FIXME - Current mixer implementation only supports stereo output: Always
1736    // Allocate a stereo buffer even if HW output is mono.
1737    if (mMixBuffer != NULL) delete[] mMixBuffer;
1738    mMixBuffer = new int16_t[mFrameCount * 2];
1739    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1740
1741    // force reconfiguration of effect chains and engines to take new buffer size and audio
1742    // parameters into account
1743    // Note that mLock is not held when readOutputParameters() is called from the constructor
1744    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1745    // matter.
1746    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1747    Vector< sp<EffectChain> > effectChains = mEffectChains;
1748    for (size_t i = 0; i < effectChains.size(); i ++) {
1749        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1750    }
1751}
1752
1753status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1754{
1755    if (halFrames == 0 || dspFrames == 0) {
1756        return BAD_VALUE;
1757    }
1758    Mutex::Autolock _l(mLock);
1759    if (initCheck() != NO_ERROR) {
1760        return INVALID_OPERATION;
1761    }
1762    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1763
1764    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1765}
1766
1767uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1768{
1769    Mutex::Autolock _l(mLock);
1770    uint32_t result = 0;
1771    if (getEffectChain_l(sessionId) != 0) {
1772        result = EFFECT_SESSION;
1773    }
1774
1775    for (size_t i = 0; i < mTracks.size(); ++i) {
1776        sp<Track> track = mTracks[i];
1777        if (sessionId == track->sessionId() &&
1778                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1779            result |= TRACK_SESSION;
1780            break;
1781        }
1782    }
1783
1784    return result;
1785}
1786
1787uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1788{
1789    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1790    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1791    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1792        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1793    }
1794    for (size_t i = 0; i < mTracks.size(); i++) {
1795        sp<Track> track = mTracks[i];
1796        if (sessionId == track->sessionId() &&
1797                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1798            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1799        }
1800    }
1801    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1802}
1803
1804
1805AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1806{
1807    Mutex::Autolock _l(mLock);
1808    return mOutput;
1809}
1810
1811AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1812{
1813    Mutex::Autolock _l(mLock);
1814    AudioStreamOut *output = mOutput;
1815    mOutput = NULL;
1816    return output;
1817}
1818
1819// this method must always be called either with ThreadBase mLock held or inside the thread loop
1820audio_stream_t* AudioFlinger::PlaybackThread::stream()
1821{
1822    if (mOutput == NULL) {
1823        return NULL;
1824    }
1825    return &mOutput->stream->common;
1826}
1827
1828uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
1829{
1830    // A2DP output latency is not due only to buffering capacity. It also reflects encoding,
1831    // decoding and transfer time. So sleeping for half of the latency would likely cause
1832    // underruns
1833    if (audio_is_a2dp_device((audio_devices_t)mDevice)) {
1834        return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000);
1835    } else {
1836        return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
1837    }
1838}
1839
1840// ----------------------------------------------------------------------------
1841
1842AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1843    :   PlaybackThread(audioFlinger, output, id, device),
1844        mAudioMixer(NULL)
1845{
1846    mType = ThreadBase::MIXER;
1847    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1848
1849    // FIXME - Current mixer implementation only supports stereo output
1850    if (mChannelCount == 1) {
1851        ALOGE("Invalid audio hardware channel count");
1852    }
1853}
1854
1855AudioFlinger::MixerThread::~MixerThread()
1856{
1857    delete mAudioMixer;
1858}
1859
1860bool AudioFlinger::MixerThread::threadLoop()
1861{
1862    Vector< sp<Track> > tracksToRemove;
1863    uint32_t mixerStatus = MIXER_IDLE;
1864    nsecs_t standbyTime = systemTime();
1865    size_t mixBufferSize = mFrameCount * mFrameSize;
1866    // FIXME: Relaxed timing because of a certain device that can't meet latency
1867    // Should be reduced to 2x after the vendor fixes the driver issue
1868    // increase threshold again due to low power audio mode. The way this warning threshold is
1869    // calculated and its usefulness should be reconsidered anyway.
1870    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1871    nsecs_t lastWarning = 0;
1872    bool longStandbyExit = false;
1873    uint32_t activeSleepTime = activeSleepTimeUs();
1874    uint32_t idleSleepTime = idleSleepTimeUs();
1875    uint32_t sleepTime = idleSleepTime;
1876    uint32_t sleepTimeShift = 0;
1877    Vector< sp<EffectChain> > effectChains;
1878#ifdef DEBUG_CPU_USAGE
1879    ThreadCpuUsage cpu;
1880    const CentralTendencyStatistics& stats = cpu.statistics();
1881#endif
1882
1883    acquireWakeLock();
1884
1885    while (!exitPending())
1886    {
1887#ifdef DEBUG_CPU_USAGE
1888        cpu.sampleAndEnable();
1889        unsigned n = stats.n();
1890        // cpu.elapsed() is expensive, so don't call it every loop
1891        if ((n & 127) == 1) {
1892            long long elapsed = cpu.elapsed();
1893            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1894                double perLoop = elapsed / (double) n;
1895                double perLoop100 = perLoop * 0.01;
1896                double mean = stats.mean();
1897                double stddev = stats.stddev();
1898                double minimum = stats.minimum();
1899                double maximum = stats.maximum();
1900                cpu.resetStatistics();
1901                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",
1902                        elapsed * .000000001, n, perLoop * .000001,
1903                        mean * .001,
1904                        stddev * .001,
1905                        minimum * .001,
1906                        maximum * .001,
1907                        mean / perLoop100,
1908                        stddev / perLoop100,
1909                        minimum / perLoop100,
1910                        maximum / perLoop100);
1911            }
1912        }
1913#endif
1914        processConfigEvents();
1915
1916        mixerStatus = MIXER_IDLE;
1917        { // scope for mLock
1918
1919            Mutex::Autolock _l(mLock);
1920
1921            if (checkForNewParameters_l()) {
1922                mixBufferSize = mFrameCount * mFrameSize;
1923                // FIXME: Relaxed timing because of a certain device that can't meet latency
1924                // Should be reduced to 2x after the vendor fixes the driver issue
1925                // increase threshold again due to low power audio mode. The way this warning
1926                // threshold is calculated and its usefulness should be reconsidered anyway.
1927                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1928                activeSleepTime = activeSleepTimeUs();
1929                idleSleepTime = idleSleepTimeUs();
1930            }
1931
1932            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1933
1934            // put audio hardware into standby after short delay
1935            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1936                        mSuspended)) {
1937                if (!mStandby) {
1938                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1939                    mOutput->stream->common.standby(&mOutput->stream->common);
1940                    mStandby = true;
1941                    mBytesWritten = 0;
1942                }
1943
1944                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1945                    // we're about to wait, flush the binder command buffer
1946                    IPCThreadState::self()->flushCommands();
1947
1948                    if (exitPending()) break;
1949
1950                    releaseWakeLock_l();
1951                    // wait until we have something to do...
1952                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1953                    mWaitWorkCV.wait(mLock);
1954                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1955                    acquireWakeLock_l();
1956
1957                    if (mMasterMute == false) {
1958                        char value[PROPERTY_VALUE_MAX];
1959                        property_get("ro.audio.silent", value, "0");
1960                        if (atoi(value)) {
1961                            ALOGD("Silence is golden");
1962                            setMasterMute(true);
1963                        }
1964                    }
1965
1966                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1967                    sleepTime = idleSleepTime;
1968                    sleepTimeShift = 0;
1969                    continue;
1970                }
1971            }
1972
1973            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1974
1975            // prevent any changes in effect chain list and in each effect chain
1976            // during mixing and effect process as the audio buffers could be deleted
1977            // or modified if an effect is created or deleted
1978            lockEffectChains_l(effectChains);
1979        }
1980
1981        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1982            // mix buffers...
1983            mAudioMixer->process();
1984            sleepTime = 0;
1985            // increase sleep time progressively when application underrun condition clears
1986            if (sleepTimeShift > 0) {
1987                sleepTimeShift--;
1988            }
1989            standbyTime = systemTime() + kStandbyTimeInNsecs;
1990            //TODO: delay standby when effects have a tail
1991        } else {
1992            // If no tracks are ready, sleep once for the duration of an output
1993            // buffer size, then write 0s to the output
1994            if (sleepTime == 0) {
1995                if (mixerStatus == MIXER_TRACKS_ENABLED) {
1996                    sleepTime = activeSleepTime >> sleepTimeShift;
1997                    if (sleepTime < kMinThreadSleepTimeUs) {
1998                        sleepTime = kMinThreadSleepTimeUs;
1999                    }
2000                    // reduce sleep time in case of consecutive application underruns to avoid
2001                    // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
2002                    // duration we would end up writing less data than needed by the audio HAL if
2003                    // the condition persists.
2004                    if (sleepTimeShift < kMaxThreadSleepTimeShift) {
2005                        sleepTimeShift++;
2006                    }
2007                } else {
2008                    sleepTime = idleSleepTime;
2009                }
2010            } else if (mBytesWritten != 0 ||
2011                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
2012                memset (mMixBuffer, 0, mixBufferSize);
2013                sleepTime = 0;
2014                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
2015            }
2016            // TODO add standby time extension fct of effect tail
2017        }
2018
2019        if (mSuspended) {
2020            sleepTime = suspendSleepTimeUs();
2021        }
2022        // sleepTime == 0 means we must write to audio hardware
2023        if (sleepTime == 0) {
2024            for (size_t i = 0; i < effectChains.size(); i ++) {
2025                effectChains[i]->process_l();
2026            }
2027            // enable changes in effect chain
2028            unlockEffectChains(effectChains);
2029            mLastWriteTime = systemTime();
2030            mInWrite = true;
2031            mBytesWritten += mixBufferSize;
2032
2033            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2034            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2035            mNumWrites++;
2036            mInWrite = false;
2037            nsecs_t now = systemTime();
2038            nsecs_t delta = now - mLastWriteTime;
2039            if (!mStandby && delta > maxPeriod) {
2040                mNumDelayedWrites++;
2041                if ((now - lastWarning) > kWarningThrottleNs) {
2042                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2043                            ns2ms(delta), mNumDelayedWrites, this);
2044                    lastWarning = now;
2045                }
2046                if (mStandby) {
2047                    longStandbyExit = true;
2048                }
2049            }
2050            mStandby = false;
2051        } else {
2052            // enable changes in effect chain
2053            unlockEffectChains(effectChains);
2054            usleep(sleepTime);
2055        }
2056
2057        // finally let go of all our tracks, without the lock held
2058        // since we can't guarantee the destructors won't acquire that
2059        // same lock.
2060        tracksToRemove.clear();
2061
2062        // Effect chains will be actually deleted here if they were removed from
2063        // mEffectChains list during mixing or effects processing
2064        effectChains.clear();
2065    }
2066
2067    if (!mStandby) {
2068        mOutput->stream->common.standby(&mOutput->stream->common);
2069    }
2070
2071    releaseWakeLock();
2072
2073    ALOGV("MixerThread %p exiting", this);
2074    return false;
2075}
2076
2077// prepareTracks_l() must be called with ThreadBase::mLock held
2078uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2079{
2080
2081    uint32_t mixerStatus = MIXER_IDLE;
2082    // find out which tracks need to be processed
2083    size_t count = activeTracks.size();
2084    size_t mixedTracks = 0;
2085    size_t tracksWithEffect = 0;
2086
2087    float masterVolume = mMasterVolume;
2088    bool  masterMute = mMasterMute;
2089
2090    if (masterMute) {
2091        masterVolume = 0;
2092    }
2093    // Delegate master volume control to effect in output mix effect chain if needed
2094    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2095    if (chain != 0) {
2096        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2097        chain->setVolume_l(&v, &v);
2098        masterVolume = (float)((v + (1 << 23)) >> 24);
2099        chain.clear();
2100    }
2101
2102    for (size_t i=0 ; i<count ; i++) {
2103        sp<Track> t = activeTracks[i].promote();
2104        if (t == 0) continue;
2105
2106        // this const just means the local variable doesn't change
2107        Track* const track = t.get();
2108        audio_track_cblk_t* cblk = track->cblk();
2109
2110        // The first time a track is added we wait
2111        // for all its buffers to be filled before processing it
2112        int name = track->name();
2113        // make sure that we have enough frames to mix one full buffer.
2114        // enforce this condition only once to enable draining the buffer in case the client
2115        // app does not call stop() and relies on underrun to stop:
2116        // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed
2117        // during last round
2118        uint32_t minFrames = 1;
2119        if (!track->isStopped() && !track->isPausing() &&
2120                (track->mRetryCount >= kMaxTrackRetries)) {
2121            if (t->sampleRate() == (int)mSampleRate) {
2122                minFrames = mFrameCount;
2123            } else {
2124                // +1 for rounding and +1 for additional sample needed for interpolation
2125                minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
2126                // add frames already consumed but not yet released by the resampler
2127                // because cblk->framesReady() will  include these frames
2128                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
2129                // the minimum track buffer size is normally twice the number of frames necessary
2130                // to fill one buffer and the resampler should not leave more than one buffer worth
2131                // of unreleased frames after each pass, but just in case...
2132                ALOG_ASSERT(minFrames <= cblk->frameCount);
2133            }
2134        }
2135        if ((cblk->framesReady() >= minFrames) && track->isReady() &&
2136                !track->isPaused() && !track->isTerminated())
2137        {
2138            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
2139
2140            mixedTracks++;
2141
2142            // track->mainBuffer() != mMixBuffer means there is an effect chain
2143            // connected to the track
2144            chain.clear();
2145            if (track->mainBuffer() != mMixBuffer) {
2146                chain = getEffectChain_l(track->sessionId());
2147                // Delegate volume control to effect in track effect chain if needed
2148                if (chain != 0) {
2149                    tracksWithEffect++;
2150                } else {
2151                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
2152                            name, track->sessionId());
2153                }
2154            }
2155
2156
2157            int param = AudioMixer::VOLUME;
2158            if (track->mFillingUpStatus == Track::FS_FILLED) {
2159                // no ramp for the first volume setting
2160                track->mFillingUpStatus = Track::FS_ACTIVE;
2161                if (track->mState == TrackBase::RESUMING) {
2162                    track->mState = TrackBase::ACTIVE;
2163                    param = AudioMixer::RAMP_VOLUME;
2164                }
2165                mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2166            } else if (cblk->server != 0) {
2167                // If the track is stopped before the first frame was mixed,
2168                // do not apply ramp
2169                param = AudioMixer::RAMP_VOLUME;
2170            }
2171
2172            // compute volume for this track
2173            uint32_t vl, vr, va;
2174            if (track->isMuted() || track->isPausing() ||
2175                mStreamTypes[track->type()].mute) {
2176                vl = vr = va = 0;
2177                if (track->isPausing()) {
2178                    track->setPaused();
2179                }
2180            } else {
2181
2182                // read original volumes with volume control
2183                float typeVolume = mStreamTypes[track->type()].volume;
2184                float v = masterVolume * typeVolume;
2185                vl = (uint32_t)(v * cblk->volume[0]) << 12;
2186                vr = (uint32_t)(v * cblk->volume[1]) << 12;
2187
2188                va = (uint32_t)(v * cblk->sendLevel);
2189            }
2190            // Delegate volume control to effect in track effect chain if needed
2191            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2192                // Do not ramp volume if volume is controlled by effect
2193                param = AudioMixer::VOLUME;
2194                track->mHasVolumeController = true;
2195            } else {
2196                // force no volume ramp when volume controller was just disabled or removed
2197                // from effect chain to avoid volume spike
2198                if (track->mHasVolumeController) {
2199                    param = AudioMixer::VOLUME;
2200                }
2201                track->mHasVolumeController = false;
2202            }
2203
2204            // Convert volumes from 8.24 to 4.12 format
2205            int16_t left, right, aux;
2206            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2207            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2208            left = int16_t(v_clamped);
2209            v_clamped = (vr + (1 << 11)) >> 12;
2210            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2211            right = int16_t(v_clamped);
2212
2213            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2214            aux = int16_t(va);
2215
2216            // XXX: these things DON'T need to be done each time
2217            mAudioMixer->setBufferProvider(name, track);
2218            mAudioMixer->enable(name);
2219
2220            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left);
2221            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right);
2222            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux);
2223            mAudioMixer->setParameter(
2224                name,
2225                AudioMixer::TRACK,
2226                AudioMixer::FORMAT, (void *)track->format());
2227            mAudioMixer->setParameter(
2228                name,
2229                AudioMixer::TRACK,
2230                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2231            mAudioMixer->setParameter(
2232                name,
2233                AudioMixer::RESAMPLE,
2234                AudioMixer::SAMPLE_RATE,
2235                (void *)(cblk->sampleRate));
2236            mAudioMixer->setParameter(
2237                name,
2238                AudioMixer::TRACK,
2239                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2240            mAudioMixer->setParameter(
2241                name,
2242                AudioMixer::TRACK,
2243                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2244
2245            // reset retry count
2246            track->mRetryCount = kMaxTrackRetries;
2247            mixerStatus = MIXER_TRACKS_READY;
2248        } else {
2249            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
2250            if (track->isStopped()) {
2251                track->reset();
2252            }
2253            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2254                // We have consumed all the buffers of this track.
2255                // Remove it from the list of active tracks.
2256                tracksToRemove->add(track);
2257            } else {
2258                // No buffers for this track. Give it a few chances to
2259                // fill a buffer, then remove it from active list.
2260                if (--(track->mRetryCount) <= 0) {
2261                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
2262                    tracksToRemove->add(track);
2263                    // indicate to client process that the track was disabled because of underrun
2264                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2265                } else if (mixerStatus != MIXER_TRACKS_READY) {
2266                    mixerStatus = MIXER_TRACKS_ENABLED;
2267                }
2268            }
2269            mAudioMixer->disable(name);
2270        }
2271    }
2272
2273    // remove all the tracks that need to be...
2274    count = tracksToRemove->size();
2275    if (CC_UNLIKELY(count)) {
2276        for (size_t i=0 ; i<count ; i++) {
2277            const sp<Track>& track = tracksToRemove->itemAt(i);
2278            mActiveTracks.remove(track);
2279            if (track->mainBuffer() != mMixBuffer) {
2280                chain = getEffectChain_l(track->sessionId());
2281                if (chain != 0) {
2282                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2283                    chain->decActiveTrackCnt();
2284                }
2285            }
2286            if (track->isTerminated()) {
2287                removeTrack_l(track);
2288            }
2289        }
2290    }
2291
2292    // mix buffer must be cleared if all tracks are connected to an
2293    // effect chain as in this case the mixer will not write to
2294    // mix buffer and track effects will accumulate into it
2295    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2296        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2297    }
2298
2299    return mixerStatus;
2300}
2301
2302void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType)
2303{
2304    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2305            this,  streamType, mTracks.size());
2306    Mutex::Autolock _l(mLock);
2307
2308    size_t size = mTracks.size();
2309    for (size_t i = 0; i < size; i++) {
2310        sp<Track> t = mTracks[i];
2311        if (t->type() == streamType) {
2312            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2313            t->mCblk->cv.signal();
2314        }
2315    }
2316}
2317
2318void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid)
2319{
2320    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2321            this,  streamType, valid);
2322    Mutex::Autolock _l(mLock);
2323
2324    mStreamTypes[streamType].valid = valid;
2325}
2326
2327// getTrackName_l() must be called with ThreadBase::mLock held
2328int AudioFlinger::MixerThread::getTrackName_l()
2329{
2330    return mAudioMixer->getTrackName();
2331}
2332
2333// deleteTrackName_l() must be called with ThreadBase::mLock held
2334void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2335{
2336    ALOGV("remove track (%d) and delete from mixer", name);
2337    mAudioMixer->deleteTrackName(name);
2338}
2339
2340// checkForNewParameters_l() must be called with ThreadBase::mLock held
2341bool AudioFlinger::MixerThread::checkForNewParameters_l()
2342{
2343    bool reconfig = false;
2344
2345    while (!mNewParameters.isEmpty()) {
2346        status_t status = NO_ERROR;
2347        String8 keyValuePair = mNewParameters[0];
2348        AudioParameter param = AudioParameter(keyValuePair);
2349        int value;
2350
2351        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2352            reconfig = true;
2353        }
2354        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2355            if (value != AUDIO_FORMAT_PCM_16_BIT) {
2356                status = BAD_VALUE;
2357            } else {
2358                reconfig = true;
2359            }
2360        }
2361        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2362            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2363                status = BAD_VALUE;
2364            } else {
2365                reconfig = true;
2366            }
2367        }
2368        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2369            // do not accept frame count changes if tracks are open as the track buffer
2370            // size depends on frame count and correct behavior would not be guaranteed
2371            // if frame count is changed after track creation
2372            if (!mTracks.isEmpty()) {
2373                status = INVALID_OPERATION;
2374            } else {
2375                reconfig = true;
2376            }
2377        }
2378        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2379            // when changing the audio output device, call addBatteryData to notify
2380            // the change
2381            if ((int)mDevice != value) {
2382                uint32_t params = 0;
2383                // check whether speaker is on
2384                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2385                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2386                }
2387
2388                int deviceWithoutSpeaker
2389                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2390                // check if any other device (except speaker) is on
2391                if (value & deviceWithoutSpeaker ) {
2392                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2393                }
2394
2395                if (params != 0) {
2396                    addBatteryData(params);
2397                }
2398            }
2399
2400            // forward device change to effects that have requested to be
2401            // aware of attached audio device.
2402            mDevice = (uint32_t)value;
2403            for (size_t i = 0; i < mEffectChains.size(); i++) {
2404                mEffectChains[i]->setDevice_l(mDevice);
2405            }
2406        }
2407
2408        if (status == NO_ERROR) {
2409            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2410                                                    keyValuePair.string());
2411            if (!mStandby && status == INVALID_OPERATION) {
2412               mOutput->stream->common.standby(&mOutput->stream->common);
2413               mStandby = true;
2414               mBytesWritten = 0;
2415               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2416                                                       keyValuePair.string());
2417            }
2418            if (status == NO_ERROR && reconfig) {
2419                delete mAudioMixer;
2420                readOutputParameters();
2421                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2422                for (size_t i = 0; i < mTracks.size() ; i++) {
2423                    int name = getTrackName_l();
2424                    if (name < 0) break;
2425                    mTracks[i]->mName = name;
2426                    // limit track sample rate to 2 x new output sample rate
2427                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2428                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2429                    }
2430                }
2431                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2432            }
2433        }
2434
2435        mNewParameters.removeAt(0);
2436
2437        mParamStatus = status;
2438        mParamCond.signal();
2439        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2440        // already timed out waiting for the status and will never signal the condition.
2441        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2442    }
2443    return reconfig;
2444}
2445
2446status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2447{
2448    const size_t SIZE = 256;
2449    char buffer[SIZE];
2450    String8 result;
2451
2452    PlaybackThread::dumpInternals(fd, args);
2453
2454    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2455    result.append(buffer);
2456    write(fd, result.string(), result.size());
2457    return NO_ERROR;
2458}
2459
2460uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2461{
2462    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2463}
2464
2465uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2466{
2467    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2468}
2469
2470// ----------------------------------------------------------------------------
2471AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2472    :   PlaybackThread(audioFlinger, output, id, device)
2473{
2474    mType = ThreadBase::DIRECT;
2475}
2476
2477AudioFlinger::DirectOutputThread::~DirectOutputThread()
2478{
2479}
2480
2481static inline
2482int32_t mul(int16_t in, int16_t v)
2483{
2484#if defined(__arm__) && !defined(__thumb__)
2485    int32_t out;
2486    asm( "smulbb %[out], %[in], %[v] \n"
2487         : [out]"=r"(out)
2488         : [in]"%r"(in), [v]"r"(v)
2489         : );
2490    return out;
2491#else
2492    return in * int32_t(v);
2493#endif
2494}
2495
2496void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2497{
2498    // Do not apply volume on compressed audio
2499    if (!audio_is_linear_pcm(mFormat)) {
2500        return;
2501    }
2502
2503    // convert to signed 16 bit before volume calculation
2504    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2505        size_t count = mFrameCount * mChannelCount;
2506        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2507        int16_t *dst = mMixBuffer + count-1;
2508        while(count--) {
2509            *dst-- = (int16_t)(*src--^0x80) << 8;
2510        }
2511    }
2512
2513    size_t frameCount = mFrameCount;
2514    int16_t *out = mMixBuffer;
2515    if (ramp) {
2516        if (mChannelCount == 1) {
2517            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2518            int32_t vlInc = d / (int32_t)frameCount;
2519            int32_t vl = ((int32_t)mLeftVolShort << 16);
2520            do {
2521                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2522                out++;
2523                vl += vlInc;
2524            } while (--frameCount);
2525
2526        } else {
2527            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2528            int32_t vlInc = d / (int32_t)frameCount;
2529            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2530            int32_t vrInc = d / (int32_t)frameCount;
2531            int32_t vl = ((int32_t)mLeftVolShort << 16);
2532            int32_t vr = ((int32_t)mRightVolShort << 16);
2533            do {
2534                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2535                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2536                out += 2;
2537                vl += vlInc;
2538                vr += vrInc;
2539            } while (--frameCount);
2540        }
2541    } else {
2542        if (mChannelCount == 1) {
2543            do {
2544                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2545                out++;
2546            } while (--frameCount);
2547        } else {
2548            do {
2549                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2550                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2551                out += 2;
2552            } while (--frameCount);
2553        }
2554    }
2555
2556    // convert back to unsigned 8 bit after volume calculation
2557    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2558        size_t count = mFrameCount * mChannelCount;
2559        int16_t *src = mMixBuffer;
2560        uint8_t *dst = (uint8_t *)mMixBuffer;
2561        while(count--) {
2562            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2563        }
2564    }
2565
2566    mLeftVolShort = leftVol;
2567    mRightVolShort = rightVol;
2568}
2569
2570bool AudioFlinger::DirectOutputThread::threadLoop()
2571{
2572    uint32_t mixerStatus = MIXER_IDLE;
2573    sp<Track> trackToRemove;
2574    sp<Track> activeTrack;
2575    nsecs_t standbyTime = systemTime();
2576    int8_t *curBuf;
2577    size_t mixBufferSize = mFrameCount*mFrameSize;
2578    uint32_t activeSleepTime = activeSleepTimeUs();
2579    uint32_t idleSleepTime = idleSleepTimeUs();
2580    uint32_t sleepTime = idleSleepTime;
2581    // use shorter standby delay as on normal output to release
2582    // hardware resources as soon as possible
2583    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2584
2585    acquireWakeLock();
2586
2587    while (!exitPending())
2588    {
2589        bool rampVolume;
2590        uint16_t leftVol;
2591        uint16_t rightVol;
2592        Vector< sp<EffectChain> > effectChains;
2593
2594        processConfigEvents();
2595
2596        mixerStatus = MIXER_IDLE;
2597
2598        { // scope for the mLock
2599
2600            Mutex::Autolock _l(mLock);
2601
2602            if (checkForNewParameters_l()) {
2603                mixBufferSize = mFrameCount*mFrameSize;
2604                activeSleepTime = activeSleepTimeUs();
2605                idleSleepTime = idleSleepTimeUs();
2606                standbyDelay = microseconds(activeSleepTime*2);
2607            }
2608
2609            // put audio hardware into standby after short delay
2610            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2611                        mSuspended)) {
2612                // wait until we have something to do...
2613                if (!mStandby) {
2614                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2615                    mOutput->stream->common.standby(&mOutput->stream->common);
2616                    mStandby = true;
2617                    mBytesWritten = 0;
2618                }
2619
2620                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2621                    // we're about to wait, flush the binder command buffer
2622                    IPCThreadState::self()->flushCommands();
2623
2624                    if (exitPending()) break;
2625
2626                    releaseWakeLock_l();
2627                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2628                    mWaitWorkCV.wait(mLock);
2629                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2630                    acquireWakeLock_l();
2631
2632                    if (mMasterMute == false) {
2633                        char value[PROPERTY_VALUE_MAX];
2634                        property_get("ro.audio.silent", value, "0");
2635                        if (atoi(value)) {
2636                            ALOGD("Silence is golden");
2637                            setMasterMute(true);
2638                        }
2639                    }
2640
2641                    standbyTime = systemTime() + standbyDelay;
2642                    sleepTime = idleSleepTime;
2643                    continue;
2644                }
2645            }
2646
2647            effectChains = mEffectChains;
2648
2649            // find out which tracks need to be processed
2650            if (mActiveTracks.size() != 0) {
2651                sp<Track> t = mActiveTracks[0].promote();
2652                if (t == 0) continue;
2653
2654                Track* const track = t.get();
2655                audio_track_cblk_t* cblk = track->cblk();
2656
2657                // The first time a track is added we wait
2658                // for all its buffers to be filled before processing it
2659                if (cblk->framesReady() && track->isReady() &&
2660                        !track->isPaused() && !track->isTerminated())
2661                {
2662                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2663
2664                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2665                        track->mFillingUpStatus = Track::FS_ACTIVE;
2666                        mLeftVolFloat = mRightVolFloat = 0;
2667                        mLeftVolShort = mRightVolShort = 0;
2668                        if (track->mState == TrackBase::RESUMING) {
2669                            track->mState = TrackBase::ACTIVE;
2670                            rampVolume = true;
2671                        }
2672                    } else if (cblk->server != 0) {
2673                        // If the track is stopped before the first frame was mixed,
2674                        // do not apply ramp
2675                        rampVolume = true;
2676                    }
2677                    // compute volume for this track
2678                    float left, right;
2679                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2680                        mStreamTypes[track->type()].mute) {
2681                        left = right = 0;
2682                        if (track->isPausing()) {
2683                            track->setPaused();
2684                        }
2685                    } else {
2686                        float typeVolume = mStreamTypes[track->type()].volume;
2687                        float v = mMasterVolume * typeVolume;
2688                        float v_clamped = v * cblk->volume[0];
2689                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2690                        left = v_clamped/MAX_GAIN;
2691                        v_clamped = v * cblk->volume[1];
2692                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2693                        right = v_clamped/MAX_GAIN;
2694                    }
2695
2696                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2697                        mLeftVolFloat = left;
2698                        mRightVolFloat = right;
2699
2700                        // If audio HAL implements volume control,
2701                        // force software volume to nominal value
2702                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2703                            left = 1.0f;
2704                            right = 1.0f;
2705                        }
2706
2707                        // Convert volumes from float to 8.24
2708                        uint32_t vl = (uint32_t)(left * (1 << 24));
2709                        uint32_t vr = (uint32_t)(right * (1 << 24));
2710
2711                        // Delegate volume control to effect in track effect chain if needed
2712                        // only one effect chain can be present on DirectOutputThread, so if
2713                        // there is one, the track is connected to it
2714                        if (!effectChains.isEmpty()) {
2715                            // Do not ramp volume if volume is controlled by effect
2716                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2717                                rampVolume = false;
2718                            }
2719                        }
2720
2721                        // Convert volumes from 8.24 to 4.12 format
2722                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2723                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2724                        leftVol = (uint16_t)v_clamped;
2725                        v_clamped = (vr + (1 << 11)) >> 12;
2726                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2727                        rightVol = (uint16_t)v_clamped;
2728                    } else {
2729                        leftVol = mLeftVolShort;
2730                        rightVol = mRightVolShort;
2731                        rampVolume = false;
2732                    }
2733
2734                    // reset retry count
2735                    track->mRetryCount = kMaxTrackRetriesDirect;
2736                    activeTrack = t;
2737                    mixerStatus = MIXER_TRACKS_READY;
2738                } else {
2739                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2740                    if (track->isStopped()) {
2741                        track->reset();
2742                    }
2743                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2744                        // We have consumed all the buffers of this track.
2745                        // Remove it from the list of active tracks.
2746                        trackToRemove = track;
2747                    } else {
2748                        // No buffers for this track. Give it a few chances to
2749                        // fill a buffer, then remove it from active list.
2750                        if (--(track->mRetryCount) <= 0) {
2751                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2752                            trackToRemove = track;
2753                        } else {
2754                            mixerStatus = MIXER_TRACKS_ENABLED;
2755                        }
2756                    }
2757                }
2758            }
2759
2760            // remove all the tracks that need to be...
2761            if (CC_UNLIKELY(trackToRemove != 0)) {
2762                mActiveTracks.remove(trackToRemove);
2763                if (!effectChains.isEmpty()) {
2764                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2765                            trackToRemove->sessionId());
2766                    effectChains[0]->decActiveTrackCnt();
2767                }
2768                if (trackToRemove->isTerminated()) {
2769                    removeTrack_l(trackToRemove);
2770                }
2771            }
2772
2773            lockEffectChains_l(effectChains);
2774       }
2775
2776        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2777            AudioBufferProvider::Buffer buffer;
2778            size_t frameCount = mFrameCount;
2779            curBuf = (int8_t *)mMixBuffer;
2780            // output audio to hardware
2781            while (frameCount) {
2782                buffer.frameCount = frameCount;
2783                activeTrack->getNextBuffer(&buffer);
2784                if (CC_UNLIKELY(buffer.raw == NULL)) {
2785                    memset(curBuf, 0, frameCount * mFrameSize);
2786                    break;
2787                }
2788                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2789                frameCount -= buffer.frameCount;
2790                curBuf += buffer.frameCount * mFrameSize;
2791                activeTrack->releaseBuffer(&buffer);
2792            }
2793            sleepTime = 0;
2794            standbyTime = systemTime() + standbyDelay;
2795        } else {
2796            if (sleepTime == 0) {
2797                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2798                    sleepTime = activeSleepTime;
2799                } else {
2800                    sleepTime = idleSleepTime;
2801                }
2802            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2803                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2804                sleepTime = 0;
2805            }
2806        }
2807
2808        if (mSuspended) {
2809            sleepTime = suspendSleepTimeUs();
2810        }
2811        // sleepTime == 0 means we must write to audio hardware
2812        if (sleepTime == 0) {
2813            if (mixerStatus == MIXER_TRACKS_READY) {
2814                applyVolume(leftVol, rightVol, rampVolume);
2815            }
2816            for (size_t i = 0; i < effectChains.size(); i ++) {
2817                effectChains[i]->process_l();
2818            }
2819            unlockEffectChains(effectChains);
2820
2821            mLastWriteTime = systemTime();
2822            mInWrite = true;
2823            mBytesWritten += mixBufferSize;
2824            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2825            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2826            mNumWrites++;
2827            mInWrite = false;
2828            mStandby = false;
2829        } else {
2830            unlockEffectChains(effectChains);
2831            usleep(sleepTime);
2832        }
2833
2834        // finally let go of removed track, without the lock held
2835        // since we can't guarantee the destructors won't acquire that
2836        // same lock.
2837        trackToRemove.clear();
2838        activeTrack.clear();
2839
2840        // Effect chains will be actually deleted here if they were removed from
2841        // mEffectChains list during mixing or effects processing
2842        effectChains.clear();
2843    }
2844
2845    if (!mStandby) {
2846        mOutput->stream->common.standby(&mOutput->stream->common);
2847    }
2848
2849    releaseWakeLock();
2850
2851    ALOGV("DirectOutputThread %p exiting", this);
2852    return false;
2853}
2854
2855// getTrackName_l() must be called with ThreadBase::mLock held
2856int AudioFlinger::DirectOutputThread::getTrackName_l()
2857{
2858    return 0;
2859}
2860
2861// deleteTrackName_l() must be called with ThreadBase::mLock held
2862void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2863{
2864}
2865
2866// checkForNewParameters_l() must be called with ThreadBase::mLock held
2867bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2868{
2869    bool reconfig = false;
2870
2871    while (!mNewParameters.isEmpty()) {
2872        status_t status = NO_ERROR;
2873        String8 keyValuePair = mNewParameters[0];
2874        AudioParameter param = AudioParameter(keyValuePair);
2875        int value;
2876
2877        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2878            // do not accept frame count changes if tracks are open as the track buffer
2879            // size depends on frame count and correct behavior would not be garantied
2880            // if frame count is changed after track creation
2881            if (!mTracks.isEmpty()) {
2882                status = INVALID_OPERATION;
2883            } else {
2884                reconfig = true;
2885            }
2886        }
2887        if (status == NO_ERROR) {
2888            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2889                                                    keyValuePair.string());
2890            if (!mStandby && status == INVALID_OPERATION) {
2891               mOutput->stream->common.standby(&mOutput->stream->common);
2892               mStandby = true;
2893               mBytesWritten = 0;
2894               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2895                                                       keyValuePair.string());
2896            }
2897            if (status == NO_ERROR && reconfig) {
2898                readOutputParameters();
2899                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2900            }
2901        }
2902
2903        mNewParameters.removeAt(0);
2904
2905        mParamStatus = status;
2906        mParamCond.signal();
2907        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2908        // already timed out waiting for the status and will never signal the condition.
2909        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2910    }
2911    return reconfig;
2912}
2913
2914uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2915{
2916    uint32_t time;
2917    if (audio_is_linear_pcm(mFormat)) {
2918        time = PlaybackThread::activeSleepTimeUs();
2919    } else {
2920        time = 10000;
2921    }
2922    return time;
2923}
2924
2925uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2926{
2927    uint32_t time;
2928    if (audio_is_linear_pcm(mFormat)) {
2929        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2930    } else {
2931        time = 10000;
2932    }
2933    return time;
2934}
2935
2936uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2937{
2938    uint32_t time;
2939    if (audio_is_linear_pcm(mFormat)) {
2940        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2941    } else {
2942        time = 10000;
2943    }
2944    return time;
2945}
2946
2947
2948// ----------------------------------------------------------------------------
2949
2950AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2951    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2952{
2953    mType = ThreadBase::DUPLICATING;
2954    addOutputTrack(mainThread);
2955}
2956
2957AudioFlinger::DuplicatingThread::~DuplicatingThread()
2958{
2959    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2960        mOutputTracks[i]->destroy();
2961    }
2962    mOutputTracks.clear();
2963}
2964
2965bool AudioFlinger::DuplicatingThread::threadLoop()
2966{
2967    Vector< sp<Track> > tracksToRemove;
2968    uint32_t mixerStatus = MIXER_IDLE;
2969    nsecs_t standbyTime = systemTime();
2970    size_t mixBufferSize = mFrameCount*mFrameSize;
2971    SortedVector< sp<OutputTrack> > outputTracks;
2972    uint32_t writeFrames = 0;
2973    uint32_t activeSleepTime = activeSleepTimeUs();
2974    uint32_t idleSleepTime = idleSleepTimeUs();
2975    uint32_t sleepTime = idleSleepTime;
2976    Vector< sp<EffectChain> > effectChains;
2977
2978    acquireWakeLock();
2979
2980    while (!exitPending())
2981    {
2982        processConfigEvents();
2983
2984        mixerStatus = MIXER_IDLE;
2985        { // scope for the mLock
2986
2987            Mutex::Autolock _l(mLock);
2988
2989            if (checkForNewParameters_l()) {
2990                mixBufferSize = mFrameCount*mFrameSize;
2991                updateWaitTime();
2992                activeSleepTime = activeSleepTimeUs();
2993                idleSleepTime = idleSleepTimeUs();
2994            }
2995
2996            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
2997
2998            for (size_t i = 0; i < mOutputTracks.size(); i++) {
2999                outputTracks.add(mOutputTracks[i]);
3000            }
3001
3002            // put audio hardware into standby after short delay
3003            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
3004                         mSuspended)) {
3005                if (!mStandby) {
3006                    for (size_t i = 0; i < outputTracks.size(); i++) {
3007                        outputTracks[i]->stop();
3008                    }
3009                    mStandby = true;
3010                    mBytesWritten = 0;
3011                }
3012
3013                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
3014                    // we're about to wait, flush the binder command buffer
3015                    IPCThreadState::self()->flushCommands();
3016                    outputTracks.clear();
3017
3018                    if (exitPending()) break;
3019
3020                    releaseWakeLock_l();
3021                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3022                    mWaitWorkCV.wait(mLock);
3023                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3024                    acquireWakeLock_l();
3025
3026                    if (mMasterMute == false) {
3027                        char value[PROPERTY_VALUE_MAX];
3028                        property_get("ro.audio.silent", value, "0");
3029                        if (atoi(value)) {
3030                            ALOGD("Silence is golden");
3031                            setMasterMute(true);
3032                        }
3033                    }
3034
3035                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3036                    sleepTime = idleSleepTime;
3037                    continue;
3038                }
3039            }
3040
3041            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3042
3043            // prevent any changes in effect chain list and in each effect chain
3044            // during mixing and effect process as the audio buffers could be deleted
3045            // or modified if an effect is created or deleted
3046            lockEffectChains_l(effectChains);
3047        }
3048
3049        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3050            // mix buffers...
3051            if (outputsReady(outputTracks)) {
3052                mAudioMixer->process();
3053            } else {
3054                memset(mMixBuffer, 0, mixBufferSize);
3055            }
3056            sleepTime = 0;
3057            writeFrames = mFrameCount;
3058        } else {
3059            if (sleepTime == 0) {
3060                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3061                    sleepTime = activeSleepTime;
3062                } else {
3063                    sleepTime = idleSleepTime;
3064                }
3065            } else if (mBytesWritten != 0) {
3066                // flush remaining overflow buffers in output tracks
3067                for (size_t i = 0; i < outputTracks.size(); i++) {
3068                    if (outputTracks[i]->isActive()) {
3069                        sleepTime = 0;
3070                        writeFrames = 0;
3071                        memset(mMixBuffer, 0, mixBufferSize);
3072                        break;
3073                    }
3074                }
3075            }
3076        }
3077
3078        if (mSuspended) {
3079            sleepTime = suspendSleepTimeUs();
3080        }
3081        // sleepTime == 0 means we must write to audio hardware
3082        if (sleepTime == 0) {
3083            for (size_t i = 0; i < effectChains.size(); i ++) {
3084                effectChains[i]->process_l();
3085            }
3086            // enable changes in effect chain
3087            unlockEffectChains(effectChains);
3088
3089            standbyTime = systemTime() + kStandbyTimeInNsecs;
3090            for (size_t i = 0; i < outputTracks.size(); i++) {
3091                outputTracks[i]->write(mMixBuffer, writeFrames);
3092            }
3093            mStandby = false;
3094            mBytesWritten += mixBufferSize;
3095        } else {
3096            // enable changes in effect chain
3097            unlockEffectChains(effectChains);
3098            usleep(sleepTime);
3099        }
3100
3101        // finally let go of all our tracks, without the lock held
3102        // since we can't guarantee the destructors won't acquire that
3103        // same lock.
3104        tracksToRemove.clear();
3105        outputTracks.clear();
3106
3107        // Effect chains will be actually deleted here if they were removed from
3108        // mEffectChains list during mixing or effects processing
3109        effectChains.clear();
3110    }
3111
3112    releaseWakeLock();
3113
3114    return false;
3115}
3116
3117void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3118{
3119    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3120    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3121                                            this,
3122                                            mSampleRate,
3123                                            mFormat,
3124                                            mChannelMask,
3125                                            frameCount);
3126    if (outputTrack->cblk() != NULL) {
3127        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3128        mOutputTracks.add(outputTrack);
3129        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3130        updateWaitTime();
3131    }
3132}
3133
3134void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3135{
3136    Mutex::Autolock _l(mLock);
3137    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3138        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3139            mOutputTracks[i]->destroy();
3140            mOutputTracks.removeAt(i);
3141            updateWaitTime();
3142            return;
3143        }
3144    }
3145    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3146}
3147
3148void AudioFlinger::DuplicatingThread::updateWaitTime()
3149{
3150    mWaitTimeMs = UINT_MAX;
3151    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3152        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3153        if (strong != NULL) {
3154            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3155            if (waitTimeMs < mWaitTimeMs) {
3156                mWaitTimeMs = waitTimeMs;
3157            }
3158        }
3159    }
3160}
3161
3162
3163bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3164{
3165    for (size_t i = 0; i < outputTracks.size(); i++) {
3166        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3167        if (thread == 0) {
3168            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3169            return false;
3170        }
3171        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3172        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3173            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3174            return false;
3175        }
3176    }
3177    return true;
3178}
3179
3180uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3181{
3182    return (mWaitTimeMs * 1000) / 2;
3183}
3184
3185// ----------------------------------------------------------------------------
3186
3187// TrackBase constructor must be called with AudioFlinger::mLock held
3188AudioFlinger::ThreadBase::TrackBase::TrackBase(
3189            const wp<ThreadBase>& thread,
3190            const sp<Client>& client,
3191            uint32_t sampleRate,
3192            uint32_t format,
3193            uint32_t channelMask,
3194            int frameCount,
3195            uint32_t flags,
3196            const sp<IMemory>& sharedBuffer,
3197            int sessionId)
3198    :   RefBase(),
3199        mThread(thread),
3200        mClient(client),
3201        mCblk(0),
3202        mFrameCount(0),
3203        mState(IDLE),
3204        mClientTid(-1),
3205        mFormat(format),
3206        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3207        mSessionId(sessionId)
3208{
3209    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3210
3211    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3212   size_t size = sizeof(audio_track_cblk_t);
3213   uint8_t channelCount = popcount(channelMask);
3214   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3215   if (sharedBuffer == 0) {
3216       size += bufferSize;
3217   }
3218
3219   if (client != NULL) {
3220        mCblkMemory = client->heap()->allocate(size);
3221        if (mCblkMemory != 0) {
3222            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3223            if (mCblk) { // construct the shared structure in-place.
3224                new(mCblk) audio_track_cblk_t();
3225                // clear all buffers
3226                mCblk->frameCount = frameCount;
3227                mCblk->sampleRate = sampleRate;
3228                mChannelCount = channelCount;
3229                mChannelMask = channelMask;
3230                if (sharedBuffer == 0) {
3231                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3232                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3233                    // Force underrun condition to avoid false underrun callback until first data is
3234                    // written to buffer (other flags are cleared)
3235                    mCblk->flags = CBLK_UNDERRUN_ON;
3236                } else {
3237                    mBuffer = sharedBuffer->pointer();
3238                }
3239                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3240            }
3241        } else {
3242            ALOGE("not enough memory for AudioTrack size=%u", size);
3243            client->heap()->dump("AudioTrack");
3244            return;
3245        }
3246   } else {
3247       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3248           // construct the shared structure in-place.
3249           new(mCblk) audio_track_cblk_t();
3250           // clear all buffers
3251           mCblk->frameCount = frameCount;
3252           mCblk->sampleRate = sampleRate;
3253           mChannelCount = channelCount;
3254           mChannelMask = channelMask;
3255           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3256           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3257           // Force underrun condition to avoid false underrun callback until first data is
3258           // written to buffer (other flags are cleared)
3259           mCblk->flags = CBLK_UNDERRUN_ON;
3260           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3261   }
3262}
3263
3264AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3265{
3266    if (mCblk) {
3267        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3268        if (mClient == NULL) {
3269            delete mCblk;
3270        }
3271    }
3272    mCblkMemory.clear();            // and free the shared memory
3273    if (mClient != NULL) {
3274        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3275        mClient.clear();
3276    }
3277}
3278
3279void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3280{
3281    buffer->raw = NULL;
3282    mFrameCount = buffer->frameCount;
3283    step();
3284    buffer->frameCount = 0;
3285}
3286
3287bool AudioFlinger::ThreadBase::TrackBase::step() {
3288    bool result;
3289    audio_track_cblk_t* cblk = this->cblk();
3290
3291    result = cblk->stepServer(mFrameCount);
3292    if (!result) {
3293        ALOGV("stepServer failed acquiring cblk mutex");
3294        mFlags |= STEPSERVER_FAILED;
3295    }
3296    return result;
3297}
3298
3299void AudioFlinger::ThreadBase::TrackBase::reset() {
3300    audio_track_cblk_t* cblk = this->cblk();
3301
3302    cblk->user = 0;
3303    cblk->server = 0;
3304    cblk->userBase = 0;
3305    cblk->serverBase = 0;
3306    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3307    ALOGV("TrackBase::reset");
3308}
3309
3310sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3311{
3312    return mCblkMemory;
3313}
3314
3315int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3316    return (int)mCblk->sampleRate;
3317}
3318
3319int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3320    return (const int)mChannelCount;
3321}
3322
3323uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3324    return mChannelMask;
3325}
3326
3327void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3328    audio_track_cblk_t* cblk = this->cblk();
3329    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
3330    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
3331
3332    // Check validity of returned pointer in case the track control block would have been corrupted.
3333    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3334        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
3335        ALOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3336                server %d, serverBase %d, user %d, userBase %d",
3337                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3338                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3339        return 0;
3340    }
3341
3342    return bufferStart;
3343}
3344
3345// ----------------------------------------------------------------------------
3346
3347// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3348AudioFlinger::PlaybackThread::Track::Track(
3349            const wp<ThreadBase>& thread,
3350            const sp<Client>& client,
3351            audio_stream_type_t streamType,
3352            uint32_t sampleRate,
3353            uint32_t format,
3354            uint32_t channelMask,
3355            int frameCount,
3356            const sp<IMemory>& sharedBuffer,
3357            int sessionId)
3358    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3359    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3360    mAuxEffectId(0), mHasVolumeController(false)
3361{
3362    if (mCblk != NULL) {
3363        sp<ThreadBase> baseThread = thread.promote();
3364        if (baseThread != 0) {
3365            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3366            mName = playbackThread->getTrackName_l();
3367            mMainBuffer = playbackThread->mixBuffer();
3368        }
3369        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3370        if (mName < 0) {
3371            ALOGE("no more track names available");
3372        }
3373        mVolume[0] = 1.0f;
3374        mVolume[1] = 1.0f;
3375        mStreamType = streamType;
3376        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3377        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3378        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3379    }
3380}
3381
3382AudioFlinger::PlaybackThread::Track::~Track()
3383{
3384    ALOGV("PlaybackThread::Track destructor");
3385    sp<ThreadBase> thread = mThread.promote();
3386    if (thread != 0) {
3387        Mutex::Autolock _l(thread->mLock);
3388        mState = TERMINATED;
3389    }
3390}
3391
3392void AudioFlinger::PlaybackThread::Track::destroy()
3393{
3394    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3395    // by removing it from mTracks vector, so there is a risk that this Tracks's
3396    // desctructor is called. As the destructor needs to lock mLock,
3397    // we must acquire a strong reference on this Track before locking mLock
3398    // here so that the destructor is called only when exiting this function.
3399    // On the other hand, as long as Track::destroy() is only called by
3400    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3401    // this Track with its member mTrack.
3402    sp<Track> keep(this);
3403    { // scope for mLock
3404        sp<ThreadBase> thread = mThread.promote();
3405        if (thread != 0) {
3406            if (!isOutputTrack()) {
3407                if (mState == ACTIVE || mState == RESUMING) {
3408                    AudioSystem::stopOutput(thread->id(),
3409                                            (audio_stream_type_t)mStreamType,
3410                                            mSessionId);
3411
3412                    // to track the speaker usage
3413                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3414                }
3415                AudioSystem::releaseOutput(thread->id());
3416            }
3417            Mutex::Autolock _l(thread->mLock);
3418            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3419            playbackThread->destroyTrack_l(this);
3420        }
3421    }
3422}
3423
3424void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3425{
3426    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",
3427            mName - AudioMixer::TRACK0,
3428            (mClient == NULL) ? getpid() : mClient->pid(),
3429            mStreamType,
3430            mFormat,
3431            mChannelMask,
3432            mSessionId,
3433            mFrameCount,
3434            mState,
3435            mMute,
3436            mFillingUpStatus,
3437            mCblk->sampleRate,
3438            mCblk->volume[0],
3439            mCblk->volume[1],
3440            mCblk->server,
3441            mCblk->user,
3442            (int)mMainBuffer,
3443            (int)mAuxBuffer);
3444}
3445
3446status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3447{
3448     audio_track_cblk_t* cblk = this->cblk();
3449     uint32_t framesReady;
3450     uint32_t framesReq = buffer->frameCount;
3451
3452     // Check if last stepServer failed, try to step now
3453     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3454         if (!step())  goto getNextBuffer_exit;
3455         ALOGV("stepServer recovered");
3456         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3457     }
3458
3459     framesReady = cblk->framesReady();
3460
3461     if (CC_LIKELY(framesReady)) {
3462        uint32_t s = cblk->server;
3463        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3464
3465        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3466        if (framesReq > framesReady) {
3467            framesReq = framesReady;
3468        }
3469        if (s + framesReq > bufferEnd) {
3470            framesReq = bufferEnd - s;
3471        }
3472
3473         buffer->raw = getBuffer(s, framesReq);
3474         if (buffer->raw == NULL) goto getNextBuffer_exit;
3475
3476         buffer->frameCount = framesReq;
3477        return NO_ERROR;
3478     }
3479
3480getNextBuffer_exit:
3481     buffer->raw = NULL;
3482     buffer->frameCount = 0;
3483     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3484     return NOT_ENOUGH_DATA;
3485}
3486
3487bool AudioFlinger::PlaybackThread::Track::isReady() const {
3488    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3489
3490    if (mCblk->framesReady() >= mCblk->frameCount ||
3491            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3492        mFillingUpStatus = FS_FILLED;
3493        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3494        return true;
3495    }
3496    return false;
3497}
3498
3499status_t AudioFlinger::PlaybackThread::Track::start()
3500{
3501    status_t status = NO_ERROR;
3502    ALOGV("start(%d), calling thread %d session %d",
3503            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3504    sp<ThreadBase> thread = mThread.promote();
3505    if (thread != 0) {
3506        Mutex::Autolock _l(thread->mLock);
3507        int state = mState;
3508        // here the track could be either new, or restarted
3509        // in both cases "unstop" the track
3510        if (mState == PAUSED) {
3511            mState = TrackBase::RESUMING;
3512            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3513        } else {
3514            mState = TrackBase::ACTIVE;
3515            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3516        }
3517
3518        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3519            thread->mLock.unlock();
3520            status = AudioSystem::startOutput(thread->id(),
3521                                              (audio_stream_type_t)mStreamType,
3522                                              mSessionId);
3523            thread->mLock.lock();
3524
3525            // to track the speaker usage
3526            if (status == NO_ERROR) {
3527                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3528            }
3529        }
3530        if (status == NO_ERROR) {
3531            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3532            playbackThread->addTrack_l(this);
3533        } else {
3534            mState = state;
3535        }
3536    } else {
3537        status = BAD_VALUE;
3538    }
3539    return status;
3540}
3541
3542void AudioFlinger::PlaybackThread::Track::stop()
3543{
3544    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3545    sp<ThreadBase> thread = mThread.promote();
3546    if (thread != 0) {
3547        Mutex::Autolock _l(thread->mLock);
3548        int state = mState;
3549        if (mState > STOPPED) {
3550            mState = STOPPED;
3551            // If the track is not active (PAUSED and buffers full), flush buffers
3552            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3553            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3554                reset();
3555            }
3556            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3557        }
3558        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3559            thread->mLock.unlock();
3560            AudioSystem::stopOutput(thread->id(),
3561                                    (audio_stream_type_t)mStreamType,
3562                                    mSessionId);
3563            thread->mLock.lock();
3564
3565            // to track the speaker usage
3566            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3567        }
3568    }
3569}
3570
3571void AudioFlinger::PlaybackThread::Track::pause()
3572{
3573    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3574    sp<ThreadBase> thread = mThread.promote();
3575    if (thread != 0) {
3576        Mutex::Autolock _l(thread->mLock);
3577        if (mState == ACTIVE || mState == RESUMING) {
3578            mState = PAUSING;
3579            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3580            if (!isOutputTrack()) {
3581                thread->mLock.unlock();
3582                AudioSystem::stopOutput(thread->id(),
3583                                        (audio_stream_type_t)mStreamType,
3584                                        mSessionId);
3585                thread->mLock.lock();
3586
3587                // to track the speaker usage
3588                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3589            }
3590        }
3591    }
3592}
3593
3594void AudioFlinger::PlaybackThread::Track::flush()
3595{
3596    ALOGV("flush(%d)", mName);
3597    sp<ThreadBase> thread = mThread.promote();
3598    if (thread != 0) {
3599        Mutex::Autolock _l(thread->mLock);
3600        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3601            return;
3602        }
3603        // No point remaining in PAUSED state after a flush => go to
3604        // STOPPED state
3605        mState = STOPPED;
3606
3607        // do not reset the track if it is still in the process of being stopped or paused.
3608        // this will be done by prepareTracks_l() when the track is stopped.
3609        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3610        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3611            reset();
3612        }
3613    }
3614}
3615
3616void AudioFlinger::PlaybackThread::Track::reset()
3617{
3618    // Do not reset twice to avoid discarding data written just after a flush and before
3619    // the audioflinger thread detects the track is stopped.
3620    if (!mResetDone) {
3621        TrackBase::reset();
3622        // Force underrun condition to avoid false underrun callback until first data is
3623        // written to buffer
3624        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3625        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3626        mFillingUpStatus = FS_FILLING;
3627        mResetDone = true;
3628    }
3629}
3630
3631void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3632{
3633    mMute = muted;
3634}
3635
3636void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3637{
3638    mVolume[0] = left;
3639    mVolume[1] = right;
3640}
3641
3642status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3643{
3644    status_t status = DEAD_OBJECT;
3645    sp<ThreadBase> thread = mThread.promote();
3646    if (thread != 0) {
3647       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3648       status = playbackThread->attachAuxEffect(this, EffectId);
3649    }
3650    return status;
3651}
3652
3653void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3654{
3655    mAuxEffectId = EffectId;
3656    mAuxBuffer = buffer;
3657}
3658
3659// ----------------------------------------------------------------------------
3660
3661// RecordTrack constructor must be called with AudioFlinger::mLock held
3662AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3663            const wp<ThreadBase>& thread,
3664            const sp<Client>& client,
3665            uint32_t sampleRate,
3666            uint32_t format,
3667            uint32_t channelMask,
3668            int frameCount,
3669            uint32_t flags,
3670            int sessionId)
3671    :   TrackBase(thread, client, sampleRate, format,
3672                  channelMask, frameCount, flags, 0, sessionId),
3673        mOverflow(false)
3674{
3675    if (mCblk != NULL) {
3676       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3677       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3678           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3679       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3680           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3681       } else {
3682           mCblk->frameSize = sizeof(int8_t);
3683       }
3684    }
3685}
3686
3687AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3688{
3689    sp<ThreadBase> thread = mThread.promote();
3690    if (thread != 0) {
3691        AudioSystem::releaseInput(thread->id());
3692    }
3693}
3694
3695status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3696{
3697    audio_track_cblk_t* cblk = this->cblk();
3698    uint32_t framesAvail;
3699    uint32_t framesReq = buffer->frameCount;
3700
3701     // Check if last stepServer failed, try to step now
3702    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3703        if (!step()) goto getNextBuffer_exit;
3704        ALOGV("stepServer recovered");
3705        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3706    }
3707
3708    framesAvail = cblk->framesAvailable_l();
3709
3710    if (CC_LIKELY(framesAvail)) {
3711        uint32_t s = cblk->server;
3712        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3713
3714        if (framesReq > framesAvail) {
3715            framesReq = framesAvail;
3716        }
3717        if (s + framesReq > bufferEnd) {
3718            framesReq = bufferEnd - s;
3719        }
3720
3721        buffer->raw = getBuffer(s, framesReq);
3722        if (buffer->raw == NULL) goto getNextBuffer_exit;
3723
3724        buffer->frameCount = framesReq;
3725        return NO_ERROR;
3726    }
3727
3728getNextBuffer_exit:
3729    buffer->raw = NULL;
3730    buffer->frameCount = 0;
3731    return NOT_ENOUGH_DATA;
3732}
3733
3734status_t AudioFlinger::RecordThread::RecordTrack::start()
3735{
3736    sp<ThreadBase> thread = mThread.promote();
3737    if (thread != 0) {
3738        RecordThread *recordThread = (RecordThread *)thread.get();
3739        return recordThread->start(this);
3740    } else {
3741        return BAD_VALUE;
3742    }
3743}
3744
3745void AudioFlinger::RecordThread::RecordTrack::stop()
3746{
3747    sp<ThreadBase> thread = mThread.promote();
3748    if (thread != 0) {
3749        RecordThread *recordThread = (RecordThread *)thread.get();
3750        recordThread->stop(this);
3751        TrackBase::reset();
3752        // Force overerrun condition to avoid false overrun callback until first data is
3753        // read from buffer
3754        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3755    }
3756}
3757
3758void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3759{
3760    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3761            (mClient == NULL) ? getpid() : mClient->pid(),
3762            mFormat,
3763            mChannelMask,
3764            mSessionId,
3765            mFrameCount,
3766            mState,
3767            mCblk->sampleRate,
3768            mCblk->server,
3769            mCblk->user);
3770}
3771
3772
3773// ----------------------------------------------------------------------------
3774
3775AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3776            const wp<ThreadBase>& thread,
3777            DuplicatingThread *sourceThread,
3778            uint32_t sampleRate,
3779            uint32_t format,
3780            uint32_t channelMask,
3781            int frameCount)
3782    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3783    mActive(false), mSourceThread(sourceThread)
3784{
3785
3786    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3787    if (mCblk != NULL) {
3788        mCblk->flags |= CBLK_DIRECTION_OUT;
3789        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3790        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
3791        mOutBuffer.frameCount = 0;
3792        playbackThread->mTracks.add(this);
3793        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3794                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3795                mCblk, mBuffer, mCblk->buffers,
3796                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3797    } else {
3798        ALOGW("Error creating output track on thread %p", playbackThread);
3799    }
3800}
3801
3802AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3803{
3804    clearBufferQueue();
3805}
3806
3807status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3808{
3809    status_t status = Track::start();
3810    if (status != NO_ERROR) {
3811        return status;
3812    }
3813
3814    mActive = true;
3815    mRetryCount = 127;
3816    return status;
3817}
3818
3819void AudioFlinger::PlaybackThread::OutputTrack::stop()
3820{
3821    Track::stop();
3822    clearBufferQueue();
3823    mOutBuffer.frameCount = 0;
3824    mActive = false;
3825}
3826
3827bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3828{
3829    Buffer *pInBuffer;
3830    Buffer inBuffer;
3831    uint32_t channelCount = mChannelCount;
3832    bool outputBufferFull = false;
3833    inBuffer.frameCount = frames;
3834    inBuffer.i16 = data;
3835
3836    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3837
3838    if (!mActive && frames != 0) {
3839        start();
3840        sp<ThreadBase> thread = mThread.promote();
3841        if (thread != 0) {
3842            MixerThread *mixerThread = (MixerThread *)thread.get();
3843            if (mCblk->frameCount > frames){
3844                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3845                    uint32_t startFrames = (mCblk->frameCount - frames);
3846                    pInBuffer = new Buffer;
3847                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3848                    pInBuffer->frameCount = startFrames;
3849                    pInBuffer->i16 = pInBuffer->mBuffer;
3850                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3851                    mBufferQueue.add(pInBuffer);
3852                } else {
3853                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
3854                }
3855            }
3856        }
3857    }
3858
3859    while (waitTimeLeftMs) {
3860        // First write pending buffers, then new data
3861        if (mBufferQueue.size()) {
3862            pInBuffer = mBufferQueue.itemAt(0);
3863        } else {
3864            pInBuffer = &inBuffer;
3865        }
3866
3867        if (pInBuffer->frameCount == 0) {
3868            break;
3869        }
3870
3871        if (mOutBuffer.frameCount == 0) {
3872            mOutBuffer.frameCount = pInBuffer->frameCount;
3873            nsecs_t startTime = systemTime();
3874            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
3875                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3876                outputBufferFull = true;
3877                break;
3878            }
3879            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3880            if (waitTimeLeftMs >= waitTimeMs) {
3881                waitTimeLeftMs -= waitTimeMs;
3882            } else {
3883                waitTimeLeftMs = 0;
3884            }
3885        }
3886
3887        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3888        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3889        mCblk->stepUser(outFrames);
3890        pInBuffer->frameCount -= outFrames;
3891        pInBuffer->i16 += outFrames * channelCount;
3892        mOutBuffer.frameCount -= outFrames;
3893        mOutBuffer.i16 += outFrames * channelCount;
3894
3895        if (pInBuffer->frameCount == 0) {
3896            if (mBufferQueue.size()) {
3897                mBufferQueue.removeAt(0);
3898                delete [] pInBuffer->mBuffer;
3899                delete pInBuffer;
3900                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3901            } else {
3902                break;
3903            }
3904        }
3905    }
3906
3907    // If we could not write all frames, allocate a buffer and queue it for next time.
3908    if (inBuffer.frameCount) {
3909        sp<ThreadBase> thread = mThread.promote();
3910        if (thread != 0 && !thread->standby()) {
3911            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3912                pInBuffer = new Buffer;
3913                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3914                pInBuffer->frameCount = inBuffer.frameCount;
3915                pInBuffer->i16 = pInBuffer->mBuffer;
3916                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3917                mBufferQueue.add(pInBuffer);
3918                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3919            } else {
3920                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3921            }
3922        }
3923    }
3924
3925    // Calling write() with a 0 length buffer, means that no more data will be written:
3926    // If no more buffers are pending, fill output track buffer to make sure it is started
3927    // by output mixer.
3928    if (frames == 0 && mBufferQueue.size() == 0) {
3929        if (mCblk->user < mCblk->frameCount) {
3930            frames = mCblk->frameCount - mCblk->user;
3931            pInBuffer = new Buffer;
3932            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3933            pInBuffer->frameCount = frames;
3934            pInBuffer->i16 = pInBuffer->mBuffer;
3935            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3936            mBufferQueue.add(pInBuffer);
3937        } else if (mActive) {
3938            stop();
3939        }
3940    }
3941
3942    return outputBufferFull;
3943}
3944
3945status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3946{
3947    int active;
3948    status_t result;
3949    audio_track_cblk_t* cblk = mCblk;
3950    uint32_t framesReq = buffer->frameCount;
3951
3952//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3953    buffer->frameCount  = 0;
3954
3955    uint32_t framesAvail = cblk->framesAvailable();
3956
3957
3958    if (framesAvail == 0) {
3959        Mutex::Autolock _l(cblk->lock);
3960        goto start_loop_here;
3961        while (framesAvail == 0) {
3962            active = mActive;
3963            if (CC_UNLIKELY(!active)) {
3964                ALOGV("Not active and NO_MORE_BUFFERS");
3965                return AudioTrack::NO_MORE_BUFFERS;
3966            }
3967            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
3968            if (result != NO_ERROR) {
3969                return AudioTrack::NO_MORE_BUFFERS;
3970            }
3971            // read the server count again
3972        start_loop_here:
3973            framesAvail = cblk->framesAvailable_l();
3974        }
3975    }
3976
3977//    if (framesAvail < framesReq) {
3978//        return AudioTrack::NO_MORE_BUFFERS;
3979//    }
3980
3981    if (framesReq > framesAvail) {
3982        framesReq = framesAvail;
3983    }
3984
3985    uint32_t u = cblk->user;
3986    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
3987
3988    if (u + framesReq > bufferEnd) {
3989        framesReq = bufferEnd - u;
3990    }
3991
3992    buffer->frameCount  = framesReq;
3993    buffer->raw         = (void *)cblk->buffer(u);
3994    return NO_ERROR;
3995}
3996
3997
3998void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
3999{
4000    size_t size = mBufferQueue.size();
4001    Buffer *pBuffer;
4002
4003    for (size_t i = 0; i < size; i++) {
4004        pBuffer = mBufferQueue.itemAt(i);
4005        delete [] pBuffer->mBuffer;
4006        delete pBuffer;
4007    }
4008    mBufferQueue.clear();
4009}
4010
4011// ----------------------------------------------------------------------------
4012
4013AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
4014    :   RefBase(),
4015        mAudioFlinger(audioFlinger),
4016        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
4017        mPid(pid)
4018{
4019    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
4020}
4021
4022// Client destructor must be called with AudioFlinger::mLock held
4023AudioFlinger::Client::~Client()
4024{
4025    mAudioFlinger->removeClient_l(mPid);
4026}
4027
4028const sp<MemoryDealer>& AudioFlinger::Client::heap() const
4029{
4030    return mMemoryDealer;
4031}
4032
4033// ----------------------------------------------------------------------------
4034
4035AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4036                                                     const sp<IAudioFlingerClient>& client,
4037                                                     pid_t pid)
4038    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
4039{
4040}
4041
4042AudioFlinger::NotificationClient::~NotificationClient()
4043{
4044    mClient.clear();
4045}
4046
4047void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4048{
4049    sp<NotificationClient> keep(this);
4050    {
4051        mAudioFlinger->removeNotificationClient(mPid);
4052    }
4053}
4054
4055// ----------------------------------------------------------------------------
4056
4057AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4058    : BnAudioTrack(),
4059      mTrack(track)
4060{
4061}
4062
4063AudioFlinger::TrackHandle::~TrackHandle() {
4064    // just stop the track on deletion, associated resources
4065    // will be freed from the main thread once all pending buffers have
4066    // been played. Unless it's not in the active track list, in which
4067    // case we free everything now...
4068    mTrack->destroy();
4069}
4070
4071status_t AudioFlinger::TrackHandle::start() {
4072    return mTrack->start();
4073}
4074
4075void AudioFlinger::TrackHandle::stop() {
4076    mTrack->stop();
4077}
4078
4079void AudioFlinger::TrackHandle::flush() {
4080    mTrack->flush();
4081}
4082
4083void AudioFlinger::TrackHandle::mute(bool e) {
4084    mTrack->mute(e);
4085}
4086
4087void AudioFlinger::TrackHandle::pause() {
4088    mTrack->pause();
4089}
4090
4091void AudioFlinger::TrackHandle::setVolume(float left, float right) {
4092    mTrack->setVolume(left, right);
4093}
4094
4095sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4096    return mTrack->getCblk();
4097}
4098
4099status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4100{
4101    return mTrack->attachAuxEffect(EffectId);
4102}
4103
4104status_t AudioFlinger::TrackHandle::onTransact(
4105    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4106{
4107    return BnAudioTrack::onTransact(code, data, reply, flags);
4108}
4109
4110// ----------------------------------------------------------------------------
4111
4112sp<IAudioRecord> AudioFlinger::openRecord(
4113        pid_t pid,
4114        int input,
4115        uint32_t sampleRate,
4116        uint32_t format,
4117        uint32_t channelMask,
4118        int frameCount,
4119        uint32_t flags,
4120        int *sessionId,
4121        status_t *status)
4122{
4123    sp<RecordThread::RecordTrack> recordTrack;
4124    sp<RecordHandle> recordHandle;
4125    sp<Client> client;
4126    wp<Client> wclient;
4127    status_t lStatus;
4128    RecordThread *thread;
4129    size_t inFrameCount;
4130    int lSessionId;
4131
4132    // check calling permissions
4133    if (!recordingAllowed()) {
4134        lStatus = PERMISSION_DENIED;
4135        goto Exit;
4136    }
4137
4138    // add client to list
4139    { // scope for mLock
4140        Mutex::Autolock _l(mLock);
4141        thread = checkRecordThread_l(input);
4142        if (thread == NULL) {
4143            lStatus = BAD_VALUE;
4144            goto Exit;
4145        }
4146
4147        wclient = mClients.valueFor(pid);
4148        if (wclient != NULL) {
4149            client = wclient.promote();
4150        } else {
4151            client = new Client(this, pid);
4152            mClients.add(pid, client);
4153        }
4154
4155        // If no audio session id is provided, create one here
4156        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4157            lSessionId = *sessionId;
4158        } else {
4159            lSessionId = nextUniqueId();
4160            if (sessionId != NULL) {
4161                *sessionId = lSessionId;
4162            }
4163        }
4164        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4165        recordTrack = thread->createRecordTrack_l(client,
4166                                                sampleRate,
4167                                                format,
4168                                                channelMask,
4169                                                frameCount,
4170                                                flags,
4171                                                lSessionId,
4172                                                &lStatus);
4173    }
4174    if (lStatus != NO_ERROR) {
4175        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4176        // destructor is called by the TrackBase destructor with mLock held
4177        client.clear();
4178        recordTrack.clear();
4179        goto Exit;
4180    }
4181
4182    // return to handle to client
4183    recordHandle = new RecordHandle(recordTrack);
4184    lStatus = NO_ERROR;
4185
4186Exit:
4187    if (status) {
4188        *status = lStatus;
4189    }
4190    return recordHandle;
4191}
4192
4193// ----------------------------------------------------------------------------
4194
4195AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4196    : BnAudioRecord(),
4197    mRecordTrack(recordTrack)
4198{
4199}
4200
4201AudioFlinger::RecordHandle::~RecordHandle() {
4202    stop();
4203}
4204
4205status_t AudioFlinger::RecordHandle::start() {
4206    ALOGV("RecordHandle::start()");
4207    return mRecordTrack->start();
4208}
4209
4210void AudioFlinger::RecordHandle::stop() {
4211    ALOGV("RecordHandle::stop()");
4212    mRecordTrack->stop();
4213}
4214
4215sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4216    return mRecordTrack->getCblk();
4217}
4218
4219status_t AudioFlinger::RecordHandle::onTransact(
4220    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4221{
4222    return BnAudioRecord::onTransact(code, data, reply, flags);
4223}
4224
4225// ----------------------------------------------------------------------------
4226
4227AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4228                                         AudioStreamIn *input,
4229                                         uint32_t sampleRate,
4230                                         uint32_t channels,
4231                                         int id,
4232                                         uint32_t device) :
4233    ThreadBase(audioFlinger, id, device),
4234    mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
4235{
4236    mType = ThreadBase::RECORD;
4237
4238    snprintf(mName, kNameLength, "AudioIn_%d", id);
4239
4240    mReqChannelCount = popcount(channels);
4241    mReqSampleRate = sampleRate;
4242    readInputParameters();
4243}
4244
4245
4246AudioFlinger::RecordThread::~RecordThread()
4247{
4248    delete[] mRsmpInBuffer;
4249    if (mResampler != NULL) {
4250        delete mResampler;
4251        delete[] mRsmpOutBuffer;
4252    }
4253}
4254
4255void AudioFlinger::RecordThread::onFirstRef()
4256{
4257    run(mName, PRIORITY_URGENT_AUDIO);
4258}
4259
4260status_t AudioFlinger::RecordThread::readyToRun()
4261{
4262    status_t status = initCheck();
4263    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4264    return status;
4265}
4266
4267bool AudioFlinger::RecordThread::threadLoop()
4268{
4269    AudioBufferProvider::Buffer buffer;
4270    sp<RecordTrack> activeTrack;
4271    Vector< sp<EffectChain> > effectChains;
4272
4273    nsecs_t lastWarning = 0;
4274
4275    acquireWakeLock();
4276
4277    // start recording
4278    while (!exitPending()) {
4279
4280        processConfigEvents();
4281
4282        { // scope for mLock
4283            Mutex::Autolock _l(mLock);
4284            checkForNewParameters_l();
4285            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4286                if (!mStandby) {
4287                    mInput->stream->common.standby(&mInput->stream->common);
4288                    mStandby = true;
4289                }
4290
4291                if (exitPending()) break;
4292
4293                releaseWakeLock_l();
4294                ALOGV("RecordThread: loop stopping");
4295                // go to sleep
4296                mWaitWorkCV.wait(mLock);
4297                ALOGV("RecordThread: loop starting");
4298                acquireWakeLock_l();
4299                continue;
4300            }
4301            if (mActiveTrack != 0) {
4302                if (mActiveTrack->mState == TrackBase::PAUSING) {
4303                    if (!mStandby) {
4304                        mInput->stream->common.standby(&mInput->stream->common);
4305                        mStandby = true;
4306                    }
4307                    mActiveTrack.clear();
4308                    mStartStopCond.broadcast();
4309                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4310                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4311                        mActiveTrack.clear();
4312                        mStartStopCond.broadcast();
4313                    } else if (mBytesRead != 0) {
4314                        // record start succeeds only if first read from audio input
4315                        // succeeds
4316                        if (mBytesRead > 0) {
4317                            mActiveTrack->mState = TrackBase::ACTIVE;
4318                        } else {
4319                            mActiveTrack.clear();
4320                        }
4321                        mStartStopCond.broadcast();
4322                    }
4323                    mStandby = false;
4324                }
4325            }
4326            lockEffectChains_l(effectChains);
4327        }
4328
4329        if (mActiveTrack != 0) {
4330            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4331                mActiveTrack->mState != TrackBase::RESUMING) {
4332                unlockEffectChains(effectChains);
4333                usleep(kRecordThreadSleepUs);
4334                continue;
4335            }
4336            for (size_t i = 0; i < effectChains.size(); i ++) {
4337                effectChains[i]->process_l();
4338            }
4339
4340            buffer.frameCount = mFrameCount;
4341            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4342                size_t framesOut = buffer.frameCount;
4343                if (mResampler == NULL) {
4344                    // no resampling
4345                    while (framesOut) {
4346                        size_t framesIn = mFrameCount - mRsmpInIndex;
4347                        if (framesIn) {
4348                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4349                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4350                            if (framesIn > framesOut)
4351                                framesIn = framesOut;
4352                            mRsmpInIndex += framesIn;
4353                            framesOut -= framesIn;
4354                            if ((int)mChannelCount == mReqChannelCount ||
4355                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4356                                memcpy(dst, src, framesIn * mFrameSize);
4357                            } else {
4358                                int16_t *src16 = (int16_t *)src;
4359                                int16_t *dst16 = (int16_t *)dst;
4360                                if (mChannelCount == 1) {
4361                                    while (framesIn--) {
4362                                        *dst16++ = *src16;
4363                                        *dst16++ = *src16++;
4364                                    }
4365                                } else {
4366                                    while (framesIn--) {
4367                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4368                                        src16 += 2;
4369                                    }
4370                                }
4371                            }
4372                        }
4373                        if (framesOut && mFrameCount == mRsmpInIndex) {
4374                            if (framesOut == mFrameCount &&
4375                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4376                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4377                                framesOut = 0;
4378                            } else {
4379                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4380                                mRsmpInIndex = 0;
4381                            }
4382                            if (mBytesRead < 0) {
4383                                ALOGE("Error reading audio input");
4384                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4385                                    // Force input into standby so that it tries to
4386                                    // recover at next read attempt
4387                                    mInput->stream->common.standby(&mInput->stream->common);
4388                                    usleep(kRecordThreadSleepUs);
4389                                }
4390                                mRsmpInIndex = mFrameCount;
4391                                framesOut = 0;
4392                                buffer.frameCount = 0;
4393                            }
4394                        }
4395                    }
4396                } else {
4397                    // resampling
4398
4399                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4400                    // alter output frame count as if we were expecting stereo samples
4401                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4402                        framesOut >>= 1;
4403                    }
4404                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4405                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4406                    // are 32 bit aligned which should be always true.
4407                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4408                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4409                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4410                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4411                        int16_t *dst = buffer.i16;
4412                        while (framesOut--) {
4413                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4414                            src += 2;
4415                        }
4416                    } else {
4417                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4418                    }
4419
4420                }
4421                mActiveTrack->releaseBuffer(&buffer);
4422                mActiveTrack->overflow();
4423            }
4424            // client isn't retrieving buffers fast enough
4425            else {
4426                if (!mActiveTrack->setOverflow()) {
4427                    nsecs_t now = systemTime();
4428                    if ((now - lastWarning) > kWarningThrottleNs) {
4429                        ALOGW("RecordThread: buffer overflow");
4430                        lastWarning = now;
4431                    }
4432                }
4433                // Release the processor for a while before asking for a new buffer.
4434                // This will give the application more chance to read from the buffer and
4435                // clear the overflow.
4436                usleep(kRecordThreadSleepUs);
4437            }
4438        }
4439        // enable changes in effect chain
4440        unlockEffectChains(effectChains);
4441        effectChains.clear();
4442    }
4443
4444    if (!mStandby) {
4445        mInput->stream->common.standby(&mInput->stream->common);
4446    }
4447    mActiveTrack.clear();
4448
4449    mStartStopCond.broadcast();
4450
4451    releaseWakeLock();
4452
4453    ALOGV("RecordThread %p exiting", this);
4454    return false;
4455}
4456
4457
4458sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4459        const sp<AudioFlinger::Client>& client,
4460        uint32_t sampleRate,
4461        int format,
4462        int channelMask,
4463        int frameCount,
4464        uint32_t flags,
4465        int sessionId,
4466        status_t *status)
4467{
4468    sp<RecordTrack> track;
4469    status_t lStatus;
4470
4471    lStatus = initCheck();
4472    if (lStatus != NO_ERROR) {
4473        ALOGE("Audio driver not initialized.");
4474        goto Exit;
4475    }
4476
4477    { // scope for mLock
4478        Mutex::Autolock _l(mLock);
4479
4480        track = new RecordTrack(this, client, sampleRate,
4481                      format, channelMask, frameCount, flags, sessionId);
4482
4483        if (track->getCblk() == NULL) {
4484            lStatus = NO_MEMORY;
4485            goto Exit;
4486        }
4487
4488        mTrack = track.get();
4489        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4490        bool suspend = audio_is_bluetooth_sco_device(
4491                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4492        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4493        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4494    }
4495    lStatus = NO_ERROR;
4496
4497Exit:
4498    if (status) {
4499        *status = lStatus;
4500    }
4501    return track;
4502}
4503
4504status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4505{
4506    ALOGV("RecordThread::start");
4507    sp <ThreadBase> strongMe = this;
4508    status_t status = NO_ERROR;
4509    {
4510        AutoMutex lock(mLock);
4511        if (mActiveTrack != 0) {
4512            if (recordTrack != mActiveTrack.get()) {
4513                status = -EBUSY;
4514            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4515                mActiveTrack->mState = TrackBase::ACTIVE;
4516            }
4517            return status;
4518        }
4519
4520        recordTrack->mState = TrackBase::IDLE;
4521        mActiveTrack = recordTrack;
4522        mLock.unlock();
4523        status_t status = AudioSystem::startInput(mId);
4524        mLock.lock();
4525        if (status != NO_ERROR) {
4526            mActiveTrack.clear();
4527            return status;
4528        }
4529        mRsmpInIndex = mFrameCount;
4530        mBytesRead = 0;
4531        if (mResampler != NULL) {
4532            mResampler->reset();
4533        }
4534        mActiveTrack->mState = TrackBase::RESUMING;
4535        // signal thread to start
4536        ALOGV("Signal record thread");
4537        mWaitWorkCV.signal();
4538        // do not wait for mStartStopCond if exiting
4539        if (mExiting) {
4540            mActiveTrack.clear();
4541            status = INVALID_OPERATION;
4542            goto startError;
4543        }
4544        mStartStopCond.wait(mLock);
4545        if (mActiveTrack == 0) {
4546            ALOGV("Record failed to start");
4547            status = BAD_VALUE;
4548            goto startError;
4549        }
4550        ALOGV("Record started OK");
4551        return status;
4552    }
4553startError:
4554    AudioSystem::stopInput(mId);
4555    return status;
4556}
4557
4558void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4559    ALOGV("RecordThread::stop");
4560    sp <ThreadBase> strongMe = this;
4561    {
4562        AutoMutex lock(mLock);
4563        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4564            mActiveTrack->mState = TrackBase::PAUSING;
4565            // do not wait for mStartStopCond if exiting
4566            if (mExiting) {
4567                return;
4568            }
4569            mStartStopCond.wait(mLock);
4570            // if we have been restarted, recordTrack == mActiveTrack.get() here
4571            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4572                mLock.unlock();
4573                AudioSystem::stopInput(mId);
4574                mLock.lock();
4575                ALOGV("Record stopped OK");
4576            }
4577        }
4578    }
4579}
4580
4581status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4582{
4583    const size_t SIZE = 256;
4584    char buffer[SIZE];
4585    String8 result;
4586    pid_t pid = 0;
4587
4588    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4589    result.append(buffer);
4590
4591    if (mActiveTrack != 0) {
4592        result.append("Active Track:\n");
4593        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4594        mActiveTrack->dump(buffer, SIZE);
4595        result.append(buffer);
4596
4597        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4598        result.append(buffer);
4599        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4600        result.append(buffer);
4601        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4602        result.append(buffer);
4603        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4604        result.append(buffer);
4605        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4606        result.append(buffer);
4607
4608
4609    } else {
4610        result.append("No record client\n");
4611    }
4612    write(fd, result.string(), result.size());
4613
4614    dumpBase(fd, args);
4615    dumpEffectChains(fd, args);
4616
4617    return NO_ERROR;
4618}
4619
4620status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4621{
4622    size_t framesReq = buffer->frameCount;
4623    size_t framesReady = mFrameCount - mRsmpInIndex;
4624    int channelCount;
4625
4626    if (framesReady == 0) {
4627        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4628        if (mBytesRead < 0) {
4629            ALOGE("RecordThread::getNextBuffer() Error reading audio input");
4630            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4631                // Force input into standby so that it tries to
4632                // recover at next read attempt
4633                mInput->stream->common.standby(&mInput->stream->common);
4634                usleep(kRecordThreadSleepUs);
4635            }
4636            buffer->raw = NULL;
4637            buffer->frameCount = 0;
4638            return NOT_ENOUGH_DATA;
4639        }
4640        mRsmpInIndex = 0;
4641        framesReady = mFrameCount;
4642    }
4643
4644    if (framesReq > framesReady) {
4645        framesReq = framesReady;
4646    }
4647
4648    if (mChannelCount == 1 && mReqChannelCount == 2) {
4649        channelCount = 1;
4650    } else {
4651        channelCount = 2;
4652    }
4653    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4654    buffer->frameCount = framesReq;
4655    return NO_ERROR;
4656}
4657
4658void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4659{
4660    mRsmpInIndex += buffer->frameCount;
4661    buffer->frameCount = 0;
4662}
4663
4664bool AudioFlinger::RecordThread::checkForNewParameters_l()
4665{
4666    bool reconfig = false;
4667
4668    while (!mNewParameters.isEmpty()) {
4669        status_t status = NO_ERROR;
4670        String8 keyValuePair = mNewParameters[0];
4671        AudioParameter param = AudioParameter(keyValuePair);
4672        int value;
4673        int reqFormat = mFormat;
4674        int reqSamplingRate = mReqSampleRate;
4675        int reqChannelCount = mReqChannelCount;
4676
4677        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4678            reqSamplingRate = value;
4679            reconfig = true;
4680        }
4681        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4682            reqFormat = value;
4683            reconfig = true;
4684        }
4685        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4686            reqChannelCount = popcount(value);
4687            reconfig = true;
4688        }
4689        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4690            // do not accept frame count changes if tracks are open as the track buffer
4691            // size depends on frame count and correct behavior would not be garantied
4692            // if frame count is changed after track creation
4693            if (mActiveTrack != 0) {
4694                status = INVALID_OPERATION;
4695            } else {
4696                reconfig = true;
4697            }
4698        }
4699        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4700            // forward device change to effects that have requested to be
4701            // aware of attached audio device.
4702            for (size_t i = 0; i < mEffectChains.size(); i++) {
4703                mEffectChains[i]->setDevice_l(value);
4704            }
4705            // store input device and output device but do not forward output device to audio HAL.
4706            // Note that status is ignored by the caller for output device
4707            // (see AudioFlinger::setParameters()
4708            if (value & AUDIO_DEVICE_OUT_ALL) {
4709                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4710                status = BAD_VALUE;
4711            } else {
4712                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4713                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4714                if (mTrack != NULL) {
4715                    bool suspend = audio_is_bluetooth_sco_device(
4716                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4717                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4718                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4719                }
4720            }
4721            mDevice |= (uint32_t)value;
4722        }
4723        if (status == NO_ERROR) {
4724            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4725            if (status == INVALID_OPERATION) {
4726               mInput->stream->common.standby(&mInput->stream->common);
4727               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4728            }
4729            if (reconfig) {
4730                if (status == BAD_VALUE &&
4731                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4732                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4733                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4734                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4735                    (reqChannelCount < 3)) {
4736                    status = NO_ERROR;
4737                }
4738                if (status == NO_ERROR) {
4739                    readInputParameters();
4740                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4741                }
4742            }
4743        }
4744
4745        mNewParameters.removeAt(0);
4746
4747        mParamStatus = status;
4748        mParamCond.signal();
4749        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4750        // already timed out waiting for the status and will never signal the condition.
4751        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4752    }
4753    return reconfig;
4754}
4755
4756String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4757{
4758    char *s;
4759    String8 out_s8 = String8();
4760
4761    Mutex::Autolock _l(mLock);
4762    if (initCheck() != NO_ERROR) {
4763        return out_s8;
4764    }
4765
4766    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4767    out_s8 = String8(s);
4768    free(s);
4769    return out_s8;
4770}
4771
4772void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4773    AudioSystem::OutputDescriptor desc;
4774    void *param2 = 0;
4775
4776    switch (event) {
4777    case AudioSystem::INPUT_OPENED:
4778    case AudioSystem::INPUT_CONFIG_CHANGED:
4779        desc.channels = mChannelMask;
4780        desc.samplingRate = mSampleRate;
4781        desc.format = mFormat;
4782        desc.frameCount = mFrameCount;
4783        desc.latency = 0;
4784        param2 = &desc;
4785        break;
4786
4787    case AudioSystem::INPUT_CLOSED:
4788    default:
4789        break;
4790    }
4791    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4792}
4793
4794void AudioFlinger::RecordThread::readInputParameters()
4795{
4796    if (mRsmpInBuffer) delete mRsmpInBuffer;
4797    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4798    if (mResampler) delete mResampler;
4799    mResampler = NULL;
4800
4801    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4802    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4803    mChannelCount = (uint16_t)popcount(mChannelMask);
4804    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4805    mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
4806    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4807    mFrameCount = mInputBytes / mFrameSize;
4808    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4809
4810    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4811    {
4812        int channelCount;
4813         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4814         // stereo to mono post process as the resampler always outputs stereo.
4815        if (mChannelCount == 1 && mReqChannelCount == 2) {
4816            channelCount = 1;
4817        } else {
4818            channelCount = 2;
4819        }
4820        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4821        mResampler->setSampleRate(mSampleRate);
4822        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4823        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4824
4825        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4826        if (mChannelCount == 1 && mReqChannelCount == 1) {
4827            mFrameCount >>= 1;
4828        }
4829
4830    }
4831    mRsmpInIndex = mFrameCount;
4832}
4833
4834unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4835{
4836    Mutex::Autolock _l(mLock);
4837    if (initCheck() != NO_ERROR) {
4838        return 0;
4839    }
4840
4841    return mInput->stream->get_input_frames_lost(mInput->stream);
4842}
4843
4844uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4845{
4846    Mutex::Autolock _l(mLock);
4847    uint32_t result = 0;
4848    if (getEffectChain_l(sessionId) != 0) {
4849        result = EFFECT_SESSION;
4850    }
4851
4852    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4853        result |= TRACK_SESSION;
4854    }
4855
4856    return result;
4857}
4858
4859AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4860{
4861    Mutex::Autolock _l(mLock);
4862    return mTrack;
4863}
4864
4865AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
4866{
4867    Mutex::Autolock _l(mLock);
4868    return mInput;
4869}
4870
4871AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4872{
4873    Mutex::Autolock _l(mLock);
4874    AudioStreamIn *input = mInput;
4875    mInput = NULL;
4876    return input;
4877}
4878
4879// this method must always be called either with ThreadBase mLock held or inside the thread loop
4880audio_stream_t* AudioFlinger::RecordThread::stream()
4881{
4882    if (mInput == NULL) {
4883        return NULL;
4884    }
4885    return &mInput->stream->common;
4886}
4887
4888
4889// ----------------------------------------------------------------------------
4890
4891int AudioFlinger::openOutput(uint32_t *pDevices,
4892                                uint32_t *pSamplingRate,
4893                                uint32_t *pFormat,
4894                                uint32_t *pChannels,
4895                                uint32_t *pLatencyMs,
4896                                uint32_t flags)
4897{
4898    status_t status;
4899    PlaybackThread *thread = NULL;
4900    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4901    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4902    uint32_t format = pFormat ? *pFormat : 0;
4903    uint32_t channels = pChannels ? *pChannels : 0;
4904    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4905    audio_stream_out_t *outStream;
4906    audio_hw_device_t *outHwDev;
4907
4908    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4909            pDevices ? *pDevices : 0,
4910            samplingRate,
4911            format,
4912            channels,
4913            flags);
4914
4915    if (pDevices == NULL || *pDevices == 0) {
4916        return 0;
4917    }
4918
4919    Mutex::Autolock _l(mLock);
4920
4921    outHwDev = findSuitableHwDev_l(*pDevices);
4922    if (outHwDev == NULL)
4923        return 0;
4924
4925    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
4926                                          &channels, &samplingRate, &outStream);
4927    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4928            outStream,
4929            samplingRate,
4930            format,
4931            channels,
4932            status);
4933
4934    mHardwareStatus = AUDIO_HW_IDLE;
4935    if (outStream != NULL) {
4936        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4937        int id = nextUniqueId();
4938
4939        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4940            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4941            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4942            thread = new DirectOutputThread(this, output, id, *pDevices);
4943            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4944        } else {
4945            thread = new MixerThread(this, output, id, *pDevices);
4946            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4947        }
4948        mPlaybackThreads.add(id, thread);
4949
4950        if (pSamplingRate) *pSamplingRate = samplingRate;
4951        if (pFormat) *pFormat = format;
4952        if (pChannels) *pChannels = channels;
4953        if (pLatencyMs) *pLatencyMs = thread->latency();
4954
4955        // notify client processes of the new output creation
4956        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4957        return id;
4958    }
4959
4960    return 0;
4961}
4962
4963int AudioFlinger::openDuplicateOutput(int output1, int output2)
4964{
4965    Mutex::Autolock _l(mLock);
4966    MixerThread *thread1 = checkMixerThread_l(output1);
4967    MixerThread *thread2 = checkMixerThread_l(output2);
4968
4969    if (thread1 == NULL || thread2 == NULL) {
4970        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
4971        return 0;
4972    }
4973
4974    int id = nextUniqueId();
4975    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
4976    thread->addOutputTrack(thread2);
4977    mPlaybackThreads.add(id, thread);
4978    // notify client processes of the new output creation
4979    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4980    return id;
4981}
4982
4983status_t AudioFlinger::closeOutput(int output)
4984{
4985    // keep strong reference on the playback thread so that
4986    // it is not destroyed while exit() is executed
4987    sp <PlaybackThread> thread;
4988    {
4989        Mutex::Autolock _l(mLock);
4990        thread = checkPlaybackThread_l(output);
4991        if (thread == NULL) {
4992            return BAD_VALUE;
4993        }
4994
4995        ALOGV("closeOutput() %d", output);
4996
4997        if (thread->type() == ThreadBase::MIXER) {
4998            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4999                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
5000                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
5001                    dupThread->removeOutputTrack((MixerThread *)thread.get());
5002                }
5003            }
5004        }
5005        void *param2 = 0;
5006        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
5007        mPlaybackThreads.removeItem(output);
5008    }
5009    thread->exit();
5010
5011    if (thread->type() != ThreadBase::DUPLICATING) {
5012        AudioStreamOut *out = thread->clearOutput();
5013        // from now on thread->mOutput is NULL
5014        out->hwDev->close_output_stream(out->hwDev, out->stream);
5015        delete out;
5016    }
5017    return NO_ERROR;
5018}
5019
5020status_t AudioFlinger::suspendOutput(int output)
5021{
5022    Mutex::Autolock _l(mLock);
5023    PlaybackThread *thread = checkPlaybackThread_l(output);
5024
5025    if (thread == NULL) {
5026        return BAD_VALUE;
5027    }
5028
5029    ALOGV("suspendOutput() %d", output);
5030    thread->suspend();
5031
5032    return NO_ERROR;
5033}
5034
5035status_t AudioFlinger::restoreOutput(int output)
5036{
5037    Mutex::Autolock _l(mLock);
5038    PlaybackThread *thread = checkPlaybackThread_l(output);
5039
5040    if (thread == NULL) {
5041        return BAD_VALUE;
5042    }
5043
5044    ALOGV("restoreOutput() %d", output);
5045
5046    thread->restore();
5047
5048    return NO_ERROR;
5049}
5050
5051int AudioFlinger::openInput(uint32_t *pDevices,
5052                                uint32_t *pSamplingRate,
5053                                uint32_t *pFormat,
5054                                uint32_t *pChannels,
5055                                uint32_t acoustics)
5056{
5057    status_t status;
5058    RecordThread *thread = NULL;
5059    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5060    uint32_t format = pFormat ? *pFormat : 0;
5061    uint32_t channels = pChannels ? *pChannels : 0;
5062    uint32_t reqSamplingRate = samplingRate;
5063    uint32_t reqFormat = format;
5064    uint32_t reqChannels = channels;
5065    audio_stream_in_t *inStream;
5066    audio_hw_device_t *inHwDev;
5067
5068    if (pDevices == NULL || *pDevices == 0) {
5069        return 0;
5070    }
5071
5072    Mutex::Autolock _l(mLock);
5073
5074    inHwDev = findSuitableHwDev_l(*pDevices);
5075    if (inHwDev == NULL)
5076        return 0;
5077
5078    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5079                                        &channels, &samplingRate,
5080                                        (audio_in_acoustics_t)acoustics,
5081                                        &inStream);
5082    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5083            inStream,
5084            samplingRate,
5085            format,
5086            channels,
5087            acoustics,
5088            status);
5089
5090    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5091    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5092    // or stereo to mono conversions on 16 bit PCM inputs.
5093    if (inStream == NULL && status == BAD_VALUE &&
5094        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5095        (samplingRate <= 2 * reqSamplingRate) &&
5096        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5097        ALOGV("openInput() reopening with proposed sampling rate and channels");
5098        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5099                                            &channels, &samplingRate,
5100                                            (audio_in_acoustics_t)acoustics,
5101                                            &inStream);
5102    }
5103
5104    if (inStream != NULL) {
5105        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5106
5107        int id = nextUniqueId();
5108        // Start record thread
5109        // RecorThread require both input and output device indication to forward to audio
5110        // pre processing modules
5111        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5112        thread = new RecordThread(this,
5113                                  input,
5114                                  reqSamplingRate,
5115                                  reqChannels,
5116                                  id,
5117                                  device);
5118        mRecordThreads.add(id, thread);
5119        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5120        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5121        if (pFormat) *pFormat = format;
5122        if (pChannels) *pChannels = reqChannels;
5123
5124        input->stream->common.standby(&input->stream->common);
5125
5126        // notify client processes of the new input creation
5127        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5128        return id;
5129    }
5130
5131    return 0;
5132}
5133
5134status_t AudioFlinger::closeInput(int input)
5135{
5136    // keep strong reference on the record thread so that
5137    // it is not destroyed while exit() is executed
5138    sp <RecordThread> thread;
5139    {
5140        Mutex::Autolock _l(mLock);
5141        thread = checkRecordThread_l(input);
5142        if (thread == NULL) {
5143            return BAD_VALUE;
5144        }
5145
5146        ALOGV("closeInput() %d", input);
5147        void *param2 = 0;
5148        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5149        mRecordThreads.removeItem(input);
5150    }
5151    thread->exit();
5152
5153    AudioStreamIn *in = thread->clearInput();
5154    // from now on thread->mInput is NULL
5155    in->hwDev->close_input_stream(in->hwDev, in->stream);
5156    delete in;
5157
5158    return NO_ERROR;
5159}
5160
5161status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output)
5162{
5163    Mutex::Autolock _l(mLock);
5164    MixerThread *dstThread = checkMixerThread_l(output);
5165    if (dstThread == NULL) {
5166        ALOGW("setStreamOutput() bad output id %d", output);
5167        return BAD_VALUE;
5168    }
5169
5170    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5171    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5172
5173    dstThread->setStreamValid(stream, true);
5174
5175    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5176        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5177        if (thread != dstThread &&
5178            thread->type() != ThreadBase::DIRECT) {
5179            MixerThread *srcThread = (MixerThread *)thread;
5180            srcThread->setStreamValid(stream, false);
5181            srcThread->invalidateTracks(stream);
5182        }
5183    }
5184
5185    return NO_ERROR;
5186}
5187
5188
5189int AudioFlinger::newAudioSessionId()
5190{
5191    return nextUniqueId();
5192}
5193
5194void AudioFlinger::acquireAudioSessionId(int audioSession)
5195{
5196    Mutex::Autolock _l(mLock);
5197    int caller = IPCThreadState::self()->getCallingPid();
5198    ALOGV("acquiring %d from %d", audioSession, caller);
5199    int num = mAudioSessionRefs.size();
5200    for (int i = 0; i< num; i++) {
5201        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5202        if (ref->sessionid == audioSession && ref->pid == caller) {
5203            ref->cnt++;
5204            ALOGV(" incremented refcount to %d", ref->cnt);
5205            return;
5206        }
5207    }
5208    AudioSessionRef *ref = new AudioSessionRef();
5209    ref->sessionid = audioSession;
5210    ref->pid = caller;
5211    ref->cnt = 1;
5212    mAudioSessionRefs.push(ref);
5213    ALOGV(" added new entry for %d", ref->sessionid);
5214}
5215
5216void AudioFlinger::releaseAudioSessionId(int audioSession)
5217{
5218    Mutex::Autolock _l(mLock);
5219    int caller = IPCThreadState::self()->getCallingPid();
5220    ALOGV("releasing %d from %d", audioSession, caller);
5221    int num = mAudioSessionRefs.size();
5222    for (int i = 0; i< num; i++) {
5223        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5224        if (ref->sessionid == audioSession && ref->pid == caller) {
5225            ref->cnt--;
5226            ALOGV(" decremented refcount to %d", ref->cnt);
5227            if (ref->cnt == 0) {
5228                mAudioSessionRefs.removeAt(i);
5229                delete ref;
5230                purgeStaleEffects_l();
5231            }
5232            return;
5233        }
5234    }
5235    ALOGW("session id %d not found for pid %d", audioSession, caller);
5236}
5237
5238void AudioFlinger::purgeStaleEffects_l() {
5239
5240    ALOGV("purging stale effects");
5241
5242    Vector< sp<EffectChain> > chains;
5243
5244    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5245        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5246        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5247            sp<EffectChain> ec = t->mEffectChains[j];
5248            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5249                chains.push(ec);
5250            }
5251        }
5252    }
5253    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5254        sp<RecordThread> t = mRecordThreads.valueAt(i);
5255        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5256            sp<EffectChain> ec = t->mEffectChains[j];
5257            chains.push(ec);
5258        }
5259    }
5260
5261    for (size_t i = 0; i < chains.size(); i++) {
5262        sp<EffectChain> ec = chains[i];
5263        int sessionid = ec->sessionId();
5264        sp<ThreadBase> t = ec->mThread.promote();
5265        if (t == 0) {
5266            continue;
5267        }
5268        size_t numsessionrefs = mAudioSessionRefs.size();
5269        bool found = false;
5270        for (size_t k = 0; k < numsessionrefs; k++) {
5271            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5272            if (ref->sessionid == sessionid) {
5273                ALOGV(" session %d still exists for %d with %d refs",
5274                     sessionid, ref->pid, ref->cnt);
5275                found = true;
5276                break;
5277            }
5278        }
5279        if (!found) {
5280            // remove all effects from the chain
5281            while (ec->mEffects.size()) {
5282                sp<EffectModule> effect = ec->mEffects[0];
5283                effect->unPin();
5284                Mutex::Autolock _l (t->mLock);
5285                t->removeEffect_l(effect);
5286                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5287                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5288                    if (handle != 0) {
5289                        handle->mEffect.clear();
5290                        if (handle->mHasControl && handle->mEnabled) {
5291                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5292                        }
5293                    }
5294                }
5295                AudioSystem::unregisterEffect(effect->id());
5296            }
5297        }
5298    }
5299    return;
5300}
5301
5302// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5303AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5304{
5305    PlaybackThread *thread = NULL;
5306    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5307        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5308    }
5309    return thread;
5310}
5311
5312// checkMixerThread_l() must be called with AudioFlinger::mLock held
5313AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5314{
5315    PlaybackThread *thread = checkPlaybackThread_l(output);
5316    if (thread != NULL) {
5317        if (thread->type() == ThreadBase::DIRECT) {
5318            thread = NULL;
5319        }
5320    }
5321    return (MixerThread *)thread;
5322}
5323
5324// checkRecordThread_l() must be called with AudioFlinger::mLock held
5325AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5326{
5327    RecordThread *thread = NULL;
5328    if (mRecordThreads.indexOfKey(input) >= 0) {
5329        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5330    }
5331    return thread;
5332}
5333
5334uint32_t AudioFlinger::nextUniqueId()
5335{
5336    return android_atomic_inc(&mNextUniqueId);
5337}
5338
5339AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5340{
5341    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5342        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5343        AudioStreamOut *output = thread->getOutput();
5344        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5345            return thread;
5346        }
5347    }
5348    return NULL;
5349}
5350
5351uint32_t AudioFlinger::primaryOutputDevice_l()
5352{
5353    PlaybackThread *thread = primaryPlaybackThread_l();
5354
5355    if (thread == NULL) {
5356        return 0;
5357    }
5358
5359    return thread->device();
5360}
5361
5362
5363// ----------------------------------------------------------------------------
5364//  Effect management
5365// ----------------------------------------------------------------------------
5366
5367
5368status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5369{
5370    Mutex::Autolock _l(mLock);
5371    return EffectQueryNumberEffects(numEffects);
5372}
5373
5374status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5375{
5376    Mutex::Autolock _l(mLock);
5377    return EffectQueryEffect(index, descriptor);
5378}
5379
5380status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5381{
5382    Mutex::Autolock _l(mLock);
5383    return EffectGetDescriptor(pUuid, descriptor);
5384}
5385
5386
5387sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5388        effect_descriptor_t *pDesc,
5389        const sp<IEffectClient>& effectClient,
5390        int32_t priority,
5391        int io,
5392        int sessionId,
5393        status_t *status,
5394        int *id,
5395        int *enabled)
5396{
5397    status_t lStatus = NO_ERROR;
5398    sp<EffectHandle> handle;
5399    effect_descriptor_t desc;
5400    sp<Client> client;
5401    wp<Client> wclient;
5402
5403    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5404            pid, effectClient.get(), priority, sessionId, io);
5405
5406    if (pDesc == NULL) {
5407        lStatus = BAD_VALUE;
5408        goto Exit;
5409    }
5410
5411    // check audio settings permission for global effects
5412    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5413        lStatus = PERMISSION_DENIED;
5414        goto Exit;
5415    }
5416
5417    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5418    // that can only be created by audio policy manager (running in same process)
5419    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5420        lStatus = PERMISSION_DENIED;
5421        goto Exit;
5422    }
5423
5424    if (io == 0) {
5425        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5426            // output must be specified by AudioPolicyManager when using session
5427            // AUDIO_SESSION_OUTPUT_STAGE
5428            lStatus = BAD_VALUE;
5429            goto Exit;
5430        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5431            // if the output returned by getOutputForEffect() is removed before we lock the
5432            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5433            // and we will exit safely
5434            io = AudioSystem::getOutputForEffect(&desc);
5435        }
5436    }
5437
5438    {
5439        Mutex::Autolock _l(mLock);
5440
5441
5442        if (!EffectIsNullUuid(&pDesc->uuid)) {
5443            // if uuid is specified, request effect descriptor
5444            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5445            if (lStatus < 0) {
5446                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5447                goto Exit;
5448            }
5449        } else {
5450            // if uuid is not specified, look for an available implementation
5451            // of the required type in effect factory
5452            if (EffectIsNullUuid(&pDesc->type)) {
5453                ALOGW("createEffect() no effect type");
5454                lStatus = BAD_VALUE;
5455                goto Exit;
5456            }
5457            uint32_t numEffects = 0;
5458            effect_descriptor_t d;
5459            d.flags = 0; // prevent compiler warning
5460            bool found = false;
5461
5462            lStatus = EffectQueryNumberEffects(&numEffects);
5463            if (lStatus < 0) {
5464                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5465                goto Exit;
5466            }
5467            for (uint32_t i = 0; i < numEffects; i++) {
5468                lStatus = EffectQueryEffect(i, &desc);
5469                if (lStatus < 0) {
5470                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5471                    continue;
5472                }
5473                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5474                    // If matching type found save effect descriptor. If the session is
5475                    // 0 and the effect is not auxiliary, continue enumeration in case
5476                    // an auxiliary version of this effect type is available
5477                    found = true;
5478                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5479                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5480                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5481                        break;
5482                    }
5483                }
5484            }
5485            if (!found) {
5486                lStatus = BAD_VALUE;
5487                ALOGW("createEffect() effect not found");
5488                goto Exit;
5489            }
5490            // For same effect type, chose auxiliary version over insert version if
5491            // connect to output mix (Compliance to OpenSL ES)
5492            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5493                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5494                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5495            }
5496        }
5497
5498        // Do not allow auxiliary effects on a session different from 0 (output mix)
5499        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5500             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5501            lStatus = INVALID_OPERATION;
5502            goto Exit;
5503        }
5504
5505        // check recording permission for visualizer
5506        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5507            !recordingAllowed()) {
5508            lStatus = PERMISSION_DENIED;
5509            goto Exit;
5510        }
5511
5512        // return effect descriptor
5513        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5514
5515        // If output is not specified try to find a matching audio session ID in one of the
5516        // output threads.
5517        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5518        // because of code checking output when entering the function.
5519        // Note: io is never 0 when creating an effect on an input
5520        if (io == 0) {
5521             // look for the thread where the specified audio session is present
5522            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5523                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5524                    io = mPlaybackThreads.keyAt(i);
5525                    break;
5526                }
5527            }
5528            if (io == 0) {
5529               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5530                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5531                       io = mRecordThreads.keyAt(i);
5532                       break;
5533                   }
5534               }
5535            }
5536            // If no output thread contains the requested session ID, default to
5537            // first output. The effect chain will be moved to the correct output
5538            // thread when a track with the same session ID is created
5539            if (io == 0 && mPlaybackThreads.size()) {
5540                io = mPlaybackThreads.keyAt(0);
5541            }
5542            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5543        }
5544        ThreadBase *thread = checkRecordThread_l(io);
5545        if (thread == NULL) {
5546            thread = checkPlaybackThread_l(io);
5547            if (thread == NULL) {
5548                ALOGE("createEffect() unknown output thread");
5549                lStatus = BAD_VALUE;
5550                goto Exit;
5551            }
5552        }
5553
5554        wclient = mClients.valueFor(pid);
5555
5556        if (wclient != NULL) {
5557            client = wclient.promote();
5558        } else {
5559            client = new Client(this, pid);
5560            mClients.add(pid, client);
5561        }
5562
5563        // create effect on selected output thread
5564        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5565                &desc, enabled, &lStatus);
5566        if (handle != 0 && id != NULL) {
5567            *id = handle->id();
5568        }
5569    }
5570
5571Exit:
5572    if(status) {
5573        *status = lStatus;
5574    }
5575    return handle;
5576}
5577
5578status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5579{
5580    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5581            sessionId, srcOutput, dstOutput);
5582    Mutex::Autolock _l(mLock);
5583    if (srcOutput == dstOutput) {
5584        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
5585        return NO_ERROR;
5586    }
5587    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5588    if (srcThread == NULL) {
5589        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
5590        return BAD_VALUE;
5591    }
5592    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5593    if (dstThread == NULL) {
5594        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
5595        return BAD_VALUE;
5596    }
5597
5598    Mutex::Autolock _dl(dstThread->mLock);
5599    Mutex::Autolock _sl(srcThread->mLock);
5600    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5601
5602    return NO_ERROR;
5603}
5604
5605// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5606status_t AudioFlinger::moveEffectChain_l(int sessionId,
5607                                   AudioFlinger::PlaybackThread *srcThread,
5608                                   AudioFlinger::PlaybackThread *dstThread,
5609                                   bool reRegister)
5610{
5611    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5612            sessionId, srcThread, dstThread);
5613
5614    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5615    if (chain == 0) {
5616        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5617                sessionId, srcThread);
5618        return INVALID_OPERATION;
5619    }
5620
5621    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5622    // so that a new chain is created with correct parameters when first effect is added. This is
5623    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5624    // removed.
5625    srcThread->removeEffectChain_l(chain);
5626
5627    // transfer all effects one by one so that new effect chain is created on new thread with
5628    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5629    int dstOutput = dstThread->id();
5630    sp<EffectChain> dstChain;
5631    uint32_t strategy = 0; // prevent compiler warning
5632    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5633    while (effect != 0) {
5634        srcThread->removeEffect_l(effect);
5635        dstThread->addEffect_l(effect);
5636        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5637        if (effect->state() == EffectModule::ACTIVE ||
5638                effect->state() == EffectModule::STOPPING) {
5639            effect->start();
5640        }
5641        // if the move request is not received from audio policy manager, the effect must be
5642        // re-registered with the new strategy and output
5643        if (dstChain == 0) {
5644            dstChain = effect->chain().promote();
5645            if (dstChain == 0) {
5646                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5647                srcThread->addEffect_l(effect);
5648                return NO_INIT;
5649            }
5650            strategy = dstChain->strategy();
5651        }
5652        if (reRegister) {
5653            AudioSystem::unregisterEffect(effect->id());
5654            AudioSystem::registerEffect(&effect->desc(),
5655                                        dstOutput,
5656                                        strategy,
5657                                        sessionId,
5658                                        effect->id());
5659        }
5660        effect = chain->getEffectFromId_l(0);
5661    }
5662
5663    return NO_ERROR;
5664}
5665
5666
5667// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5668sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5669        const sp<AudioFlinger::Client>& client,
5670        const sp<IEffectClient>& effectClient,
5671        int32_t priority,
5672        int sessionId,
5673        effect_descriptor_t *desc,
5674        int *enabled,
5675        status_t *status
5676        )
5677{
5678    sp<EffectModule> effect;
5679    sp<EffectHandle> handle;
5680    status_t lStatus;
5681    sp<EffectChain> chain;
5682    bool chainCreated = false;
5683    bool effectCreated = false;
5684    bool effectRegistered = false;
5685
5686    lStatus = initCheck();
5687    if (lStatus != NO_ERROR) {
5688        ALOGW("createEffect_l() Audio driver not initialized.");
5689        goto Exit;
5690    }
5691
5692    // Do not allow effects with session ID 0 on direct output or duplicating threads
5693    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5694    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5695        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5696                desc->name, sessionId);
5697        lStatus = BAD_VALUE;
5698        goto Exit;
5699    }
5700    // Only Pre processor effects are allowed on input threads and only on input threads
5701    if ((mType == RECORD &&
5702            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5703            (mType != RECORD &&
5704                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5705        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5706                desc->name, desc->flags, mType);
5707        lStatus = BAD_VALUE;
5708        goto Exit;
5709    }
5710
5711    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5712
5713    { // scope for mLock
5714        Mutex::Autolock _l(mLock);
5715
5716        // check for existing effect chain with the requested audio session
5717        chain = getEffectChain_l(sessionId);
5718        if (chain == 0) {
5719            // create a new chain for this session
5720            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5721            chain = new EffectChain(this, sessionId);
5722            addEffectChain_l(chain);
5723            chain->setStrategy(getStrategyForSession_l(sessionId));
5724            chainCreated = true;
5725        } else {
5726            effect = chain->getEffectFromDesc_l(desc);
5727        }
5728
5729        ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
5730
5731        if (effect == 0) {
5732            int id = mAudioFlinger->nextUniqueId();
5733            // Check CPU and memory usage
5734            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5735            if (lStatus != NO_ERROR) {
5736                goto Exit;
5737            }
5738            effectRegistered = true;
5739            // create a new effect module if none present in the chain
5740            effect = new EffectModule(this, chain, desc, id, sessionId);
5741            lStatus = effect->status();
5742            if (lStatus != NO_ERROR) {
5743                goto Exit;
5744            }
5745            lStatus = chain->addEffect_l(effect);
5746            if (lStatus != NO_ERROR) {
5747                goto Exit;
5748            }
5749            effectCreated = true;
5750
5751            effect->setDevice(mDevice);
5752            effect->setMode(mAudioFlinger->getMode());
5753        }
5754        // create effect handle and connect it to effect module
5755        handle = new EffectHandle(effect, client, effectClient, priority);
5756        lStatus = effect->addHandle(handle);
5757        if (enabled) {
5758            *enabled = (int)effect->isEnabled();
5759        }
5760    }
5761
5762Exit:
5763    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5764        Mutex::Autolock _l(mLock);
5765        if (effectCreated) {
5766            chain->removeEffect_l(effect);
5767        }
5768        if (effectRegistered) {
5769            AudioSystem::unregisterEffect(effect->id());
5770        }
5771        if (chainCreated) {
5772            removeEffectChain_l(chain);
5773        }
5774        handle.clear();
5775    }
5776
5777    if(status) {
5778        *status = lStatus;
5779    }
5780    return handle;
5781}
5782
5783sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5784{
5785    sp<EffectModule> effect;
5786
5787    sp<EffectChain> chain = getEffectChain_l(sessionId);
5788    if (chain != 0) {
5789        effect = chain->getEffectFromId_l(effectId);
5790    }
5791    return effect;
5792}
5793
5794// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5795// PlaybackThread::mLock held
5796status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5797{
5798    // check for existing effect chain with the requested audio session
5799    int sessionId = effect->sessionId();
5800    sp<EffectChain> chain = getEffectChain_l(sessionId);
5801    bool chainCreated = false;
5802
5803    if (chain == 0) {
5804        // create a new chain for this session
5805        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5806        chain = new EffectChain(this, sessionId);
5807        addEffectChain_l(chain);
5808        chain->setStrategy(getStrategyForSession_l(sessionId));
5809        chainCreated = true;
5810    }
5811    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5812
5813    if (chain->getEffectFromId_l(effect->id()) != 0) {
5814        ALOGW("addEffect_l() %p effect %s already present in chain %p",
5815                this, effect->desc().name, chain.get());
5816        return BAD_VALUE;
5817    }
5818
5819    status_t status = chain->addEffect_l(effect);
5820    if (status != NO_ERROR) {
5821        if (chainCreated) {
5822            removeEffectChain_l(chain);
5823        }
5824        return status;
5825    }
5826
5827    effect->setDevice(mDevice);
5828    effect->setMode(mAudioFlinger->getMode());
5829    return NO_ERROR;
5830}
5831
5832void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5833
5834    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5835    effect_descriptor_t desc = effect->desc();
5836    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5837        detachAuxEffect_l(effect->id());
5838    }
5839
5840    sp<EffectChain> chain = effect->chain().promote();
5841    if (chain != 0) {
5842        // remove effect chain if removing last effect
5843        if (chain->removeEffect_l(effect) == 0) {
5844            removeEffectChain_l(chain);
5845        }
5846    } else {
5847        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5848    }
5849}
5850
5851void AudioFlinger::ThreadBase::lockEffectChains_l(
5852        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5853{
5854    effectChains = mEffectChains;
5855    for (size_t i = 0; i < mEffectChains.size(); i++) {
5856        mEffectChains[i]->lock();
5857    }
5858}
5859
5860void AudioFlinger::ThreadBase::unlockEffectChains(
5861        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5862{
5863    for (size_t i = 0; i < effectChains.size(); i++) {
5864        effectChains[i]->unlock();
5865    }
5866}
5867
5868sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5869{
5870    Mutex::Autolock _l(mLock);
5871    return getEffectChain_l(sessionId);
5872}
5873
5874sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5875{
5876    sp<EffectChain> chain;
5877
5878    size_t size = mEffectChains.size();
5879    for (size_t i = 0; i < size; i++) {
5880        if (mEffectChains[i]->sessionId() == sessionId) {
5881            chain = mEffectChains[i];
5882            break;
5883        }
5884    }
5885    return chain;
5886}
5887
5888void AudioFlinger::ThreadBase::setMode(uint32_t mode)
5889{
5890    Mutex::Autolock _l(mLock);
5891    size_t size = mEffectChains.size();
5892    for (size_t i = 0; i < size; i++) {
5893        mEffectChains[i]->setMode_l(mode);
5894    }
5895}
5896
5897void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5898                                                    const wp<EffectHandle>& handle,
5899                                                    bool unpiniflast) {
5900
5901    Mutex::Autolock _l(mLock);
5902    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5903    // delete the effect module if removing last handle on it
5904    if (effect->removeHandle(handle) == 0) {
5905        if (!effect->isPinned() || unpiniflast) {
5906            removeEffect_l(effect);
5907            AudioSystem::unregisterEffect(effect->id());
5908        }
5909    }
5910}
5911
5912status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5913{
5914    int session = chain->sessionId();
5915    int16_t *buffer = mMixBuffer;
5916    bool ownsBuffer = false;
5917
5918    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5919    if (session > 0) {
5920        // Only one effect chain can be present in direct output thread and it uses
5921        // the mix buffer as input
5922        if (mType != DIRECT) {
5923            size_t numSamples = mFrameCount * mChannelCount;
5924            buffer = new int16_t[numSamples];
5925            memset(buffer, 0, numSamples * sizeof(int16_t));
5926            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5927            ownsBuffer = true;
5928        }
5929
5930        // Attach all tracks with same session ID to this chain.
5931        for (size_t i = 0; i < mTracks.size(); ++i) {
5932            sp<Track> track = mTracks[i];
5933            if (session == track->sessionId()) {
5934                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5935                track->setMainBuffer(buffer);
5936                chain->incTrackCnt();
5937            }
5938        }
5939
5940        // indicate all active tracks in the chain
5941        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5942            sp<Track> track = mActiveTracks[i].promote();
5943            if (track == 0) continue;
5944            if (session == track->sessionId()) {
5945                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5946                chain->incActiveTrackCnt();
5947            }
5948        }
5949    }
5950
5951    chain->setInBuffer(buffer, ownsBuffer);
5952    chain->setOutBuffer(mMixBuffer);
5953    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5954    // chains list in order to be processed last as it contains output stage effects
5955    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5956    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5957    // after track specific effects and before output stage
5958    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5959    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5960    // Effect chain for other sessions are inserted at beginning of effect
5961    // chains list to be processed before output mix effects. Relative order between other
5962    // sessions is not important
5963    size_t size = mEffectChains.size();
5964    size_t i = 0;
5965    for (i = 0; i < size; i++) {
5966        if (mEffectChains[i]->sessionId() < session) break;
5967    }
5968    mEffectChains.insertAt(chain, i);
5969    checkSuspendOnAddEffectChain_l(chain);
5970
5971    return NO_ERROR;
5972}
5973
5974size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
5975{
5976    int session = chain->sessionId();
5977
5978    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
5979
5980    for (size_t i = 0; i < mEffectChains.size(); i++) {
5981        if (chain == mEffectChains[i]) {
5982            mEffectChains.removeAt(i);
5983            // detach all active tracks from the chain
5984            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5985                sp<Track> track = mActiveTracks[i].promote();
5986                if (track == 0) continue;
5987                if (session == track->sessionId()) {
5988                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
5989                            chain.get(), session);
5990                    chain->decActiveTrackCnt();
5991                }
5992            }
5993
5994            // detach all tracks with same session ID from this chain
5995            for (size_t i = 0; i < mTracks.size(); ++i) {
5996                sp<Track> track = mTracks[i];
5997                if (session == track->sessionId()) {
5998                    track->setMainBuffer(mMixBuffer);
5999                    chain->decTrackCnt();
6000                }
6001            }
6002            break;
6003        }
6004    }
6005    return mEffectChains.size();
6006}
6007
6008status_t AudioFlinger::PlaybackThread::attachAuxEffect(
6009        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6010{
6011    Mutex::Autolock _l(mLock);
6012    return attachAuxEffect_l(track, EffectId);
6013}
6014
6015status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6016        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6017{
6018    status_t status = NO_ERROR;
6019
6020    if (EffectId == 0) {
6021        track->setAuxBuffer(0, NULL);
6022    } else {
6023        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6024        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6025        if (effect != 0) {
6026            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6027                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6028            } else {
6029                status = INVALID_OPERATION;
6030            }
6031        } else {
6032            status = BAD_VALUE;
6033        }
6034    }
6035    return status;
6036}
6037
6038void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6039{
6040     for (size_t i = 0; i < mTracks.size(); ++i) {
6041        sp<Track> track = mTracks[i];
6042        if (track->auxEffectId() == effectId) {
6043            attachAuxEffect_l(track, 0);
6044        }
6045    }
6046}
6047
6048status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6049{
6050    // only one chain per input thread
6051    if (mEffectChains.size() != 0) {
6052        return INVALID_OPERATION;
6053    }
6054    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6055
6056    chain->setInBuffer(NULL);
6057    chain->setOutBuffer(NULL);
6058
6059    checkSuspendOnAddEffectChain_l(chain);
6060
6061    mEffectChains.add(chain);
6062
6063    return NO_ERROR;
6064}
6065
6066size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6067{
6068    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6069    ALOGW_IF(mEffectChains.size() != 1,
6070            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6071            chain.get(), mEffectChains.size(), this);
6072    if (mEffectChains.size() == 1) {
6073        mEffectChains.removeAt(0);
6074    }
6075    return 0;
6076}
6077
6078// ----------------------------------------------------------------------------
6079//  EffectModule implementation
6080// ----------------------------------------------------------------------------
6081
6082#undef LOG_TAG
6083#define LOG_TAG "AudioFlinger::EffectModule"
6084
6085AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6086                                        const wp<AudioFlinger::EffectChain>& chain,
6087                                        effect_descriptor_t *desc,
6088                                        int id,
6089                                        int sessionId)
6090    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6091      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6092{
6093    ALOGV("Constructor %p", this);
6094    int lStatus;
6095    sp<ThreadBase> thread = mThread.promote();
6096    if (thread == 0) {
6097        return;
6098    }
6099
6100    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6101
6102    // create effect engine from effect factory
6103    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6104
6105    if (mStatus != NO_ERROR) {
6106        return;
6107    }
6108    lStatus = init();
6109    if (lStatus < 0) {
6110        mStatus = lStatus;
6111        goto Error;
6112    }
6113
6114    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6115        mPinned = true;
6116    }
6117    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6118    return;
6119Error:
6120    EffectRelease(mEffectInterface);
6121    mEffectInterface = NULL;
6122    ALOGV("Constructor Error %d", mStatus);
6123}
6124
6125AudioFlinger::EffectModule::~EffectModule()
6126{
6127    ALOGV("Destructor %p", this);
6128    if (mEffectInterface != NULL) {
6129        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6130                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6131            sp<ThreadBase> thread = mThread.promote();
6132            if (thread != 0) {
6133                audio_stream_t *stream = thread->stream();
6134                if (stream != NULL) {
6135                    stream->remove_audio_effect(stream, mEffectInterface);
6136                }
6137            }
6138        }
6139        // release effect engine
6140        EffectRelease(mEffectInterface);
6141    }
6142}
6143
6144status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6145{
6146    status_t status;
6147
6148    Mutex::Autolock _l(mLock);
6149    // First handle in mHandles has highest priority and controls the effect module
6150    int priority = handle->priority();
6151    size_t size = mHandles.size();
6152    sp<EffectHandle> h;
6153    size_t i;
6154    for (i = 0; i < size; i++) {
6155        h = mHandles[i].promote();
6156        if (h == 0) continue;
6157        if (h->priority() <= priority) break;
6158    }
6159    // if inserted in first place, move effect control from previous owner to this handle
6160    if (i == 0) {
6161        bool enabled = false;
6162        if (h != 0) {
6163            enabled = h->enabled();
6164            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6165        }
6166        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6167        status = NO_ERROR;
6168    } else {
6169        status = ALREADY_EXISTS;
6170    }
6171    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6172    mHandles.insertAt(handle, i);
6173    return status;
6174}
6175
6176size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6177{
6178    Mutex::Autolock _l(mLock);
6179    size_t size = mHandles.size();
6180    size_t i;
6181    for (i = 0; i < size; i++) {
6182        if (mHandles[i] == handle) break;
6183    }
6184    if (i == size) {
6185        return size;
6186    }
6187    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6188
6189    bool enabled = false;
6190    EffectHandle *hdl = handle.unsafe_get();
6191    if (hdl) {
6192        ALOGV("removeHandle() unsafe_get OK");
6193        enabled = hdl->enabled();
6194    }
6195    mHandles.removeAt(i);
6196    size = mHandles.size();
6197    // if removed from first place, move effect control from this handle to next in line
6198    if (i == 0 && size != 0) {
6199        sp<EffectHandle> h = mHandles[0].promote();
6200        if (h != 0) {
6201            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6202        }
6203    }
6204
6205    // Prevent calls to process() and other functions on effect interface from now on.
6206    // The effect engine will be released by the destructor when the last strong reference on
6207    // this object is released which can happen after next process is called.
6208    if (size == 0 && !mPinned) {
6209        mState = DESTROYED;
6210    }
6211
6212    return size;
6213}
6214
6215sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6216{
6217    Mutex::Autolock _l(mLock);
6218    sp<EffectHandle> handle;
6219    if (mHandles.size() != 0) {
6220        handle = mHandles[0].promote();
6221    }
6222    return handle;
6223}
6224
6225void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6226{
6227    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6228    // keep a strong reference on this EffectModule to avoid calling the
6229    // destructor before we exit
6230    sp<EffectModule> keep(this);
6231    {
6232        sp<ThreadBase> thread = mThread.promote();
6233        if (thread != 0) {
6234            thread->disconnectEffect(keep, handle, unpiniflast);
6235        }
6236    }
6237}
6238
6239void AudioFlinger::EffectModule::updateState() {
6240    Mutex::Autolock _l(mLock);
6241
6242    switch (mState) {
6243    case RESTART:
6244        reset_l();
6245        // FALL THROUGH
6246
6247    case STARTING:
6248        // clear auxiliary effect input buffer for next accumulation
6249        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6250            memset(mConfig.inputCfg.buffer.raw,
6251                   0,
6252                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6253        }
6254        start_l();
6255        mState = ACTIVE;
6256        break;
6257    case STOPPING:
6258        stop_l();
6259        mDisableWaitCnt = mMaxDisableWaitCnt;
6260        mState = STOPPED;
6261        break;
6262    case STOPPED:
6263        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6264        // turn off sequence.
6265        if (--mDisableWaitCnt == 0) {
6266            reset_l();
6267            mState = IDLE;
6268        }
6269        break;
6270    default: //IDLE , ACTIVE, DESTROYED
6271        break;
6272    }
6273}
6274
6275void AudioFlinger::EffectModule::process()
6276{
6277    Mutex::Autolock _l(mLock);
6278
6279    if (mState == DESTROYED || mEffectInterface == NULL ||
6280            mConfig.inputCfg.buffer.raw == NULL ||
6281            mConfig.outputCfg.buffer.raw == NULL) {
6282        return;
6283    }
6284
6285    if (isProcessEnabled()) {
6286        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6287        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6288            ditherAndClamp(mConfig.inputCfg.buffer.s32,
6289                                        mConfig.inputCfg.buffer.s32,
6290                                        mConfig.inputCfg.buffer.frameCount/2);
6291        }
6292
6293        // do the actual processing in the effect engine
6294        int ret = (*mEffectInterface)->process(mEffectInterface,
6295                                               &mConfig.inputCfg.buffer,
6296                                               &mConfig.outputCfg.buffer);
6297
6298        // force transition to IDLE state when engine is ready
6299        if (mState == STOPPED && ret == -ENODATA) {
6300            mDisableWaitCnt = 1;
6301        }
6302
6303        // clear auxiliary effect input buffer for next accumulation
6304        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6305            memset(mConfig.inputCfg.buffer.raw, 0,
6306                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6307        }
6308    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6309                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6310        // If an insert effect is idle and input buffer is different from output buffer,
6311        // accumulate input onto output
6312        sp<EffectChain> chain = mChain.promote();
6313        if (chain != 0 && chain->activeTrackCnt() != 0) {
6314            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6315            int16_t *in = mConfig.inputCfg.buffer.s16;
6316            int16_t *out = mConfig.outputCfg.buffer.s16;
6317            for (size_t i = 0; i < frameCnt; i++) {
6318                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6319            }
6320        }
6321    }
6322}
6323
6324void AudioFlinger::EffectModule::reset_l()
6325{
6326    if (mEffectInterface == NULL) {
6327        return;
6328    }
6329    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6330}
6331
6332status_t AudioFlinger::EffectModule::configure()
6333{
6334    uint32_t channels;
6335    if (mEffectInterface == NULL) {
6336        return NO_INIT;
6337    }
6338
6339    sp<ThreadBase> thread = mThread.promote();
6340    if (thread == 0) {
6341        return DEAD_OBJECT;
6342    }
6343
6344    // TODO: handle configuration of effects replacing track process
6345    if (thread->channelCount() == 1) {
6346        channels = AUDIO_CHANNEL_OUT_MONO;
6347    } else {
6348        channels = AUDIO_CHANNEL_OUT_STEREO;
6349    }
6350
6351    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6352        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6353    } else {
6354        mConfig.inputCfg.channels = channels;
6355    }
6356    mConfig.outputCfg.channels = channels;
6357    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6358    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6359    mConfig.inputCfg.samplingRate = thread->sampleRate();
6360    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6361    mConfig.inputCfg.bufferProvider.cookie = NULL;
6362    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6363    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6364    mConfig.outputCfg.bufferProvider.cookie = NULL;
6365    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6366    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6367    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6368    // Insert effect:
6369    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6370    // always overwrites output buffer: input buffer == output buffer
6371    // - in other sessions:
6372    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6373    //      other effect: overwrites output buffer: input buffer == output buffer
6374    // Auxiliary effect:
6375    //      accumulates in output buffer: input buffer != output buffer
6376    // Therefore: accumulate <=> input buffer != output buffer
6377    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6378        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6379    } else {
6380        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6381    }
6382    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6383    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6384    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6385    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6386
6387    ALOGV("configure() %p thread %p buffer %p framecount %d",
6388            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6389
6390    status_t cmdStatus;
6391    uint32_t size = sizeof(int);
6392    status_t status = (*mEffectInterface)->command(mEffectInterface,
6393                                                   EFFECT_CMD_SET_CONFIG,
6394                                                   sizeof(effect_config_t),
6395                                                   &mConfig,
6396                                                   &size,
6397                                                   &cmdStatus);
6398    if (status == 0) {
6399        status = cmdStatus;
6400    }
6401
6402    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6403            (1000 * mConfig.outputCfg.buffer.frameCount);
6404
6405    return status;
6406}
6407
6408status_t AudioFlinger::EffectModule::init()
6409{
6410    Mutex::Autolock _l(mLock);
6411    if (mEffectInterface == NULL) {
6412        return NO_INIT;
6413    }
6414    status_t cmdStatus;
6415    uint32_t size = sizeof(status_t);
6416    status_t status = (*mEffectInterface)->command(mEffectInterface,
6417                                                   EFFECT_CMD_INIT,
6418                                                   0,
6419                                                   NULL,
6420                                                   &size,
6421                                                   &cmdStatus);
6422    if (status == 0) {
6423        status = cmdStatus;
6424    }
6425    return status;
6426}
6427
6428status_t AudioFlinger::EffectModule::start()
6429{
6430    Mutex::Autolock _l(mLock);
6431    return start_l();
6432}
6433
6434status_t AudioFlinger::EffectModule::start_l()
6435{
6436    if (mEffectInterface == NULL) {
6437        return NO_INIT;
6438    }
6439    status_t cmdStatus;
6440    uint32_t size = sizeof(status_t);
6441    status_t status = (*mEffectInterface)->command(mEffectInterface,
6442                                                   EFFECT_CMD_ENABLE,
6443                                                   0,
6444                                                   NULL,
6445                                                   &size,
6446                                                   &cmdStatus);
6447    if (status == 0) {
6448        status = cmdStatus;
6449    }
6450    if (status == 0 &&
6451            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6452             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6453        sp<ThreadBase> thread = mThread.promote();
6454        if (thread != 0) {
6455            audio_stream_t *stream = thread->stream();
6456            if (stream != NULL) {
6457                stream->add_audio_effect(stream, mEffectInterface);
6458            }
6459        }
6460    }
6461    return status;
6462}
6463
6464status_t AudioFlinger::EffectModule::stop()
6465{
6466    Mutex::Autolock _l(mLock);
6467    return stop_l();
6468}
6469
6470status_t AudioFlinger::EffectModule::stop_l()
6471{
6472    if (mEffectInterface == NULL) {
6473        return NO_INIT;
6474    }
6475    status_t cmdStatus;
6476    uint32_t size = sizeof(status_t);
6477    status_t status = (*mEffectInterface)->command(mEffectInterface,
6478                                                   EFFECT_CMD_DISABLE,
6479                                                   0,
6480                                                   NULL,
6481                                                   &size,
6482                                                   &cmdStatus);
6483    if (status == 0) {
6484        status = cmdStatus;
6485    }
6486    if (status == 0 &&
6487            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6488             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6489        sp<ThreadBase> thread = mThread.promote();
6490        if (thread != 0) {
6491            audio_stream_t *stream = thread->stream();
6492            if (stream != NULL) {
6493                stream->remove_audio_effect(stream, mEffectInterface);
6494            }
6495        }
6496    }
6497    return status;
6498}
6499
6500status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6501                                             uint32_t cmdSize,
6502                                             void *pCmdData,
6503                                             uint32_t *replySize,
6504                                             void *pReplyData)
6505{
6506    Mutex::Autolock _l(mLock);
6507//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6508
6509    if (mState == DESTROYED || mEffectInterface == NULL) {
6510        return NO_INIT;
6511    }
6512    status_t status = (*mEffectInterface)->command(mEffectInterface,
6513                                                   cmdCode,
6514                                                   cmdSize,
6515                                                   pCmdData,
6516                                                   replySize,
6517                                                   pReplyData);
6518    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6519        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6520        for (size_t i = 1; i < mHandles.size(); i++) {
6521            sp<EffectHandle> h = mHandles[i].promote();
6522            if (h != 0) {
6523                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6524            }
6525        }
6526    }
6527    return status;
6528}
6529
6530status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6531{
6532
6533    Mutex::Autolock _l(mLock);
6534    ALOGV("setEnabled %p enabled %d", this, enabled);
6535
6536    if (enabled != isEnabled()) {
6537        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6538        if (enabled && status != NO_ERROR) {
6539            return status;
6540        }
6541
6542        switch (mState) {
6543        // going from disabled to enabled
6544        case IDLE:
6545            mState = STARTING;
6546            break;
6547        case STOPPED:
6548            mState = RESTART;
6549            break;
6550        case STOPPING:
6551            mState = ACTIVE;
6552            break;
6553
6554        // going from enabled to disabled
6555        case RESTART:
6556            mState = STOPPED;
6557            break;
6558        case STARTING:
6559            mState = IDLE;
6560            break;
6561        case ACTIVE:
6562            mState = STOPPING;
6563            break;
6564        case DESTROYED:
6565            return NO_ERROR; // simply ignore as we are being destroyed
6566        }
6567        for (size_t i = 1; i < mHandles.size(); i++) {
6568            sp<EffectHandle> h = mHandles[i].promote();
6569            if (h != 0) {
6570                h->setEnabled(enabled);
6571            }
6572        }
6573    }
6574    return NO_ERROR;
6575}
6576
6577bool AudioFlinger::EffectModule::isEnabled()
6578{
6579    switch (mState) {
6580    case RESTART:
6581    case STARTING:
6582    case ACTIVE:
6583        return true;
6584    case IDLE:
6585    case STOPPING:
6586    case STOPPED:
6587    case DESTROYED:
6588    default:
6589        return false;
6590    }
6591}
6592
6593bool AudioFlinger::EffectModule::isProcessEnabled()
6594{
6595    switch (mState) {
6596    case RESTART:
6597    case ACTIVE:
6598    case STOPPING:
6599    case STOPPED:
6600        return true;
6601    case IDLE:
6602    case STARTING:
6603    case DESTROYED:
6604    default:
6605        return false;
6606    }
6607}
6608
6609status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6610{
6611    Mutex::Autolock _l(mLock);
6612    status_t status = NO_ERROR;
6613
6614    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6615    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6616    if (isProcessEnabled() &&
6617            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6618            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6619        status_t cmdStatus;
6620        uint32_t volume[2];
6621        uint32_t *pVolume = NULL;
6622        uint32_t size = sizeof(volume);
6623        volume[0] = *left;
6624        volume[1] = *right;
6625        if (controller) {
6626            pVolume = volume;
6627        }
6628        status = (*mEffectInterface)->command(mEffectInterface,
6629                                              EFFECT_CMD_SET_VOLUME,
6630                                              size,
6631                                              volume,
6632                                              &size,
6633                                              pVolume);
6634        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6635            *left = volume[0];
6636            *right = volume[1];
6637        }
6638    }
6639    return status;
6640}
6641
6642status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6643{
6644    Mutex::Autolock _l(mLock);
6645    status_t status = NO_ERROR;
6646    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6647        // audio pre processing modules on RecordThread can receive both output and
6648        // input device indication in the same call
6649        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6650        if (dev) {
6651            status_t cmdStatus;
6652            uint32_t size = sizeof(status_t);
6653
6654            status = (*mEffectInterface)->command(mEffectInterface,
6655                                                  EFFECT_CMD_SET_DEVICE,
6656                                                  sizeof(uint32_t),
6657                                                  &dev,
6658                                                  &size,
6659                                                  &cmdStatus);
6660            if (status == NO_ERROR) {
6661                status = cmdStatus;
6662            }
6663        }
6664        dev = device & AUDIO_DEVICE_IN_ALL;
6665        if (dev) {
6666            status_t cmdStatus;
6667            uint32_t size = sizeof(status_t);
6668
6669            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6670                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6671                                                  sizeof(uint32_t),
6672                                                  &dev,
6673                                                  &size,
6674                                                  &cmdStatus);
6675            if (status2 == NO_ERROR) {
6676                status2 = cmdStatus;
6677            }
6678            if (status == NO_ERROR) {
6679                status = status2;
6680            }
6681        }
6682    }
6683    return status;
6684}
6685
6686status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
6687{
6688    Mutex::Autolock _l(mLock);
6689    status_t status = NO_ERROR;
6690    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6691        status_t cmdStatus;
6692        uint32_t size = sizeof(status_t);
6693        status = (*mEffectInterface)->command(mEffectInterface,
6694                                              EFFECT_CMD_SET_AUDIO_MODE,
6695                                              sizeof(int),
6696                                              &mode,
6697                                              &size,
6698                                              &cmdStatus);
6699        if (status == NO_ERROR) {
6700            status = cmdStatus;
6701        }
6702    }
6703    return status;
6704}
6705
6706void AudioFlinger::EffectModule::setSuspended(bool suspended)
6707{
6708    Mutex::Autolock _l(mLock);
6709    mSuspended = suspended;
6710}
6711
6712bool AudioFlinger::EffectModule::suspended() const
6713{
6714    Mutex::Autolock _l(mLock);
6715    return mSuspended;
6716}
6717
6718status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6719{
6720    const size_t SIZE = 256;
6721    char buffer[SIZE];
6722    String8 result;
6723
6724    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6725    result.append(buffer);
6726
6727    bool locked = tryLock(mLock);
6728    // failed to lock - AudioFlinger is probably deadlocked
6729    if (!locked) {
6730        result.append("\t\tCould not lock Fx mutex:\n");
6731    }
6732
6733    result.append("\t\tSession Status State Engine:\n");
6734    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6735            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6736    result.append(buffer);
6737
6738    result.append("\t\tDescriptor:\n");
6739    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6740            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6741            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6742            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6743    result.append(buffer);
6744    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6745                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6746                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6747                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6748    result.append(buffer);
6749    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6750            mDescriptor.apiVersion,
6751            mDescriptor.flags);
6752    result.append(buffer);
6753    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6754            mDescriptor.name);
6755    result.append(buffer);
6756    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6757            mDescriptor.implementor);
6758    result.append(buffer);
6759
6760    result.append("\t\t- Input configuration:\n");
6761    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6762    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6763            (uint32_t)mConfig.inputCfg.buffer.raw,
6764            mConfig.inputCfg.buffer.frameCount,
6765            mConfig.inputCfg.samplingRate,
6766            mConfig.inputCfg.channels,
6767            mConfig.inputCfg.format);
6768    result.append(buffer);
6769
6770    result.append("\t\t- Output configuration:\n");
6771    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6772    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6773            (uint32_t)mConfig.outputCfg.buffer.raw,
6774            mConfig.outputCfg.buffer.frameCount,
6775            mConfig.outputCfg.samplingRate,
6776            mConfig.outputCfg.channels,
6777            mConfig.outputCfg.format);
6778    result.append(buffer);
6779
6780    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6781    result.append(buffer);
6782    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6783    for (size_t i = 0; i < mHandles.size(); ++i) {
6784        sp<EffectHandle> handle = mHandles[i].promote();
6785        if (handle != 0) {
6786            handle->dump(buffer, SIZE);
6787            result.append(buffer);
6788        }
6789    }
6790
6791    result.append("\n");
6792
6793    write(fd, result.string(), result.length());
6794
6795    if (locked) {
6796        mLock.unlock();
6797    }
6798
6799    return NO_ERROR;
6800}
6801
6802// ----------------------------------------------------------------------------
6803//  EffectHandle implementation
6804// ----------------------------------------------------------------------------
6805
6806#undef LOG_TAG
6807#define LOG_TAG "AudioFlinger::EffectHandle"
6808
6809AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6810                                        const sp<AudioFlinger::Client>& client,
6811                                        const sp<IEffectClient>& effectClient,
6812                                        int32_t priority)
6813    : BnEffect(),
6814    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6815    mPriority(priority), mHasControl(false), mEnabled(false)
6816{
6817    ALOGV("constructor %p", this);
6818
6819    if (client == 0) {
6820        return;
6821    }
6822    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6823    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6824    if (mCblkMemory != 0) {
6825        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6826
6827        if (mCblk) {
6828            new(mCblk) effect_param_cblk_t();
6829            mBuffer = (uint8_t *)mCblk + bufOffset;
6830         }
6831    } else {
6832        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6833        return;
6834    }
6835}
6836
6837AudioFlinger::EffectHandle::~EffectHandle()
6838{
6839    ALOGV("Destructor %p", this);
6840    disconnect(false);
6841    ALOGV("Destructor DONE %p", this);
6842}
6843
6844status_t AudioFlinger::EffectHandle::enable()
6845{
6846    ALOGV("enable %p", this);
6847    if (!mHasControl) return INVALID_OPERATION;
6848    if (mEffect == 0) return DEAD_OBJECT;
6849
6850    if (mEnabled) {
6851        return NO_ERROR;
6852    }
6853
6854    mEnabled = true;
6855
6856    sp<ThreadBase> thread = mEffect->thread().promote();
6857    if (thread != 0) {
6858        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6859    }
6860
6861    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6862    if (mEffect->suspended()) {
6863        return NO_ERROR;
6864    }
6865
6866    status_t status = mEffect->setEnabled(true);
6867    if (status != NO_ERROR) {
6868        if (thread != 0) {
6869            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6870        }
6871        mEnabled = false;
6872    }
6873    return status;
6874}
6875
6876status_t AudioFlinger::EffectHandle::disable()
6877{
6878    ALOGV("disable %p", this);
6879    if (!mHasControl) return INVALID_OPERATION;
6880    if (mEffect == 0) return DEAD_OBJECT;
6881
6882    if (!mEnabled) {
6883        return NO_ERROR;
6884    }
6885    mEnabled = false;
6886
6887    if (mEffect->suspended()) {
6888        return NO_ERROR;
6889    }
6890
6891    status_t status = mEffect->setEnabled(false);
6892
6893    sp<ThreadBase> thread = mEffect->thread().promote();
6894    if (thread != 0) {
6895        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6896    }
6897
6898    return status;
6899}
6900
6901void AudioFlinger::EffectHandle::disconnect()
6902{
6903    disconnect(true);
6904}
6905
6906void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6907{
6908    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6909    if (mEffect == 0) {
6910        return;
6911    }
6912    mEffect->disconnect(this, unpiniflast);
6913
6914    if (mHasControl && mEnabled) {
6915        sp<ThreadBase> thread = mEffect->thread().promote();
6916        if (thread != 0) {
6917            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6918        }
6919    }
6920
6921    // release sp on module => module destructor can be called now
6922    mEffect.clear();
6923    if (mClient != 0) {
6924        if (mCblk) {
6925            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6926        }
6927        mCblkMemory.clear();            // and free the shared memory
6928        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6929        mClient.clear();
6930    }
6931}
6932
6933status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6934                                             uint32_t cmdSize,
6935                                             void *pCmdData,
6936                                             uint32_t *replySize,
6937                                             void *pReplyData)
6938{
6939//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6940//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6941
6942    // only get parameter command is permitted for applications not controlling the effect
6943    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6944        return INVALID_OPERATION;
6945    }
6946    if (mEffect == 0) return DEAD_OBJECT;
6947    if (mClient == 0) return INVALID_OPERATION;
6948
6949    // handle commands that are not forwarded transparently to effect engine
6950    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6951        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6952        // no risk to block the whole media server process or mixer threads is we are stuck here
6953        Mutex::Autolock _l(mCblk->lock);
6954        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6955            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6956            mCblk->serverIndex = 0;
6957            mCblk->clientIndex = 0;
6958            return BAD_VALUE;
6959        }
6960        status_t status = NO_ERROR;
6961        while (mCblk->serverIndex < mCblk->clientIndex) {
6962            int reply;
6963            uint32_t rsize = sizeof(int);
6964            int *p = (int *)(mBuffer + mCblk->serverIndex);
6965            int size = *p++;
6966            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
6967                ALOGW("command(): invalid parameter block size");
6968                break;
6969            }
6970            effect_param_t *param = (effect_param_t *)p;
6971            if (param->psize == 0 || param->vsize == 0) {
6972                ALOGW("command(): null parameter or value size");
6973                mCblk->serverIndex += size;
6974                continue;
6975            }
6976            uint32_t psize = sizeof(effect_param_t) +
6977                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
6978                             param->vsize;
6979            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
6980                                            psize,
6981                                            p,
6982                                            &rsize,
6983                                            &reply);
6984            // stop at first error encountered
6985            if (ret != NO_ERROR) {
6986                status = ret;
6987                *(int *)pReplyData = reply;
6988                break;
6989            } else if (reply != NO_ERROR) {
6990                *(int *)pReplyData = reply;
6991                break;
6992            }
6993            mCblk->serverIndex += size;
6994        }
6995        mCblk->serverIndex = 0;
6996        mCblk->clientIndex = 0;
6997        return status;
6998    } else if (cmdCode == EFFECT_CMD_ENABLE) {
6999        *(int *)pReplyData = NO_ERROR;
7000        return enable();
7001    } else if (cmdCode == EFFECT_CMD_DISABLE) {
7002        *(int *)pReplyData = NO_ERROR;
7003        return disable();
7004    }
7005
7006    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7007}
7008
7009sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
7010    return mCblkMemory;
7011}
7012
7013void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
7014{
7015    ALOGV("setControl %p control %d", this, hasControl);
7016
7017    mHasControl = hasControl;
7018    mEnabled = enabled;
7019
7020    if (signal && mEffectClient != 0) {
7021        mEffectClient->controlStatusChanged(hasControl);
7022    }
7023}
7024
7025void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7026                                                 uint32_t cmdSize,
7027                                                 void *pCmdData,
7028                                                 uint32_t replySize,
7029                                                 void *pReplyData)
7030{
7031    if (mEffectClient != 0) {
7032        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7033    }
7034}
7035
7036
7037
7038void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7039{
7040    if (mEffectClient != 0) {
7041        mEffectClient->enableStatusChanged(enabled);
7042    }
7043}
7044
7045status_t AudioFlinger::EffectHandle::onTransact(
7046    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7047{
7048    return BnEffect::onTransact(code, data, reply, flags);
7049}
7050
7051
7052void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7053{
7054    bool locked = mCblk ? tryLock(mCblk->lock) : false;
7055
7056    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7057            (mClient == NULL) ? getpid() : mClient->pid(),
7058            mPriority,
7059            mHasControl,
7060            !locked,
7061            mCblk ? mCblk->clientIndex : 0,
7062            mCblk ? mCblk->serverIndex : 0
7063            );
7064
7065    if (locked) {
7066        mCblk->lock.unlock();
7067    }
7068}
7069
7070#undef LOG_TAG
7071#define LOG_TAG "AudioFlinger::EffectChain"
7072
7073AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7074                                        int sessionId)
7075    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7076      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7077      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7078{
7079    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7080    sp<ThreadBase> thread = mThread.promote();
7081    if (thread == 0) {
7082        return;
7083    }
7084    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7085                                    thread->frameCount();
7086}
7087
7088AudioFlinger::EffectChain::~EffectChain()
7089{
7090    if (mOwnInBuffer) {
7091        delete mInBuffer;
7092    }
7093
7094}
7095
7096// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7097sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7098{
7099    sp<EffectModule> effect;
7100    size_t size = mEffects.size();
7101
7102    for (size_t i = 0; i < size; i++) {
7103        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7104            effect = mEffects[i];
7105            break;
7106        }
7107    }
7108    return effect;
7109}
7110
7111// getEffectFromId_l() must be called with ThreadBase::mLock held
7112sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7113{
7114    sp<EffectModule> effect;
7115    size_t size = mEffects.size();
7116
7117    for (size_t i = 0; i < size; i++) {
7118        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7119        if (id == 0 || mEffects[i]->id() == id) {
7120            effect = mEffects[i];
7121            break;
7122        }
7123    }
7124    return effect;
7125}
7126
7127// getEffectFromType_l() must be called with ThreadBase::mLock held
7128sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7129        const effect_uuid_t *type)
7130{
7131    sp<EffectModule> effect;
7132    size_t size = mEffects.size();
7133
7134    for (size_t i = 0; i < size; i++) {
7135        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7136            effect = mEffects[i];
7137            break;
7138        }
7139    }
7140    return effect;
7141}
7142
7143// Must be called with EffectChain::mLock locked
7144void AudioFlinger::EffectChain::process_l()
7145{
7146    sp<ThreadBase> thread = mThread.promote();
7147    if (thread == 0) {
7148        ALOGW("process_l(): cannot promote mixer thread");
7149        return;
7150    }
7151    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7152            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7153    // always process effects unless no more tracks are on the session and the effect tail
7154    // has been rendered
7155    bool doProcess = true;
7156    if (!isGlobalSession) {
7157        bool tracksOnSession = (trackCnt() != 0);
7158
7159        if (!tracksOnSession && mTailBufferCount == 0) {
7160            doProcess = false;
7161        }
7162
7163        if (activeTrackCnt() == 0) {
7164            // if no track is active and the effect tail has not been rendered,
7165            // the input buffer must be cleared here as the mixer process will not do it
7166            if (tracksOnSession || mTailBufferCount > 0) {
7167                size_t numSamples = thread->frameCount() * thread->channelCount();
7168                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7169                if (mTailBufferCount > 0) {
7170                    mTailBufferCount--;
7171                }
7172            }
7173        }
7174    }
7175
7176    size_t size = mEffects.size();
7177    if (doProcess) {
7178        for (size_t i = 0; i < size; i++) {
7179            mEffects[i]->process();
7180        }
7181    }
7182    for (size_t i = 0; i < size; i++) {
7183        mEffects[i]->updateState();
7184    }
7185}
7186
7187// addEffect_l() must be called with PlaybackThread::mLock held
7188status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7189{
7190    effect_descriptor_t desc = effect->desc();
7191    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7192
7193    Mutex::Autolock _l(mLock);
7194    effect->setChain(this);
7195    sp<ThreadBase> thread = mThread.promote();
7196    if (thread == 0) {
7197        return NO_INIT;
7198    }
7199    effect->setThread(thread);
7200
7201    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7202        // Auxiliary effects are inserted at the beginning of mEffects vector as
7203        // they are processed first and accumulated in chain input buffer
7204        mEffects.insertAt(effect, 0);
7205
7206        // the input buffer for auxiliary effect contains mono samples in
7207        // 32 bit format. This is to avoid saturation in AudoMixer
7208        // accumulation stage. Saturation is done in EffectModule::process() before
7209        // calling the process in effect engine
7210        size_t numSamples = thread->frameCount();
7211        int32_t *buffer = new int32_t[numSamples];
7212        memset(buffer, 0, numSamples * sizeof(int32_t));
7213        effect->setInBuffer((int16_t *)buffer);
7214        // auxiliary effects output samples to chain input buffer for further processing
7215        // by insert effects
7216        effect->setOutBuffer(mInBuffer);
7217    } else {
7218        // Insert effects are inserted at the end of mEffects vector as they are processed
7219        //  after track and auxiliary effects.
7220        // Insert effect order as a function of indicated preference:
7221        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7222        //  another effect is present
7223        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7224        //  last effect claiming first position
7225        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7226        //  first effect claiming last position
7227        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7228        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7229        // already present
7230
7231        int size = (int)mEffects.size();
7232        int idx_insert = size;
7233        int idx_insert_first = -1;
7234        int idx_insert_last = -1;
7235
7236        for (int i = 0; i < size; i++) {
7237            effect_descriptor_t d = mEffects[i]->desc();
7238            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7239            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7240            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7241                // check invalid effect chaining combinations
7242                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7243                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7244                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7245                    return INVALID_OPERATION;
7246                }
7247                // remember position of first insert effect and by default
7248                // select this as insert position for new effect
7249                if (idx_insert == size) {
7250                    idx_insert = i;
7251                }
7252                // remember position of last insert effect claiming
7253                // first position
7254                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7255                    idx_insert_first = i;
7256                }
7257                // remember position of first insert effect claiming
7258                // last position
7259                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7260                    idx_insert_last == -1) {
7261                    idx_insert_last = i;
7262                }
7263            }
7264        }
7265
7266        // modify idx_insert from first position if needed
7267        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7268            if (idx_insert_last != -1) {
7269                idx_insert = idx_insert_last;
7270            } else {
7271                idx_insert = size;
7272            }
7273        } else {
7274            if (idx_insert_first != -1) {
7275                idx_insert = idx_insert_first + 1;
7276            }
7277        }
7278
7279        // always read samples from chain input buffer
7280        effect->setInBuffer(mInBuffer);
7281
7282        // if last effect in the chain, output samples to chain
7283        // output buffer, otherwise to chain input buffer
7284        if (idx_insert == size) {
7285            if (idx_insert != 0) {
7286                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7287                mEffects[idx_insert-1]->configure();
7288            }
7289            effect->setOutBuffer(mOutBuffer);
7290        } else {
7291            effect->setOutBuffer(mInBuffer);
7292        }
7293        mEffects.insertAt(effect, idx_insert);
7294
7295        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7296    }
7297    effect->configure();
7298    return NO_ERROR;
7299}
7300
7301// removeEffect_l() must be called with PlaybackThread::mLock held
7302size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7303{
7304    Mutex::Autolock _l(mLock);
7305    int size = (int)mEffects.size();
7306    int i;
7307    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7308
7309    for (i = 0; i < size; i++) {
7310        if (effect == mEffects[i]) {
7311            // calling stop here will remove pre-processing effect from the audio HAL.
7312            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7313            // the middle of a read from audio HAL
7314            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7315                    mEffects[i]->state() == EffectModule::STOPPING) {
7316                mEffects[i]->stop();
7317            }
7318            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7319                delete[] effect->inBuffer();
7320            } else {
7321                if (i == size - 1 && i != 0) {
7322                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7323                    mEffects[i - 1]->configure();
7324                }
7325            }
7326            mEffects.removeAt(i);
7327            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7328            break;
7329        }
7330    }
7331
7332    return mEffects.size();
7333}
7334
7335// setDevice_l() must be called with PlaybackThread::mLock held
7336void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7337{
7338    size_t size = mEffects.size();
7339    for (size_t i = 0; i < size; i++) {
7340        mEffects[i]->setDevice(device);
7341    }
7342}
7343
7344// setMode_l() must be called with PlaybackThread::mLock held
7345void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
7346{
7347    size_t size = mEffects.size();
7348    for (size_t i = 0; i < size; i++) {
7349        mEffects[i]->setMode(mode);
7350    }
7351}
7352
7353// setVolume_l() must be called with PlaybackThread::mLock held
7354bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7355{
7356    uint32_t newLeft = *left;
7357    uint32_t newRight = *right;
7358    bool hasControl = false;
7359    int ctrlIdx = -1;
7360    size_t size = mEffects.size();
7361
7362    // first update volume controller
7363    for (size_t i = size; i > 0; i--) {
7364        if (mEffects[i - 1]->isProcessEnabled() &&
7365            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7366            ctrlIdx = i - 1;
7367            hasControl = true;
7368            break;
7369        }
7370    }
7371
7372    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7373        if (hasControl) {
7374            *left = mNewLeftVolume;
7375            *right = mNewRightVolume;
7376        }
7377        return hasControl;
7378    }
7379
7380    mVolumeCtrlIdx = ctrlIdx;
7381    mLeftVolume = newLeft;
7382    mRightVolume = newRight;
7383
7384    // second get volume update from volume controller
7385    if (ctrlIdx >= 0) {
7386        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7387        mNewLeftVolume = newLeft;
7388        mNewRightVolume = newRight;
7389    }
7390    // then indicate volume to all other effects in chain.
7391    // Pass altered volume to effects before volume controller
7392    // and requested volume to effects after controller
7393    uint32_t lVol = newLeft;
7394    uint32_t rVol = newRight;
7395
7396    for (size_t i = 0; i < size; i++) {
7397        if ((int)i == ctrlIdx) continue;
7398        // this also works for ctrlIdx == -1 when there is no volume controller
7399        if ((int)i > ctrlIdx) {
7400            lVol = *left;
7401            rVol = *right;
7402        }
7403        mEffects[i]->setVolume(&lVol, &rVol, false);
7404    }
7405    *left = newLeft;
7406    *right = newRight;
7407
7408    return hasControl;
7409}
7410
7411status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7412{
7413    const size_t SIZE = 256;
7414    char buffer[SIZE];
7415    String8 result;
7416
7417    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7418    result.append(buffer);
7419
7420    bool locked = tryLock(mLock);
7421    // failed to lock - AudioFlinger is probably deadlocked
7422    if (!locked) {
7423        result.append("\tCould not lock mutex:\n");
7424    }
7425
7426    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7427    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7428            mEffects.size(),
7429            (uint32_t)mInBuffer,
7430            (uint32_t)mOutBuffer,
7431            mActiveTrackCnt);
7432    result.append(buffer);
7433    write(fd, result.string(), result.size());
7434
7435    for (size_t i = 0; i < mEffects.size(); ++i) {
7436        sp<EffectModule> effect = mEffects[i];
7437        if (effect != 0) {
7438            effect->dump(fd, args);
7439        }
7440    }
7441
7442    if (locked) {
7443        mLock.unlock();
7444    }
7445
7446    return NO_ERROR;
7447}
7448
7449// must be called with ThreadBase::mLock held
7450void AudioFlinger::EffectChain::setEffectSuspended_l(
7451        const effect_uuid_t *type, bool suspend)
7452{
7453    sp<SuspendedEffectDesc> desc;
7454    // use effect type UUID timelow as key as there is no real risk of identical
7455    // timeLow fields among effect type UUIDs.
7456    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7457    if (suspend) {
7458        if (index >= 0) {
7459            desc = mSuspendedEffects.valueAt(index);
7460        } else {
7461            desc = new SuspendedEffectDesc();
7462            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7463            mSuspendedEffects.add(type->timeLow, desc);
7464            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7465        }
7466        if (desc->mRefCount++ == 0) {
7467            sp<EffectModule> effect = getEffectIfEnabled(type);
7468            if (effect != 0) {
7469                desc->mEffect = effect;
7470                effect->setSuspended(true);
7471                effect->setEnabled(false);
7472            }
7473        }
7474    } else {
7475        if (index < 0) {
7476            return;
7477        }
7478        desc = mSuspendedEffects.valueAt(index);
7479        if (desc->mRefCount <= 0) {
7480            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7481            desc->mRefCount = 1;
7482        }
7483        if (--desc->mRefCount == 0) {
7484            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7485            if (desc->mEffect != 0) {
7486                sp<EffectModule> effect = desc->mEffect.promote();
7487                if (effect != 0) {
7488                    effect->setSuspended(false);
7489                    sp<EffectHandle> handle = effect->controlHandle();
7490                    if (handle != 0) {
7491                        effect->setEnabled(handle->enabled());
7492                    }
7493                }
7494                desc->mEffect.clear();
7495            }
7496            mSuspendedEffects.removeItemsAt(index);
7497        }
7498    }
7499}
7500
7501// must be called with ThreadBase::mLock held
7502void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7503{
7504    sp<SuspendedEffectDesc> desc;
7505
7506    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7507    if (suspend) {
7508        if (index >= 0) {
7509            desc = mSuspendedEffects.valueAt(index);
7510        } else {
7511            desc = new SuspendedEffectDesc();
7512            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7513            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7514        }
7515        if (desc->mRefCount++ == 0) {
7516            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7517            for (size_t i = 0; i < effects.size(); i++) {
7518                setEffectSuspended_l(&effects[i]->desc().type, true);
7519            }
7520        }
7521    } else {
7522        if (index < 0) {
7523            return;
7524        }
7525        desc = mSuspendedEffects.valueAt(index);
7526        if (desc->mRefCount <= 0) {
7527            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7528            desc->mRefCount = 1;
7529        }
7530        if (--desc->mRefCount == 0) {
7531            Vector<const effect_uuid_t *> types;
7532            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7533                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7534                    continue;
7535                }
7536                types.add(&mSuspendedEffects.valueAt(i)->mType);
7537            }
7538            for (size_t i = 0; i < types.size(); i++) {
7539                setEffectSuspended_l(types[i], false);
7540            }
7541            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7542            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7543        }
7544    }
7545}
7546
7547
7548// The volume effect is used for automated tests only
7549#ifndef OPENSL_ES_H_
7550static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7551                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7552const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7553#endif //OPENSL_ES_H_
7554
7555bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7556{
7557    // auxiliary effects and visualizer are never suspended on output mix
7558    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7559        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7560         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7561         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7562        return false;
7563    }
7564    return true;
7565}
7566
7567Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7568{
7569    Vector< sp<EffectModule> > effects;
7570    for (size_t i = 0; i < mEffects.size(); i++) {
7571        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7572            continue;
7573        }
7574        effects.add(mEffects[i]);
7575    }
7576    return effects;
7577}
7578
7579sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7580                                                            const effect_uuid_t *type)
7581{
7582    sp<EffectModule> effect;
7583    effect = getEffectFromType_l(type);
7584    if (effect != 0 && !effect->isEnabled()) {
7585        effect.clear();
7586    }
7587    return effect;
7588}
7589
7590void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7591                                                            bool enabled)
7592{
7593    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7594    if (enabled) {
7595        if (index < 0) {
7596            // if the effect is not suspend check if all effects are suspended
7597            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7598            if (index < 0) {
7599                return;
7600            }
7601            if (!isEffectEligibleForSuspend(effect->desc())) {
7602                return;
7603            }
7604            setEffectSuspended_l(&effect->desc().type, enabled);
7605            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7606            if (index < 0) {
7607                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7608                return;
7609            }
7610        }
7611        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7612             effect->desc().type.timeLow);
7613        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7614        // if effect is requested to suspended but was not yet enabled, supend it now.
7615        if (desc->mEffect == 0) {
7616            desc->mEffect = effect;
7617            effect->setEnabled(false);
7618            effect->setSuspended(true);
7619        }
7620    } else {
7621        if (index < 0) {
7622            return;
7623        }
7624        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7625             effect->desc().type.timeLow);
7626        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7627        desc->mEffect.clear();
7628        effect->setSuspended(false);
7629    }
7630}
7631
7632#undef LOG_TAG
7633#define LOG_TAG "AudioFlinger"
7634
7635// ----------------------------------------------------------------------------
7636
7637status_t AudioFlinger::onTransact(
7638        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7639{
7640    return BnAudioFlinger::onTransact(code, data, reply, flags);
7641}
7642
7643}; // namespace android
7644