AudioFlinger.cpp revision d1e672acd8fa1af899f85ee2321327237028adf8 
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include <math.h>
23#include <signal.h>
24#include <sys/time.h>
25#include <sys/resource.h>
26
27#include <binder/IPCThreadState.h>
28#include <binder/IServiceManager.h>
29#include <utils/Log.h>
30#include <binder/Parcel.h>
31#include <binder/IPCThreadState.h>
32#include <utils/String16.h>
33#include <utils/threads.h>
34#include <utils/Atomic.h>
35
36#include <cutils/bitops.h>
37#include <cutils/properties.h>
38
39#include <media/AudioTrack.h>
40#include <media/AudioRecord.h>
41#include <media/IMediaPlayerService.h>
42
43#include <private/media/AudioTrackShared.h>
44#include <private/media/AudioEffectShared.h>
45
46#include <system/audio.h>
47#include <hardware/audio.h>
48
49#include "AudioMixer.h"
50#include "AudioFlinger.h"
51
52#include <media/EffectsFactoryApi.h>
53#include <audio_effects/effect_visualizer.h>
54#include <audio_effects/effect_ns.h>
55#include <audio_effects/effect_aec.h>
56
57#include <audio_utils/primitives.h>
58
59#include <cpustats/ThreadCpuUsage.h>
60#include <powermanager/PowerManager.h>
61// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
62
63// ----------------------------------------------------------------------------
64
65
66namespace android {
67
68static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
69static const char kHardwareLockedString[] = "Hardware lock is taken\n";
70
71//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
72static const float MAX_GAIN = 4096.0f;
73static const float MAX_GAIN_INT = 0x1000;
74
75// retry counts for buffer fill timeout
76// 50 * ~20msecs = 1 second
77static const int8_t kMaxTrackRetries = 50;
78static const int8_t kMaxTrackStartupRetries = 50;
79// allow less retry attempts on direct output thread.
80// direct outputs can be a scarce resource in audio hardware and should
81// be released as quickly as possible.
82static const int8_t kMaxTrackRetriesDirect = 2;
83
84static const int kDumpLockRetries = 50;
85static const int kDumpLockSleepUs = 20000;
86
87// don't warn about blocked writes or record buffer overflows more often than this
88static const nsecs_t kWarningThrottleNs = seconds(5);
89
90// RecordThread loop sleep time upon application overrun or audio HAL read error
91static const int kRecordThreadSleepUs = 5000;
92
93// maximum time to wait for setParameters to complete
94static const nsecs_t kSetParametersTimeoutNs = seconds(2);
95
96// minimum sleep time for the mixer thread loop when tracks are active but in underrun
97static const uint32_t kMinThreadSleepTimeUs = 5000;
98// maximum divider applied to the active sleep time in the mixer thread loop
99static const uint32_t kMaxThreadSleepTimeShift = 2;
100
101
102// ----------------------------------------------------------------------------
103
104static bool recordingAllowed() {
105    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
106    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
107    if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
108    return ok;
109}
110
111static bool settingsAllowed() {
112    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
113    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
114    if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
115    return ok;
116}
117
118// To collect the amplifier usage
119static void addBatteryData(uint32_t params) {
120    sp<IBinder> binder =
121        defaultServiceManager()->getService(String16("media.player"));
122    sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder);
123    if (service.get() == NULL) {
124        LOGW("Cannot connect to the MediaPlayerService for battery tracking");
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    LOGE_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        LOGI("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            LOGI("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        LOGE("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    int hardwareStatus = mHardwareStatus;
298
299    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
300    result.append(buffer);
301    write(fd, result.string(), result.size());
302    return NO_ERROR;
303}
304
305status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
306{
307    const size_t SIZE = 256;
308    char buffer[SIZE];
309    String8 result;
310    snprintf(buffer, SIZE, "Permission Denial: "
311            "can't dump AudioFlinger from pid=%d, uid=%d\n",
312            IPCThreadState::self()->getCallingPid(),
313            IPCThreadState::self()->getCallingUid());
314    result.append(buffer);
315    write(fd, result.string(), result.size());
316    return NO_ERROR;
317}
318
319static bool tryLock(Mutex& mutex)
320{
321    bool locked = false;
322    for (int i = 0; i < kDumpLockRetries; ++i) {
323        if (mutex.tryLock() == NO_ERROR) {
324            locked = true;
325            break;
326        }
327        usleep(kDumpLockSleepUs);
328    }
329    return locked;
330}
331
332status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
333{
334    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
335        dumpPermissionDenial(fd, args);
336    } else {
337        // get state of hardware lock
338        bool hardwareLocked = tryLock(mHardwareLock);
339        if (!hardwareLocked) {
340            String8 result(kHardwareLockedString);
341            write(fd, result.string(), result.size());
342        } else {
343            mHardwareLock.unlock();
344        }
345
346        bool locked = tryLock(mLock);
347
348        // failed to lock - AudioFlinger is probably deadlocked
349        if (!locked) {
350            String8 result(kDeadlockedString);
351            write(fd, result.string(), result.size());
352        }
353
354        dumpClients(fd, args);
355        dumpInternals(fd, args);
356
357        // dump playback threads
358        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
359            mPlaybackThreads.valueAt(i)->dump(fd, args);
360        }
361
362        // dump record threads
363        for (size_t i = 0; i < mRecordThreads.size(); i++) {
364            mRecordThreads.valueAt(i)->dump(fd, args);
365        }
366
367        // dump all hardware devs
368        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
369            audio_hw_device_t *dev = mAudioHwDevs[i];
370            dev->dump(dev, fd);
371        }
372        if (locked) mLock.unlock();
373    }
374    return NO_ERROR;
375}
376
377
378// IAudioFlinger interface
379
380
381sp<IAudioTrack> AudioFlinger::createTrack(
382        pid_t pid,
383        int streamType,
384        uint32_t sampleRate,
385        uint32_t format,
386        uint32_t channelMask,
387        int frameCount,
388        uint32_t flags,
389        const sp<IMemory>& sharedBuffer,
390        int output,
391        int *sessionId,
392        status_t *status)
393{
394    sp<PlaybackThread::Track> track;
395    sp<TrackHandle> trackHandle;
396    sp<Client> client;
397    wp<Client> wclient;
398    status_t lStatus;
399    int lSessionId;
400
401    if (streamType >= AUDIO_STREAM_CNT) {
402        LOGE("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            LOGE("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                        LOGE("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        LOGW("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        LOGW("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        LOGW("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        LOGW("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        LOGW("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 ((mode < 0) || (mode >= AUDIO_MODE_CNT)) {
578        LOGW("Illegal value: setMode(%d)", mode);
579        return BAD_VALUE;
580    }
581
582    { // scope for the lock
583        AutoMutex lock(mHardwareLock);
584        mHardwareStatus = AUDIO_HW_SET_MODE;
585        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
586        mHardwareStatus = AUDIO_HW_IDLE;
587    }
588
589    if (NO_ERROR == ret) {
590        Mutex::Autolock _l(mLock);
591        mMode = mode;
592        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
593           mPlaybackThreads.valueAt(i)->setMode(mode);
594    }
595
596    return ret;
597}
598
599status_t AudioFlinger::setMicMute(bool state)
600{
601    status_t ret = initCheck();
602    if (ret != NO_ERROR) {
603        return ret;
604    }
605
606    // check calling permissions
607    if (!settingsAllowed()) {
608        return PERMISSION_DENIED;
609    }
610
611    AutoMutex lock(mHardwareLock);
612    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
613    ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
614    mHardwareStatus = AUDIO_HW_IDLE;
615    return ret;
616}
617
618bool AudioFlinger::getMicMute() const
619{
620    status_t ret = initCheck();
621    if (ret != NO_ERROR) {
622        return false;
623    }
624
625    bool state = AUDIO_MODE_INVALID;
626    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
627    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
628    mHardwareStatus = AUDIO_HW_IDLE;
629    return state;
630}
631
632status_t AudioFlinger::setMasterMute(bool muted)
633{
634    // check calling permissions
635    if (!settingsAllowed()) {
636        return PERMISSION_DENIED;
637    }
638
639    Mutex::Autolock _l(mLock);
640    mMasterMute = muted;
641    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
642       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
643
644    return NO_ERROR;
645}
646
647float AudioFlinger::masterVolume() const
648{
649    return mMasterVolume;
650}
651
652bool AudioFlinger::masterMute() const
653{
654    return mMasterMute;
655}
656
657status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
658{
659    // check calling permissions
660    if (!settingsAllowed()) {
661        return PERMISSION_DENIED;
662    }
663
664    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
665        LOGE("setStreamVolume() invalid stream %d", stream);
666        return BAD_VALUE;
667    }
668
669    AutoMutex lock(mLock);
670    PlaybackThread *thread = NULL;
671    if (output) {
672        thread = checkPlaybackThread_l(output);
673        if (thread == NULL) {
674            return BAD_VALUE;
675        }
676    }
677
678    mStreamTypes[stream].volume = value;
679
680    if (thread == NULL) {
681        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
682           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
683        }
684    } else {
685        thread->setStreamVolume(stream, value);
686    }
687
688    return NO_ERROR;
689}
690
691status_t AudioFlinger::setStreamMute(int stream, bool muted)
692{
693    // check calling permissions
694    if (!settingsAllowed()) {
695        return PERMISSION_DENIED;
696    }
697
698    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
699        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
700        LOGE("setStreamMute() invalid stream %d", stream);
701        return BAD_VALUE;
702    }
703
704    AutoMutex lock(mLock);
705    mStreamTypes[stream].mute = muted;
706    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
707       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
708
709    return NO_ERROR;
710}
711
712float AudioFlinger::streamVolume(int stream, int output) const
713{
714    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
715        return 0.0f;
716    }
717
718    AutoMutex lock(mLock);
719    float volume;
720    if (output) {
721        PlaybackThread *thread = checkPlaybackThread_l(output);
722        if (thread == NULL) {
723            return 0.0f;
724        }
725        volume = thread->streamVolume(stream);
726    } else {
727        volume = mStreamTypes[stream].volume;
728    }
729
730    return volume;
731}
732
733bool AudioFlinger::streamMute(int stream) const
734{
735    if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
736        return true;
737    }
738
739    return mStreamTypes[stream].mute;
740}
741
742status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
743{
744    status_t result;
745
746    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
747            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
748    // check calling permissions
749    if (!settingsAllowed()) {
750        return PERMISSION_DENIED;
751    }
752
753    // ioHandle == 0 means the parameters are global to the audio hardware interface
754    if (ioHandle == 0) {
755        AutoMutex lock(mHardwareLock);
756        mHardwareStatus = AUDIO_SET_PARAMETER;
757        status_t final_result = NO_ERROR;
758        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
759            audio_hw_device_t *dev = mAudioHwDevs[i];
760            result = dev->set_parameters(dev, keyValuePairs.string());
761            final_result = result ?: final_result;
762        }
763        mHardwareStatus = AUDIO_HW_IDLE;
764        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
765        AudioParameter param = AudioParameter(keyValuePairs);
766        String8 value;
767        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
768            Mutex::Autolock _l(mLock);
769            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
770            if (mBtNrecIsOff != btNrecIsOff) {
771                for (size_t i = 0; i < mRecordThreads.size(); i++) {
772                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
773                    RecordThread::RecordTrack *track = thread->track();
774                    if (track != NULL) {
775                        audio_devices_t device = (audio_devices_t)(
776                                thread->device() & AUDIO_DEVICE_IN_ALL);
777                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
778                        thread->setEffectSuspended(FX_IID_AEC,
779                                                   suspend,
780                                                   track->sessionId());
781                        thread->setEffectSuspended(FX_IID_NS,
782                                                   suspend,
783                                                   track->sessionId());
784                    }
785                }
786                mBtNrecIsOff = btNrecIsOff;
787            }
788        }
789        return final_result;
790    }
791
792    // hold a strong ref on thread in case closeOutput() or closeInput() is called
793    // and the thread is exited once the lock is released
794    sp<ThreadBase> thread;
795    {
796        Mutex::Autolock _l(mLock);
797        thread = checkPlaybackThread_l(ioHandle);
798        if (thread == NULL) {
799            thread = checkRecordThread_l(ioHandle);
800        } else if (thread.get() == primaryPlaybackThread_l()) {
801            // indicate output device change to all input threads for pre processing
802            AudioParameter param = AudioParameter(keyValuePairs);
803            int value;
804            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
805                for (size_t i = 0; i < mRecordThreads.size(); i++) {
806                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
807                }
808            }
809        }
810    }
811    if (thread != NULL) {
812        result = thread->setParameters(keyValuePairs);
813        return result;
814    }
815    return BAD_VALUE;
816}
817
818String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
819{
820//    ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
821//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
822
823    if (ioHandle == 0) {
824        String8 out_s8;
825
826        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
827            audio_hw_device_t *dev = mAudioHwDevs[i];
828            char *s = dev->get_parameters(dev, keys.string());
829            out_s8 += String8(s);
830            free(s);
831        }
832        return out_s8;
833    }
834
835    Mutex::Autolock _l(mLock);
836
837    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
838    if (playbackThread != NULL) {
839        return playbackThread->getParameters(keys);
840    }
841    RecordThread *recordThread = checkRecordThread_l(ioHandle);
842    if (recordThread != NULL) {
843        return recordThread->getParameters(keys);
844    }
845    return String8("");
846}
847
848size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
849{
850    status_t ret = initCheck();
851    if (ret != NO_ERROR) {
852        return 0;
853    }
854
855    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
856}
857
858unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
859{
860    if (ioHandle == 0) {
861        return 0;
862    }
863
864    Mutex::Autolock _l(mLock);
865
866    RecordThread *recordThread = checkRecordThread_l(ioHandle);
867    if (recordThread != NULL) {
868        return recordThread->getInputFramesLost();
869    }
870    return 0;
871}
872
873status_t AudioFlinger::setVoiceVolume(float value)
874{
875    status_t ret = initCheck();
876    if (ret != NO_ERROR) {
877        return ret;
878    }
879
880    // check calling permissions
881    if (!settingsAllowed()) {
882        return PERMISSION_DENIED;
883    }
884
885    AutoMutex lock(mHardwareLock);
886    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
887    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
888    mHardwareStatus = AUDIO_HW_IDLE;
889
890    return ret;
891}
892
893status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
894{
895    status_t status;
896
897    Mutex::Autolock _l(mLock);
898
899    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
900    if (playbackThread != NULL) {
901        return playbackThread->getRenderPosition(halFrames, dspFrames);
902    }
903
904    return BAD_VALUE;
905}
906
907void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
908{
909
910    Mutex::Autolock _l(mLock);
911
912    int pid = IPCThreadState::self()->getCallingPid();
913    if (mNotificationClients.indexOfKey(pid) < 0) {
914        sp<NotificationClient> notificationClient = new NotificationClient(this,
915                                                                            client,
916                                                                            pid);
917        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
918
919        mNotificationClients.add(pid, notificationClient);
920
921        sp<IBinder> binder = client->asBinder();
922        binder->linkToDeath(notificationClient);
923
924        // the config change is always sent from playback or record threads to avoid deadlock
925        // with AudioSystem::gLock
926        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
927            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
928        }
929
930        for (size_t i = 0; i < mRecordThreads.size(); i++) {
931            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
932        }
933    }
934}
935
936void AudioFlinger::removeNotificationClient(pid_t pid)
937{
938    Mutex::Autolock _l(mLock);
939
940    int index = mNotificationClients.indexOfKey(pid);
941    if (index >= 0) {
942        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
943        ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
944        mNotificationClients.removeItem(pid);
945    }
946
947    ALOGV("%d died, releasing its sessions", pid);
948    int num = mAudioSessionRefs.size();
949    bool removed = false;
950    for (int i = 0; i< num; i++) {
951        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
952        ALOGV(" pid %d @ %d", ref->pid, i);
953        if (ref->pid == pid) {
954            ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
955            mAudioSessionRefs.removeAt(i);
956            delete ref;
957            removed = true;
958            i--;
959            num--;
960        }
961    }
962    if (removed) {
963        purgeStaleEffects_l();
964    }
965}
966
967// audioConfigChanged_l() must be called with AudioFlinger::mLock held
968void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
969{
970    size_t size = mNotificationClients.size();
971    for (size_t i = 0; i < size; i++) {
972        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
973    }
974}
975
976// removeClient_l() must be called with AudioFlinger::mLock held
977void AudioFlinger::removeClient_l(pid_t pid)
978{
979    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
980    mClients.removeItem(pid);
981}
982
983
984// ----------------------------------------------------------------------------
985
986AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
987    :   Thread(false),
988        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
989        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
990        mDevice(device)
991{
992    mDeathRecipient = new PMDeathRecipient(this);
993}
994
995AudioFlinger::ThreadBase::~ThreadBase()
996{
997    mParamCond.broadcast();
998    // do not lock the mutex in destructor
999    releaseWakeLock_l();
1000    if (mPowerManager != 0) {
1001        sp<IBinder> binder = mPowerManager->asBinder();
1002        binder->unlinkToDeath(mDeathRecipient);
1003    }
1004}
1005
1006void AudioFlinger::ThreadBase::exit()
1007{
1008    // keep a strong ref on ourself so that we won't get
1009    // destroyed in the middle of requestExitAndWait()
1010    sp <ThreadBase> strongMe = this;
1011
1012    ALOGV("ThreadBase::exit");
1013    {
1014        AutoMutex lock(&mLock);
1015        mExiting = true;
1016        requestExit();
1017        mWaitWorkCV.signal();
1018    }
1019    requestExitAndWait();
1020}
1021
1022uint32_t AudioFlinger::ThreadBase::sampleRate() const
1023{
1024    return mSampleRate;
1025}
1026
1027int AudioFlinger::ThreadBase::channelCount() const
1028{
1029    return (int)mChannelCount;
1030}
1031
1032uint32_t AudioFlinger::ThreadBase::format() const
1033{
1034    return mFormat;
1035}
1036
1037size_t AudioFlinger::ThreadBase::frameCount() const
1038{
1039    return mFrameCount;
1040}
1041
1042status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1043{
1044    status_t status;
1045
1046    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1047    Mutex::Autolock _l(mLock);
1048
1049    mNewParameters.add(keyValuePairs);
1050    mWaitWorkCV.signal();
1051    // wait condition with timeout in case the thread loop has exited
1052    // before the request could be processed
1053    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
1054        status = mParamStatus;
1055        mWaitWorkCV.signal();
1056    } else {
1057        status = TIMED_OUT;
1058    }
1059    return status;
1060}
1061
1062void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1063{
1064    Mutex::Autolock _l(mLock);
1065    sendConfigEvent_l(event, param);
1066}
1067
1068// sendConfigEvent_l() must be called with ThreadBase::mLock held
1069void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1070{
1071    ConfigEvent configEvent;
1072    configEvent.mEvent = event;
1073    configEvent.mParam = param;
1074    mConfigEvents.add(configEvent);
1075    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1076    mWaitWorkCV.signal();
1077}
1078
1079void AudioFlinger::ThreadBase::processConfigEvents()
1080{
1081    mLock.lock();
1082    while(!mConfigEvents.isEmpty()) {
1083        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1084        ConfigEvent configEvent = mConfigEvents[0];
1085        mConfigEvents.removeAt(0);
1086        // release mLock before locking AudioFlinger mLock: lock order is always
1087        // AudioFlinger then ThreadBase to avoid cross deadlock
1088        mLock.unlock();
1089        mAudioFlinger->mLock.lock();
1090        audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
1091        mAudioFlinger->mLock.unlock();
1092        mLock.lock();
1093    }
1094    mLock.unlock();
1095}
1096
1097status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1098{
1099    const size_t SIZE = 256;
1100    char buffer[SIZE];
1101    String8 result;
1102
1103    bool locked = tryLock(mLock);
1104    if (!locked) {
1105        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1106        write(fd, buffer, strlen(buffer));
1107    }
1108
1109    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1110    result.append(buffer);
1111    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1112    result.append(buffer);
1113    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1114    result.append(buffer);
1115    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1116    result.append(buffer);
1117    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1118    result.append(buffer);
1119    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1120    result.append(buffer);
1121    snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
1122    result.append(buffer);
1123
1124    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1125    result.append(buffer);
1126    result.append(" Index Command");
1127    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1128        snprintf(buffer, SIZE, "\n %02d    ", i);
1129        result.append(buffer);
1130        result.append(mNewParameters[i]);
1131    }
1132
1133    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1134    result.append(buffer);
1135    snprintf(buffer, SIZE, " Index event param\n");
1136    result.append(buffer);
1137    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1138        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
1139        result.append(buffer);
1140    }
1141    result.append("\n");
1142
1143    write(fd, result.string(), result.size());
1144
1145    if (locked) {
1146        mLock.unlock();
1147    }
1148    return NO_ERROR;
1149}
1150
1151status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1152{
1153    const size_t SIZE = 256;
1154    char buffer[SIZE];
1155    String8 result;
1156
1157    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1158    write(fd, buffer, strlen(buffer));
1159
1160    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1161        sp<EffectChain> chain = mEffectChains[i];
1162        if (chain != 0) {
1163            chain->dump(fd, args);
1164        }
1165    }
1166    return NO_ERROR;
1167}
1168
1169void AudioFlinger::ThreadBase::acquireWakeLock()
1170{
1171    Mutex::Autolock _l(mLock);
1172    acquireWakeLock_l();
1173}
1174
1175void AudioFlinger::ThreadBase::acquireWakeLock_l()
1176{
1177    if (mPowerManager == 0) {
1178        // use checkService() to avoid blocking if power service is not up yet
1179        sp<IBinder> binder =
1180            defaultServiceManager()->checkService(String16("power"));
1181        if (binder == 0) {
1182            LOGW("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    LOGW("power manager service died !!!");
1231}
1232
1233void AudioFlinger::ThreadBase::setEffectSuspended(
1234        const effect_uuid_t *type, bool suspend, int sessionId)
1235{
1236    Mutex::Autolock _l(mLock);
1237    setEffectSuspended_l(type, suspend, sessionId);
1238}
1239
1240void AudioFlinger::ThreadBase::setEffectSuspended_l(
1241        const effect_uuid_t *type, bool suspend, int sessionId)
1242{
1243    sp<EffectChain> chain;
1244    chain = getEffectChain_l(sessionId);
1245    if (chain != 0) {
1246        if (type != NULL) {
1247            chain->setEffectSuspended_l(type, suspend);
1248        } else {
1249            chain->setEffectSuspendedAll_l(suspend);
1250        }
1251    }
1252
1253    updateSuspendedSessions_l(type, suspend, sessionId);
1254}
1255
1256void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1257{
1258    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1259    if (index < 0) {
1260        return;
1261    }
1262
1263    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1264            mSuspendedSessions.editValueAt(index);
1265
1266    for (size_t i = 0; i < sessionEffects.size(); i++) {
1267        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1268        for (int j = 0; j < desc->mRefCount; j++) {
1269            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1270                chain->setEffectSuspendedAll_l(true);
1271            } else {
1272                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1273                     desc->mType.timeLow);
1274                chain->setEffectSuspended_l(&desc->mType, true);
1275            }
1276        }
1277    }
1278}
1279
1280void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1281                                                         bool suspend,
1282                                                         int sessionId)
1283{
1284    int index = mSuspendedSessions.indexOfKey(sessionId);
1285
1286    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1287
1288    if (suspend) {
1289        if (index >= 0) {
1290            sessionEffects = mSuspendedSessions.editValueAt(index);
1291        } else {
1292            mSuspendedSessions.add(sessionId, sessionEffects);
1293        }
1294    } else {
1295        if (index < 0) {
1296            return;
1297        }
1298        sessionEffects = mSuspendedSessions.editValueAt(index);
1299    }
1300
1301
1302    int key = EffectChain::kKeyForSuspendAll;
1303    if (type != NULL) {
1304        key = type->timeLow;
1305    }
1306    index = sessionEffects.indexOfKey(key);
1307
1308    sp <SuspendedSessionDesc> desc;
1309    if (suspend) {
1310        if (index >= 0) {
1311            desc = sessionEffects.valueAt(index);
1312        } else {
1313            desc = new SuspendedSessionDesc();
1314            if (type != NULL) {
1315                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1316            }
1317            sessionEffects.add(key, desc);
1318            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1319        }
1320        desc->mRefCount++;
1321    } else {
1322        if (index < 0) {
1323            return;
1324        }
1325        desc = sessionEffects.valueAt(index);
1326        if (--desc->mRefCount == 0) {
1327            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1328            sessionEffects.removeItemsAt(index);
1329            if (sessionEffects.isEmpty()) {
1330                ALOGV("updateSuspendedSessions_l() restore removing session %d",
1331                                 sessionId);
1332                mSuspendedSessions.removeItem(sessionId);
1333            }
1334        }
1335    }
1336    if (!sessionEffects.isEmpty()) {
1337        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1338    }
1339}
1340
1341void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1342                                                            bool enabled,
1343                                                            int sessionId)
1344{
1345    Mutex::Autolock _l(mLock);
1346    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1347}
1348
1349void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1350                                                            bool enabled,
1351                                                            int sessionId)
1352{
1353    if (mType != RECORD) {
1354        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1355        // another session. This gives the priority to well behaved effect control panels
1356        // and applications not using global effects.
1357        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1358            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1359        }
1360    }
1361
1362    sp<EffectChain> chain = getEffectChain_l(sessionId);
1363    if (chain != 0) {
1364        chain->checkSuspendOnEffectEnabled(effect, enabled);
1365    }
1366}
1367
1368// ----------------------------------------------------------------------------
1369
1370AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1371                                             AudioStreamOut* output,
1372                                             int id,
1373                                             uint32_t device)
1374    :   ThreadBase(audioFlinger, id, device),
1375        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output),
1376        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1377{
1378    snprintf(mName, kNameLength, "AudioOut_%d", id);
1379
1380    readOutputParameters();
1381
1382    mMasterVolume = mAudioFlinger->masterVolume();
1383    mMasterMute = mAudioFlinger->masterMute();
1384
1385    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
1386        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1387        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1388        mStreamTypes[stream].valid = true;
1389    }
1390}
1391
1392AudioFlinger::PlaybackThread::~PlaybackThread()
1393{
1394    delete [] mMixBuffer;
1395}
1396
1397status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1398{
1399    dumpInternals(fd, args);
1400    dumpTracks(fd, args);
1401    dumpEffectChains(fd, args);
1402    return NO_ERROR;
1403}
1404
1405status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1406{
1407    const size_t SIZE = 256;
1408    char buffer[SIZE];
1409    String8 result;
1410
1411    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1412    result.append(buffer);
1413    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1414    for (size_t i = 0; i < mTracks.size(); ++i) {
1415        sp<Track> track = mTracks[i];
1416        if (track != 0) {
1417            track->dump(buffer, SIZE);
1418            result.append(buffer);
1419        }
1420    }
1421
1422    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1423    result.append(buffer);
1424    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1425    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1426        wp<Track> wTrack = mActiveTracks[i];
1427        if (wTrack != 0) {
1428            sp<Track> track = wTrack.promote();
1429            if (track != 0) {
1430                track->dump(buffer, SIZE);
1431                result.append(buffer);
1432            }
1433        }
1434    }
1435    write(fd, result.string(), result.size());
1436    return NO_ERROR;
1437}
1438
1439status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1440{
1441    const size_t SIZE = 256;
1442    char buffer[SIZE];
1443    String8 result;
1444
1445    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1446    result.append(buffer);
1447    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1448    result.append(buffer);
1449    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1450    result.append(buffer);
1451    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1452    result.append(buffer);
1453    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1454    result.append(buffer);
1455    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1456    result.append(buffer);
1457    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1458    result.append(buffer);
1459    write(fd, result.string(), result.size());
1460
1461    dumpBase(fd, args);
1462
1463    return NO_ERROR;
1464}
1465
1466// Thread virtuals
1467status_t AudioFlinger::PlaybackThread::readyToRun()
1468{
1469    status_t status = initCheck();
1470    if (status == NO_ERROR) {
1471        LOGI("AudioFlinger's thread %p ready to run", this);
1472    } else {
1473        LOGE("No working audio driver found.");
1474    }
1475    return status;
1476}
1477
1478void AudioFlinger::PlaybackThread::onFirstRef()
1479{
1480    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1481}
1482
1483// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1484sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1485        const sp<AudioFlinger::Client>& client,
1486        int streamType,
1487        uint32_t sampleRate,
1488        uint32_t format,
1489        uint32_t channelMask,
1490        int frameCount,
1491        const sp<IMemory>& sharedBuffer,
1492        int sessionId,
1493        status_t *status)
1494{
1495    sp<Track> track;
1496    status_t lStatus;
1497
1498    if (mType == DIRECT) {
1499        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1500            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1501                LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1502                        "for output %p with format %d",
1503                        sampleRate, format, channelMask, mOutput, mFormat);
1504                lStatus = BAD_VALUE;
1505                goto Exit;
1506            }
1507        }
1508    } else {
1509        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1510        if (sampleRate > mSampleRate*2) {
1511            LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1512            lStatus = BAD_VALUE;
1513            goto Exit;
1514        }
1515    }
1516
1517    lStatus = initCheck();
1518    if (lStatus != NO_ERROR) {
1519        LOGE("Audio driver not initialized.");
1520        goto Exit;
1521    }
1522
1523    { // scope for mLock
1524        Mutex::Autolock _l(mLock);
1525
1526        // all tracks in same audio session must share the same routing strategy otherwise
1527        // conflicts will happen when tracks are moved from one output to another by audio policy
1528        // manager
1529        uint32_t strategy =
1530                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1531        for (size_t i = 0; i < mTracks.size(); ++i) {
1532            sp<Track> t = mTracks[i];
1533            if (t != 0) {
1534                uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
1535                if (sessionId == t->sessionId() && strategy != actual) {
1536                    LOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1537                            strategy, actual);
1538                    lStatus = BAD_VALUE;
1539                    goto Exit;
1540                }
1541            }
1542        }
1543
1544        track = new Track(this, client, streamType, sampleRate, format,
1545                channelMask, frameCount, sharedBuffer, sessionId);
1546        if (track->getCblk() == NULL || track->name() < 0) {
1547            lStatus = NO_MEMORY;
1548            goto Exit;
1549        }
1550        mTracks.add(track);
1551
1552        sp<EffectChain> chain = getEffectChain_l(sessionId);
1553        if (chain != 0) {
1554            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1555            track->setMainBuffer(chain->inBuffer());
1556            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1557            chain->incTrackCnt();
1558        }
1559
1560        // invalidate track immediately if the stream type was moved to another thread since
1561        // createTrack() was called by the client process.
1562        if (!mStreamTypes[streamType].valid) {
1563            LOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1564                 this, streamType);
1565            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1566        }
1567    }
1568    lStatus = NO_ERROR;
1569
1570Exit:
1571    if(status) {
1572        *status = lStatus;
1573    }
1574    return track;
1575}
1576
1577uint32_t AudioFlinger::PlaybackThread::latency() const
1578{
1579    Mutex::Autolock _l(mLock);
1580    if (initCheck() == NO_ERROR) {
1581        return mOutput->stream->get_latency(mOutput->stream);
1582    } else {
1583        return 0;
1584    }
1585}
1586
1587status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1588{
1589    mMasterVolume = value;
1590    return NO_ERROR;
1591}
1592
1593status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1594{
1595    mMasterMute = muted;
1596    return NO_ERROR;
1597}
1598
1599float AudioFlinger::PlaybackThread::masterVolume() const
1600{
1601    return mMasterVolume;
1602}
1603
1604bool AudioFlinger::PlaybackThread::masterMute() const
1605{
1606    return mMasterMute;
1607}
1608
1609status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
1610{
1611    mStreamTypes[stream].volume = value;
1612    return NO_ERROR;
1613}
1614
1615status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
1616{
1617    mStreamTypes[stream].mute = muted;
1618    return NO_ERROR;
1619}
1620
1621float AudioFlinger::PlaybackThread::streamVolume(int stream) const
1622{
1623    return mStreamTypes[stream].volume;
1624}
1625
1626bool AudioFlinger::PlaybackThread::streamMute(int stream) const
1627{
1628    return mStreamTypes[stream].mute;
1629}
1630
1631// addTrack_l() must be called with ThreadBase::mLock held
1632status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1633{
1634    status_t status = ALREADY_EXISTS;
1635
1636    // set retry count for buffer fill
1637    track->mRetryCount = kMaxTrackStartupRetries;
1638    if (mActiveTracks.indexOf(track) < 0) {
1639        // the track is newly added, make sure it fills up all its
1640        // buffers before playing. This is to ensure the client will
1641        // effectively get the latency it requested.
1642        track->mFillingUpStatus = Track::FS_FILLING;
1643        track->mResetDone = false;
1644        mActiveTracks.add(track);
1645        if (track->mainBuffer() != mMixBuffer) {
1646            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1647            if (chain != 0) {
1648                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1649                chain->incActiveTrackCnt();
1650            }
1651        }
1652
1653        status = NO_ERROR;
1654    }
1655
1656    ALOGV("mWaitWorkCV.broadcast");
1657    mWaitWorkCV.broadcast();
1658
1659    return status;
1660}
1661
1662// destroyTrack_l() must be called with ThreadBase::mLock held
1663void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1664{
1665    track->mState = TrackBase::TERMINATED;
1666    if (mActiveTracks.indexOf(track) < 0) {
1667        removeTrack_l(track);
1668    }
1669}
1670
1671void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1672{
1673    mTracks.remove(track);
1674    deleteTrackName_l(track->name());
1675    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1676    if (chain != 0) {
1677        chain->decTrackCnt();
1678    }
1679}
1680
1681String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1682{
1683    String8 out_s8 = String8("");
1684    char *s;
1685
1686    Mutex::Autolock _l(mLock);
1687    if (initCheck() != NO_ERROR) {
1688        return out_s8;
1689    }
1690
1691    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1692    out_s8 = String8(s);
1693    free(s);
1694    return out_s8;
1695}
1696
1697// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1698void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1699    AudioSystem::OutputDescriptor desc;
1700    void *param2 = 0;
1701
1702    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1703
1704    switch (event) {
1705    case AudioSystem::OUTPUT_OPENED:
1706    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1707        desc.channels = mChannelMask;
1708        desc.samplingRate = mSampleRate;
1709        desc.format = mFormat;
1710        desc.frameCount = mFrameCount;
1711        desc.latency = latency();
1712        param2 = &desc;
1713        break;
1714
1715    case AudioSystem::STREAM_CONFIG_CHANGED:
1716        param2 = &param;
1717    case AudioSystem::OUTPUT_CLOSED:
1718    default:
1719        break;
1720    }
1721    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1722}
1723
1724void AudioFlinger::PlaybackThread::readOutputParameters()
1725{
1726    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1727    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1728    mChannelCount = (uint16_t)popcount(mChannelMask);
1729    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1730    mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
1731    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1732
1733    // FIXME - Current mixer implementation only supports stereo output: Always
1734    // Allocate a stereo buffer even if HW output is mono.
1735    if (mMixBuffer != NULL) delete[] mMixBuffer;
1736    mMixBuffer = new int16_t[mFrameCount * 2];
1737    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1738
1739    // force reconfiguration of effect chains and engines to take new buffer size and audio
1740    // parameters into account
1741    // Note that mLock is not held when readOutputParameters() is called from the constructor
1742    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1743    // matter.
1744    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1745    Vector< sp<EffectChain> > effectChains = mEffectChains;
1746    for (size_t i = 0; i < effectChains.size(); i ++) {
1747        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1748    }
1749}
1750
1751status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1752{
1753    if (halFrames == 0 || dspFrames == 0) {
1754        return BAD_VALUE;
1755    }
1756    Mutex::Autolock _l(mLock);
1757    if (initCheck() != NO_ERROR) {
1758        return INVALID_OPERATION;
1759    }
1760    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1761
1762    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1763}
1764
1765uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1766{
1767    Mutex::Autolock _l(mLock);
1768    uint32_t result = 0;
1769    if (getEffectChain_l(sessionId) != 0) {
1770        result = EFFECT_SESSION;
1771    }
1772
1773    for (size_t i = 0; i < mTracks.size(); ++i) {
1774        sp<Track> track = mTracks[i];
1775        if (sessionId == track->sessionId() &&
1776                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1777            result |= TRACK_SESSION;
1778            break;
1779        }
1780    }
1781
1782    return result;
1783}
1784
1785uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1786{
1787    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1788    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1789    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1790        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1791    }
1792    for (size_t i = 0; i < mTracks.size(); i++) {
1793        sp<Track> track = mTracks[i];
1794        if (sessionId == track->sessionId() &&
1795                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1796            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1797        }
1798    }
1799    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1800}
1801
1802
1803AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1804{
1805    Mutex::Autolock _l(mLock);
1806    return mOutput;
1807}
1808
1809AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1810{
1811    Mutex::Autolock _l(mLock);
1812    AudioStreamOut *output = mOutput;
1813    mOutput = NULL;
1814    return output;
1815}
1816
1817// this method must always be called either with ThreadBase mLock held or inside the thread loop
1818audio_stream_t* AudioFlinger::PlaybackThread::stream()
1819{
1820    if (mOutput == NULL) {
1821        return NULL;
1822    }
1823    return &mOutput->stream->common;
1824}
1825
1826uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
1827{
1828    // A2DP output latency is not due only to buffering capacity. It also reflects encoding,
1829    // decoding and transfer time. So sleeping for half of the latency would likely cause
1830    // underruns
1831    if (audio_is_a2dp_device((audio_devices_t)mDevice)) {
1832        return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000);
1833    } else {
1834        return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
1835    }
1836}
1837
1838// ----------------------------------------------------------------------------
1839
1840AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1841    :   PlaybackThread(audioFlinger, output, id, device),
1842        mAudioMixer(NULL)
1843{
1844    mType = ThreadBase::MIXER;
1845    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1846
1847    // FIXME - Current mixer implementation only supports stereo output
1848    if (mChannelCount == 1) {
1849        LOGE("Invalid audio hardware channel count");
1850    }
1851}
1852
1853AudioFlinger::MixerThread::~MixerThread()
1854{
1855    delete mAudioMixer;
1856}
1857
1858bool AudioFlinger::MixerThread::threadLoop()
1859{
1860    Vector< sp<Track> > tracksToRemove;
1861    uint32_t mixerStatus = MIXER_IDLE;
1862    nsecs_t standbyTime = systemTime();
1863    size_t mixBufferSize = mFrameCount * mFrameSize;
1864    // FIXME: Relaxed timing because of a certain device that can't meet latency
1865    // Should be reduced to 2x after the vendor fixes the driver issue
1866    // increase threshold again due to low power audio mode. The way this warning threshold is
1867    // calculated and its usefulness should be reconsidered anyway.
1868    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1869    nsecs_t lastWarning = 0;
1870    bool longStandbyExit = false;
1871    uint32_t activeSleepTime = activeSleepTimeUs();
1872    uint32_t idleSleepTime = idleSleepTimeUs();
1873    uint32_t sleepTime = idleSleepTime;
1874    uint32_t sleepTimeShift = 0;
1875    Vector< sp<EffectChain> > effectChains;
1876#ifdef DEBUG_CPU_USAGE
1877    ThreadCpuUsage cpu;
1878    const CentralTendencyStatistics& stats = cpu.statistics();
1879#endif
1880
1881    acquireWakeLock();
1882
1883    while (!exitPending())
1884    {
1885#ifdef DEBUG_CPU_USAGE
1886        cpu.sampleAndEnable();
1887        unsigned n = stats.n();
1888        // cpu.elapsed() is expensive, so don't call it every loop
1889        if ((n & 127) == 1) {
1890            long long elapsed = cpu.elapsed();
1891            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1892                double perLoop = elapsed / (double) n;
1893                double perLoop100 = perLoop * 0.01;
1894                double mean = stats.mean();
1895                double stddev = stats.stddev();
1896                double minimum = stats.minimum();
1897                double maximum = stats.maximum();
1898                cpu.resetStatistics();
1899                LOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
1900                        elapsed * .000000001, n, perLoop * .000001,
1901                        mean * .001,
1902                        stddev * .001,
1903                        minimum * .001,
1904                        maximum * .001,
1905                        mean / perLoop100,
1906                        stddev / perLoop100,
1907                        minimum / perLoop100,
1908                        maximum / perLoop100);
1909            }
1910        }
1911#endif
1912        processConfigEvents();
1913
1914        mixerStatus = MIXER_IDLE;
1915        { // scope for mLock
1916
1917            Mutex::Autolock _l(mLock);
1918
1919            if (checkForNewParameters_l()) {
1920                mixBufferSize = mFrameCount * mFrameSize;
1921                // FIXME: Relaxed timing because of a certain device that can't meet latency
1922                // Should be reduced to 2x after the vendor fixes the driver issue
1923                // increase threshold again due to low power audio mode. The way this warning
1924                // threshold is calculated and its usefulness should be reconsidered anyway.
1925                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1926                activeSleepTime = activeSleepTimeUs();
1927                idleSleepTime = idleSleepTimeUs();
1928            }
1929
1930            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1931
1932            // put audio hardware into standby after short delay
1933            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1934                        mSuspended) {
1935                if (!mStandby) {
1936                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1937                    mOutput->stream->common.standby(&mOutput->stream->common);
1938                    mStandby = true;
1939                    mBytesWritten = 0;
1940                }
1941
1942                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1943                    // we're about to wait, flush the binder command buffer
1944                    IPCThreadState::self()->flushCommands();
1945
1946                    if (exitPending()) break;
1947
1948                    releaseWakeLock_l();
1949                    // wait until we have something to do...
1950                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1951                    mWaitWorkCV.wait(mLock);
1952                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1953                    acquireWakeLock_l();
1954
1955                    if (mMasterMute == false) {
1956                        char value[PROPERTY_VALUE_MAX];
1957                        property_get("ro.audio.silent", value, "0");
1958                        if (atoi(value)) {
1959                            LOGD("Silence is golden");
1960                            setMasterMute(true);
1961                        }
1962                    }
1963
1964                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1965                    sleepTime = idleSleepTime;
1966                    sleepTimeShift = 0;
1967                    continue;
1968                }
1969            }
1970
1971            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1972
1973            // prevent any changes in effect chain list and in each effect chain
1974            // during mixing and effect process as the audio buffers could be deleted
1975            // or modified if an effect is created or deleted
1976            lockEffectChains_l(effectChains);
1977        }
1978
1979        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1980            // mix buffers...
1981            mAudioMixer->process();
1982            sleepTime = 0;
1983            // increase sleep time progressively when application underrun condition clears
1984            if (sleepTimeShift > 0) {
1985                sleepTimeShift--;
1986            }
1987            standbyTime = systemTime() + kStandbyTimeInNsecs;
1988            //TODO: delay standby when effects have a tail
1989        } else {
1990            // If no tracks are ready, sleep once for the duration of an output
1991            // buffer size, then write 0s to the output
1992            if (sleepTime == 0) {
1993                if (mixerStatus == MIXER_TRACKS_ENABLED) {
1994                    sleepTime = activeSleepTime >> sleepTimeShift;
1995                    if (sleepTime < kMinThreadSleepTimeUs) {
1996                        sleepTime = kMinThreadSleepTimeUs;
1997                    }
1998                    // reduce sleep time in case of consecutive application underruns to avoid
1999                    // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
2000                    // duration we would end up writing less data than needed by the audio HAL if
2001                    // the condition persists.
2002                    if (sleepTimeShift < kMaxThreadSleepTimeShift) {
2003                        sleepTimeShift++;
2004                    }
2005                } else {
2006                    sleepTime = idleSleepTime;
2007                }
2008            } else if (mBytesWritten != 0 ||
2009                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
2010                memset (mMixBuffer, 0, mixBufferSize);
2011                sleepTime = 0;
2012                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
2013            }
2014            // TODO add standby time extension fct of effect tail
2015        }
2016
2017        if (mSuspended) {
2018            sleepTime = suspendSleepTimeUs();
2019        }
2020        // sleepTime == 0 means we must write to audio hardware
2021        if (sleepTime == 0) {
2022            for (size_t i = 0; i < effectChains.size(); i ++) {
2023                effectChains[i]->process_l();
2024            }
2025            // enable changes in effect chain
2026            unlockEffectChains(effectChains);
2027            mLastWriteTime = systemTime();
2028            mInWrite = true;
2029            mBytesWritten += mixBufferSize;
2030
2031            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2032            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2033            mNumWrites++;
2034            mInWrite = false;
2035            nsecs_t now = systemTime();
2036            nsecs_t delta = now - mLastWriteTime;
2037            if (!mStandby && delta > maxPeriod) {
2038                mNumDelayedWrites++;
2039                if ((now - lastWarning) > kWarningThrottleNs) {
2040                    LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2041                            ns2ms(delta), mNumDelayedWrites, this);
2042                    lastWarning = now;
2043                }
2044                if (mStandby) {
2045                    longStandbyExit = true;
2046                }
2047            }
2048            mStandby = false;
2049        } else {
2050            // enable changes in effect chain
2051            unlockEffectChains(effectChains);
2052            usleep(sleepTime);
2053        }
2054
2055        // finally let go of all our tracks, without the lock held
2056        // since we can't guarantee the destructors won't acquire that
2057        // same lock.
2058        tracksToRemove.clear();
2059
2060        // Effect chains will be actually deleted here if they were removed from
2061        // mEffectChains list during mixing or effects processing
2062        effectChains.clear();
2063    }
2064
2065    if (!mStandby) {
2066        mOutput->stream->common.standby(&mOutput->stream->common);
2067    }
2068
2069    releaseWakeLock();
2070
2071    ALOGV("MixerThread %p exiting", this);
2072    return false;
2073}
2074
2075// prepareTracks_l() must be called with ThreadBase::mLock held
2076uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2077{
2078
2079    uint32_t mixerStatus = MIXER_IDLE;
2080    // find out which tracks need to be processed
2081    size_t count = activeTracks.size();
2082    size_t mixedTracks = 0;
2083    size_t tracksWithEffect = 0;
2084
2085    float masterVolume = mMasterVolume;
2086    bool  masterMute = mMasterMute;
2087
2088    if (masterMute) {
2089        masterVolume = 0;
2090    }
2091    // Delegate master volume control to effect in output mix effect chain if needed
2092    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2093    if (chain != 0) {
2094        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2095        chain->setVolume_l(&v, &v);
2096        masterVolume = (float)((v + (1 << 23)) >> 24);
2097        chain.clear();
2098    }
2099
2100    for (size_t i=0 ; i<count ; i++) {
2101        sp<Track> t = activeTracks[i].promote();
2102        if (t == 0) continue;
2103
2104        Track* const track = t.get();
2105        audio_track_cblk_t* cblk = track->cblk();
2106
2107        // The first time a track is added we wait
2108        // for all its buffers to be filled before processing it
2109        mAudioMixer->setActiveTrack(track->name());
2110        // make sure that we have enough frames to mix one full buffer.
2111        // enforce this condition only once to enable draining the buffer in case the client
2112        // app does not call stop() and relies on underrun to stop:
2113        // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed
2114        // during last round
2115        uint32_t minFrames = 1;
2116        if (!track->isStopped() && !track->isPausing() &&
2117                (track->mRetryCount >= kMaxTrackRetries)) {
2118            if (t->sampleRate() == (int)mSampleRate) {
2119                minFrames = mFrameCount;
2120            } else {
2121                minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1;
2122            }
2123        }
2124        if ((cblk->framesReady() >= minFrames) && track->isReady() &&
2125                !track->isPaused() && !track->isTerminated())
2126        {
2127            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
2128
2129            mixedTracks++;
2130
2131            // track->mainBuffer() != mMixBuffer means there is an effect chain
2132            // connected to the track
2133            chain.clear();
2134            if (track->mainBuffer() != mMixBuffer) {
2135                chain = getEffectChain_l(track->sessionId());
2136                // Delegate volume control to effect in track effect chain if needed
2137                if (chain != 0) {
2138                    tracksWithEffect++;
2139                } else {
2140                    LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
2141                            track->name(), track->sessionId());
2142                }
2143            }
2144
2145
2146            int param = AudioMixer::VOLUME;
2147            if (track->mFillingUpStatus == Track::FS_FILLED) {
2148                // no ramp for the first volume setting
2149                track->mFillingUpStatus = Track::FS_ACTIVE;
2150                if (track->mState == TrackBase::RESUMING) {
2151                    track->mState = TrackBase::ACTIVE;
2152                    param = AudioMixer::RAMP_VOLUME;
2153                }
2154                mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2155            } else if (cblk->server != 0) {
2156                // If the track is stopped before the first frame was mixed,
2157                // do not apply ramp
2158                param = AudioMixer::RAMP_VOLUME;
2159            }
2160
2161            // compute volume for this track
2162            uint32_t vl, vr, va;
2163            if (track->isMuted() || track->isPausing() ||
2164                mStreamTypes[track->type()].mute) {
2165                vl = vr = va = 0;
2166                if (track->isPausing()) {
2167                    track->setPaused();
2168                }
2169            } else {
2170
2171                // read original volumes with volume control
2172                float typeVolume = mStreamTypes[track->type()].volume;
2173                float v = masterVolume * typeVolume;
2174                vl = (uint32_t)(v * cblk->volume[0]) << 12;
2175                vr = (uint32_t)(v * cblk->volume[1]) << 12;
2176
2177                va = (uint32_t)(v * cblk->sendLevel);
2178            }
2179            // Delegate volume control to effect in track effect chain if needed
2180            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2181                // Do not ramp volume if volume is controlled by effect
2182                param = AudioMixer::VOLUME;
2183                track->mHasVolumeController = true;
2184            } else {
2185                // force no volume ramp when volume controller was just disabled or removed
2186                // from effect chain to avoid volume spike
2187                if (track->mHasVolumeController) {
2188                    param = AudioMixer::VOLUME;
2189                }
2190                track->mHasVolumeController = false;
2191            }
2192
2193            // Convert volumes from 8.24 to 4.12 format
2194            int16_t left, right, aux;
2195            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2196            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2197            left = int16_t(v_clamped);
2198            v_clamped = (vr + (1 << 11)) >> 12;
2199            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2200            right = int16_t(v_clamped);
2201
2202            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2203            aux = int16_t(va);
2204
2205            // XXX: these things DON'T need to be done each time
2206            mAudioMixer->setBufferProvider(track);
2207            mAudioMixer->enable();
2208
2209            mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
2210            mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
2211            mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
2212            mAudioMixer->setParameter(
2213                AudioMixer::TRACK,
2214                AudioMixer::FORMAT, (void *)track->format());
2215            mAudioMixer->setParameter(
2216                AudioMixer::TRACK,
2217                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2218            mAudioMixer->setParameter(
2219                AudioMixer::RESAMPLE,
2220                AudioMixer::SAMPLE_RATE,
2221                (void *)(cblk->sampleRate));
2222            mAudioMixer->setParameter(
2223                AudioMixer::TRACK,
2224                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2225            mAudioMixer->setParameter(
2226                AudioMixer::TRACK,
2227                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2228
2229            // reset retry count
2230            track->mRetryCount = kMaxTrackRetries;
2231            mixerStatus = MIXER_TRACKS_READY;
2232        } else {
2233            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
2234            if (track->isStopped()) {
2235                track->reset();
2236            }
2237            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2238                // We have consumed all the buffers of this track.
2239                // Remove it from the list of active tracks.
2240                tracksToRemove->add(track);
2241            } else {
2242                // No buffers for this track. Give it a few chances to
2243                // fill a buffer, then remove it from active list.
2244                if (--(track->mRetryCount) <= 0) {
2245                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
2246                    tracksToRemove->add(track);
2247                    // indicate to client process that the track was disabled because of underrun
2248                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2249                } else if (mixerStatus != MIXER_TRACKS_READY) {
2250                    mixerStatus = MIXER_TRACKS_ENABLED;
2251                }
2252            }
2253            mAudioMixer->disable();
2254        }
2255    }
2256
2257    // remove all the tracks that need to be...
2258    count = tracksToRemove->size();
2259    if (UNLIKELY(count)) {
2260        for (size_t i=0 ; i<count ; i++) {
2261            const sp<Track>& track = tracksToRemove->itemAt(i);
2262            mActiveTracks.remove(track);
2263            if (track->mainBuffer() != mMixBuffer) {
2264                chain = getEffectChain_l(track->sessionId());
2265                if (chain != 0) {
2266                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2267                    chain->decActiveTrackCnt();
2268                }
2269            }
2270            if (track->isTerminated()) {
2271                removeTrack_l(track);
2272            }
2273        }
2274    }
2275
2276    // mix buffer must be cleared if all tracks are connected to an
2277    // effect chain as in this case the mixer will not write to
2278    // mix buffer and track effects will accumulate into it
2279    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2280        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2281    }
2282
2283    return mixerStatus;
2284}
2285
2286void AudioFlinger::MixerThread::invalidateTracks(int streamType)
2287{
2288    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2289            this,  streamType, mTracks.size());
2290    Mutex::Autolock _l(mLock);
2291
2292    size_t size = mTracks.size();
2293    for (size_t i = 0; i < size; i++) {
2294        sp<Track> t = mTracks[i];
2295        if (t->type() == streamType) {
2296            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2297            t->mCblk->cv.signal();
2298        }
2299    }
2300}
2301
2302void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid)
2303{
2304    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2305            this,  streamType, valid);
2306    Mutex::Autolock _l(mLock);
2307
2308    mStreamTypes[streamType].valid = valid;
2309}
2310
2311// getTrackName_l() must be called with ThreadBase::mLock held
2312int AudioFlinger::MixerThread::getTrackName_l()
2313{
2314    return mAudioMixer->getTrackName();
2315}
2316
2317// deleteTrackName_l() must be called with ThreadBase::mLock held
2318void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2319{
2320    ALOGV("remove track (%d) and delete from mixer", name);
2321    mAudioMixer->deleteTrackName(name);
2322}
2323
2324// checkForNewParameters_l() must be called with ThreadBase::mLock held
2325bool AudioFlinger::MixerThread::checkForNewParameters_l()
2326{
2327    bool reconfig = false;
2328
2329    while (!mNewParameters.isEmpty()) {
2330        status_t status = NO_ERROR;
2331        String8 keyValuePair = mNewParameters[0];
2332        AudioParameter param = AudioParameter(keyValuePair);
2333        int value;
2334
2335        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2336            reconfig = true;
2337        }
2338        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2339            if (value != AUDIO_FORMAT_PCM_16_BIT) {
2340                status = BAD_VALUE;
2341            } else {
2342                reconfig = true;
2343            }
2344        }
2345        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2346            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2347                status = BAD_VALUE;
2348            } else {
2349                reconfig = true;
2350            }
2351        }
2352        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2353            // do not accept frame count changes if tracks are open as the track buffer
2354            // size depends on frame count and correct behavior would not be guaranteed
2355            // if frame count is changed after track creation
2356            if (!mTracks.isEmpty()) {
2357                status = INVALID_OPERATION;
2358            } else {
2359                reconfig = true;
2360            }
2361        }
2362        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2363            // when changing the audio output device, call addBatteryData to notify
2364            // the change
2365            if ((int)mDevice != value) {
2366                uint32_t params = 0;
2367                // check whether speaker is on
2368                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2369                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2370                }
2371
2372                int deviceWithoutSpeaker
2373                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2374                // check if any other device (except speaker) is on
2375                if (value & deviceWithoutSpeaker ) {
2376                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2377                }
2378
2379                if (params != 0) {
2380                    addBatteryData(params);
2381                }
2382            }
2383
2384            // forward device change to effects that have requested to be
2385            // aware of attached audio device.
2386            mDevice = (uint32_t)value;
2387            for (size_t i = 0; i < mEffectChains.size(); i++) {
2388                mEffectChains[i]->setDevice_l(mDevice);
2389            }
2390        }
2391
2392        if (status == NO_ERROR) {
2393            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2394                                                    keyValuePair.string());
2395            if (!mStandby && status == INVALID_OPERATION) {
2396               mOutput->stream->common.standby(&mOutput->stream->common);
2397               mStandby = true;
2398               mBytesWritten = 0;
2399               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2400                                                       keyValuePair.string());
2401            }
2402            if (status == NO_ERROR && reconfig) {
2403                delete mAudioMixer;
2404                readOutputParameters();
2405                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2406                for (size_t i = 0; i < mTracks.size() ; i++) {
2407                    int name = getTrackName_l();
2408                    if (name < 0) break;
2409                    mTracks[i]->mName = name;
2410                    // limit track sample rate to 2 x new output sample rate
2411                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2412                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2413                    }
2414                }
2415                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2416            }
2417        }
2418
2419        mNewParameters.removeAt(0);
2420
2421        mParamStatus = status;
2422        mParamCond.signal();
2423        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2424        // already timed out waiting for the status and will never signal the condition.
2425        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2426    }
2427    return reconfig;
2428}
2429
2430status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2431{
2432    const size_t SIZE = 256;
2433    char buffer[SIZE];
2434    String8 result;
2435
2436    PlaybackThread::dumpInternals(fd, args);
2437
2438    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2439    result.append(buffer);
2440    write(fd, result.string(), result.size());
2441    return NO_ERROR;
2442}
2443
2444uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2445{
2446    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2447}
2448
2449uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2450{
2451    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2452}
2453
2454// ----------------------------------------------------------------------------
2455AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2456    :   PlaybackThread(audioFlinger, output, id, device)
2457{
2458    mType = ThreadBase::DIRECT;
2459}
2460
2461AudioFlinger::DirectOutputThread::~DirectOutputThread()
2462{
2463}
2464
2465static inline
2466int32_t mul(int16_t in, int16_t v)
2467{
2468#if defined(__arm__) && !defined(__thumb__)
2469    int32_t out;
2470    asm( "smulbb %[out], %[in], %[v] \n"
2471         : [out]"=r"(out)
2472         : [in]"%r"(in), [v]"r"(v)
2473         : );
2474    return out;
2475#else
2476    return in * int32_t(v);
2477#endif
2478}
2479
2480void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2481{
2482    // Do not apply volume on compressed audio
2483    if (!audio_is_linear_pcm(mFormat)) {
2484        return;
2485    }
2486
2487    // convert to signed 16 bit before volume calculation
2488    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2489        size_t count = mFrameCount * mChannelCount;
2490        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2491        int16_t *dst = mMixBuffer + count-1;
2492        while(count--) {
2493            *dst-- = (int16_t)(*src--^0x80) << 8;
2494        }
2495    }
2496
2497    size_t frameCount = mFrameCount;
2498    int16_t *out = mMixBuffer;
2499    if (ramp) {
2500        if (mChannelCount == 1) {
2501            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2502            int32_t vlInc = d / (int32_t)frameCount;
2503            int32_t vl = ((int32_t)mLeftVolShort << 16);
2504            do {
2505                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2506                out++;
2507                vl += vlInc;
2508            } while (--frameCount);
2509
2510        } else {
2511            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2512            int32_t vlInc = d / (int32_t)frameCount;
2513            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2514            int32_t vrInc = d / (int32_t)frameCount;
2515            int32_t vl = ((int32_t)mLeftVolShort << 16);
2516            int32_t vr = ((int32_t)mRightVolShort << 16);
2517            do {
2518                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2519                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2520                out += 2;
2521                vl += vlInc;
2522                vr += vrInc;
2523            } while (--frameCount);
2524        }
2525    } else {
2526        if (mChannelCount == 1) {
2527            do {
2528                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2529                out++;
2530            } while (--frameCount);
2531        } else {
2532            do {
2533                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2534                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2535                out += 2;
2536            } while (--frameCount);
2537        }
2538    }
2539
2540    // convert back to unsigned 8 bit after volume calculation
2541    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2542        size_t count = mFrameCount * mChannelCount;
2543        int16_t *src = mMixBuffer;
2544        uint8_t *dst = (uint8_t *)mMixBuffer;
2545        while(count--) {
2546            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2547        }
2548    }
2549
2550    mLeftVolShort = leftVol;
2551    mRightVolShort = rightVol;
2552}
2553
2554bool AudioFlinger::DirectOutputThread::threadLoop()
2555{
2556    uint32_t mixerStatus = MIXER_IDLE;
2557    sp<Track> trackToRemove;
2558    sp<Track> activeTrack;
2559    nsecs_t standbyTime = systemTime();
2560    int8_t *curBuf;
2561    size_t mixBufferSize = mFrameCount*mFrameSize;
2562    uint32_t activeSleepTime = activeSleepTimeUs();
2563    uint32_t idleSleepTime = idleSleepTimeUs();
2564    uint32_t sleepTime = idleSleepTime;
2565    // use shorter standby delay as on normal output to release
2566    // hardware resources as soon as possible
2567    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2568
2569    acquireWakeLock();
2570
2571    while (!exitPending())
2572    {
2573        bool rampVolume;
2574        uint16_t leftVol;
2575        uint16_t rightVol;
2576        Vector< sp<EffectChain> > effectChains;
2577
2578        processConfigEvents();
2579
2580        mixerStatus = MIXER_IDLE;
2581
2582        { // scope for the mLock
2583
2584            Mutex::Autolock _l(mLock);
2585
2586            if (checkForNewParameters_l()) {
2587                mixBufferSize = mFrameCount*mFrameSize;
2588                activeSleepTime = activeSleepTimeUs();
2589                idleSleepTime = idleSleepTimeUs();
2590                standbyDelay = microseconds(activeSleepTime*2);
2591            }
2592
2593            // put audio hardware into standby after short delay
2594            if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2595                        mSuspended) {
2596                // wait until we have something to do...
2597                if (!mStandby) {
2598                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2599                    mOutput->stream->common.standby(&mOutput->stream->common);
2600                    mStandby = true;
2601                    mBytesWritten = 0;
2602                }
2603
2604                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2605                    // we're about to wait, flush the binder command buffer
2606                    IPCThreadState::self()->flushCommands();
2607
2608                    if (exitPending()) break;
2609
2610                    releaseWakeLock_l();
2611                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2612                    mWaitWorkCV.wait(mLock);
2613                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2614                    acquireWakeLock_l();
2615
2616                    if (mMasterMute == false) {
2617                        char value[PROPERTY_VALUE_MAX];
2618                        property_get("ro.audio.silent", value, "0");
2619                        if (atoi(value)) {
2620                            LOGD("Silence is golden");
2621                            setMasterMute(true);
2622                        }
2623                    }
2624
2625                    standbyTime = systemTime() + standbyDelay;
2626                    sleepTime = idleSleepTime;
2627                    continue;
2628                }
2629            }
2630
2631            effectChains = mEffectChains;
2632
2633            // find out which tracks need to be processed
2634            if (mActiveTracks.size() != 0) {
2635                sp<Track> t = mActiveTracks[0].promote();
2636                if (t == 0) continue;
2637
2638                Track* const track = t.get();
2639                audio_track_cblk_t* cblk = track->cblk();
2640
2641                // The first time a track is added we wait
2642                // for all its buffers to be filled before processing it
2643                if (cblk->framesReady() && track->isReady() &&
2644                        !track->isPaused() && !track->isTerminated())
2645                {
2646                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2647
2648                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2649                        track->mFillingUpStatus = Track::FS_ACTIVE;
2650                        mLeftVolFloat = mRightVolFloat = 0;
2651                        mLeftVolShort = mRightVolShort = 0;
2652                        if (track->mState == TrackBase::RESUMING) {
2653                            track->mState = TrackBase::ACTIVE;
2654                            rampVolume = true;
2655                        }
2656                    } else if (cblk->server != 0) {
2657                        // If the track is stopped before the first frame was mixed,
2658                        // do not apply ramp
2659                        rampVolume = true;
2660                    }
2661                    // compute volume for this track
2662                    float left, right;
2663                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2664                        mStreamTypes[track->type()].mute) {
2665                        left = right = 0;
2666                        if (track->isPausing()) {
2667                            track->setPaused();
2668                        }
2669                    } else {
2670                        float typeVolume = mStreamTypes[track->type()].volume;
2671                        float v = mMasterVolume * typeVolume;
2672                        float v_clamped = v * cblk->volume[0];
2673                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2674                        left = v_clamped/MAX_GAIN;
2675                        v_clamped = v * cblk->volume[1];
2676                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2677                        right = v_clamped/MAX_GAIN;
2678                    }
2679
2680                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2681                        mLeftVolFloat = left;
2682                        mRightVolFloat = right;
2683
2684                        // If audio HAL implements volume control,
2685                        // force software volume to nominal value
2686                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2687                            left = 1.0f;
2688                            right = 1.0f;
2689                        }
2690
2691                        // Convert volumes from float to 8.24
2692                        uint32_t vl = (uint32_t)(left * (1 << 24));
2693                        uint32_t vr = (uint32_t)(right * (1 << 24));
2694
2695                        // Delegate volume control to effect in track effect chain if needed
2696                        // only one effect chain can be present on DirectOutputThread, so if
2697                        // there is one, the track is connected to it
2698                        if (!effectChains.isEmpty()) {
2699                            // Do not ramp volume if volume is controlled by effect
2700                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2701                                rampVolume = false;
2702                            }
2703                        }
2704
2705                        // Convert volumes from 8.24 to 4.12 format
2706                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2707                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2708                        leftVol = (uint16_t)v_clamped;
2709                        v_clamped = (vr + (1 << 11)) >> 12;
2710                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2711                        rightVol = (uint16_t)v_clamped;
2712                    } else {
2713                        leftVol = mLeftVolShort;
2714                        rightVol = mRightVolShort;
2715                        rampVolume = false;
2716                    }
2717
2718                    // reset retry count
2719                    track->mRetryCount = kMaxTrackRetriesDirect;
2720                    activeTrack = t;
2721                    mixerStatus = MIXER_TRACKS_READY;
2722                } else {
2723                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2724                    if (track->isStopped()) {
2725                        track->reset();
2726                    }
2727                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2728                        // We have consumed all the buffers of this track.
2729                        // Remove it from the list of active tracks.
2730                        trackToRemove = track;
2731                    } else {
2732                        // No buffers for this track. Give it a few chances to
2733                        // fill a buffer, then remove it from active list.
2734                        if (--(track->mRetryCount) <= 0) {
2735                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2736                            trackToRemove = track;
2737                        } else {
2738                            mixerStatus = MIXER_TRACKS_ENABLED;
2739                        }
2740                    }
2741                }
2742            }
2743
2744            // remove all the tracks that need to be...
2745            if (UNLIKELY(trackToRemove != 0)) {
2746                mActiveTracks.remove(trackToRemove);
2747                if (!effectChains.isEmpty()) {
2748                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2749                            trackToRemove->sessionId());
2750                    effectChains[0]->decActiveTrackCnt();
2751                }
2752                if (trackToRemove->isTerminated()) {
2753                    removeTrack_l(trackToRemove);
2754                }
2755            }
2756
2757            lockEffectChains_l(effectChains);
2758       }
2759
2760        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2761            AudioBufferProvider::Buffer buffer;
2762            size_t frameCount = mFrameCount;
2763            curBuf = (int8_t *)mMixBuffer;
2764            // output audio to hardware
2765            while (frameCount) {
2766                buffer.frameCount = frameCount;
2767                activeTrack->getNextBuffer(&buffer);
2768                if (UNLIKELY(buffer.raw == NULL)) {
2769                    memset(curBuf, 0, frameCount * mFrameSize);
2770                    break;
2771                }
2772                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2773                frameCount -= buffer.frameCount;
2774                curBuf += buffer.frameCount * mFrameSize;
2775                activeTrack->releaseBuffer(&buffer);
2776            }
2777            sleepTime = 0;
2778            standbyTime = systemTime() + standbyDelay;
2779        } else {
2780            if (sleepTime == 0) {
2781                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2782                    sleepTime = activeSleepTime;
2783                } else {
2784                    sleepTime = idleSleepTime;
2785                }
2786            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2787                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2788                sleepTime = 0;
2789            }
2790        }
2791
2792        if (mSuspended) {
2793            sleepTime = suspendSleepTimeUs();
2794        }
2795        // sleepTime == 0 means we must write to audio hardware
2796        if (sleepTime == 0) {
2797            if (mixerStatus == MIXER_TRACKS_READY) {
2798                applyVolume(leftVol, rightVol, rampVolume);
2799            }
2800            for (size_t i = 0; i < effectChains.size(); i ++) {
2801                effectChains[i]->process_l();
2802            }
2803            unlockEffectChains(effectChains);
2804
2805            mLastWriteTime = systemTime();
2806            mInWrite = true;
2807            mBytesWritten += mixBufferSize;
2808            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2809            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2810            mNumWrites++;
2811            mInWrite = false;
2812            mStandby = false;
2813        } else {
2814            unlockEffectChains(effectChains);
2815            usleep(sleepTime);
2816        }
2817
2818        // finally let go of removed track, without the lock held
2819        // since we can't guarantee the destructors won't acquire that
2820        // same lock.
2821        trackToRemove.clear();
2822        activeTrack.clear();
2823
2824        // Effect chains will be actually deleted here if they were removed from
2825        // mEffectChains list during mixing or effects processing
2826        effectChains.clear();
2827    }
2828
2829    if (!mStandby) {
2830        mOutput->stream->common.standby(&mOutput->stream->common);
2831    }
2832
2833    releaseWakeLock();
2834
2835    ALOGV("DirectOutputThread %p exiting", this);
2836    return false;
2837}
2838
2839// getTrackName_l() must be called with ThreadBase::mLock held
2840int AudioFlinger::DirectOutputThread::getTrackName_l()
2841{
2842    return 0;
2843}
2844
2845// deleteTrackName_l() must be called with ThreadBase::mLock held
2846void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2847{
2848}
2849
2850// checkForNewParameters_l() must be called with ThreadBase::mLock held
2851bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2852{
2853    bool reconfig = false;
2854
2855    while (!mNewParameters.isEmpty()) {
2856        status_t status = NO_ERROR;
2857        String8 keyValuePair = mNewParameters[0];
2858        AudioParameter param = AudioParameter(keyValuePair);
2859        int value;
2860
2861        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2862            // do not accept frame count changes if tracks are open as the track buffer
2863            // size depends on frame count and correct behavior would not be garantied
2864            // if frame count is changed after track creation
2865            if (!mTracks.isEmpty()) {
2866                status = INVALID_OPERATION;
2867            } else {
2868                reconfig = true;
2869            }
2870        }
2871        if (status == NO_ERROR) {
2872            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2873                                                    keyValuePair.string());
2874            if (!mStandby && status == INVALID_OPERATION) {
2875               mOutput->stream->common.standby(&mOutput->stream->common);
2876               mStandby = true;
2877               mBytesWritten = 0;
2878               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2879                                                       keyValuePair.string());
2880            }
2881            if (status == NO_ERROR && reconfig) {
2882                readOutputParameters();
2883                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2884            }
2885        }
2886
2887        mNewParameters.removeAt(0);
2888
2889        mParamStatus = status;
2890        mParamCond.signal();
2891        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2892        // already timed out waiting for the status and will never signal the condition.
2893        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2894    }
2895    return reconfig;
2896}
2897
2898uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2899{
2900    uint32_t time;
2901    if (audio_is_linear_pcm(mFormat)) {
2902        time = PlaybackThread::activeSleepTimeUs();
2903    } else {
2904        time = 10000;
2905    }
2906    return time;
2907}
2908
2909uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2910{
2911    uint32_t time;
2912    if (audio_is_linear_pcm(mFormat)) {
2913        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2914    } else {
2915        time = 10000;
2916    }
2917    return time;
2918}
2919
2920uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2921{
2922    uint32_t time;
2923    if (audio_is_linear_pcm(mFormat)) {
2924        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2925    } else {
2926        time = 10000;
2927    }
2928    return time;
2929}
2930
2931
2932// ----------------------------------------------------------------------------
2933
2934AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2935    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2936{
2937    mType = ThreadBase::DUPLICATING;
2938    addOutputTrack(mainThread);
2939}
2940
2941AudioFlinger::DuplicatingThread::~DuplicatingThread()
2942{
2943    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2944        mOutputTracks[i]->destroy();
2945    }
2946    mOutputTracks.clear();
2947}
2948
2949bool AudioFlinger::DuplicatingThread::threadLoop()
2950{
2951    Vector< sp<Track> > tracksToRemove;
2952    uint32_t mixerStatus = MIXER_IDLE;
2953    nsecs_t standbyTime = systemTime();
2954    size_t mixBufferSize = mFrameCount*mFrameSize;
2955    SortedVector< sp<OutputTrack> > outputTracks;
2956    uint32_t writeFrames = 0;
2957    uint32_t activeSleepTime = activeSleepTimeUs();
2958    uint32_t idleSleepTime = idleSleepTimeUs();
2959    uint32_t sleepTime = idleSleepTime;
2960    Vector< sp<EffectChain> > effectChains;
2961
2962    acquireWakeLock();
2963
2964    while (!exitPending())
2965    {
2966        processConfigEvents();
2967
2968        mixerStatus = MIXER_IDLE;
2969        { // scope for the mLock
2970
2971            Mutex::Autolock _l(mLock);
2972
2973            if (checkForNewParameters_l()) {
2974                mixBufferSize = mFrameCount*mFrameSize;
2975                updateWaitTime();
2976                activeSleepTime = activeSleepTimeUs();
2977                idleSleepTime = idleSleepTimeUs();
2978            }
2979
2980            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
2981
2982            for (size_t i = 0; i < mOutputTracks.size(); i++) {
2983                outputTracks.add(mOutputTracks[i]);
2984            }
2985
2986            // put audio hardware into standby after short delay
2987            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
2988                         mSuspended) {
2989                if (!mStandby) {
2990                    for (size_t i = 0; i < outputTracks.size(); i++) {
2991                        outputTracks[i]->stop();
2992                    }
2993                    mStandby = true;
2994                    mBytesWritten = 0;
2995                }
2996
2997                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
2998                    // we're about to wait, flush the binder command buffer
2999                    IPCThreadState::self()->flushCommands();
3000                    outputTracks.clear();
3001
3002                    if (exitPending()) break;
3003
3004                    releaseWakeLock_l();
3005                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3006                    mWaitWorkCV.wait(mLock);
3007                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3008                    acquireWakeLock_l();
3009
3010                    if (mMasterMute == false) {
3011                        char value[PROPERTY_VALUE_MAX];
3012                        property_get("ro.audio.silent", value, "0");
3013                        if (atoi(value)) {
3014                            LOGD("Silence is golden");
3015                            setMasterMute(true);
3016                        }
3017                    }
3018
3019                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3020                    sleepTime = idleSleepTime;
3021                    continue;
3022                }
3023            }
3024
3025            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3026
3027            // prevent any changes in effect chain list and in each effect chain
3028            // during mixing and effect process as the audio buffers could be deleted
3029            // or modified if an effect is created or deleted
3030            lockEffectChains_l(effectChains);
3031        }
3032
3033        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3034            // mix buffers...
3035            if (outputsReady(outputTracks)) {
3036                mAudioMixer->process();
3037            } else {
3038                memset(mMixBuffer, 0, mixBufferSize);
3039            }
3040            sleepTime = 0;
3041            writeFrames = mFrameCount;
3042        } else {
3043            if (sleepTime == 0) {
3044                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3045                    sleepTime = activeSleepTime;
3046                } else {
3047                    sleepTime = idleSleepTime;
3048                }
3049            } else if (mBytesWritten != 0) {
3050                // flush remaining overflow buffers in output tracks
3051                for (size_t i = 0; i < outputTracks.size(); i++) {
3052                    if (outputTracks[i]->isActive()) {
3053                        sleepTime = 0;
3054                        writeFrames = 0;
3055                        memset(mMixBuffer, 0, mixBufferSize);
3056                        break;
3057                    }
3058                }
3059            }
3060        }
3061
3062        if (mSuspended) {
3063            sleepTime = suspendSleepTimeUs();
3064        }
3065        // sleepTime == 0 means we must write to audio hardware
3066        if (sleepTime == 0) {
3067            for (size_t i = 0; i < effectChains.size(); i ++) {
3068                effectChains[i]->process_l();
3069            }
3070            // enable changes in effect chain
3071            unlockEffectChains(effectChains);
3072
3073            standbyTime = systemTime() + kStandbyTimeInNsecs;
3074            for (size_t i = 0; i < outputTracks.size(); i++) {
3075                outputTracks[i]->write(mMixBuffer, writeFrames);
3076            }
3077            mStandby = false;
3078            mBytesWritten += mixBufferSize;
3079        } else {
3080            // enable changes in effect chain
3081            unlockEffectChains(effectChains);
3082            usleep(sleepTime);
3083        }
3084
3085        // finally let go of all our tracks, without the lock held
3086        // since we can't guarantee the destructors won't acquire that
3087        // same lock.
3088        tracksToRemove.clear();
3089        outputTracks.clear();
3090
3091        // Effect chains will be actually deleted here if they were removed from
3092        // mEffectChains list during mixing or effects processing
3093        effectChains.clear();
3094    }
3095
3096    releaseWakeLock();
3097
3098    return false;
3099}
3100
3101void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3102{
3103    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3104    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3105                                            this,
3106                                            mSampleRate,
3107                                            mFormat,
3108                                            mChannelMask,
3109                                            frameCount);
3110    if (outputTrack->cblk() != NULL) {
3111        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3112        mOutputTracks.add(outputTrack);
3113        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3114        updateWaitTime();
3115    }
3116}
3117
3118void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3119{
3120    Mutex::Autolock _l(mLock);
3121    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3122        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3123            mOutputTracks[i]->destroy();
3124            mOutputTracks.removeAt(i);
3125            updateWaitTime();
3126            return;
3127        }
3128    }
3129    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3130}
3131
3132void AudioFlinger::DuplicatingThread::updateWaitTime()
3133{
3134    mWaitTimeMs = UINT_MAX;
3135    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3136        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3137        if (strong != NULL) {
3138            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3139            if (waitTimeMs < mWaitTimeMs) {
3140                mWaitTimeMs = waitTimeMs;
3141            }
3142        }
3143    }
3144}
3145
3146
3147bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3148{
3149    for (size_t i = 0; i < outputTracks.size(); i++) {
3150        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3151        if (thread == 0) {
3152            LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3153            return false;
3154        }
3155        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3156        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3157            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3158            return false;
3159        }
3160    }
3161    return true;
3162}
3163
3164uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3165{
3166    return (mWaitTimeMs * 1000) / 2;
3167}
3168
3169// ----------------------------------------------------------------------------
3170
3171// TrackBase constructor must be called with AudioFlinger::mLock held
3172AudioFlinger::ThreadBase::TrackBase::TrackBase(
3173            const wp<ThreadBase>& thread,
3174            const sp<Client>& client,
3175            uint32_t sampleRate,
3176            uint32_t format,
3177            uint32_t channelMask,
3178            int frameCount,
3179            uint32_t flags,
3180            const sp<IMemory>& sharedBuffer,
3181            int sessionId)
3182    :   RefBase(),
3183        mThread(thread),
3184        mClient(client),
3185        mCblk(0),
3186        mFrameCount(0),
3187        mState(IDLE),
3188        mClientTid(-1),
3189        mFormat(format),
3190        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3191        mSessionId(sessionId)
3192{
3193    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3194
3195    // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3196   size_t size = sizeof(audio_track_cblk_t);
3197   uint8_t channelCount = popcount(channelMask);
3198   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3199   if (sharedBuffer == 0) {
3200       size += bufferSize;
3201   }
3202
3203   if (client != NULL) {
3204        mCblkMemory = client->heap()->allocate(size);
3205        if (mCblkMemory != 0) {
3206            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3207            if (mCblk) { // construct the shared structure in-place.
3208                new(mCblk) audio_track_cblk_t();
3209                // clear all buffers
3210                mCblk->frameCount = frameCount;
3211                mCblk->sampleRate = sampleRate;
3212                mChannelCount = channelCount;
3213                mChannelMask = channelMask;
3214                if (sharedBuffer == 0) {
3215                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3216                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3217                    // Force underrun condition to avoid false underrun callback until first data is
3218                    // written to buffer (other flags are cleared)
3219                    mCblk->flags = CBLK_UNDERRUN_ON;
3220                } else {
3221                    mBuffer = sharedBuffer->pointer();
3222                }
3223                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3224            }
3225        } else {
3226            LOGE("not enough memory for AudioTrack size=%u", size);
3227            client->heap()->dump("AudioTrack");
3228            return;
3229        }
3230   } else {
3231       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3232       if (mCblk) { // construct the shared structure in-place.
3233           new(mCblk) audio_track_cblk_t();
3234           // clear all buffers
3235           mCblk->frameCount = frameCount;
3236           mCblk->sampleRate = sampleRate;
3237           mChannelCount = channelCount;
3238           mChannelMask = channelMask;
3239           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3240           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3241           // Force underrun condition to avoid false underrun callback until first data is
3242           // written to buffer (other flags are cleared)
3243           mCblk->flags = CBLK_UNDERRUN_ON;
3244           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3245       }
3246   }
3247}
3248
3249AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3250{
3251    if (mCblk) {
3252        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3253        if (mClient == NULL) {
3254            delete mCblk;
3255        }
3256    }
3257    mCblkMemory.clear();            // and free the shared memory
3258    if (mClient != NULL) {
3259        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3260        mClient.clear();
3261    }
3262}
3263
3264void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3265{
3266    buffer->raw = NULL;
3267    mFrameCount = buffer->frameCount;
3268    step();
3269    buffer->frameCount = 0;
3270}
3271
3272bool AudioFlinger::ThreadBase::TrackBase::step() {
3273    bool result;
3274    audio_track_cblk_t* cblk = this->cblk();
3275
3276    result = cblk->stepServer(mFrameCount);
3277    if (!result) {
3278        ALOGV("stepServer failed acquiring cblk mutex");
3279        mFlags |= STEPSERVER_FAILED;
3280    }
3281    return result;
3282}
3283
3284void AudioFlinger::ThreadBase::TrackBase::reset() {
3285    audio_track_cblk_t* cblk = this->cblk();
3286
3287    cblk->user = 0;
3288    cblk->server = 0;
3289    cblk->userBase = 0;
3290    cblk->serverBase = 0;
3291    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3292    ALOGV("TrackBase::reset");
3293}
3294
3295sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3296{
3297    return mCblkMemory;
3298}
3299
3300int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3301    return (int)mCblk->sampleRate;
3302}
3303
3304int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3305    return (const int)mChannelCount;
3306}
3307
3308uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3309    return mChannelMask;
3310}
3311
3312void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3313    audio_track_cblk_t* cblk = this->cblk();
3314    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
3315    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
3316
3317    // Check validity of returned pointer in case the track control block would have been corrupted.
3318    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3319        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
3320        LOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3321                server %d, serverBase %d, user %d, userBase %d",
3322                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3323                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3324        return 0;
3325    }
3326
3327    return bufferStart;
3328}
3329
3330// ----------------------------------------------------------------------------
3331
3332// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3333AudioFlinger::PlaybackThread::Track::Track(
3334            const wp<ThreadBase>& thread,
3335            const sp<Client>& client,
3336            int streamType,
3337            uint32_t sampleRate,
3338            uint32_t format,
3339            uint32_t channelMask,
3340            int frameCount,
3341            const sp<IMemory>& sharedBuffer,
3342            int sessionId)
3343    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3344    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3345    mAuxEffectId(0), mHasVolumeController(false)
3346{
3347    if (mCblk != NULL) {
3348        sp<ThreadBase> baseThread = thread.promote();
3349        if (baseThread != 0) {
3350            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3351            mName = playbackThread->getTrackName_l();
3352            mMainBuffer = playbackThread->mixBuffer();
3353        }
3354        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3355        if (mName < 0) {
3356            LOGE("no more track names available");
3357        }
3358        mVolume[0] = 1.0f;
3359        mVolume[1] = 1.0f;
3360        mStreamType = streamType;
3361        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3362        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3363        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3364    }
3365}
3366
3367AudioFlinger::PlaybackThread::Track::~Track()
3368{
3369    ALOGV("PlaybackThread::Track destructor");
3370    sp<ThreadBase> thread = mThread.promote();
3371    if (thread != 0) {
3372        Mutex::Autolock _l(thread->mLock);
3373        mState = TERMINATED;
3374    }
3375}
3376
3377void AudioFlinger::PlaybackThread::Track::destroy()
3378{
3379    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3380    // by removing it from mTracks vector, so there is a risk that this Tracks's
3381    // desctructor is called. As the destructor needs to lock mLock,
3382    // we must acquire a strong reference on this Track before locking mLock
3383    // here so that the destructor is called only when exiting this function.
3384    // On the other hand, as long as Track::destroy() is only called by
3385    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3386    // this Track with its member mTrack.
3387    sp<Track> keep(this);
3388    { // scope for mLock
3389        sp<ThreadBase> thread = mThread.promote();
3390        if (thread != 0) {
3391            if (!isOutputTrack()) {
3392                if (mState == ACTIVE || mState == RESUMING) {
3393                    AudioSystem::stopOutput(thread->id(),
3394                                            (audio_stream_type_t)mStreamType,
3395                                            mSessionId);
3396
3397                    // to track the speaker usage
3398                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3399                }
3400                AudioSystem::releaseOutput(thread->id());
3401            }
3402            Mutex::Autolock _l(thread->mLock);
3403            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3404            playbackThread->destroyTrack_l(this);
3405        }
3406    }
3407}
3408
3409void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3410{
3411    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",
3412            mName - AudioMixer::TRACK0,
3413            (mClient == NULL) ? getpid() : mClient->pid(),
3414            mStreamType,
3415            mFormat,
3416            mChannelMask,
3417            mSessionId,
3418            mFrameCount,
3419            mState,
3420            mMute,
3421            mFillingUpStatus,
3422            mCblk->sampleRate,
3423            mCblk->volume[0],
3424            mCblk->volume[1],
3425            mCblk->server,
3426            mCblk->user,
3427            (int)mMainBuffer,
3428            (int)mAuxBuffer);
3429}
3430
3431status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3432{
3433     audio_track_cblk_t* cblk = this->cblk();
3434     uint32_t framesReady;
3435     uint32_t framesReq = buffer->frameCount;
3436
3437     // Check if last stepServer failed, try to step now
3438     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3439         if (!step())  goto getNextBuffer_exit;
3440         ALOGV("stepServer recovered");
3441         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3442     }
3443
3444     framesReady = cblk->framesReady();
3445
3446     if (LIKELY(framesReady)) {
3447        uint32_t s = cblk->server;
3448        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3449
3450        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3451        if (framesReq > framesReady) {
3452            framesReq = framesReady;
3453        }
3454        if (s + framesReq > bufferEnd) {
3455            framesReq = bufferEnd - s;
3456        }
3457
3458         buffer->raw = getBuffer(s, framesReq);
3459         if (buffer->raw == NULL) goto getNextBuffer_exit;
3460
3461         buffer->frameCount = framesReq;
3462        return NO_ERROR;
3463     }
3464
3465getNextBuffer_exit:
3466     buffer->raw = NULL;
3467     buffer->frameCount = 0;
3468     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3469     return NOT_ENOUGH_DATA;
3470}
3471
3472bool AudioFlinger::PlaybackThread::Track::isReady() const {
3473    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3474
3475    if (mCblk->framesReady() >= mCblk->frameCount ||
3476            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3477        mFillingUpStatus = FS_FILLED;
3478        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3479        return true;
3480    }
3481    return false;
3482}
3483
3484status_t AudioFlinger::PlaybackThread::Track::start()
3485{
3486    status_t status = NO_ERROR;
3487    ALOGV("start(%d), calling thread %d session %d",
3488            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3489    sp<ThreadBase> thread = mThread.promote();
3490    if (thread != 0) {
3491        Mutex::Autolock _l(thread->mLock);
3492        int state = mState;
3493        // here the track could be either new, or restarted
3494        // in both cases "unstop" the track
3495        if (mState == PAUSED) {
3496            mState = TrackBase::RESUMING;
3497            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3498        } else {
3499            mState = TrackBase::ACTIVE;
3500            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3501        }
3502
3503        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3504            thread->mLock.unlock();
3505            status = AudioSystem::startOutput(thread->id(),
3506                                              (audio_stream_type_t)mStreamType,
3507                                              mSessionId);
3508            thread->mLock.lock();
3509
3510            // to track the speaker usage
3511            if (status == NO_ERROR) {
3512                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3513            }
3514        }
3515        if (status == NO_ERROR) {
3516            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3517            playbackThread->addTrack_l(this);
3518        } else {
3519            mState = state;
3520        }
3521    } else {
3522        status = BAD_VALUE;
3523    }
3524    return status;
3525}
3526
3527void AudioFlinger::PlaybackThread::Track::stop()
3528{
3529    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3530    sp<ThreadBase> thread = mThread.promote();
3531    if (thread != 0) {
3532        Mutex::Autolock _l(thread->mLock);
3533        int state = mState;
3534        if (mState > STOPPED) {
3535            mState = STOPPED;
3536            // If the track is not active (PAUSED and buffers full), flush buffers
3537            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3538            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3539                reset();
3540            }
3541            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3542        }
3543        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3544            thread->mLock.unlock();
3545            AudioSystem::stopOutput(thread->id(),
3546                                    (audio_stream_type_t)mStreamType,
3547                                    mSessionId);
3548            thread->mLock.lock();
3549
3550            // to track the speaker usage
3551            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3552        }
3553    }
3554}
3555
3556void AudioFlinger::PlaybackThread::Track::pause()
3557{
3558    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3559    sp<ThreadBase> thread = mThread.promote();
3560    if (thread != 0) {
3561        Mutex::Autolock _l(thread->mLock);
3562        if (mState == ACTIVE || mState == RESUMING) {
3563            mState = PAUSING;
3564            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3565            if (!isOutputTrack()) {
3566                thread->mLock.unlock();
3567                AudioSystem::stopOutput(thread->id(),
3568                                        (audio_stream_type_t)mStreamType,
3569                                        mSessionId);
3570                thread->mLock.lock();
3571
3572                // to track the speaker usage
3573                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3574            }
3575        }
3576    }
3577}
3578
3579void AudioFlinger::PlaybackThread::Track::flush()
3580{
3581    ALOGV("flush(%d)", mName);
3582    sp<ThreadBase> thread = mThread.promote();
3583    if (thread != 0) {
3584        Mutex::Autolock _l(thread->mLock);
3585        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3586            return;
3587        }
3588        // No point remaining in PAUSED state after a flush => go to
3589        // STOPPED state
3590        mState = STOPPED;
3591
3592        // do not reset the track if it is still in the process of being stopped or paused.
3593        // this will be done by prepareTracks_l() when the track is stopped.
3594        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3595        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3596            reset();
3597        }
3598    }
3599}
3600
3601void AudioFlinger::PlaybackThread::Track::reset()
3602{
3603    // Do not reset twice to avoid discarding data written just after a flush and before
3604    // the audioflinger thread detects the track is stopped.
3605    if (!mResetDone) {
3606        TrackBase::reset();
3607        // Force underrun condition to avoid false underrun callback until first data is
3608        // written to buffer
3609        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3610        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3611        mFillingUpStatus = FS_FILLING;
3612        mResetDone = true;
3613    }
3614}
3615
3616void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3617{
3618    mMute = muted;
3619}
3620
3621void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3622{
3623    mVolume[0] = left;
3624    mVolume[1] = right;
3625}
3626
3627status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3628{
3629    status_t status = DEAD_OBJECT;
3630    sp<ThreadBase> thread = mThread.promote();
3631    if (thread != 0) {
3632       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3633       status = playbackThread->attachAuxEffect(this, EffectId);
3634    }
3635    return status;
3636}
3637
3638void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3639{
3640    mAuxEffectId = EffectId;
3641    mAuxBuffer = buffer;
3642}
3643
3644// ----------------------------------------------------------------------------
3645
3646// RecordTrack constructor must be called with AudioFlinger::mLock held
3647AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3648            const wp<ThreadBase>& thread,
3649            const sp<Client>& client,
3650            uint32_t sampleRate,
3651            uint32_t format,
3652            uint32_t channelMask,
3653            int frameCount,
3654            uint32_t flags,
3655            int sessionId)
3656    :   TrackBase(thread, client, sampleRate, format,
3657                  channelMask, frameCount, flags, 0, sessionId),
3658        mOverflow(false)
3659{
3660    if (mCblk != NULL) {
3661       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3662       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3663           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3664       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3665           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3666       } else {
3667           mCblk->frameSize = sizeof(int8_t);
3668       }
3669    }
3670}
3671
3672AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3673{
3674    sp<ThreadBase> thread = mThread.promote();
3675    if (thread != 0) {
3676        AudioSystem::releaseInput(thread->id());
3677    }
3678}
3679
3680status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3681{
3682    audio_track_cblk_t* cblk = this->cblk();
3683    uint32_t framesAvail;
3684    uint32_t framesReq = buffer->frameCount;
3685
3686     // Check if last stepServer failed, try to step now
3687    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3688        if (!step()) goto getNextBuffer_exit;
3689        ALOGV("stepServer recovered");
3690        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3691    }
3692
3693    framesAvail = cblk->framesAvailable_l();
3694
3695    if (LIKELY(framesAvail)) {
3696        uint32_t s = cblk->server;
3697        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3698
3699        if (framesReq > framesAvail) {
3700            framesReq = framesAvail;
3701        }
3702        if (s + framesReq > bufferEnd) {
3703            framesReq = bufferEnd - s;
3704        }
3705
3706        buffer->raw = getBuffer(s, framesReq);
3707        if (buffer->raw == NULL) goto getNextBuffer_exit;
3708
3709        buffer->frameCount = framesReq;
3710        return NO_ERROR;
3711    }
3712
3713getNextBuffer_exit:
3714    buffer->raw = NULL;
3715    buffer->frameCount = 0;
3716    return NOT_ENOUGH_DATA;
3717}
3718
3719status_t AudioFlinger::RecordThread::RecordTrack::start()
3720{
3721    sp<ThreadBase> thread = mThread.promote();
3722    if (thread != 0) {
3723        RecordThread *recordThread = (RecordThread *)thread.get();
3724        return recordThread->start(this);
3725    } else {
3726        return BAD_VALUE;
3727    }
3728}
3729
3730void AudioFlinger::RecordThread::RecordTrack::stop()
3731{
3732    sp<ThreadBase> thread = mThread.promote();
3733    if (thread != 0) {
3734        RecordThread *recordThread = (RecordThread *)thread.get();
3735        recordThread->stop(this);
3736        TrackBase::reset();
3737        // Force overerrun condition to avoid false overrun callback until first data is
3738        // read from buffer
3739        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3740    }
3741}
3742
3743void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3744{
3745    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3746            (mClient == NULL) ? getpid() : mClient->pid(),
3747            mFormat,
3748            mChannelMask,
3749            mSessionId,
3750            mFrameCount,
3751            mState,
3752            mCblk->sampleRate,
3753            mCblk->server,
3754            mCblk->user);
3755}
3756
3757
3758// ----------------------------------------------------------------------------
3759
3760AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3761            const wp<ThreadBase>& thread,
3762            DuplicatingThread *sourceThread,
3763            uint32_t sampleRate,
3764            uint32_t format,
3765            uint32_t channelMask,
3766            int frameCount)
3767    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3768    mActive(false), mSourceThread(sourceThread)
3769{
3770
3771    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3772    if (mCblk != NULL) {
3773        mCblk->flags |= CBLK_DIRECTION_OUT;
3774        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3775        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
3776        mOutBuffer.frameCount = 0;
3777        playbackThread->mTracks.add(this);
3778        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3779                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3780                mCblk, mBuffer, mCblk->buffers,
3781                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3782    } else {
3783        LOGW("Error creating output track on thread %p", playbackThread);
3784    }
3785}
3786
3787AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3788{
3789    clearBufferQueue();
3790}
3791
3792status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3793{
3794    status_t status = Track::start();
3795    if (status != NO_ERROR) {
3796        return status;
3797    }
3798
3799    mActive = true;
3800    mRetryCount = 127;
3801    return status;
3802}
3803
3804void AudioFlinger::PlaybackThread::OutputTrack::stop()
3805{
3806    Track::stop();
3807    clearBufferQueue();
3808    mOutBuffer.frameCount = 0;
3809    mActive = false;
3810}
3811
3812bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3813{
3814    Buffer *pInBuffer;
3815    Buffer inBuffer;
3816    uint32_t channelCount = mChannelCount;
3817    bool outputBufferFull = false;
3818    inBuffer.frameCount = frames;
3819    inBuffer.i16 = data;
3820
3821    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3822
3823    if (!mActive && frames != 0) {
3824        start();
3825        sp<ThreadBase> thread = mThread.promote();
3826        if (thread != 0) {
3827            MixerThread *mixerThread = (MixerThread *)thread.get();
3828            if (mCblk->frameCount > frames){
3829                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3830                    uint32_t startFrames = (mCblk->frameCount - frames);
3831                    pInBuffer = new Buffer;
3832                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3833                    pInBuffer->frameCount = startFrames;
3834                    pInBuffer->i16 = pInBuffer->mBuffer;
3835                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3836                    mBufferQueue.add(pInBuffer);
3837                } else {
3838                    LOGW ("OutputTrack::write() %p no more buffers in queue", this);
3839                }
3840            }
3841        }
3842    }
3843
3844    while (waitTimeLeftMs) {
3845        // First write pending buffers, then new data
3846        if (mBufferQueue.size()) {
3847            pInBuffer = mBufferQueue.itemAt(0);
3848        } else {
3849            pInBuffer = &inBuffer;
3850        }
3851
3852        if (pInBuffer->frameCount == 0) {
3853            break;
3854        }
3855
3856        if (mOutBuffer.frameCount == 0) {
3857            mOutBuffer.frameCount = pInBuffer->frameCount;
3858            nsecs_t startTime = systemTime();
3859            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
3860                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3861                outputBufferFull = true;
3862                break;
3863            }
3864            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3865            if (waitTimeLeftMs >= waitTimeMs) {
3866                waitTimeLeftMs -= waitTimeMs;
3867            } else {
3868                waitTimeLeftMs = 0;
3869            }
3870        }
3871
3872        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3873        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3874        mCblk->stepUser(outFrames);
3875        pInBuffer->frameCount -= outFrames;
3876        pInBuffer->i16 += outFrames * channelCount;
3877        mOutBuffer.frameCount -= outFrames;
3878        mOutBuffer.i16 += outFrames * channelCount;
3879
3880        if (pInBuffer->frameCount == 0) {
3881            if (mBufferQueue.size()) {
3882                mBufferQueue.removeAt(0);
3883                delete [] pInBuffer->mBuffer;
3884                delete pInBuffer;
3885                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3886            } else {
3887                break;
3888            }
3889        }
3890    }
3891
3892    // If we could not write all frames, allocate a buffer and queue it for next time.
3893    if (inBuffer.frameCount) {
3894        sp<ThreadBase> thread = mThread.promote();
3895        if (thread != 0 && !thread->standby()) {
3896            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3897                pInBuffer = new Buffer;
3898                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3899                pInBuffer->frameCount = inBuffer.frameCount;
3900                pInBuffer->i16 = pInBuffer->mBuffer;
3901                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3902                mBufferQueue.add(pInBuffer);
3903                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3904            } else {
3905                LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3906            }
3907        }
3908    }
3909
3910    // Calling write() with a 0 length buffer, means that no more data will be written:
3911    // If no more buffers are pending, fill output track buffer to make sure it is started
3912    // by output mixer.
3913    if (frames == 0 && mBufferQueue.size() == 0) {
3914        if (mCblk->user < mCblk->frameCount) {
3915            frames = mCblk->frameCount - mCblk->user;
3916            pInBuffer = new Buffer;
3917            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3918            pInBuffer->frameCount = frames;
3919            pInBuffer->i16 = pInBuffer->mBuffer;
3920            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3921            mBufferQueue.add(pInBuffer);
3922        } else if (mActive) {
3923            stop();
3924        }
3925    }
3926
3927    return outputBufferFull;
3928}
3929
3930status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3931{
3932    int active;
3933    status_t result;
3934    audio_track_cblk_t* cblk = mCblk;
3935    uint32_t framesReq = buffer->frameCount;
3936
3937//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3938    buffer->frameCount  = 0;
3939
3940    uint32_t framesAvail = cblk->framesAvailable();
3941
3942
3943    if (framesAvail == 0) {
3944        Mutex::Autolock _l(cblk->lock);
3945        goto start_loop_here;
3946        while (framesAvail == 0) {
3947            active = mActive;
3948            if (UNLIKELY(!active)) {
3949                ALOGV("Not active and NO_MORE_BUFFERS");
3950                return AudioTrack::NO_MORE_BUFFERS;
3951            }
3952            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
3953            if (result != NO_ERROR) {
3954                return AudioTrack::NO_MORE_BUFFERS;
3955            }
3956            // read the server count again
3957        start_loop_here:
3958            framesAvail = cblk->framesAvailable_l();
3959        }
3960    }
3961
3962//    if (framesAvail < framesReq) {
3963//        return AudioTrack::NO_MORE_BUFFERS;
3964//    }
3965
3966    if (framesReq > framesAvail) {
3967        framesReq = framesAvail;
3968    }
3969
3970    uint32_t u = cblk->user;
3971    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
3972
3973    if (u + framesReq > bufferEnd) {
3974        framesReq = bufferEnd - u;
3975    }
3976
3977    buffer->frameCount  = framesReq;
3978    buffer->raw         = (void *)cblk->buffer(u);
3979    return NO_ERROR;
3980}
3981
3982
3983void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
3984{
3985    size_t size = mBufferQueue.size();
3986    Buffer *pBuffer;
3987
3988    for (size_t i = 0; i < size; i++) {
3989        pBuffer = mBufferQueue.itemAt(i);
3990        delete [] pBuffer->mBuffer;
3991        delete pBuffer;
3992    }
3993    mBufferQueue.clear();
3994}
3995
3996// ----------------------------------------------------------------------------
3997
3998AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
3999    :   RefBase(),
4000        mAudioFlinger(audioFlinger),
4001        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
4002        mPid(pid)
4003{
4004    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
4005}
4006
4007// Client destructor must be called with AudioFlinger::mLock held
4008AudioFlinger::Client::~Client()
4009{
4010    mAudioFlinger->removeClient_l(mPid);
4011}
4012
4013const sp<MemoryDealer>& AudioFlinger::Client::heap() const
4014{
4015    return mMemoryDealer;
4016}
4017
4018// ----------------------------------------------------------------------------
4019
4020AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4021                                                     const sp<IAudioFlingerClient>& client,
4022                                                     pid_t pid)
4023    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
4024{
4025}
4026
4027AudioFlinger::NotificationClient::~NotificationClient()
4028{
4029    mClient.clear();
4030}
4031
4032void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4033{
4034    sp<NotificationClient> keep(this);
4035    {
4036        mAudioFlinger->removeNotificationClient(mPid);
4037    }
4038}
4039
4040// ----------------------------------------------------------------------------
4041
4042AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4043    : BnAudioTrack(),
4044      mTrack(track)
4045{
4046}
4047
4048AudioFlinger::TrackHandle::~TrackHandle() {
4049    // just stop the track on deletion, associated resources
4050    // will be freed from the main thread once all pending buffers have
4051    // been played. Unless it's not in the active track list, in which
4052    // case we free everything now...
4053    mTrack->destroy();
4054}
4055
4056status_t AudioFlinger::TrackHandle::start() {
4057    return mTrack->start();
4058}
4059
4060void AudioFlinger::TrackHandle::stop() {
4061    mTrack->stop();
4062}
4063
4064void AudioFlinger::TrackHandle::flush() {
4065    mTrack->flush();
4066}
4067
4068void AudioFlinger::TrackHandle::mute(bool e) {
4069    mTrack->mute(e);
4070}
4071
4072void AudioFlinger::TrackHandle::pause() {
4073    mTrack->pause();
4074}
4075
4076void AudioFlinger::TrackHandle::setVolume(float left, float right) {
4077    mTrack->setVolume(left, right);
4078}
4079
4080sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4081    return mTrack->getCblk();
4082}
4083
4084status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4085{
4086    return mTrack->attachAuxEffect(EffectId);
4087}
4088
4089status_t AudioFlinger::TrackHandle::onTransact(
4090    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4091{
4092    return BnAudioTrack::onTransact(code, data, reply, flags);
4093}
4094
4095// ----------------------------------------------------------------------------
4096
4097sp<IAudioRecord> AudioFlinger::openRecord(
4098        pid_t pid,
4099        int input,
4100        uint32_t sampleRate,
4101        uint32_t format,
4102        uint32_t channelMask,
4103        int frameCount,
4104        uint32_t flags,
4105        int *sessionId,
4106        status_t *status)
4107{
4108    sp<RecordThread::RecordTrack> recordTrack;
4109    sp<RecordHandle> recordHandle;
4110    sp<Client> client;
4111    wp<Client> wclient;
4112    status_t lStatus;
4113    RecordThread *thread;
4114    size_t inFrameCount;
4115    int lSessionId;
4116
4117    // check calling permissions
4118    if (!recordingAllowed()) {
4119        lStatus = PERMISSION_DENIED;
4120        goto Exit;
4121    }
4122
4123    // add client to list
4124    { // scope for mLock
4125        Mutex::Autolock _l(mLock);
4126        thread = checkRecordThread_l(input);
4127        if (thread == NULL) {
4128            lStatus = BAD_VALUE;
4129            goto Exit;
4130        }
4131
4132        wclient = mClients.valueFor(pid);
4133        if (wclient != NULL) {
4134            client = wclient.promote();
4135        } else {
4136            client = new Client(this, pid);
4137            mClients.add(pid, client);
4138        }
4139
4140        // If no audio session id is provided, create one here
4141        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4142            lSessionId = *sessionId;
4143        } else {
4144            lSessionId = nextUniqueId();
4145            if (sessionId != NULL) {
4146                *sessionId = lSessionId;
4147            }
4148        }
4149        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4150        recordTrack = thread->createRecordTrack_l(client,
4151                                                sampleRate,
4152                                                format,
4153                                                channelMask,
4154                                                frameCount,
4155                                                flags,
4156                                                lSessionId,
4157                                                &lStatus);
4158    }
4159    if (lStatus != NO_ERROR) {
4160        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4161        // destructor is called by the TrackBase destructor with mLock held
4162        client.clear();
4163        recordTrack.clear();
4164        goto Exit;
4165    }
4166
4167    // return to handle to client
4168    recordHandle = new RecordHandle(recordTrack);
4169    lStatus = NO_ERROR;
4170
4171Exit:
4172    if (status) {
4173        *status = lStatus;
4174    }
4175    return recordHandle;
4176}
4177
4178// ----------------------------------------------------------------------------
4179
4180AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4181    : BnAudioRecord(),
4182    mRecordTrack(recordTrack)
4183{
4184}
4185
4186AudioFlinger::RecordHandle::~RecordHandle() {
4187    stop();
4188}
4189
4190status_t AudioFlinger::RecordHandle::start() {
4191    ALOGV("RecordHandle::start()");
4192    return mRecordTrack->start();
4193}
4194
4195void AudioFlinger::RecordHandle::stop() {
4196    ALOGV("RecordHandle::stop()");
4197    mRecordTrack->stop();
4198}
4199
4200sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4201    return mRecordTrack->getCblk();
4202}
4203
4204status_t AudioFlinger::RecordHandle::onTransact(
4205    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4206{
4207    return BnAudioRecord::onTransact(code, data, reply, flags);
4208}
4209
4210// ----------------------------------------------------------------------------
4211
4212AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4213                                         AudioStreamIn *input,
4214                                         uint32_t sampleRate,
4215                                         uint32_t channels,
4216                                         int id,
4217                                         uint32_t device) :
4218    ThreadBase(audioFlinger, id, device),
4219    mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
4220{
4221    mType = ThreadBase::RECORD;
4222
4223    snprintf(mName, kNameLength, "AudioIn_%d", id);
4224
4225    mReqChannelCount = popcount(channels);
4226    mReqSampleRate = sampleRate;
4227    readInputParameters();
4228}
4229
4230
4231AudioFlinger::RecordThread::~RecordThread()
4232{
4233    delete[] mRsmpInBuffer;
4234    if (mResampler != NULL) {
4235        delete mResampler;
4236        delete[] mRsmpOutBuffer;
4237    }
4238}
4239
4240void AudioFlinger::RecordThread::onFirstRef()
4241{
4242    run(mName, PRIORITY_URGENT_AUDIO);
4243}
4244
4245status_t AudioFlinger::RecordThread::readyToRun()
4246{
4247    status_t status = initCheck();
4248    LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4249    return status;
4250}
4251
4252bool AudioFlinger::RecordThread::threadLoop()
4253{
4254    AudioBufferProvider::Buffer buffer;
4255    sp<RecordTrack> activeTrack;
4256    Vector< sp<EffectChain> > effectChains;
4257
4258    nsecs_t lastWarning = 0;
4259
4260    acquireWakeLock();
4261
4262    // start recording
4263    while (!exitPending()) {
4264
4265        processConfigEvents();
4266
4267        { // scope for mLock
4268            Mutex::Autolock _l(mLock);
4269            checkForNewParameters_l();
4270            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4271                if (!mStandby) {
4272                    mInput->stream->common.standby(&mInput->stream->common);
4273                    mStandby = true;
4274                }
4275
4276                if (exitPending()) break;
4277
4278                releaseWakeLock_l();
4279                ALOGV("RecordThread: loop stopping");
4280                // go to sleep
4281                mWaitWorkCV.wait(mLock);
4282                ALOGV("RecordThread: loop starting");
4283                acquireWakeLock_l();
4284                continue;
4285            }
4286            if (mActiveTrack != 0) {
4287                if (mActiveTrack->mState == TrackBase::PAUSING) {
4288                    if (!mStandby) {
4289                        mInput->stream->common.standby(&mInput->stream->common);
4290                        mStandby = true;
4291                    }
4292                    mActiveTrack.clear();
4293                    mStartStopCond.broadcast();
4294                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4295                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4296                        mActiveTrack.clear();
4297                        mStartStopCond.broadcast();
4298                    } else if (mBytesRead != 0) {
4299                        // record start succeeds only if first read from audio input
4300                        // succeeds
4301                        if (mBytesRead > 0) {
4302                            mActiveTrack->mState = TrackBase::ACTIVE;
4303                        } else {
4304                            mActiveTrack.clear();
4305                        }
4306                        mStartStopCond.broadcast();
4307                    }
4308                    mStandby = false;
4309                }
4310            }
4311            lockEffectChains_l(effectChains);
4312        }
4313
4314        if (mActiveTrack != 0) {
4315            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4316                mActiveTrack->mState != TrackBase::RESUMING) {
4317                unlockEffectChains(effectChains);
4318                usleep(kRecordThreadSleepUs);
4319                continue;
4320            }
4321            for (size_t i = 0; i < effectChains.size(); i ++) {
4322                effectChains[i]->process_l();
4323            }
4324
4325            buffer.frameCount = mFrameCount;
4326            if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4327                size_t framesOut = buffer.frameCount;
4328                if (mResampler == NULL) {
4329                    // no resampling
4330                    while (framesOut) {
4331                        size_t framesIn = mFrameCount - mRsmpInIndex;
4332                        if (framesIn) {
4333                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4334                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4335                            if (framesIn > framesOut)
4336                                framesIn = framesOut;
4337                            mRsmpInIndex += framesIn;
4338                            framesOut -= framesIn;
4339                            if ((int)mChannelCount == mReqChannelCount ||
4340                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4341                                memcpy(dst, src, framesIn * mFrameSize);
4342                            } else {
4343                                int16_t *src16 = (int16_t *)src;
4344                                int16_t *dst16 = (int16_t *)dst;
4345                                if (mChannelCount == 1) {
4346                                    while (framesIn--) {
4347                                        *dst16++ = *src16;
4348                                        *dst16++ = *src16++;
4349                                    }
4350                                } else {
4351                                    while (framesIn--) {
4352                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4353                                        src16 += 2;
4354                                    }
4355                                }
4356                            }
4357                        }
4358                        if (framesOut && mFrameCount == mRsmpInIndex) {
4359                            if (framesOut == mFrameCount &&
4360                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4361                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4362                                framesOut = 0;
4363                            } else {
4364                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4365                                mRsmpInIndex = 0;
4366                            }
4367                            if (mBytesRead < 0) {
4368                                LOGE("Error reading audio input");
4369                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4370                                    // Force input into standby so that it tries to
4371                                    // recover at next read attempt
4372                                    mInput->stream->common.standby(&mInput->stream->common);
4373                                    usleep(kRecordThreadSleepUs);
4374                                }
4375                                mRsmpInIndex = mFrameCount;
4376                                framesOut = 0;
4377                                buffer.frameCount = 0;
4378                            }
4379                        }
4380                    }
4381                } else {
4382                    // resampling
4383
4384                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4385                    // alter output frame count as if we were expecting stereo samples
4386                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4387                        framesOut >>= 1;
4388                    }
4389                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4390                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4391                    // are 32 bit aligned which should be always true.
4392                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4393                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4394                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4395                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4396                        int16_t *dst = buffer.i16;
4397                        while (framesOut--) {
4398                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4399                            src += 2;
4400                        }
4401                    } else {
4402                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4403                    }
4404
4405                }
4406                mActiveTrack->releaseBuffer(&buffer);
4407                mActiveTrack->overflow();
4408            }
4409            // client isn't retrieving buffers fast enough
4410            else {
4411                if (!mActiveTrack->setOverflow()) {
4412                    nsecs_t now = systemTime();
4413                    if ((now - lastWarning) > kWarningThrottleNs) {
4414                        LOGW("RecordThread: buffer overflow");
4415                        lastWarning = now;
4416                    }
4417                }
4418                // Release the processor for a while before asking for a new buffer.
4419                // This will give the application more chance to read from the buffer and
4420                // clear the overflow.
4421                usleep(kRecordThreadSleepUs);
4422            }
4423        }
4424        // enable changes in effect chain
4425        unlockEffectChains(effectChains);
4426        effectChains.clear();
4427    }
4428
4429    if (!mStandby) {
4430        mInput->stream->common.standby(&mInput->stream->common);
4431    }
4432    mActiveTrack.clear();
4433
4434    mStartStopCond.broadcast();
4435
4436    releaseWakeLock();
4437
4438    ALOGV("RecordThread %p exiting", this);
4439    return false;
4440}
4441
4442
4443sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4444        const sp<AudioFlinger::Client>& client,
4445        uint32_t sampleRate,
4446        int format,
4447        int channelMask,
4448        int frameCount,
4449        uint32_t flags,
4450        int sessionId,
4451        status_t *status)
4452{
4453    sp<RecordTrack> track;
4454    status_t lStatus;
4455
4456    lStatus = initCheck();
4457    if (lStatus != NO_ERROR) {
4458        LOGE("Audio driver not initialized.");
4459        goto Exit;
4460    }
4461
4462    { // scope for mLock
4463        Mutex::Autolock _l(mLock);
4464
4465        track = new RecordTrack(this, client, sampleRate,
4466                      format, channelMask, frameCount, flags, sessionId);
4467
4468        if (track->getCblk() == NULL) {
4469            lStatus = NO_MEMORY;
4470            goto Exit;
4471        }
4472
4473        mTrack = track.get();
4474        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4475        bool suspend = audio_is_bluetooth_sco_device(
4476                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4477        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4478        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4479    }
4480    lStatus = NO_ERROR;
4481
4482Exit:
4483    if (status) {
4484        *status = lStatus;
4485    }
4486    return track;
4487}
4488
4489status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4490{
4491    ALOGV("RecordThread::start");
4492    sp <ThreadBase> strongMe = this;
4493    status_t status = NO_ERROR;
4494    {
4495        AutoMutex lock(&mLock);
4496        if (mActiveTrack != 0) {
4497            if (recordTrack != mActiveTrack.get()) {
4498                status = -EBUSY;
4499            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4500                mActiveTrack->mState = TrackBase::ACTIVE;
4501            }
4502            return status;
4503        }
4504
4505        recordTrack->mState = TrackBase::IDLE;
4506        mActiveTrack = recordTrack;
4507        mLock.unlock();
4508        status_t status = AudioSystem::startInput(mId);
4509        mLock.lock();
4510        if (status != NO_ERROR) {
4511            mActiveTrack.clear();
4512            return status;
4513        }
4514        mRsmpInIndex = mFrameCount;
4515        mBytesRead = 0;
4516        if (mResampler != NULL) {
4517            mResampler->reset();
4518        }
4519        mActiveTrack->mState = TrackBase::RESUMING;
4520        // signal thread to start
4521        ALOGV("Signal record thread");
4522        mWaitWorkCV.signal();
4523        // do not wait for mStartStopCond if exiting
4524        if (mExiting) {
4525            mActiveTrack.clear();
4526            status = INVALID_OPERATION;
4527            goto startError;
4528        }
4529        mStartStopCond.wait(mLock);
4530        if (mActiveTrack == 0) {
4531            ALOGV("Record failed to start");
4532            status = BAD_VALUE;
4533            goto startError;
4534        }
4535        ALOGV("Record started OK");
4536        return status;
4537    }
4538startError:
4539    AudioSystem::stopInput(mId);
4540    return status;
4541}
4542
4543void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4544    ALOGV("RecordThread::stop");
4545    sp <ThreadBase> strongMe = this;
4546    {
4547        AutoMutex lock(&mLock);
4548        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4549            mActiveTrack->mState = TrackBase::PAUSING;
4550            // do not wait for mStartStopCond if exiting
4551            if (mExiting) {
4552                return;
4553            }
4554            mStartStopCond.wait(mLock);
4555            // if we have been restarted, recordTrack == mActiveTrack.get() here
4556            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4557                mLock.unlock();
4558                AudioSystem::stopInput(mId);
4559                mLock.lock();
4560                ALOGV("Record stopped OK");
4561            }
4562        }
4563    }
4564}
4565
4566status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4567{
4568    const size_t SIZE = 256;
4569    char buffer[SIZE];
4570    String8 result;
4571    pid_t pid = 0;
4572
4573    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4574    result.append(buffer);
4575
4576    if (mActiveTrack != 0) {
4577        result.append("Active Track:\n");
4578        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4579        mActiveTrack->dump(buffer, SIZE);
4580        result.append(buffer);
4581
4582        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4583        result.append(buffer);
4584        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4585        result.append(buffer);
4586        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4587        result.append(buffer);
4588        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4589        result.append(buffer);
4590        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4591        result.append(buffer);
4592
4593
4594    } else {
4595        result.append("No record client\n");
4596    }
4597    write(fd, result.string(), result.size());
4598
4599    dumpBase(fd, args);
4600    dumpEffectChains(fd, args);
4601
4602    return NO_ERROR;
4603}
4604
4605status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4606{
4607    size_t framesReq = buffer->frameCount;
4608    size_t framesReady = mFrameCount - mRsmpInIndex;
4609    int channelCount;
4610
4611    if (framesReady == 0) {
4612        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4613        if (mBytesRead < 0) {
4614            LOGE("RecordThread::getNextBuffer() Error reading audio input");
4615            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4616                // Force input into standby so that it tries to
4617                // recover at next read attempt
4618                mInput->stream->common.standby(&mInput->stream->common);
4619                usleep(kRecordThreadSleepUs);
4620            }
4621            buffer->raw = NULL;
4622            buffer->frameCount = 0;
4623            return NOT_ENOUGH_DATA;
4624        }
4625        mRsmpInIndex = 0;
4626        framesReady = mFrameCount;
4627    }
4628
4629    if (framesReq > framesReady) {
4630        framesReq = framesReady;
4631    }
4632
4633    if (mChannelCount == 1 && mReqChannelCount == 2) {
4634        channelCount = 1;
4635    } else {
4636        channelCount = 2;
4637    }
4638    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4639    buffer->frameCount = framesReq;
4640    return NO_ERROR;
4641}
4642
4643void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4644{
4645    mRsmpInIndex += buffer->frameCount;
4646    buffer->frameCount = 0;
4647}
4648
4649bool AudioFlinger::RecordThread::checkForNewParameters_l()
4650{
4651    bool reconfig = false;
4652
4653    while (!mNewParameters.isEmpty()) {
4654        status_t status = NO_ERROR;
4655        String8 keyValuePair = mNewParameters[0];
4656        AudioParameter param = AudioParameter(keyValuePair);
4657        int value;
4658        int reqFormat = mFormat;
4659        int reqSamplingRate = mReqSampleRate;
4660        int reqChannelCount = mReqChannelCount;
4661
4662        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4663            reqSamplingRate = value;
4664            reconfig = true;
4665        }
4666        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4667            reqFormat = value;
4668            reconfig = true;
4669        }
4670        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4671            reqChannelCount = popcount(value);
4672            reconfig = true;
4673        }
4674        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4675            // do not accept frame count changes if tracks are open as the track buffer
4676            // size depends on frame count and correct behavior would not be garantied
4677            // if frame count is changed after track creation
4678            if (mActiveTrack != 0) {
4679                status = INVALID_OPERATION;
4680            } else {
4681                reconfig = true;
4682            }
4683        }
4684        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4685            // forward device change to effects that have requested to be
4686            // aware of attached audio device.
4687            for (size_t i = 0; i < mEffectChains.size(); i++) {
4688                mEffectChains[i]->setDevice_l(value);
4689            }
4690            // store input device and output device but do not forward output device to audio HAL.
4691            // Note that status is ignored by the caller for output device
4692            // (see AudioFlinger::setParameters()
4693            if (value & AUDIO_DEVICE_OUT_ALL) {
4694                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4695                status = BAD_VALUE;
4696            } else {
4697                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4698                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4699                if (mTrack != NULL) {
4700                    bool suspend = audio_is_bluetooth_sco_device(
4701                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4702                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4703                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4704                }
4705            }
4706            mDevice |= (uint32_t)value;
4707        }
4708        if (status == NO_ERROR) {
4709            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4710            if (status == INVALID_OPERATION) {
4711               mInput->stream->common.standby(&mInput->stream->common);
4712               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4713            }
4714            if (reconfig) {
4715                if (status == BAD_VALUE &&
4716                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4717                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4718                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4719                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4720                    (reqChannelCount < 3)) {
4721                    status = NO_ERROR;
4722                }
4723                if (status == NO_ERROR) {
4724                    readInputParameters();
4725                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4726                }
4727            }
4728        }
4729
4730        mNewParameters.removeAt(0);
4731
4732        mParamStatus = status;
4733        mParamCond.signal();
4734        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4735        // already timed out waiting for the status and will never signal the condition.
4736        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4737    }
4738    return reconfig;
4739}
4740
4741String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4742{
4743    char *s;
4744    String8 out_s8 = String8();
4745
4746    Mutex::Autolock _l(mLock);
4747    if (initCheck() != NO_ERROR) {
4748        return out_s8;
4749    }
4750
4751    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4752    out_s8 = String8(s);
4753    free(s);
4754    return out_s8;
4755}
4756
4757void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4758    AudioSystem::OutputDescriptor desc;
4759    void *param2 = 0;
4760
4761    switch (event) {
4762    case AudioSystem::INPUT_OPENED:
4763    case AudioSystem::INPUT_CONFIG_CHANGED:
4764        desc.channels = mChannelMask;
4765        desc.samplingRate = mSampleRate;
4766        desc.format = mFormat;
4767        desc.frameCount = mFrameCount;
4768        desc.latency = 0;
4769        param2 = &desc;
4770        break;
4771
4772    case AudioSystem::INPUT_CLOSED:
4773    default:
4774        break;
4775    }
4776    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4777}
4778
4779void AudioFlinger::RecordThread::readInputParameters()
4780{
4781    if (mRsmpInBuffer) delete mRsmpInBuffer;
4782    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4783    if (mResampler) delete mResampler;
4784    mResampler = NULL;
4785
4786    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4787    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4788    mChannelCount = (uint16_t)popcount(mChannelMask);
4789    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4790    mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
4791    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4792    mFrameCount = mInputBytes / mFrameSize;
4793    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4794
4795    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4796    {
4797        int channelCount;
4798         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4799         // stereo to mono post process as the resampler always outputs stereo.
4800        if (mChannelCount == 1 && mReqChannelCount == 2) {
4801            channelCount = 1;
4802        } else {
4803            channelCount = 2;
4804        }
4805        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4806        mResampler->setSampleRate(mSampleRate);
4807        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4808        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4809
4810        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4811        if (mChannelCount == 1 && mReqChannelCount == 1) {
4812            mFrameCount >>= 1;
4813        }
4814
4815    }
4816    mRsmpInIndex = mFrameCount;
4817}
4818
4819unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4820{
4821    Mutex::Autolock _l(mLock);
4822    if (initCheck() != NO_ERROR) {
4823        return 0;
4824    }
4825
4826    return mInput->stream->get_input_frames_lost(mInput->stream);
4827}
4828
4829uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4830{
4831    Mutex::Autolock _l(mLock);
4832    uint32_t result = 0;
4833    if (getEffectChain_l(sessionId) != 0) {
4834        result = EFFECT_SESSION;
4835    }
4836
4837    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4838        result |= TRACK_SESSION;
4839    }
4840
4841    return result;
4842}
4843
4844AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4845{
4846    Mutex::Autolock _l(mLock);
4847    return mTrack;
4848}
4849
4850AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
4851{
4852    Mutex::Autolock _l(mLock);
4853    return mInput;
4854}
4855
4856AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4857{
4858    Mutex::Autolock _l(mLock);
4859    AudioStreamIn *input = mInput;
4860    mInput = NULL;
4861    return input;
4862}
4863
4864// this method must always be called either with ThreadBase mLock held or inside the thread loop
4865audio_stream_t* AudioFlinger::RecordThread::stream()
4866{
4867    if (mInput == NULL) {
4868        return NULL;
4869    }
4870    return &mInput->stream->common;
4871}
4872
4873
4874// ----------------------------------------------------------------------------
4875
4876int AudioFlinger::openOutput(uint32_t *pDevices,
4877                                uint32_t *pSamplingRate,
4878                                uint32_t *pFormat,
4879                                uint32_t *pChannels,
4880                                uint32_t *pLatencyMs,
4881                                uint32_t flags)
4882{
4883    status_t status;
4884    PlaybackThread *thread = NULL;
4885    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4886    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4887    uint32_t format = pFormat ? *pFormat : 0;
4888    uint32_t channels = pChannels ? *pChannels : 0;
4889    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4890    audio_stream_out_t *outStream;
4891    audio_hw_device_t *outHwDev;
4892
4893    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4894            pDevices ? *pDevices : 0,
4895            samplingRate,
4896            format,
4897            channels,
4898            flags);
4899
4900    if (pDevices == NULL || *pDevices == 0) {
4901        return 0;
4902    }
4903
4904    Mutex::Autolock _l(mLock);
4905
4906    outHwDev = findSuitableHwDev_l(*pDevices);
4907    if (outHwDev == NULL)
4908        return 0;
4909
4910    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
4911                                          &channels, &samplingRate, &outStream);
4912    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4913            outStream,
4914            samplingRate,
4915            format,
4916            channels,
4917            status);
4918
4919    mHardwareStatus = AUDIO_HW_IDLE;
4920    if (outStream != NULL) {
4921        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4922        int id = nextUniqueId();
4923
4924        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4925            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4926            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4927            thread = new DirectOutputThread(this, output, id, *pDevices);
4928            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4929        } else {
4930            thread = new MixerThread(this, output, id, *pDevices);
4931            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4932        }
4933        mPlaybackThreads.add(id, thread);
4934
4935        if (pSamplingRate) *pSamplingRate = samplingRate;
4936        if (pFormat) *pFormat = format;
4937        if (pChannels) *pChannels = channels;
4938        if (pLatencyMs) *pLatencyMs = thread->latency();
4939
4940        // notify client processes of the new output creation
4941        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4942        return id;
4943    }
4944
4945    return 0;
4946}
4947
4948int AudioFlinger::openDuplicateOutput(int output1, int output2)
4949{
4950    Mutex::Autolock _l(mLock);
4951    MixerThread *thread1 = checkMixerThread_l(output1);
4952    MixerThread *thread2 = checkMixerThread_l(output2);
4953
4954    if (thread1 == NULL || thread2 == NULL) {
4955        LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
4956        return 0;
4957    }
4958
4959    int id = nextUniqueId();
4960    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
4961    thread->addOutputTrack(thread2);
4962    mPlaybackThreads.add(id, thread);
4963    // notify client processes of the new output creation
4964    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4965    return id;
4966}
4967
4968status_t AudioFlinger::closeOutput(int output)
4969{
4970    // keep strong reference on the playback thread so that
4971    // it is not destroyed while exit() is executed
4972    sp <PlaybackThread> thread;
4973    {
4974        Mutex::Autolock _l(mLock);
4975        thread = checkPlaybackThread_l(output);
4976        if (thread == NULL) {
4977            return BAD_VALUE;
4978        }
4979
4980        ALOGV("closeOutput() %d", output);
4981
4982        if (thread->type() == ThreadBase::MIXER) {
4983            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4984                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
4985                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
4986                    dupThread->removeOutputTrack((MixerThread *)thread.get());
4987                }
4988            }
4989        }
4990        void *param2 = 0;
4991        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
4992        mPlaybackThreads.removeItem(output);
4993    }
4994    thread->exit();
4995
4996    if (thread->type() != ThreadBase::DUPLICATING) {
4997        AudioStreamOut *out = thread->clearOutput();
4998        // from now on thread->mOutput is NULL
4999        out->hwDev->close_output_stream(out->hwDev, out->stream);
5000        delete out;
5001    }
5002    return NO_ERROR;
5003}
5004
5005status_t AudioFlinger::suspendOutput(int output)
5006{
5007    Mutex::Autolock _l(mLock);
5008    PlaybackThread *thread = checkPlaybackThread_l(output);
5009
5010    if (thread == NULL) {
5011        return BAD_VALUE;
5012    }
5013
5014    ALOGV("suspendOutput() %d", output);
5015    thread->suspend();
5016
5017    return NO_ERROR;
5018}
5019
5020status_t AudioFlinger::restoreOutput(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("restoreOutput() %d", output);
5030
5031    thread->restore();
5032
5033    return NO_ERROR;
5034}
5035
5036int AudioFlinger::openInput(uint32_t *pDevices,
5037                                uint32_t *pSamplingRate,
5038                                uint32_t *pFormat,
5039                                uint32_t *pChannels,
5040                                uint32_t acoustics)
5041{
5042    status_t status;
5043    RecordThread *thread = NULL;
5044    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5045    uint32_t format = pFormat ? *pFormat : 0;
5046    uint32_t channels = pChannels ? *pChannels : 0;
5047    uint32_t reqSamplingRate = samplingRate;
5048    uint32_t reqFormat = format;
5049    uint32_t reqChannels = channels;
5050    audio_stream_in_t *inStream;
5051    audio_hw_device_t *inHwDev;
5052
5053    if (pDevices == NULL || *pDevices == 0) {
5054        return 0;
5055    }
5056
5057    Mutex::Autolock _l(mLock);
5058
5059    inHwDev = findSuitableHwDev_l(*pDevices);
5060    if (inHwDev == NULL)
5061        return 0;
5062
5063    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5064                                        &channels, &samplingRate,
5065                                        (audio_in_acoustics_t)acoustics,
5066                                        &inStream);
5067    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5068            inStream,
5069            samplingRate,
5070            format,
5071            channels,
5072            acoustics,
5073            status);
5074
5075    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5076    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5077    // or stereo to mono conversions on 16 bit PCM inputs.
5078    if (inStream == NULL && status == BAD_VALUE &&
5079        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5080        (samplingRate <= 2 * reqSamplingRate) &&
5081        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5082        ALOGV("openInput() reopening with proposed sampling rate and channels");
5083        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5084                                            &channels, &samplingRate,
5085                                            (audio_in_acoustics_t)acoustics,
5086                                            &inStream);
5087    }
5088
5089    if (inStream != NULL) {
5090        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5091
5092        int id = nextUniqueId();
5093        // Start record thread
5094        // RecorThread require both input and output device indication to forward to audio
5095        // pre processing modules
5096        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5097        thread = new RecordThread(this,
5098                                  input,
5099                                  reqSamplingRate,
5100                                  reqChannels,
5101                                  id,
5102                                  device);
5103        mRecordThreads.add(id, thread);
5104        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5105        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5106        if (pFormat) *pFormat = format;
5107        if (pChannels) *pChannels = reqChannels;
5108
5109        input->stream->common.standby(&input->stream->common);
5110
5111        // notify client processes of the new input creation
5112        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5113        return id;
5114    }
5115
5116    return 0;
5117}
5118
5119status_t AudioFlinger::closeInput(int input)
5120{
5121    // keep strong reference on the record thread so that
5122    // it is not destroyed while exit() is executed
5123    sp <RecordThread> thread;
5124    {
5125        Mutex::Autolock _l(mLock);
5126        thread = checkRecordThread_l(input);
5127        if (thread == NULL) {
5128            return BAD_VALUE;
5129        }
5130
5131        ALOGV("closeInput() %d", input);
5132        void *param2 = 0;
5133        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5134        mRecordThreads.removeItem(input);
5135    }
5136    thread->exit();
5137
5138    AudioStreamIn *in = thread->clearInput();
5139    // from now on thread->mInput is NULL
5140    in->hwDev->close_input_stream(in->hwDev, in->stream);
5141    delete in;
5142
5143    return NO_ERROR;
5144}
5145
5146status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
5147{
5148    Mutex::Autolock _l(mLock);
5149    MixerThread *dstThread = checkMixerThread_l(output);
5150    if (dstThread == NULL) {
5151        LOGW("setStreamOutput() bad output id %d", output);
5152        return BAD_VALUE;
5153    }
5154
5155    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5156    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5157
5158    dstThread->setStreamValid(stream, true);
5159
5160    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5161        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5162        if (thread != dstThread &&
5163            thread->type() != ThreadBase::DIRECT) {
5164            MixerThread *srcThread = (MixerThread *)thread;
5165            srcThread->setStreamValid(stream, false);
5166            srcThread->invalidateTracks(stream);
5167        }
5168    }
5169
5170    return NO_ERROR;
5171}
5172
5173
5174int AudioFlinger::newAudioSessionId()
5175{
5176    return nextUniqueId();
5177}
5178
5179void AudioFlinger::acquireAudioSessionId(int audioSession)
5180{
5181    Mutex::Autolock _l(mLock);
5182    int caller = IPCThreadState::self()->getCallingPid();
5183    ALOGV("acquiring %d from %d", audioSession, caller);
5184    int num = mAudioSessionRefs.size();
5185    for (int i = 0; i< num; i++) {
5186        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5187        if (ref->sessionid == audioSession && ref->pid == caller) {
5188            ref->cnt++;
5189            ALOGV(" incremented refcount to %d", ref->cnt);
5190            return;
5191        }
5192    }
5193    AudioSessionRef *ref = new AudioSessionRef();
5194    ref->sessionid = audioSession;
5195    ref->pid = caller;
5196    ref->cnt = 1;
5197    mAudioSessionRefs.push(ref);
5198    ALOGV(" added new entry for %d", ref->sessionid);
5199}
5200
5201void AudioFlinger::releaseAudioSessionId(int audioSession)
5202{
5203    Mutex::Autolock _l(mLock);
5204    int caller = IPCThreadState::self()->getCallingPid();
5205    ALOGV("releasing %d from %d", audioSession, caller);
5206    int num = mAudioSessionRefs.size();
5207    for (int i = 0; i< num; i++) {
5208        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5209        if (ref->sessionid == audioSession && ref->pid == caller) {
5210            ref->cnt--;
5211            ALOGV(" decremented refcount to %d", ref->cnt);
5212            if (ref->cnt == 0) {
5213                mAudioSessionRefs.removeAt(i);
5214                delete ref;
5215                purgeStaleEffects_l();
5216            }
5217            return;
5218        }
5219    }
5220    LOGW("session id %d not found for pid %d", audioSession, caller);
5221}
5222
5223void AudioFlinger::purgeStaleEffects_l() {
5224
5225    ALOGV("purging stale effects");
5226
5227    Vector< sp<EffectChain> > chains;
5228
5229    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5230        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5231        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5232            sp<EffectChain> ec = t->mEffectChains[j];
5233            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5234                chains.push(ec);
5235            }
5236        }
5237    }
5238    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5239        sp<RecordThread> t = mRecordThreads.valueAt(i);
5240        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5241            sp<EffectChain> ec = t->mEffectChains[j];
5242            chains.push(ec);
5243        }
5244    }
5245
5246    for (size_t i = 0; i < chains.size(); i++) {
5247        sp<EffectChain> ec = chains[i];
5248        int sessionid = ec->sessionId();
5249        sp<ThreadBase> t = ec->mThread.promote();
5250        if (t == 0) {
5251            continue;
5252        }
5253        size_t numsessionrefs = mAudioSessionRefs.size();
5254        bool found = false;
5255        for (size_t k = 0; k < numsessionrefs; k++) {
5256            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5257            if (ref->sessionid == sessionid) {
5258                ALOGV(" session %d still exists for %d with %d refs",
5259                     sessionid, ref->pid, ref->cnt);
5260                found = true;
5261                break;
5262            }
5263        }
5264        if (!found) {
5265            // remove all effects from the chain
5266            while (ec->mEffects.size()) {
5267                sp<EffectModule> effect = ec->mEffects[0];
5268                effect->unPin();
5269                Mutex::Autolock _l (t->mLock);
5270                t->removeEffect_l(effect);
5271                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5272                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5273                    if (handle != 0) {
5274                        handle->mEffect.clear();
5275                        if (handle->mHasControl && handle->mEnabled) {
5276                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5277                        }
5278                    }
5279                }
5280                AudioSystem::unregisterEffect(effect->id());
5281            }
5282        }
5283    }
5284    return;
5285}
5286
5287// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5288AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5289{
5290    PlaybackThread *thread = NULL;
5291    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5292        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5293    }
5294    return thread;
5295}
5296
5297// checkMixerThread_l() must be called with AudioFlinger::mLock held
5298AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5299{
5300    PlaybackThread *thread = checkPlaybackThread_l(output);
5301    if (thread != NULL) {
5302        if (thread->type() == ThreadBase::DIRECT) {
5303            thread = NULL;
5304        }
5305    }
5306    return (MixerThread *)thread;
5307}
5308
5309// checkRecordThread_l() must be called with AudioFlinger::mLock held
5310AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5311{
5312    RecordThread *thread = NULL;
5313    if (mRecordThreads.indexOfKey(input) >= 0) {
5314        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5315    }
5316    return thread;
5317}
5318
5319uint32_t AudioFlinger::nextUniqueId()
5320{
5321    return android_atomic_inc(&mNextUniqueId);
5322}
5323
5324AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5325{
5326    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5327        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5328        AudioStreamOut *output = thread->getOutput();
5329        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5330            return thread;
5331        }
5332    }
5333    return NULL;
5334}
5335
5336uint32_t AudioFlinger::primaryOutputDevice_l()
5337{
5338    PlaybackThread *thread = primaryPlaybackThread_l();
5339
5340    if (thread == NULL) {
5341        return 0;
5342    }
5343
5344    return thread->device();
5345}
5346
5347
5348// ----------------------------------------------------------------------------
5349//  Effect management
5350// ----------------------------------------------------------------------------
5351
5352
5353status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5354{
5355    Mutex::Autolock _l(mLock);
5356    return EffectQueryNumberEffects(numEffects);
5357}
5358
5359status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5360{
5361    Mutex::Autolock _l(mLock);
5362    return EffectQueryEffect(index, descriptor);
5363}
5364
5365status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5366{
5367    Mutex::Autolock _l(mLock);
5368    return EffectGetDescriptor(pUuid, descriptor);
5369}
5370
5371
5372sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5373        effect_descriptor_t *pDesc,
5374        const sp<IEffectClient>& effectClient,
5375        int32_t priority,
5376        int io,
5377        int sessionId,
5378        status_t *status,
5379        int *id,
5380        int *enabled)
5381{
5382    status_t lStatus = NO_ERROR;
5383    sp<EffectHandle> handle;
5384    effect_descriptor_t desc;
5385    sp<Client> client;
5386    wp<Client> wclient;
5387
5388    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5389            pid, effectClient.get(), priority, sessionId, io);
5390
5391    if (pDesc == NULL) {
5392        lStatus = BAD_VALUE;
5393        goto Exit;
5394    }
5395
5396    // check audio settings permission for global effects
5397    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5398        lStatus = PERMISSION_DENIED;
5399        goto Exit;
5400    }
5401
5402    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5403    // that can only be created by audio policy manager (running in same process)
5404    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5405        lStatus = PERMISSION_DENIED;
5406        goto Exit;
5407    }
5408
5409    if (io == 0) {
5410        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5411            // output must be specified by AudioPolicyManager when using session
5412            // AUDIO_SESSION_OUTPUT_STAGE
5413            lStatus = BAD_VALUE;
5414            goto Exit;
5415        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5416            // if the output returned by getOutputForEffect() is removed before we lock the
5417            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5418            // and we will exit safely
5419            io = AudioSystem::getOutputForEffect(&desc);
5420        }
5421    }
5422
5423    {
5424        Mutex::Autolock _l(mLock);
5425
5426
5427        if (!EffectIsNullUuid(&pDesc->uuid)) {
5428            // if uuid is specified, request effect descriptor
5429            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5430            if (lStatus < 0) {
5431                LOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5432                goto Exit;
5433            }
5434        } else {
5435            // if uuid is not specified, look for an available implementation
5436            // of the required type in effect factory
5437            if (EffectIsNullUuid(&pDesc->type)) {
5438                LOGW("createEffect() no effect type");
5439                lStatus = BAD_VALUE;
5440                goto Exit;
5441            }
5442            uint32_t numEffects = 0;
5443            effect_descriptor_t d;
5444            d.flags = 0; // prevent compiler warning
5445            bool found = false;
5446
5447            lStatus = EffectQueryNumberEffects(&numEffects);
5448            if (lStatus < 0) {
5449                LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5450                goto Exit;
5451            }
5452            for (uint32_t i = 0; i < numEffects; i++) {
5453                lStatus = EffectQueryEffect(i, &desc);
5454                if (lStatus < 0) {
5455                    LOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5456                    continue;
5457                }
5458                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5459                    // If matching type found save effect descriptor. If the session is
5460                    // 0 and the effect is not auxiliary, continue enumeration in case
5461                    // an auxiliary version of this effect type is available
5462                    found = true;
5463                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5464                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5465                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5466                        break;
5467                    }
5468                }
5469            }
5470            if (!found) {
5471                lStatus = BAD_VALUE;
5472                LOGW("createEffect() effect not found");
5473                goto Exit;
5474            }
5475            // For same effect type, chose auxiliary version over insert version if
5476            // connect to output mix (Compliance to OpenSL ES)
5477            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5478                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5479                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5480            }
5481        }
5482
5483        // Do not allow auxiliary effects on a session different from 0 (output mix)
5484        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5485             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5486            lStatus = INVALID_OPERATION;
5487            goto Exit;
5488        }
5489
5490        // check recording permission for visualizer
5491        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5492            !recordingAllowed()) {
5493            lStatus = PERMISSION_DENIED;
5494            goto Exit;
5495        }
5496
5497        // return effect descriptor
5498        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5499
5500        // If output is not specified try to find a matching audio session ID in one of the
5501        // output threads.
5502        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5503        // because of code checking output when entering the function.
5504        // Note: io is never 0 when creating an effect on an input
5505        if (io == 0) {
5506             // look for the thread where the specified audio session is present
5507            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5508                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5509                    io = mPlaybackThreads.keyAt(i);
5510                    break;
5511                }
5512            }
5513            if (io == 0) {
5514               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5515                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5516                       io = mRecordThreads.keyAt(i);
5517                       break;
5518                   }
5519               }
5520            }
5521            // If no output thread contains the requested session ID, default to
5522            // first output. The effect chain will be moved to the correct output
5523            // thread when a track with the same session ID is created
5524            if (io == 0 && mPlaybackThreads.size()) {
5525                io = mPlaybackThreads.keyAt(0);
5526            }
5527            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5528        }
5529        ThreadBase *thread = checkRecordThread_l(io);
5530        if (thread == NULL) {
5531            thread = checkPlaybackThread_l(io);
5532            if (thread == NULL) {
5533                LOGE("createEffect() unknown output thread");
5534                lStatus = BAD_VALUE;
5535                goto Exit;
5536            }
5537        }
5538
5539        wclient = mClients.valueFor(pid);
5540
5541        if (wclient != NULL) {
5542            client = wclient.promote();
5543        } else {
5544            client = new Client(this, pid);
5545            mClients.add(pid, client);
5546        }
5547
5548        // create effect on selected output thread
5549        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5550                &desc, enabled, &lStatus);
5551        if (handle != 0 && id != NULL) {
5552            *id = handle->id();
5553        }
5554    }
5555
5556Exit:
5557    if(status) {
5558        *status = lStatus;
5559    }
5560    return handle;
5561}
5562
5563status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5564{
5565    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5566            sessionId, srcOutput, dstOutput);
5567    Mutex::Autolock _l(mLock);
5568    if (srcOutput == dstOutput) {
5569        LOGW("moveEffects() same dst and src outputs %d", dstOutput);
5570        return NO_ERROR;
5571    }
5572    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5573    if (srcThread == NULL) {
5574        LOGW("moveEffects() bad srcOutput %d", srcOutput);
5575        return BAD_VALUE;
5576    }
5577    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5578    if (dstThread == NULL) {
5579        LOGW("moveEffects() bad dstOutput %d", dstOutput);
5580        return BAD_VALUE;
5581    }
5582
5583    Mutex::Autolock _dl(dstThread->mLock);
5584    Mutex::Autolock _sl(srcThread->mLock);
5585    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5586
5587    return NO_ERROR;
5588}
5589
5590// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5591status_t AudioFlinger::moveEffectChain_l(int sessionId,
5592                                   AudioFlinger::PlaybackThread *srcThread,
5593                                   AudioFlinger::PlaybackThread *dstThread,
5594                                   bool reRegister)
5595{
5596    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5597            sessionId, srcThread, dstThread);
5598
5599    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5600    if (chain == 0) {
5601        LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5602                sessionId, srcThread);
5603        return INVALID_OPERATION;
5604    }
5605
5606    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5607    // so that a new chain is created with correct parameters when first effect is added. This is
5608    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5609    // removed.
5610    srcThread->removeEffectChain_l(chain);
5611
5612    // transfer all effects one by one so that new effect chain is created on new thread with
5613    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5614    int dstOutput = dstThread->id();
5615    sp<EffectChain> dstChain;
5616    uint32_t strategy = 0; // prevent compiler warning
5617    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5618    while (effect != 0) {
5619        srcThread->removeEffect_l(effect);
5620        dstThread->addEffect_l(effect);
5621        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5622        if (effect->state() == EffectModule::ACTIVE ||
5623                effect->state() == EffectModule::STOPPING) {
5624            effect->start();
5625        }
5626        // if the move request is not received from audio policy manager, the effect must be
5627        // re-registered with the new strategy and output
5628        if (dstChain == 0) {
5629            dstChain = effect->chain().promote();
5630            if (dstChain == 0) {
5631                LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5632                srcThread->addEffect_l(effect);
5633                return NO_INIT;
5634            }
5635            strategy = dstChain->strategy();
5636        }
5637        if (reRegister) {
5638            AudioSystem::unregisterEffect(effect->id());
5639            AudioSystem::registerEffect(&effect->desc(),
5640                                        dstOutput,
5641                                        strategy,
5642                                        sessionId,
5643                                        effect->id());
5644        }
5645        effect = chain->getEffectFromId_l(0);
5646    }
5647
5648    return NO_ERROR;
5649}
5650
5651
5652// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5653sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5654        const sp<AudioFlinger::Client>& client,
5655        const sp<IEffectClient>& effectClient,
5656        int32_t priority,
5657        int sessionId,
5658        effect_descriptor_t *desc,
5659        int *enabled,
5660        status_t *status
5661        )
5662{
5663    sp<EffectModule> effect;
5664    sp<EffectHandle> handle;
5665    status_t lStatus;
5666    sp<EffectChain> chain;
5667    bool chainCreated = false;
5668    bool effectCreated = false;
5669    bool effectRegistered = false;
5670
5671    lStatus = initCheck();
5672    if (lStatus != NO_ERROR) {
5673        LOGW("createEffect_l() Audio driver not initialized.");
5674        goto Exit;
5675    }
5676
5677    // Do not allow effects with session ID 0 on direct output or duplicating threads
5678    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5679    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5680        LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5681                desc->name, sessionId);
5682        lStatus = BAD_VALUE;
5683        goto Exit;
5684    }
5685    // Only Pre processor effects are allowed on input threads and only on input threads
5686    if ((mType == RECORD &&
5687            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5688            (mType != RECORD &&
5689                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5690        LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5691                desc->name, desc->flags, mType);
5692        lStatus = BAD_VALUE;
5693        goto Exit;
5694    }
5695
5696    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5697
5698    { // scope for mLock
5699        Mutex::Autolock _l(mLock);
5700
5701        // check for existing effect chain with the requested audio session
5702        chain = getEffectChain_l(sessionId);
5703        if (chain == 0) {
5704            // create a new chain for this session
5705            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5706            chain = new EffectChain(this, sessionId);
5707            addEffectChain_l(chain);
5708            chain->setStrategy(getStrategyForSession_l(sessionId));
5709            chainCreated = true;
5710        } else {
5711            effect = chain->getEffectFromDesc_l(desc);
5712        }
5713
5714        ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
5715
5716        if (effect == 0) {
5717            int id = mAudioFlinger->nextUniqueId();
5718            // Check CPU and memory usage
5719            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5720            if (lStatus != NO_ERROR) {
5721                goto Exit;
5722            }
5723            effectRegistered = true;
5724            // create a new effect module if none present in the chain
5725            effect = new EffectModule(this, chain, desc, id, sessionId);
5726            lStatus = effect->status();
5727            if (lStatus != NO_ERROR) {
5728                goto Exit;
5729            }
5730            lStatus = chain->addEffect_l(effect);
5731            if (lStatus != NO_ERROR) {
5732                goto Exit;
5733            }
5734            effectCreated = true;
5735
5736            effect->setDevice(mDevice);
5737            effect->setMode(mAudioFlinger->getMode());
5738        }
5739        // create effect handle and connect it to effect module
5740        handle = new EffectHandle(effect, client, effectClient, priority);
5741        lStatus = effect->addHandle(handle);
5742        if (enabled) {
5743            *enabled = (int)effect->isEnabled();
5744        }
5745    }
5746
5747Exit:
5748    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5749        Mutex::Autolock _l(mLock);
5750        if (effectCreated) {
5751            chain->removeEffect_l(effect);
5752        }
5753        if (effectRegistered) {
5754            AudioSystem::unregisterEffect(effect->id());
5755        }
5756        if (chainCreated) {
5757            removeEffectChain_l(chain);
5758        }
5759        handle.clear();
5760    }
5761
5762    if(status) {
5763        *status = lStatus;
5764    }
5765    return handle;
5766}
5767
5768sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5769{
5770    sp<EffectModule> effect;
5771
5772    sp<EffectChain> chain = getEffectChain_l(sessionId);
5773    if (chain != 0) {
5774        effect = chain->getEffectFromId_l(effectId);
5775    }
5776    return effect;
5777}
5778
5779// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5780// PlaybackThread::mLock held
5781status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5782{
5783    // check for existing effect chain with the requested audio session
5784    int sessionId = effect->sessionId();
5785    sp<EffectChain> chain = getEffectChain_l(sessionId);
5786    bool chainCreated = false;
5787
5788    if (chain == 0) {
5789        // create a new chain for this session
5790        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5791        chain = new EffectChain(this, sessionId);
5792        addEffectChain_l(chain);
5793        chain->setStrategy(getStrategyForSession_l(sessionId));
5794        chainCreated = true;
5795    }
5796    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5797
5798    if (chain->getEffectFromId_l(effect->id()) != 0) {
5799        LOGW("addEffect_l() %p effect %s already present in chain %p",
5800                this, effect->desc().name, chain.get());
5801        return BAD_VALUE;
5802    }
5803
5804    status_t status = chain->addEffect_l(effect);
5805    if (status != NO_ERROR) {
5806        if (chainCreated) {
5807            removeEffectChain_l(chain);
5808        }
5809        return status;
5810    }
5811
5812    effect->setDevice(mDevice);
5813    effect->setMode(mAudioFlinger->getMode());
5814    return NO_ERROR;
5815}
5816
5817void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5818
5819    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5820    effect_descriptor_t desc = effect->desc();
5821    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5822        detachAuxEffect_l(effect->id());
5823    }
5824
5825    sp<EffectChain> chain = effect->chain().promote();
5826    if (chain != 0) {
5827        // remove effect chain if removing last effect
5828        if (chain->removeEffect_l(effect) == 0) {
5829            removeEffectChain_l(chain);
5830        }
5831    } else {
5832        LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5833    }
5834}
5835
5836void AudioFlinger::ThreadBase::lockEffectChains_l(
5837        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5838{
5839    effectChains = mEffectChains;
5840    for (size_t i = 0; i < mEffectChains.size(); i++) {
5841        mEffectChains[i]->lock();
5842    }
5843}
5844
5845void AudioFlinger::ThreadBase::unlockEffectChains(
5846        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5847{
5848    for (size_t i = 0; i < effectChains.size(); i++) {
5849        effectChains[i]->unlock();
5850    }
5851}
5852
5853sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5854{
5855    Mutex::Autolock _l(mLock);
5856    return getEffectChain_l(sessionId);
5857}
5858
5859sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5860{
5861    sp<EffectChain> chain;
5862
5863    size_t size = mEffectChains.size();
5864    for (size_t i = 0; i < size; i++) {
5865        if (mEffectChains[i]->sessionId() == sessionId) {
5866            chain = mEffectChains[i];
5867            break;
5868        }
5869    }
5870    return chain;
5871}
5872
5873void AudioFlinger::ThreadBase::setMode(uint32_t mode)
5874{
5875    Mutex::Autolock _l(mLock);
5876    size_t size = mEffectChains.size();
5877    for (size_t i = 0; i < size; i++) {
5878        mEffectChains[i]->setMode_l(mode);
5879    }
5880}
5881
5882void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5883                                                    const wp<EffectHandle>& handle,
5884                                                    bool unpiniflast) {
5885
5886    Mutex::Autolock _l(mLock);
5887    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5888    // delete the effect module if removing last handle on it
5889    if (effect->removeHandle(handle) == 0) {
5890        if (!effect->isPinned() || unpiniflast) {
5891            removeEffect_l(effect);
5892            AudioSystem::unregisterEffect(effect->id());
5893        }
5894    }
5895}
5896
5897status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5898{
5899    int session = chain->sessionId();
5900    int16_t *buffer = mMixBuffer;
5901    bool ownsBuffer = false;
5902
5903    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5904    if (session > 0) {
5905        // Only one effect chain can be present in direct output thread and it uses
5906        // the mix buffer as input
5907        if (mType != DIRECT) {
5908            size_t numSamples = mFrameCount * mChannelCount;
5909            buffer = new int16_t[numSamples];
5910            memset(buffer, 0, numSamples * sizeof(int16_t));
5911            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5912            ownsBuffer = true;
5913        }
5914
5915        // Attach all tracks with same session ID to this chain.
5916        for (size_t i = 0; i < mTracks.size(); ++i) {
5917            sp<Track> track = mTracks[i];
5918            if (session == track->sessionId()) {
5919                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5920                track->setMainBuffer(buffer);
5921                chain->incTrackCnt();
5922            }
5923        }
5924
5925        // indicate all active tracks in the chain
5926        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5927            sp<Track> track = mActiveTracks[i].promote();
5928            if (track == 0) continue;
5929            if (session == track->sessionId()) {
5930                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5931                chain->incActiveTrackCnt();
5932            }
5933        }
5934    }
5935
5936    chain->setInBuffer(buffer, ownsBuffer);
5937    chain->setOutBuffer(mMixBuffer);
5938    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5939    // chains list in order to be processed last as it contains output stage effects
5940    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5941    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5942    // after track specific effects and before output stage
5943    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5944    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5945    // Effect chain for other sessions are inserted at beginning of effect
5946    // chains list to be processed before output mix effects. Relative order between other
5947    // sessions is not important
5948    size_t size = mEffectChains.size();
5949    size_t i = 0;
5950    for (i = 0; i < size; i++) {
5951        if (mEffectChains[i]->sessionId() < session) break;
5952    }
5953    mEffectChains.insertAt(chain, i);
5954    checkSuspendOnAddEffectChain_l(chain);
5955
5956    return NO_ERROR;
5957}
5958
5959size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
5960{
5961    int session = chain->sessionId();
5962
5963    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
5964
5965    for (size_t i = 0; i < mEffectChains.size(); i++) {
5966        if (chain == mEffectChains[i]) {
5967            mEffectChains.removeAt(i);
5968            // detach all active tracks from the chain
5969            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5970                sp<Track> track = mActiveTracks[i].promote();
5971                if (track == 0) continue;
5972                if (session == track->sessionId()) {
5973                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
5974                            chain.get(), session);
5975                    chain->decActiveTrackCnt();
5976                }
5977            }
5978
5979            // detach all tracks with same session ID from this chain
5980            for (size_t i = 0; i < mTracks.size(); ++i) {
5981                sp<Track> track = mTracks[i];
5982                if (session == track->sessionId()) {
5983                    track->setMainBuffer(mMixBuffer);
5984                    chain->decTrackCnt();
5985                }
5986            }
5987            break;
5988        }
5989    }
5990    return mEffectChains.size();
5991}
5992
5993status_t AudioFlinger::PlaybackThread::attachAuxEffect(
5994        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
5995{
5996    Mutex::Autolock _l(mLock);
5997    return attachAuxEffect_l(track, EffectId);
5998}
5999
6000status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6001        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6002{
6003    status_t status = NO_ERROR;
6004
6005    if (EffectId == 0) {
6006        track->setAuxBuffer(0, NULL);
6007    } else {
6008        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6009        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6010        if (effect != 0) {
6011            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6012                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6013            } else {
6014                status = INVALID_OPERATION;
6015            }
6016        } else {
6017            status = BAD_VALUE;
6018        }
6019    }
6020    return status;
6021}
6022
6023void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6024{
6025     for (size_t i = 0; i < mTracks.size(); ++i) {
6026        sp<Track> track = mTracks[i];
6027        if (track->auxEffectId() == effectId) {
6028            attachAuxEffect_l(track, 0);
6029        }
6030    }
6031}
6032
6033status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6034{
6035    // only one chain per input thread
6036    if (mEffectChains.size() != 0) {
6037        return INVALID_OPERATION;
6038    }
6039    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6040
6041    chain->setInBuffer(NULL);
6042    chain->setOutBuffer(NULL);
6043
6044    checkSuspendOnAddEffectChain_l(chain);
6045
6046    mEffectChains.add(chain);
6047
6048    return NO_ERROR;
6049}
6050
6051size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6052{
6053    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6054    LOGW_IF(mEffectChains.size() != 1,
6055            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6056            chain.get(), mEffectChains.size(), this);
6057    if (mEffectChains.size() == 1) {
6058        mEffectChains.removeAt(0);
6059    }
6060    return 0;
6061}
6062
6063// ----------------------------------------------------------------------------
6064//  EffectModule implementation
6065// ----------------------------------------------------------------------------
6066
6067#undef LOG_TAG
6068#define LOG_TAG "AudioFlinger::EffectModule"
6069
6070AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6071                                        const wp<AudioFlinger::EffectChain>& chain,
6072                                        effect_descriptor_t *desc,
6073                                        int id,
6074                                        int sessionId)
6075    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6076      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6077{
6078    ALOGV("Constructor %p", this);
6079    int lStatus;
6080    sp<ThreadBase> thread = mThread.promote();
6081    if (thread == 0) {
6082        return;
6083    }
6084
6085    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6086
6087    // create effect engine from effect factory
6088    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6089
6090    if (mStatus != NO_ERROR) {
6091        return;
6092    }
6093    lStatus = init();
6094    if (lStatus < 0) {
6095        mStatus = lStatus;
6096        goto Error;
6097    }
6098
6099    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6100        mPinned = true;
6101    }
6102    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6103    return;
6104Error:
6105    EffectRelease(mEffectInterface);
6106    mEffectInterface = NULL;
6107    ALOGV("Constructor Error %d", mStatus);
6108}
6109
6110AudioFlinger::EffectModule::~EffectModule()
6111{
6112    ALOGV("Destructor %p", this);
6113    if (mEffectInterface != NULL) {
6114        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6115                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6116            sp<ThreadBase> thread = mThread.promote();
6117            if (thread != 0) {
6118                audio_stream_t *stream = thread->stream();
6119                if (stream != NULL) {
6120                    stream->remove_audio_effect(stream, mEffectInterface);
6121                }
6122            }
6123        }
6124        // release effect engine
6125        EffectRelease(mEffectInterface);
6126    }
6127}
6128
6129status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6130{
6131    status_t status;
6132
6133    Mutex::Autolock _l(mLock);
6134    // First handle in mHandles has highest priority and controls the effect module
6135    int priority = handle->priority();
6136    size_t size = mHandles.size();
6137    sp<EffectHandle> h;
6138    size_t i;
6139    for (i = 0; i < size; i++) {
6140        h = mHandles[i].promote();
6141        if (h == 0) continue;
6142        if (h->priority() <= priority) break;
6143    }
6144    // if inserted in first place, move effect control from previous owner to this handle
6145    if (i == 0) {
6146        bool enabled = false;
6147        if (h != 0) {
6148            enabled = h->enabled();
6149            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6150        }
6151        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6152        status = NO_ERROR;
6153    } else {
6154        status = ALREADY_EXISTS;
6155    }
6156    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6157    mHandles.insertAt(handle, i);
6158    return status;
6159}
6160
6161size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6162{
6163    Mutex::Autolock _l(mLock);
6164    size_t size = mHandles.size();
6165    size_t i;
6166    for (i = 0; i < size; i++) {
6167        if (mHandles[i] == handle) break;
6168    }
6169    if (i == size) {
6170        return size;
6171    }
6172    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6173
6174    bool enabled = false;
6175    EffectHandle *hdl = handle.unsafe_get();
6176    if (hdl) {
6177        ALOGV("removeHandle() unsafe_get OK");
6178        enabled = hdl->enabled();
6179    }
6180    mHandles.removeAt(i);
6181    size = mHandles.size();
6182    // if removed from first place, move effect control from this handle to next in line
6183    if (i == 0 && size != 0) {
6184        sp<EffectHandle> h = mHandles[0].promote();
6185        if (h != 0) {
6186            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6187        }
6188    }
6189
6190    // Prevent calls to process() and other functions on effect interface from now on.
6191    // The effect engine will be released by the destructor when the last strong reference on
6192    // this object is released which can happen after next process is called.
6193    if (size == 0 && !mPinned) {
6194        mState = DESTROYED;
6195    }
6196
6197    return size;
6198}
6199
6200sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6201{
6202    Mutex::Autolock _l(mLock);
6203    sp<EffectHandle> handle;
6204    if (mHandles.size() != 0) {
6205        handle = mHandles[0].promote();
6206    }
6207    return handle;
6208}
6209
6210void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6211{
6212    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6213    // keep a strong reference on this EffectModule to avoid calling the
6214    // destructor before we exit
6215    sp<EffectModule> keep(this);
6216    {
6217        sp<ThreadBase> thread = mThread.promote();
6218        if (thread != 0) {
6219            thread->disconnectEffect(keep, handle, unpiniflast);
6220        }
6221    }
6222}
6223
6224void AudioFlinger::EffectModule::updateState() {
6225    Mutex::Autolock _l(mLock);
6226
6227    switch (mState) {
6228    case RESTART:
6229        reset_l();
6230        // FALL THROUGH
6231
6232    case STARTING:
6233        // clear auxiliary effect input buffer for next accumulation
6234        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6235            memset(mConfig.inputCfg.buffer.raw,
6236                   0,
6237                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6238        }
6239        start_l();
6240        mState = ACTIVE;
6241        break;
6242    case STOPPING:
6243        stop_l();
6244        mDisableWaitCnt = mMaxDisableWaitCnt;
6245        mState = STOPPED;
6246        break;
6247    case STOPPED:
6248        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6249        // turn off sequence.
6250        if (--mDisableWaitCnt == 0) {
6251            reset_l();
6252            mState = IDLE;
6253        }
6254        break;
6255    default: //IDLE , ACTIVE, DESTROYED
6256        break;
6257    }
6258}
6259
6260void AudioFlinger::EffectModule::process()
6261{
6262    Mutex::Autolock _l(mLock);
6263
6264    if (mState == DESTROYED || mEffectInterface == NULL ||
6265            mConfig.inputCfg.buffer.raw == NULL ||
6266            mConfig.outputCfg.buffer.raw == NULL) {
6267        return;
6268    }
6269
6270    if (isProcessEnabled()) {
6271        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6272        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6273            ditherAndClamp(mConfig.inputCfg.buffer.s32,
6274                                        mConfig.inputCfg.buffer.s32,
6275                                        mConfig.inputCfg.buffer.frameCount/2);
6276        }
6277
6278        // do the actual processing in the effect engine
6279        int ret = (*mEffectInterface)->process(mEffectInterface,
6280                                               &mConfig.inputCfg.buffer,
6281                                               &mConfig.outputCfg.buffer);
6282
6283        // force transition to IDLE state when engine is ready
6284        if (mState == STOPPED && ret == -ENODATA) {
6285            mDisableWaitCnt = 1;
6286        }
6287
6288        // clear auxiliary effect input buffer for next accumulation
6289        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6290            memset(mConfig.inputCfg.buffer.raw, 0,
6291                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6292        }
6293    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6294                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6295        // If an insert effect is idle and input buffer is different from output buffer,
6296        // accumulate input onto output
6297        sp<EffectChain> chain = mChain.promote();
6298        if (chain != 0 && chain->activeTrackCnt() != 0) {
6299            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6300            int16_t *in = mConfig.inputCfg.buffer.s16;
6301            int16_t *out = mConfig.outputCfg.buffer.s16;
6302            for (size_t i = 0; i < frameCnt; i++) {
6303                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6304            }
6305        }
6306    }
6307}
6308
6309void AudioFlinger::EffectModule::reset_l()
6310{
6311    if (mEffectInterface == NULL) {
6312        return;
6313    }
6314    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6315}
6316
6317status_t AudioFlinger::EffectModule::configure()
6318{
6319    uint32_t channels;
6320    if (mEffectInterface == NULL) {
6321        return NO_INIT;
6322    }
6323
6324    sp<ThreadBase> thread = mThread.promote();
6325    if (thread == 0) {
6326        return DEAD_OBJECT;
6327    }
6328
6329    // TODO: handle configuration of effects replacing track process
6330    if (thread->channelCount() == 1) {
6331        channels = AUDIO_CHANNEL_OUT_MONO;
6332    } else {
6333        channels = AUDIO_CHANNEL_OUT_STEREO;
6334    }
6335
6336    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6337        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6338    } else {
6339        mConfig.inputCfg.channels = channels;
6340    }
6341    mConfig.outputCfg.channels = channels;
6342    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6343    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6344    mConfig.inputCfg.samplingRate = thread->sampleRate();
6345    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6346    mConfig.inputCfg.bufferProvider.cookie = NULL;
6347    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6348    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6349    mConfig.outputCfg.bufferProvider.cookie = NULL;
6350    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6351    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6352    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6353    // Insert effect:
6354    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6355    // always overwrites output buffer: input buffer == output buffer
6356    // - in other sessions:
6357    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6358    //      other effect: overwrites output buffer: input buffer == output buffer
6359    // Auxiliary effect:
6360    //      accumulates in output buffer: input buffer != output buffer
6361    // Therefore: accumulate <=> input buffer != output buffer
6362    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6363        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6364    } else {
6365        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6366    }
6367    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6368    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6369    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6370    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6371
6372    ALOGV("configure() %p thread %p buffer %p framecount %d",
6373            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6374
6375    status_t cmdStatus;
6376    uint32_t size = sizeof(int);
6377    status_t status = (*mEffectInterface)->command(mEffectInterface,
6378                                                   EFFECT_CMD_CONFIGURE,
6379                                                   sizeof(effect_config_t),
6380                                                   &mConfig,
6381                                                   &size,
6382                                                   &cmdStatus);
6383    if (status == 0) {
6384        status = cmdStatus;
6385    }
6386
6387    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6388            (1000 * mConfig.outputCfg.buffer.frameCount);
6389
6390    return status;
6391}
6392
6393status_t AudioFlinger::EffectModule::init()
6394{
6395    Mutex::Autolock _l(mLock);
6396    if (mEffectInterface == NULL) {
6397        return NO_INIT;
6398    }
6399    status_t cmdStatus;
6400    uint32_t size = sizeof(status_t);
6401    status_t status = (*mEffectInterface)->command(mEffectInterface,
6402                                                   EFFECT_CMD_INIT,
6403                                                   0,
6404                                                   NULL,
6405                                                   &size,
6406                                                   &cmdStatus);
6407    if (status == 0) {
6408        status = cmdStatus;
6409    }
6410    return status;
6411}
6412
6413status_t AudioFlinger::EffectModule::start()
6414{
6415    Mutex::Autolock _l(mLock);
6416    return start_l();
6417}
6418
6419status_t AudioFlinger::EffectModule::start_l()
6420{
6421    if (mEffectInterface == NULL) {
6422        return NO_INIT;
6423    }
6424    status_t cmdStatus;
6425    uint32_t size = sizeof(status_t);
6426    status_t status = (*mEffectInterface)->command(mEffectInterface,
6427                                                   EFFECT_CMD_ENABLE,
6428                                                   0,
6429                                                   NULL,
6430                                                   &size,
6431                                                   &cmdStatus);
6432    if (status == 0) {
6433        status = cmdStatus;
6434    }
6435    if (status == 0 &&
6436            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6437             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6438        sp<ThreadBase> thread = mThread.promote();
6439        if (thread != 0) {
6440            audio_stream_t *stream = thread->stream();
6441            if (stream != NULL) {
6442                stream->add_audio_effect(stream, mEffectInterface);
6443            }
6444        }
6445    }
6446    return status;
6447}
6448
6449status_t AudioFlinger::EffectModule::stop()
6450{
6451    Mutex::Autolock _l(mLock);
6452    return stop_l();
6453}
6454
6455status_t AudioFlinger::EffectModule::stop_l()
6456{
6457    if (mEffectInterface == NULL) {
6458        return NO_INIT;
6459    }
6460    status_t cmdStatus;
6461    uint32_t size = sizeof(status_t);
6462    status_t status = (*mEffectInterface)->command(mEffectInterface,
6463                                                   EFFECT_CMD_DISABLE,
6464                                                   0,
6465                                                   NULL,
6466                                                   &size,
6467                                                   &cmdStatus);
6468    if (status == 0) {
6469        status = cmdStatus;
6470    }
6471    if (status == 0 &&
6472            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6473             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6474        sp<ThreadBase> thread = mThread.promote();
6475        if (thread != 0) {
6476            audio_stream_t *stream = thread->stream();
6477            if (stream != NULL) {
6478                stream->remove_audio_effect(stream, mEffectInterface);
6479            }
6480        }
6481    }
6482    return status;
6483}
6484
6485status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6486                                             uint32_t cmdSize,
6487                                             void *pCmdData,
6488                                             uint32_t *replySize,
6489                                             void *pReplyData)
6490{
6491    Mutex::Autolock _l(mLock);
6492//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6493
6494    if (mState == DESTROYED || mEffectInterface == NULL) {
6495        return NO_INIT;
6496    }
6497    status_t status = (*mEffectInterface)->command(mEffectInterface,
6498                                                   cmdCode,
6499                                                   cmdSize,
6500                                                   pCmdData,
6501                                                   replySize,
6502                                                   pReplyData);
6503    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6504        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6505        for (size_t i = 1; i < mHandles.size(); i++) {
6506            sp<EffectHandle> h = mHandles[i].promote();
6507            if (h != 0) {
6508                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6509            }
6510        }
6511    }
6512    return status;
6513}
6514
6515status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6516{
6517
6518    Mutex::Autolock _l(mLock);
6519    ALOGV("setEnabled %p enabled %d", this, enabled);
6520
6521    if (enabled != isEnabled()) {
6522        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6523        if (enabled && status != NO_ERROR) {
6524            return status;
6525        }
6526
6527        switch (mState) {
6528        // going from disabled to enabled
6529        case IDLE:
6530            mState = STARTING;
6531            break;
6532        case STOPPED:
6533            mState = RESTART;
6534            break;
6535        case STOPPING:
6536            mState = ACTIVE;
6537            break;
6538
6539        // going from enabled to disabled
6540        case RESTART:
6541            mState = STOPPED;
6542            break;
6543        case STARTING:
6544            mState = IDLE;
6545            break;
6546        case ACTIVE:
6547            mState = STOPPING;
6548            break;
6549        case DESTROYED:
6550            return NO_ERROR; // simply ignore as we are being destroyed
6551        }
6552        for (size_t i = 1; i < mHandles.size(); i++) {
6553            sp<EffectHandle> h = mHandles[i].promote();
6554            if (h != 0) {
6555                h->setEnabled(enabled);
6556            }
6557        }
6558    }
6559    return NO_ERROR;
6560}
6561
6562bool AudioFlinger::EffectModule::isEnabled()
6563{
6564    switch (mState) {
6565    case RESTART:
6566    case STARTING:
6567    case ACTIVE:
6568        return true;
6569    case IDLE:
6570    case STOPPING:
6571    case STOPPED:
6572    case DESTROYED:
6573    default:
6574        return false;
6575    }
6576}
6577
6578bool AudioFlinger::EffectModule::isProcessEnabled()
6579{
6580    switch (mState) {
6581    case RESTART:
6582    case ACTIVE:
6583    case STOPPING:
6584    case STOPPED:
6585        return true;
6586    case IDLE:
6587    case STARTING:
6588    case DESTROYED:
6589    default:
6590        return false;
6591    }
6592}
6593
6594status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6595{
6596    Mutex::Autolock _l(mLock);
6597    status_t status = NO_ERROR;
6598
6599    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6600    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6601    if (isProcessEnabled() &&
6602            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6603            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6604        status_t cmdStatus;
6605        uint32_t volume[2];
6606        uint32_t *pVolume = NULL;
6607        uint32_t size = sizeof(volume);
6608        volume[0] = *left;
6609        volume[1] = *right;
6610        if (controller) {
6611            pVolume = volume;
6612        }
6613        status = (*mEffectInterface)->command(mEffectInterface,
6614                                              EFFECT_CMD_SET_VOLUME,
6615                                              size,
6616                                              volume,
6617                                              &size,
6618                                              pVolume);
6619        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6620            *left = volume[0];
6621            *right = volume[1];
6622        }
6623    }
6624    return status;
6625}
6626
6627status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6628{
6629    Mutex::Autolock _l(mLock);
6630    status_t status = NO_ERROR;
6631    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6632        // audio pre processing modules on RecordThread can receive both output and
6633        // input device indication in the same call
6634        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6635        if (dev) {
6636            status_t cmdStatus;
6637            uint32_t size = sizeof(status_t);
6638
6639            status = (*mEffectInterface)->command(mEffectInterface,
6640                                                  EFFECT_CMD_SET_DEVICE,
6641                                                  sizeof(uint32_t),
6642                                                  &dev,
6643                                                  &size,
6644                                                  &cmdStatus);
6645            if (status == NO_ERROR) {
6646                status = cmdStatus;
6647            }
6648        }
6649        dev = device & AUDIO_DEVICE_IN_ALL;
6650        if (dev) {
6651            status_t cmdStatus;
6652            uint32_t size = sizeof(status_t);
6653
6654            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6655                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6656                                                  sizeof(uint32_t),
6657                                                  &dev,
6658                                                  &size,
6659                                                  &cmdStatus);
6660            if (status2 == NO_ERROR) {
6661                status2 = cmdStatus;
6662            }
6663            if (status == NO_ERROR) {
6664                status = status2;
6665            }
6666        }
6667    }
6668    return status;
6669}
6670
6671status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
6672{
6673    Mutex::Autolock _l(mLock);
6674    status_t status = NO_ERROR;
6675    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6676        status_t cmdStatus;
6677        uint32_t size = sizeof(status_t);
6678        status = (*mEffectInterface)->command(mEffectInterface,
6679                                              EFFECT_CMD_SET_AUDIO_MODE,
6680                                              sizeof(int),
6681                                              &mode,
6682                                              &size,
6683                                              &cmdStatus);
6684        if (status == NO_ERROR) {
6685            status = cmdStatus;
6686        }
6687    }
6688    return status;
6689}
6690
6691void AudioFlinger::EffectModule::setSuspended(bool suspended)
6692{
6693    Mutex::Autolock _l(mLock);
6694    mSuspended = suspended;
6695}
6696bool AudioFlinger::EffectModule::suspended()
6697{
6698    Mutex::Autolock _l(mLock);
6699    return mSuspended;
6700}
6701
6702status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6703{
6704    const size_t SIZE = 256;
6705    char buffer[SIZE];
6706    String8 result;
6707
6708    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6709    result.append(buffer);
6710
6711    bool locked = tryLock(mLock);
6712    // failed to lock - AudioFlinger is probably deadlocked
6713    if (!locked) {
6714        result.append("\t\tCould not lock Fx mutex:\n");
6715    }
6716
6717    result.append("\t\tSession Status State Engine:\n");
6718    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6719            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6720    result.append(buffer);
6721
6722    result.append("\t\tDescriptor:\n");
6723    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6724            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6725            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6726            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6727    result.append(buffer);
6728    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6729                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6730                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6731                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6732    result.append(buffer);
6733    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6734            mDescriptor.apiVersion,
6735            mDescriptor.flags);
6736    result.append(buffer);
6737    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6738            mDescriptor.name);
6739    result.append(buffer);
6740    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6741            mDescriptor.implementor);
6742    result.append(buffer);
6743
6744    result.append("\t\t- Input configuration:\n");
6745    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6746    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6747            (uint32_t)mConfig.inputCfg.buffer.raw,
6748            mConfig.inputCfg.buffer.frameCount,
6749            mConfig.inputCfg.samplingRate,
6750            mConfig.inputCfg.channels,
6751            mConfig.inputCfg.format);
6752    result.append(buffer);
6753
6754    result.append("\t\t- Output configuration:\n");
6755    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6756    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6757            (uint32_t)mConfig.outputCfg.buffer.raw,
6758            mConfig.outputCfg.buffer.frameCount,
6759            mConfig.outputCfg.samplingRate,
6760            mConfig.outputCfg.channels,
6761            mConfig.outputCfg.format);
6762    result.append(buffer);
6763
6764    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6765    result.append(buffer);
6766    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6767    for (size_t i = 0; i < mHandles.size(); ++i) {
6768        sp<EffectHandle> handle = mHandles[i].promote();
6769        if (handle != 0) {
6770            handle->dump(buffer, SIZE);
6771            result.append(buffer);
6772        }
6773    }
6774
6775    result.append("\n");
6776
6777    write(fd, result.string(), result.length());
6778
6779    if (locked) {
6780        mLock.unlock();
6781    }
6782
6783    return NO_ERROR;
6784}
6785
6786// ----------------------------------------------------------------------------
6787//  EffectHandle implementation
6788// ----------------------------------------------------------------------------
6789
6790#undef LOG_TAG
6791#define LOG_TAG "AudioFlinger::EffectHandle"
6792
6793AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6794                                        const sp<AudioFlinger::Client>& client,
6795                                        const sp<IEffectClient>& effectClient,
6796                                        int32_t priority)
6797    : BnEffect(),
6798    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6799    mPriority(priority), mHasControl(false), mEnabled(false)
6800{
6801    ALOGV("constructor %p", this);
6802
6803    if (client == 0) {
6804        return;
6805    }
6806    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6807    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6808    if (mCblkMemory != 0) {
6809        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6810
6811        if (mCblk) {
6812            new(mCblk) effect_param_cblk_t();
6813            mBuffer = (uint8_t *)mCblk + bufOffset;
6814         }
6815    } else {
6816        LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6817        return;
6818    }
6819}
6820
6821AudioFlinger::EffectHandle::~EffectHandle()
6822{
6823    ALOGV("Destructor %p", this);
6824    disconnect(false);
6825    ALOGV("Destructor DONE %p", this);
6826}
6827
6828status_t AudioFlinger::EffectHandle::enable()
6829{
6830    ALOGV("enable %p", this);
6831    if (!mHasControl) return INVALID_OPERATION;
6832    if (mEffect == 0) return DEAD_OBJECT;
6833
6834    if (mEnabled) {
6835        return NO_ERROR;
6836    }
6837
6838    mEnabled = true;
6839
6840    sp<ThreadBase> thread = mEffect->thread().promote();
6841    if (thread != 0) {
6842        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6843    }
6844
6845    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6846    if (mEffect->suspended()) {
6847        return NO_ERROR;
6848    }
6849
6850    status_t status = mEffect->setEnabled(true);
6851    if (status != NO_ERROR) {
6852        if (thread != 0) {
6853            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6854        }
6855        mEnabled = false;
6856    }
6857    return status;
6858}
6859
6860status_t AudioFlinger::EffectHandle::disable()
6861{
6862    ALOGV("disable %p", this);
6863    if (!mHasControl) return INVALID_OPERATION;
6864    if (mEffect == 0) return DEAD_OBJECT;
6865
6866    if (!mEnabled) {
6867        return NO_ERROR;
6868    }
6869    mEnabled = false;
6870
6871    if (mEffect->suspended()) {
6872        return NO_ERROR;
6873    }
6874
6875    status_t status = mEffect->setEnabled(false);
6876
6877    sp<ThreadBase> thread = mEffect->thread().promote();
6878    if (thread != 0) {
6879        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6880    }
6881
6882    return status;
6883}
6884
6885void AudioFlinger::EffectHandle::disconnect()
6886{
6887    disconnect(true);
6888}
6889
6890void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6891{
6892    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6893    if (mEffect == 0) {
6894        return;
6895    }
6896    mEffect->disconnect(this, unpiniflast);
6897
6898    if (mHasControl && mEnabled) {
6899        sp<ThreadBase> thread = mEffect->thread().promote();
6900        if (thread != 0) {
6901            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6902        }
6903    }
6904
6905    // release sp on module => module destructor can be called now
6906    mEffect.clear();
6907    if (mClient != 0) {
6908        if (mCblk) {
6909            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6910        }
6911        mCblkMemory.clear();            // and free the shared memory
6912        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6913        mClient.clear();
6914    }
6915}
6916
6917status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6918                                             uint32_t cmdSize,
6919                                             void *pCmdData,
6920                                             uint32_t *replySize,
6921                                             void *pReplyData)
6922{
6923//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6924//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6925
6926    // only get parameter command is permitted for applications not controlling the effect
6927    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6928        return INVALID_OPERATION;
6929    }
6930    if (mEffect == 0) return DEAD_OBJECT;
6931    if (mClient == 0) return INVALID_OPERATION;
6932
6933    // handle commands that are not forwarded transparently to effect engine
6934    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6935        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6936        // no risk to block the whole media server process or mixer threads is we are stuck here
6937        Mutex::Autolock _l(mCblk->lock);
6938        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6939            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6940            mCblk->serverIndex = 0;
6941            mCblk->clientIndex = 0;
6942            return BAD_VALUE;
6943        }
6944        status_t status = NO_ERROR;
6945        while (mCblk->serverIndex < mCblk->clientIndex) {
6946            int reply;
6947            uint32_t rsize = sizeof(int);
6948            int *p = (int *)(mBuffer + mCblk->serverIndex);
6949            int size = *p++;
6950            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
6951                LOGW("command(): invalid parameter block size");
6952                break;
6953            }
6954            effect_param_t *param = (effect_param_t *)p;
6955            if (param->psize == 0 || param->vsize == 0) {
6956                LOGW("command(): null parameter or value size");
6957                mCblk->serverIndex += size;
6958                continue;
6959            }
6960            uint32_t psize = sizeof(effect_param_t) +
6961                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
6962                             param->vsize;
6963            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
6964                                            psize,
6965                                            p,
6966                                            &rsize,
6967                                            &reply);
6968            // stop at first error encountered
6969            if (ret != NO_ERROR) {
6970                status = ret;
6971                *(int *)pReplyData = reply;
6972                break;
6973            } else if (reply != NO_ERROR) {
6974                *(int *)pReplyData = reply;
6975                break;
6976            }
6977            mCblk->serverIndex += size;
6978        }
6979        mCblk->serverIndex = 0;
6980        mCblk->clientIndex = 0;
6981        return status;
6982    } else if (cmdCode == EFFECT_CMD_ENABLE) {
6983        *(int *)pReplyData = NO_ERROR;
6984        return enable();
6985    } else if (cmdCode == EFFECT_CMD_DISABLE) {
6986        *(int *)pReplyData = NO_ERROR;
6987        return disable();
6988    }
6989
6990    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
6991}
6992
6993sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
6994    return mCblkMemory;
6995}
6996
6997void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
6998{
6999    ALOGV("setControl %p control %d", this, hasControl);
7000
7001    mHasControl = hasControl;
7002    mEnabled = enabled;
7003
7004    if (signal && mEffectClient != 0) {
7005        mEffectClient->controlStatusChanged(hasControl);
7006    }
7007}
7008
7009void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7010                                                 uint32_t cmdSize,
7011                                                 void *pCmdData,
7012                                                 uint32_t replySize,
7013                                                 void *pReplyData)
7014{
7015    if (mEffectClient != 0) {
7016        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7017    }
7018}
7019
7020
7021
7022void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7023{
7024    if (mEffectClient != 0) {
7025        mEffectClient->enableStatusChanged(enabled);
7026    }
7027}
7028
7029status_t AudioFlinger::EffectHandle::onTransact(
7030    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7031{
7032    return BnEffect::onTransact(code, data, reply, flags);
7033}
7034
7035
7036void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7037{
7038    bool locked = mCblk ? tryLock(mCblk->lock) : false;
7039
7040    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7041            (mClient == NULL) ? getpid() : mClient->pid(),
7042            mPriority,
7043            mHasControl,
7044            !locked,
7045            mCblk ? mCblk->clientIndex : 0,
7046            mCblk ? mCblk->serverIndex : 0
7047            );
7048
7049    if (locked) {
7050        mCblk->lock.unlock();
7051    }
7052}
7053
7054#undef LOG_TAG
7055#define LOG_TAG "AudioFlinger::EffectChain"
7056
7057AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7058                                        int sessionId)
7059    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7060      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7061      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7062{
7063    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7064    sp<ThreadBase> thread = mThread.promote();
7065    if (thread == 0) {
7066        return;
7067    }
7068    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7069                                    thread->frameCount();
7070}
7071
7072AudioFlinger::EffectChain::~EffectChain()
7073{
7074    if (mOwnInBuffer) {
7075        delete mInBuffer;
7076    }
7077
7078}
7079
7080// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7081sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7082{
7083    sp<EffectModule> effect;
7084    size_t size = mEffects.size();
7085
7086    for (size_t i = 0; i < size; i++) {
7087        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7088            effect = mEffects[i];
7089            break;
7090        }
7091    }
7092    return effect;
7093}
7094
7095// getEffectFromId_l() must be called with ThreadBase::mLock held
7096sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7097{
7098    sp<EffectModule> effect;
7099    size_t size = mEffects.size();
7100
7101    for (size_t i = 0; i < size; i++) {
7102        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7103        if (id == 0 || mEffects[i]->id() == id) {
7104            effect = mEffects[i];
7105            break;
7106        }
7107    }
7108    return effect;
7109}
7110
7111// getEffectFromType_l() must be called with ThreadBase::mLock held
7112sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7113        const effect_uuid_t *type)
7114{
7115    sp<EffectModule> effect;
7116    size_t size = mEffects.size();
7117
7118    for (size_t i = 0; i < size; i++) {
7119        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7120            effect = mEffects[i];
7121            break;
7122        }
7123    }
7124    return effect;
7125}
7126
7127// Must be called with EffectChain::mLock locked
7128void AudioFlinger::EffectChain::process_l()
7129{
7130    sp<ThreadBase> thread = mThread.promote();
7131    if (thread == 0) {
7132        LOGW("process_l(): cannot promote mixer thread");
7133        return;
7134    }
7135    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7136            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7137    // always process effects unless no more tracks are on the session and the effect tail
7138    // has been rendered
7139    bool doProcess = true;
7140    if (!isGlobalSession) {
7141        bool tracksOnSession = (trackCnt() != 0);
7142
7143        if (!tracksOnSession && mTailBufferCount == 0) {
7144            doProcess = false;
7145        }
7146
7147        if (activeTrackCnt() == 0) {
7148            // if no track is active and the effect tail has not been rendered,
7149            // the input buffer must be cleared here as the mixer process will not do it
7150            if (tracksOnSession || mTailBufferCount > 0) {
7151                size_t numSamples = thread->frameCount() * thread->channelCount();
7152                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7153                if (mTailBufferCount > 0) {
7154                    mTailBufferCount--;
7155                }
7156            }
7157        }
7158    }
7159
7160    size_t size = mEffects.size();
7161    if (doProcess) {
7162        for (size_t i = 0; i < size; i++) {
7163            mEffects[i]->process();
7164        }
7165    }
7166    for (size_t i = 0; i < size; i++) {
7167        mEffects[i]->updateState();
7168    }
7169}
7170
7171// addEffect_l() must be called with PlaybackThread::mLock held
7172status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7173{
7174    effect_descriptor_t desc = effect->desc();
7175    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7176
7177    Mutex::Autolock _l(mLock);
7178    effect->setChain(this);
7179    sp<ThreadBase> thread = mThread.promote();
7180    if (thread == 0) {
7181        return NO_INIT;
7182    }
7183    effect->setThread(thread);
7184
7185    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7186        // Auxiliary effects are inserted at the beginning of mEffects vector as
7187        // they are processed first and accumulated in chain input buffer
7188        mEffects.insertAt(effect, 0);
7189
7190        // the input buffer for auxiliary effect contains mono samples in
7191        // 32 bit format. This is to avoid saturation in AudoMixer
7192        // accumulation stage. Saturation is done in EffectModule::process() before
7193        // calling the process in effect engine
7194        size_t numSamples = thread->frameCount();
7195        int32_t *buffer = new int32_t[numSamples];
7196        memset(buffer, 0, numSamples * sizeof(int32_t));
7197        effect->setInBuffer((int16_t *)buffer);
7198        // auxiliary effects output samples to chain input buffer for further processing
7199        // by insert effects
7200        effect->setOutBuffer(mInBuffer);
7201    } else {
7202        // Insert effects are inserted at the end of mEffects vector as they are processed
7203        //  after track and auxiliary effects.
7204        // Insert effect order as a function of indicated preference:
7205        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7206        //  another effect is present
7207        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7208        //  last effect claiming first position
7209        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7210        //  first effect claiming last position
7211        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7212        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7213        // already present
7214
7215        int size = (int)mEffects.size();
7216        int idx_insert = size;
7217        int idx_insert_first = -1;
7218        int idx_insert_last = -1;
7219
7220        for (int i = 0; i < size; i++) {
7221            effect_descriptor_t d = mEffects[i]->desc();
7222            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7223            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7224            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7225                // check invalid effect chaining combinations
7226                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7227                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7228                    LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7229                    return INVALID_OPERATION;
7230                }
7231                // remember position of first insert effect and by default
7232                // select this as insert position for new effect
7233                if (idx_insert == size) {
7234                    idx_insert = i;
7235                }
7236                // remember position of last insert effect claiming
7237                // first position
7238                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7239                    idx_insert_first = i;
7240                }
7241                // remember position of first insert effect claiming
7242                // last position
7243                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7244                    idx_insert_last == -1) {
7245                    idx_insert_last = i;
7246                }
7247            }
7248        }
7249
7250        // modify idx_insert from first position if needed
7251        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7252            if (idx_insert_last != -1) {
7253                idx_insert = idx_insert_last;
7254            } else {
7255                idx_insert = size;
7256            }
7257        } else {
7258            if (idx_insert_first != -1) {
7259                idx_insert = idx_insert_first + 1;
7260            }
7261        }
7262
7263        // always read samples from chain input buffer
7264        effect->setInBuffer(mInBuffer);
7265
7266        // if last effect in the chain, output samples to chain
7267        // output buffer, otherwise to chain input buffer
7268        if (idx_insert == size) {
7269            if (idx_insert != 0) {
7270                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7271                mEffects[idx_insert-1]->configure();
7272            }
7273            effect->setOutBuffer(mOutBuffer);
7274        } else {
7275            effect->setOutBuffer(mInBuffer);
7276        }
7277        mEffects.insertAt(effect, idx_insert);
7278
7279        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7280    }
7281    effect->configure();
7282    return NO_ERROR;
7283}
7284
7285// removeEffect_l() must be called with PlaybackThread::mLock held
7286size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7287{
7288    Mutex::Autolock _l(mLock);
7289    int size = (int)mEffects.size();
7290    int i;
7291    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7292
7293    for (i = 0; i < size; i++) {
7294        if (effect == mEffects[i]) {
7295            // calling stop here will remove pre-processing effect from the audio HAL.
7296            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7297            // the middle of a read from audio HAL
7298            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7299                    mEffects[i]->state() == EffectModule::STOPPING) {
7300                mEffects[i]->stop();
7301            }
7302            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7303                delete[] effect->inBuffer();
7304            } else {
7305                if (i == size - 1 && i != 0) {
7306                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7307                    mEffects[i - 1]->configure();
7308                }
7309            }
7310            mEffects.removeAt(i);
7311            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7312            break;
7313        }
7314    }
7315
7316    return mEffects.size();
7317}
7318
7319// setDevice_l() must be called with PlaybackThread::mLock held
7320void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7321{
7322    size_t size = mEffects.size();
7323    for (size_t i = 0; i < size; i++) {
7324        mEffects[i]->setDevice(device);
7325    }
7326}
7327
7328// setMode_l() must be called with PlaybackThread::mLock held
7329void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
7330{
7331    size_t size = mEffects.size();
7332    for (size_t i = 0; i < size; i++) {
7333        mEffects[i]->setMode(mode);
7334    }
7335}
7336
7337// setVolume_l() must be called with PlaybackThread::mLock held
7338bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7339{
7340    uint32_t newLeft = *left;
7341    uint32_t newRight = *right;
7342    bool hasControl = false;
7343    int ctrlIdx = -1;
7344    size_t size = mEffects.size();
7345
7346    // first update volume controller
7347    for (size_t i = size; i > 0; i--) {
7348        if (mEffects[i - 1]->isProcessEnabled() &&
7349            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7350            ctrlIdx = i - 1;
7351            hasControl = true;
7352            break;
7353        }
7354    }
7355
7356    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7357        if (hasControl) {
7358            *left = mNewLeftVolume;
7359            *right = mNewRightVolume;
7360        }
7361        return hasControl;
7362    }
7363
7364    mVolumeCtrlIdx = ctrlIdx;
7365    mLeftVolume = newLeft;
7366    mRightVolume = newRight;
7367
7368    // second get volume update from volume controller
7369    if (ctrlIdx >= 0) {
7370        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7371        mNewLeftVolume = newLeft;
7372        mNewRightVolume = newRight;
7373    }
7374    // then indicate volume to all other effects in chain.
7375    // Pass altered volume to effects before volume controller
7376    // and requested volume to effects after controller
7377    uint32_t lVol = newLeft;
7378    uint32_t rVol = newRight;
7379
7380    for (size_t i = 0; i < size; i++) {
7381        if ((int)i == ctrlIdx) continue;
7382        // this also works for ctrlIdx == -1 when there is no volume controller
7383        if ((int)i > ctrlIdx) {
7384            lVol = *left;
7385            rVol = *right;
7386        }
7387        mEffects[i]->setVolume(&lVol, &rVol, false);
7388    }
7389    *left = newLeft;
7390    *right = newRight;
7391
7392    return hasControl;
7393}
7394
7395status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7396{
7397    const size_t SIZE = 256;
7398    char buffer[SIZE];
7399    String8 result;
7400
7401    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7402    result.append(buffer);
7403
7404    bool locked = tryLock(mLock);
7405    // failed to lock - AudioFlinger is probably deadlocked
7406    if (!locked) {
7407        result.append("\tCould not lock mutex:\n");
7408    }
7409
7410    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7411    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7412            mEffects.size(),
7413            (uint32_t)mInBuffer,
7414            (uint32_t)mOutBuffer,
7415            mActiveTrackCnt);
7416    result.append(buffer);
7417    write(fd, result.string(), result.size());
7418
7419    for (size_t i = 0; i < mEffects.size(); ++i) {
7420        sp<EffectModule> effect = mEffects[i];
7421        if (effect != 0) {
7422            effect->dump(fd, args);
7423        }
7424    }
7425
7426    if (locked) {
7427        mLock.unlock();
7428    }
7429
7430    return NO_ERROR;
7431}
7432
7433// must be called with ThreadBase::mLock held
7434void AudioFlinger::EffectChain::setEffectSuspended_l(
7435        const effect_uuid_t *type, bool suspend)
7436{
7437    sp<SuspendedEffectDesc> desc;
7438    // use effect type UUID timelow as key as there is no real risk of identical
7439    // timeLow fields among effect type UUIDs.
7440    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7441    if (suspend) {
7442        if (index >= 0) {
7443            desc = mSuspendedEffects.valueAt(index);
7444        } else {
7445            desc = new SuspendedEffectDesc();
7446            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7447            mSuspendedEffects.add(type->timeLow, desc);
7448            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7449        }
7450        if (desc->mRefCount++ == 0) {
7451            sp<EffectModule> effect = getEffectIfEnabled(type);
7452            if (effect != 0) {
7453                desc->mEffect = effect;
7454                effect->setSuspended(true);
7455                effect->setEnabled(false);
7456            }
7457        }
7458    } else {
7459        if (index < 0) {
7460            return;
7461        }
7462        desc = mSuspendedEffects.valueAt(index);
7463        if (desc->mRefCount <= 0) {
7464            LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7465            desc->mRefCount = 1;
7466        }
7467        if (--desc->mRefCount == 0) {
7468            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7469            if (desc->mEffect != 0) {
7470                sp<EffectModule> effect = desc->mEffect.promote();
7471                if (effect != 0) {
7472                    effect->setSuspended(false);
7473                    sp<EffectHandle> handle = effect->controlHandle();
7474                    if (handle != 0) {
7475                        effect->setEnabled(handle->enabled());
7476                    }
7477                }
7478                desc->mEffect.clear();
7479            }
7480            mSuspendedEffects.removeItemsAt(index);
7481        }
7482    }
7483}
7484
7485// must be called with ThreadBase::mLock held
7486void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7487{
7488    sp<SuspendedEffectDesc> desc;
7489
7490    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7491    if (suspend) {
7492        if (index >= 0) {
7493            desc = mSuspendedEffects.valueAt(index);
7494        } else {
7495            desc = new SuspendedEffectDesc();
7496            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7497            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7498        }
7499        if (desc->mRefCount++ == 0) {
7500            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7501            for (size_t i = 0; i < effects.size(); i++) {
7502                setEffectSuspended_l(&effects[i]->desc().type, true);
7503            }
7504        }
7505    } else {
7506        if (index < 0) {
7507            return;
7508        }
7509        desc = mSuspendedEffects.valueAt(index);
7510        if (desc->mRefCount <= 0) {
7511            LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7512            desc->mRefCount = 1;
7513        }
7514        if (--desc->mRefCount == 0) {
7515            Vector<const effect_uuid_t *> types;
7516            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7517                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7518                    continue;
7519                }
7520                types.add(&mSuspendedEffects.valueAt(i)->mType);
7521            }
7522            for (size_t i = 0; i < types.size(); i++) {
7523                setEffectSuspended_l(types[i], false);
7524            }
7525            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7526            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7527        }
7528    }
7529}
7530
7531
7532// The volume effect is used for automated tests only
7533#ifndef OPENSL_ES_H_
7534static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7535                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7536const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7537#endif //OPENSL_ES_H_
7538
7539bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7540{
7541    // auxiliary effects and visualizer are never suspended on output mix
7542    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7543        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7544         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7545         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7546        return false;
7547    }
7548    return true;
7549}
7550
7551Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7552{
7553    Vector< sp<EffectModule> > effects;
7554    for (size_t i = 0; i < mEffects.size(); i++) {
7555        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7556            continue;
7557        }
7558        effects.add(mEffects[i]);
7559    }
7560    return effects;
7561}
7562
7563sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7564                                                            const effect_uuid_t *type)
7565{
7566    sp<EffectModule> effect;
7567    effect = getEffectFromType_l(type);
7568    if (effect != 0 && !effect->isEnabled()) {
7569        effect.clear();
7570    }
7571    return effect;
7572}
7573
7574void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7575                                                            bool enabled)
7576{
7577    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7578    if (enabled) {
7579        if (index < 0) {
7580            // if the effect is not suspend check if all effects are suspended
7581            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7582            if (index < 0) {
7583                return;
7584            }
7585            if (!isEffectEligibleForSuspend(effect->desc())) {
7586                return;
7587            }
7588            setEffectSuspended_l(&effect->desc().type, enabled);
7589            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7590            if (index < 0) {
7591                LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7592                return;
7593            }
7594        }
7595        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7596             effect->desc().type.timeLow);
7597        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7598        // if effect is requested to suspended but was not yet enabled, supend it now.
7599        if (desc->mEffect == 0) {
7600            desc->mEffect = effect;
7601            effect->setEnabled(false);
7602            effect->setSuspended(true);
7603        }
7604    } else {
7605        if (index < 0) {
7606            return;
7607        }
7608        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7609             effect->desc().type.timeLow);
7610        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7611        desc->mEffect.clear();
7612        effect->setSuspended(false);
7613    }
7614}
7615
7616#undef LOG_TAG
7617#define LOG_TAG "AudioFlinger"
7618
7619// ----------------------------------------------------------------------------
7620
7621status_t AudioFlinger::onTransact(
7622        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7623{
7624    return BnAudioFlinger::onTransact(code, data, reply, flags);
7625}
7626
7627}; // namespace android
7628