AudioFlinger.cpp revision 613882293184e575a44bff681a3decaefe889e69 
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include <math.h>
23#include <signal.h>
24#include <sys/time.h>
25#include <sys/resource.h>
26
27#include <binder/IPCThreadState.h>
28#include <binder/IServiceManager.h>
29#include <utils/Log.h>
30#include <binder/Parcel.h>
31#include <binder/IPCThreadState.h>
32#include <utils/String16.h>
33#include <utils/threads.h>
34#include <utils/Atomic.h>
35
36#include <cutils/bitops.h>
37#include <cutils/properties.h>
38#include <cutils/compiler.h>
39
40#include <media/AudioTrack.h>
41#include <media/AudioRecord.h>
42#include <media/IMediaPlayerService.h>
43#include <media/IMediaDeathNotifier.h>
44
45#include <private/media/AudioTrackShared.h>
46#include <private/media/AudioEffectShared.h>
47
48#include <system/audio.h>
49#include <hardware/audio.h>
50
51#include "AudioMixer.h"
52#include "AudioFlinger.h"
53
54#include <media/EffectsFactoryApi.h>
55#include <audio_effects/effect_visualizer.h>
56#include <audio_effects/effect_ns.h>
57#include <audio_effects/effect_aec.h>
58
59#include <audio_utils/primitives.h>
60
61#include <cpustats/ThreadCpuUsage.h>
62#include <powermanager/PowerManager.h>
63// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
64
65// ----------------------------------------------------------------------------
66
67
68namespace android {
69
70static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
71static const char kHardwareLockedString[] = "Hardware lock is taken\n";
72
73//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
74static const float MAX_GAIN = 4096.0f;
75static const float MAX_GAIN_INT = 0x1000;
76
77// retry counts for buffer fill timeout
78// 50 * ~20msecs = 1 second
79static const int8_t kMaxTrackRetries = 50;
80static const int8_t kMaxTrackStartupRetries = 50;
81// allow less retry attempts on direct output thread.
82// direct outputs can be a scarce resource in audio hardware and should
83// be released as quickly as possible.
84static const int8_t kMaxTrackRetriesDirect = 2;
85
86static const int kDumpLockRetries = 50;
87static const int kDumpLockSleepUs = 20000;
88
89// don't warn about blocked writes or record buffer overflows more often than this
90static const nsecs_t kWarningThrottleNs = seconds(5);
91
92// RecordThread loop sleep time upon application overrun or audio HAL read error
93static const int kRecordThreadSleepUs = 5000;
94
95// maximum time to wait for setParameters to complete
96static const nsecs_t kSetParametersTimeoutNs = seconds(2);
97
98// minimum sleep time for the mixer thread loop when tracks are active but in underrun
99static const uint32_t kMinThreadSleepTimeUs = 5000;
100// maximum divider applied to the active sleep time in the mixer thread loop
101static const uint32_t kMaxThreadSleepTimeShift = 2;
102
103
104// ----------------------------------------------------------------------------
105
106static bool recordingAllowed() {
107    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
108    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
109    if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
110    return ok;
111}
112
113static bool settingsAllowed() {
114    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
115    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
116    if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
117    return ok;
118}
119
120// To collect the amplifier usage
121static void addBatteryData(uint32_t params) {
122    sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
123    if (service == NULL) {
124        // it already logged
125        return;
126    }
127
128    service->addBatteryData(params);
129}
130
131static int load_audio_interface(const char *if_name, const hw_module_t **mod,
132                                audio_hw_device_t **dev)
133{
134    int rc;
135
136    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
137    if (rc)
138        goto out;
139
140    rc = audio_hw_device_open(*mod, dev);
141    ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
142            AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
143    if (rc)
144        goto out;
145
146    return 0;
147
148out:
149    *mod = NULL;
150    *dev = NULL;
151    return rc;
152}
153
154static const char * const audio_interfaces[] = {
155    "primary",
156    "a2dp",
157    "usb",
158};
159#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
160
161// ----------------------------------------------------------------------------
162
163AudioFlinger::AudioFlinger()
164    : BnAudioFlinger(),
165        mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
166        mBtNrecIsOff(false)
167{
168}
169
170void AudioFlinger::onFirstRef()
171{
172    int rc = 0;
173
174    Mutex::Autolock _l(mLock);
175
176    /* TODO: move all this work into an Init() function */
177    mHardwareStatus = AUDIO_HW_IDLE;
178
179    for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
180        const hw_module_t *mod;
181        audio_hw_device_t *dev;
182
183        rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
184        if (rc)
185            continue;
186
187        ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
188             mod->name, mod->id);
189        mAudioHwDevs.push(dev);
190
191        if (!mPrimaryHardwareDev) {
192            mPrimaryHardwareDev = dev;
193            ALOGI("Using '%s' (%s.%s) as the primary audio interface",
194                 mod->name, mod->id, audio_interfaces[i]);
195        }
196    }
197
198    mHardwareStatus = AUDIO_HW_INIT;
199
200    if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
201        ALOGE("Primary audio interface not found");
202        return;
203    }
204
205    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
206        audio_hw_device_t *dev = mAudioHwDevs[i];
207
208        mHardwareStatus = AUDIO_HW_INIT;
209        rc = dev->init_check(dev);
210        if (rc == 0) {
211            AutoMutex lock(mHardwareLock);
212
213            mMode = AUDIO_MODE_NORMAL;
214            mHardwareStatus = AUDIO_HW_SET_MODE;
215            dev->set_mode(dev, mMode);
216            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
217            dev->set_master_volume(dev, 1.0f);
218            mHardwareStatus = AUDIO_HW_IDLE;
219        }
220    }
221}
222
223status_t AudioFlinger::initCheck() const
224{
225    Mutex::Autolock _l(mLock);
226    if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
227        return NO_INIT;
228    return NO_ERROR;
229}
230
231AudioFlinger::~AudioFlinger()
232{
233    int num_devs = mAudioHwDevs.size();
234
235    while (!mRecordThreads.isEmpty()) {
236        // closeInput() will remove first entry from mRecordThreads
237        closeInput(mRecordThreads.keyAt(0));
238    }
239    while (!mPlaybackThreads.isEmpty()) {
240        // closeOutput() will remove first entry from mPlaybackThreads
241        closeOutput(mPlaybackThreads.keyAt(0));
242    }
243
244    for (int i = 0; i < num_devs; i++) {
245        audio_hw_device_t *dev = mAudioHwDevs[i];
246        audio_hw_device_close(dev);
247    }
248    mAudioHwDevs.clear();
249}
250
251audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
252{
253    /* first matching HW device is returned */
254    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
255        audio_hw_device_t *dev = mAudioHwDevs[i];
256        if ((dev->get_supported_devices(dev) & devices) == devices)
257            return dev;
258    }
259    return NULL;
260}
261
262status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
263{
264    const size_t SIZE = 256;
265    char buffer[SIZE];
266    String8 result;
267
268    result.append("Clients:\n");
269    for (size_t i = 0; i < mClients.size(); ++i) {
270        wp<Client> wClient = mClients.valueAt(i);
271        if (wClient != 0) {
272            sp<Client> client = wClient.promote();
273            if (client != 0) {
274                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
275                result.append(buffer);
276            }
277        }
278    }
279
280    result.append("Global session refs:\n");
281    result.append(" session pid cnt\n");
282    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
283        AudioSessionRef *r = mAudioSessionRefs[i];
284        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
285        result.append(buffer);
286    }
287    write(fd, result.string(), result.size());
288    return NO_ERROR;
289}
290
291
292status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
293{
294    const size_t SIZE = 256;
295    char buffer[SIZE];
296    String8 result;
297    hardware_call_state hardwareStatus = mHardwareStatus;
298
299    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
300    result.append(buffer);
301    write(fd, result.string(), result.size());
302    return NO_ERROR;
303}
304
305status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
306{
307    const size_t SIZE = 256;
308    char buffer[SIZE];
309    String8 result;
310    snprintf(buffer, SIZE, "Permission Denial: "
311            "can't dump AudioFlinger from pid=%d, uid=%d\n",
312            IPCThreadState::self()->getCallingPid(),
313            IPCThreadState::self()->getCallingUid());
314    result.append(buffer);
315    write(fd, result.string(), result.size());
316    return NO_ERROR;
317}
318
319static bool tryLock(Mutex& mutex)
320{
321    bool locked = false;
322    for (int i = 0; i < kDumpLockRetries; ++i) {
323        if (mutex.tryLock() == NO_ERROR) {
324            locked = true;
325            break;
326        }
327        usleep(kDumpLockSleepUs);
328    }
329    return locked;
330}
331
332status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
333{
334    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
335        dumpPermissionDenial(fd, args);
336    } else {
337        // get state of hardware lock
338        bool hardwareLocked = tryLock(mHardwareLock);
339        if (!hardwareLocked) {
340            String8 result(kHardwareLockedString);
341            write(fd, result.string(), result.size());
342        } else {
343            mHardwareLock.unlock();
344        }
345
346        bool locked = tryLock(mLock);
347
348        // failed to lock - AudioFlinger is probably deadlocked
349        if (!locked) {
350            String8 result(kDeadlockedString);
351            write(fd, result.string(), result.size());
352        }
353
354        dumpClients(fd, args);
355        dumpInternals(fd, args);
356
357        // dump playback threads
358        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
359            mPlaybackThreads.valueAt(i)->dump(fd, args);
360        }
361
362        // dump record threads
363        for (size_t i = 0; i < mRecordThreads.size(); i++) {
364            mRecordThreads.valueAt(i)->dump(fd, args);
365        }
366
367        // dump all hardware devs
368        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
369            audio_hw_device_t *dev = mAudioHwDevs[i];
370            dev->dump(dev, fd);
371        }
372        if (locked) mLock.unlock();
373    }
374    return NO_ERROR;
375}
376
377
378// IAudioFlinger interface
379
380
381sp<IAudioTrack> AudioFlinger::createTrack(
382        pid_t pid,
383        audio_stream_type_t streamType,
384        uint32_t sampleRate,
385        uint32_t format,
386        uint32_t channelMask,
387        int frameCount,
388        uint32_t flags,
389        const sp<IMemory>& sharedBuffer,
390        int output,
391        int *sessionId,
392        status_t *status)
393{
394    sp<PlaybackThread::Track> track;
395    sp<TrackHandle> trackHandle;
396    sp<Client> client;
397    wp<Client> wclient;
398    status_t lStatus;
399    int lSessionId;
400
401    if (streamType >= AUDIO_STREAM_CNT) {
402        ALOGE("createTrack() invalid stream type %d", streamType);
403        lStatus = BAD_VALUE;
404        goto Exit;
405    }
406
407    {
408        Mutex::Autolock _l(mLock);
409        PlaybackThread *thread = checkPlaybackThread_l(output);
410        PlaybackThread *effectThread = NULL;
411        if (thread == NULL) {
412            ALOGE("unknown output thread");
413            lStatus = BAD_VALUE;
414            goto Exit;
415        }
416
417        wclient = mClients.valueFor(pid);
418
419        if (wclient != NULL) {
420            client = wclient.promote();
421        } else {
422            client = new Client(this, pid);
423            mClients.add(pid, client);
424        }
425
426        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
427        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
428            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
429                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
430                if (mPlaybackThreads.keyAt(i) != output) {
431                    // prevent same audio session on different output threads
432                    uint32_t sessions = t->hasAudioSession(*sessionId);
433                    if (sessions & PlaybackThread::TRACK_SESSION) {
434                        ALOGE("createTrack() session ID %d already in use", *sessionId);
435                        lStatus = BAD_VALUE;
436                        goto Exit;
437                    }
438                    // check if an effect with same session ID is waiting for a track to be created
439                    if (sessions & PlaybackThread::EFFECT_SESSION) {
440                        effectThread = t.get();
441                    }
442                }
443            }
444            lSessionId = *sessionId;
445        } else {
446            // if no audio session id is provided, create one here
447            lSessionId = nextUniqueId();
448            if (sessionId != NULL) {
449                *sessionId = lSessionId;
450            }
451        }
452        ALOGV("createTrack() lSessionId: %d", lSessionId);
453
454        track = thread->createTrack_l(client, streamType, sampleRate, format,
455                channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
456
457        // move effect chain to this output thread if an effect on same session was waiting
458        // for a track to be created
459        if (lStatus == NO_ERROR && effectThread != NULL) {
460            Mutex::Autolock _dl(thread->mLock);
461            Mutex::Autolock _sl(effectThread->mLock);
462            moveEffectChain_l(lSessionId, effectThread, thread, true);
463        }
464    }
465    if (lStatus == NO_ERROR) {
466        trackHandle = new TrackHandle(track);
467    } else {
468        // remove local strong reference to Client before deleting the Track so that the Client
469        // destructor is called by the TrackBase destructor with mLock held
470        client.clear();
471        track.clear();
472    }
473
474Exit:
475    if(status) {
476        *status = lStatus;
477    }
478    return trackHandle;
479}
480
481uint32_t AudioFlinger::sampleRate(int output) const
482{
483    Mutex::Autolock _l(mLock);
484    PlaybackThread *thread = checkPlaybackThread_l(output);
485    if (thread == NULL) {
486        ALOGW("sampleRate() unknown thread %d", output);
487        return 0;
488    }
489    return thread->sampleRate();
490}
491
492int AudioFlinger::channelCount(int output) const
493{
494    Mutex::Autolock _l(mLock);
495    PlaybackThread *thread = checkPlaybackThread_l(output);
496    if (thread == NULL) {
497        ALOGW("channelCount() unknown thread %d", output);
498        return 0;
499    }
500    return thread->channelCount();
501}
502
503uint32_t AudioFlinger::format(int output) const
504{
505    Mutex::Autolock _l(mLock);
506    PlaybackThread *thread = checkPlaybackThread_l(output);
507    if (thread == NULL) {
508        ALOGW("format() unknown thread %d", output);
509        return 0;
510    }
511    return thread->format();
512}
513
514size_t AudioFlinger::frameCount(int output) const
515{
516    Mutex::Autolock _l(mLock);
517    PlaybackThread *thread = checkPlaybackThread_l(output);
518    if (thread == NULL) {
519        ALOGW("frameCount() unknown thread %d", output);
520        return 0;
521    }
522    return thread->frameCount();
523}
524
525uint32_t AudioFlinger::latency(int output) const
526{
527    Mutex::Autolock _l(mLock);
528    PlaybackThread *thread = checkPlaybackThread_l(output);
529    if (thread == NULL) {
530        ALOGW("latency() unknown thread %d", output);
531        return 0;
532    }
533    return thread->latency();
534}
535
536status_t AudioFlinger::setMasterVolume(float value)
537{
538    status_t ret = initCheck();
539    if (ret != NO_ERROR) {
540        return ret;
541    }
542
543    // check calling permissions
544    if (!settingsAllowed()) {
545        return PERMISSION_DENIED;
546    }
547
548    // when hw supports master volume, don't scale in sw mixer
549    { // scope for the lock
550        AutoMutex lock(mHardwareLock);
551        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
552        if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
553            value = 1.0f;
554        }
555        mHardwareStatus = AUDIO_HW_IDLE;
556    }
557
558    Mutex::Autolock _l(mLock);
559    mMasterVolume = value;
560    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
561       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
562
563    return NO_ERROR;
564}
565
566status_t AudioFlinger::setMode(int mode)
567{
568    status_t ret = initCheck();
569    if (ret != NO_ERROR) {
570        return ret;
571    }
572
573    // check calling permissions
574    if (!settingsAllowed()) {
575        return PERMISSION_DENIED;
576    }
577    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
578        ALOGW("Illegal value: setMode(%d)", mode);
579        return BAD_VALUE;
580    }
581
582    { // scope for the lock
583        AutoMutex lock(mHardwareLock);
584        mHardwareStatus = AUDIO_HW_SET_MODE;
585        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
586        mHardwareStatus = AUDIO_HW_IDLE;
587    }
588
589    if (NO_ERROR == ret) {
590        Mutex::Autolock _l(mLock);
591        mMode = mode;
592        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
593           mPlaybackThreads.valueAt(i)->setMode(mode);
594    }
595
596    return ret;
597}
598
599status_t AudioFlinger::setMicMute(bool state)
600{
601    status_t ret = initCheck();
602    if (ret != NO_ERROR) {
603        return ret;
604    }
605
606    // check calling permissions
607    if (!settingsAllowed()) {
608        return PERMISSION_DENIED;
609    }
610
611    AutoMutex lock(mHardwareLock);
612    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
613    ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
614    mHardwareStatus = AUDIO_HW_IDLE;
615    return ret;
616}
617
618bool AudioFlinger::getMicMute() const
619{
620    status_t ret = initCheck();
621    if (ret != NO_ERROR) {
622        return false;
623    }
624
625    bool state = AUDIO_MODE_INVALID;
626    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
627    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
628    mHardwareStatus = AUDIO_HW_IDLE;
629    return state;
630}
631
632status_t AudioFlinger::setMasterMute(bool muted)
633{
634    // check calling permissions
635    if (!settingsAllowed()) {
636        return PERMISSION_DENIED;
637    }
638
639    Mutex::Autolock _l(mLock);
640    mMasterMute = muted;
641    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
642       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
643
644    return NO_ERROR;
645}
646
647float AudioFlinger::masterVolume() const
648{
649    return mMasterVolume;
650}
651
652bool AudioFlinger::masterMute() const
653{
654    return mMasterMute;
655}
656
657status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output)
658{
659    // check calling permissions
660    if (!settingsAllowed()) {
661        return PERMISSION_DENIED;
662    }
663
664    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
665        ALOGE("setStreamVolume() invalid stream %d", stream);
666        return BAD_VALUE;
667    }
668
669    AutoMutex lock(mLock);
670    PlaybackThread *thread = NULL;
671    if (output) {
672        thread = checkPlaybackThread_l(output);
673        if (thread == NULL) {
674            return BAD_VALUE;
675        }
676    }
677
678    mStreamTypes[stream].volume = value;
679
680    if (thread == NULL) {
681        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
682           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
683        }
684    } else {
685        thread->setStreamVolume(stream, value);
686    }
687
688    return NO_ERROR;
689}
690
691status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
692{
693    // check calling permissions
694    if (!settingsAllowed()) {
695        return PERMISSION_DENIED;
696    }
697
698    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
699        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
700        ALOGE("setStreamMute() invalid stream %d", stream);
701        return BAD_VALUE;
702    }
703
704    AutoMutex lock(mLock);
705    mStreamTypes[stream].mute = muted;
706    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
707       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
708
709    return NO_ERROR;
710}
711
712float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const
713{
714    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
715        return 0.0f;
716    }
717
718    AutoMutex lock(mLock);
719    float volume;
720    if (output) {
721        PlaybackThread *thread = checkPlaybackThread_l(output);
722        if (thread == NULL) {
723            return 0.0f;
724        }
725        volume = thread->streamVolume(stream);
726    } else {
727        volume = mStreamTypes[stream].volume;
728    }
729
730    return volume;
731}
732
733bool AudioFlinger::streamMute(audio_stream_type_t stream) const
734{
735    if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
736        return true;
737    }
738
739    return mStreamTypes[stream].mute;
740}
741
742status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
743{
744    status_t result;
745
746    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
747            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
748    // check calling permissions
749    if (!settingsAllowed()) {
750        return PERMISSION_DENIED;
751    }
752
753    // ioHandle == 0 means the parameters are global to the audio hardware interface
754    if (ioHandle == 0) {
755        AutoMutex lock(mHardwareLock);
756        mHardwareStatus = AUDIO_SET_PARAMETER;
757        status_t final_result = NO_ERROR;
758        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
759            audio_hw_device_t *dev = mAudioHwDevs[i];
760            result = dev->set_parameters(dev, keyValuePairs.string());
761            final_result = result ?: final_result;
762        }
763        mHardwareStatus = AUDIO_HW_IDLE;
764        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
765        AudioParameter param = AudioParameter(keyValuePairs);
766        String8 value;
767        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
768            Mutex::Autolock _l(mLock);
769            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
770            if (mBtNrecIsOff != btNrecIsOff) {
771                for (size_t i = 0; i < mRecordThreads.size(); i++) {
772                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
773                    RecordThread::RecordTrack *track = thread->track();
774                    if (track != NULL) {
775                        audio_devices_t device = (audio_devices_t)(
776                                thread->device() & AUDIO_DEVICE_IN_ALL);
777                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
778                        thread->setEffectSuspended(FX_IID_AEC,
779                                                   suspend,
780                                                   track->sessionId());
781                        thread->setEffectSuspended(FX_IID_NS,
782                                                   suspend,
783                                                   track->sessionId());
784                    }
785                }
786                mBtNrecIsOff = btNrecIsOff;
787            }
788        }
789        return final_result;
790    }
791
792    // hold a strong ref on thread in case closeOutput() or closeInput() is called
793    // and the thread is exited once the lock is released
794    sp<ThreadBase> thread;
795    {
796        Mutex::Autolock _l(mLock);
797        thread = checkPlaybackThread_l(ioHandle);
798        if (thread == NULL) {
799            thread = checkRecordThread_l(ioHandle);
800        } else if (thread.get() == primaryPlaybackThread_l()) {
801            // indicate output device change to all input threads for pre processing
802            AudioParameter param = AudioParameter(keyValuePairs);
803            int value;
804            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
805                for (size_t i = 0; i < mRecordThreads.size(); i++) {
806                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
807                }
808            }
809        }
810    }
811    if (thread != NULL) {
812        result = thread->setParameters(keyValuePairs);
813        return result;
814    }
815    return BAD_VALUE;
816}
817
818String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
819{
820//    ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
821//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
822
823    if (ioHandle == 0) {
824        String8 out_s8;
825
826        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
827            audio_hw_device_t *dev = mAudioHwDevs[i];
828            char *s = dev->get_parameters(dev, keys.string());
829            out_s8 += String8(s);
830            free(s);
831        }
832        return out_s8;
833    }
834
835    Mutex::Autolock _l(mLock);
836
837    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
838    if (playbackThread != NULL) {
839        return playbackThread->getParameters(keys);
840    }
841    RecordThread *recordThread = checkRecordThread_l(ioHandle);
842    if (recordThread != NULL) {
843        return recordThread->getParameters(keys);
844    }
845    return String8("");
846}
847
848size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
849{
850    status_t ret = initCheck();
851    if (ret != NO_ERROR) {
852        return 0;
853    }
854
855    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
856}
857
858unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
859{
860    if (ioHandle == 0) {
861        return 0;
862    }
863
864    Mutex::Autolock _l(mLock);
865
866    RecordThread *recordThread = checkRecordThread_l(ioHandle);
867    if (recordThread != NULL) {
868        return recordThread->getInputFramesLost();
869    }
870    return 0;
871}
872
873status_t AudioFlinger::setVoiceVolume(float value)
874{
875    status_t ret = initCheck();
876    if (ret != NO_ERROR) {
877        return ret;
878    }
879
880    // check calling permissions
881    if (!settingsAllowed()) {
882        return PERMISSION_DENIED;
883    }
884
885    AutoMutex lock(mHardwareLock);
886    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
887    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
888    mHardwareStatus = AUDIO_HW_IDLE;
889
890    return ret;
891}
892
893status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
894{
895    status_t status;
896
897    Mutex::Autolock _l(mLock);
898
899    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
900    if (playbackThread != NULL) {
901        return playbackThread->getRenderPosition(halFrames, dspFrames);
902    }
903
904    return BAD_VALUE;
905}
906
907void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
908{
909
910    Mutex::Autolock _l(mLock);
911
912    int pid = IPCThreadState::self()->getCallingPid();
913    if (mNotificationClients.indexOfKey(pid) < 0) {
914        sp<NotificationClient> notificationClient = new NotificationClient(this,
915                                                                            client,
916                                                                            pid);
917        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
918
919        mNotificationClients.add(pid, notificationClient);
920
921        sp<IBinder> binder = client->asBinder();
922        binder->linkToDeath(notificationClient);
923
924        // the config change is always sent from playback or record threads to avoid deadlock
925        // with AudioSystem::gLock
926        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
927            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
928        }
929
930        for (size_t i = 0; i < mRecordThreads.size(); i++) {
931            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
932        }
933    }
934}
935
936void AudioFlinger::removeNotificationClient(pid_t pid)
937{
938    Mutex::Autolock _l(mLock);
939
940    int index = mNotificationClients.indexOfKey(pid);
941    if (index >= 0) {
942        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
943        ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
944        mNotificationClients.removeItem(pid);
945    }
946
947    ALOGV("%d died, releasing its sessions", pid);
948    int num = mAudioSessionRefs.size();
949    bool removed = false;
950    for (int i = 0; i< num; i++) {
951        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
952        ALOGV(" pid %d @ %d", ref->pid, i);
953        if (ref->pid == pid) {
954            ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
955            mAudioSessionRefs.removeAt(i);
956            delete ref;
957            removed = true;
958            i--;
959            num--;
960        }
961    }
962    if (removed) {
963        purgeStaleEffects_l();
964    }
965}
966
967// audioConfigChanged_l() must be called with AudioFlinger::mLock held
968void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
969{
970    size_t size = mNotificationClients.size();
971    for (size_t i = 0; i < size; i++) {
972        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
973    }
974}
975
976// removeClient_l() must be called with AudioFlinger::mLock held
977void AudioFlinger::removeClient_l(pid_t pid)
978{
979    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
980    mClients.removeItem(pid);
981}
982
983
984// ----------------------------------------------------------------------------
985
986AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
987    :   Thread(false),
988        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
989        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
990        mDevice(device)
991{
992    mDeathRecipient = new PMDeathRecipient(this);
993}
994
995AudioFlinger::ThreadBase::~ThreadBase()
996{
997    mParamCond.broadcast();
998    // do not lock the mutex in destructor
999    releaseWakeLock_l();
1000    if (mPowerManager != 0) {
1001        sp<IBinder> binder = mPowerManager->asBinder();
1002        binder->unlinkToDeath(mDeathRecipient);
1003    }
1004}
1005
1006void AudioFlinger::ThreadBase::exit()
1007{
1008    // keep a strong ref on ourself so that we won't get
1009    // destroyed in the middle of requestExitAndWait()
1010    sp <ThreadBase> strongMe = this;
1011
1012    ALOGV("ThreadBase::exit");
1013    {
1014        AutoMutex lock(mLock);
1015        mExiting = true;
1016        requestExit();
1017        mWaitWorkCV.signal();
1018    }
1019    requestExitAndWait();
1020}
1021
1022uint32_t AudioFlinger::ThreadBase::sampleRate() const
1023{
1024    return mSampleRate;
1025}
1026
1027int AudioFlinger::ThreadBase::channelCount() const
1028{
1029    return (int)mChannelCount;
1030}
1031
1032uint32_t AudioFlinger::ThreadBase::format() const
1033{
1034    return mFormat;
1035}
1036
1037size_t AudioFlinger::ThreadBase::frameCount() const
1038{
1039    return mFrameCount;
1040}
1041
1042status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1043{
1044    status_t status;
1045
1046    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1047    Mutex::Autolock _l(mLock);
1048
1049    mNewParameters.add(keyValuePairs);
1050    mWaitWorkCV.signal();
1051    // wait condition with timeout in case the thread loop has exited
1052    // before the request could be processed
1053    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
1054        status = mParamStatus;
1055        mWaitWorkCV.signal();
1056    } else {
1057        status = TIMED_OUT;
1058    }
1059    return status;
1060}
1061
1062void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1063{
1064    Mutex::Autolock _l(mLock);
1065    sendConfigEvent_l(event, param);
1066}
1067
1068// sendConfigEvent_l() must be called with ThreadBase::mLock held
1069void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1070{
1071    ConfigEvent configEvent;
1072    configEvent.mEvent = event;
1073    configEvent.mParam = param;
1074    mConfigEvents.add(configEvent);
1075    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1076    mWaitWorkCV.signal();
1077}
1078
1079void AudioFlinger::ThreadBase::processConfigEvents()
1080{
1081    mLock.lock();
1082    while(!mConfigEvents.isEmpty()) {
1083        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1084        ConfigEvent configEvent = mConfigEvents[0];
1085        mConfigEvents.removeAt(0);
1086        // release mLock before locking AudioFlinger mLock: lock order is always
1087        // AudioFlinger then ThreadBase to avoid cross deadlock
1088        mLock.unlock();
1089        mAudioFlinger->mLock.lock();
1090        audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
1091        mAudioFlinger->mLock.unlock();
1092        mLock.lock();
1093    }
1094    mLock.unlock();
1095}
1096
1097status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1098{
1099    const size_t SIZE = 256;
1100    char buffer[SIZE];
1101    String8 result;
1102
1103    bool locked = tryLock(mLock);
1104    if (!locked) {
1105        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1106        write(fd, buffer, strlen(buffer));
1107    }
1108
1109    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1110    result.append(buffer);
1111    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1112    result.append(buffer);
1113    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1114    result.append(buffer);
1115    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1116    result.append(buffer);
1117    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1118    result.append(buffer);
1119    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1120    result.append(buffer);
1121    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
1122    result.append(buffer);
1123
1124    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1125    result.append(buffer);
1126    result.append(" Index Command");
1127    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1128        snprintf(buffer, SIZE, "\n %02d    ", i);
1129        result.append(buffer);
1130        result.append(mNewParameters[i]);
1131    }
1132
1133    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1134    result.append(buffer);
1135    snprintf(buffer, SIZE, " Index event param\n");
1136    result.append(buffer);
1137    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1138        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
1139        result.append(buffer);
1140    }
1141    result.append("\n");
1142
1143    write(fd, result.string(), result.size());
1144
1145    if (locked) {
1146        mLock.unlock();
1147    }
1148    return NO_ERROR;
1149}
1150
1151status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1152{
1153    const size_t SIZE = 256;
1154    char buffer[SIZE];
1155    String8 result;
1156
1157    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1158    write(fd, buffer, strlen(buffer));
1159
1160    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1161        sp<EffectChain> chain = mEffectChains[i];
1162        if (chain != 0) {
1163            chain->dump(fd, args);
1164        }
1165    }
1166    return NO_ERROR;
1167}
1168
1169void AudioFlinger::ThreadBase::acquireWakeLock()
1170{
1171    Mutex::Autolock _l(mLock);
1172    acquireWakeLock_l();
1173}
1174
1175void AudioFlinger::ThreadBase::acquireWakeLock_l()
1176{
1177    if (mPowerManager == 0) {
1178        // use checkService() to avoid blocking if power service is not up yet
1179        sp<IBinder> binder =
1180            defaultServiceManager()->checkService(String16("power"));
1181        if (binder == 0) {
1182            ALOGW("Thread %s cannot connect to the power manager service", mName);
1183        } else {
1184            mPowerManager = interface_cast<IPowerManager>(binder);
1185            binder->linkToDeath(mDeathRecipient);
1186        }
1187    }
1188    if (mPowerManager != 0) {
1189        sp<IBinder> binder = new BBinder();
1190        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
1191                                                         binder,
1192                                                         String16(mName));
1193        if (status == NO_ERROR) {
1194            mWakeLockToken = binder;
1195        }
1196        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
1197    }
1198}
1199
1200void AudioFlinger::ThreadBase::releaseWakeLock()
1201{
1202    Mutex::Autolock _l(mLock);
1203    releaseWakeLock_l();
1204}
1205
1206void AudioFlinger::ThreadBase::releaseWakeLock_l()
1207{
1208    if (mWakeLockToken != 0) {
1209        ALOGV("releaseWakeLock_l() %s", mName);
1210        if (mPowerManager != 0) {
1211            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
1212        }
1213        mWakeLockToken.clear();
1214    }
1215}
1216
1217void AudioFlinger::ThreadBase::clearPowerManager()
1218{
1219    Mutex::Autolock _l(mLock);
1220    releaseWakeLock_l();
1221    mPowerManager.clear();
1222}
1223
1224void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
1225{
1226    sp<ThreadBase> thread = mThread.promote();
1227    if (thread != 0) {
1228        thread->clearPowerManager();
1229    }
1230    ALOGW("power manager service died !!!");
1231}
1232
1233void AudioFlinger::ThreadBase::setEffectSuspended(
1234        const effect_uuid_t *type, bool suspend, int sessionId)
1235{
1236    Mutex::Autolock _l(mLock);
1237    setEffectSuspended_l(type, suspend, sessionId);
1238}
1239
1240void AudioFlinger::ThreadBase::setEffectSuspended_l(
1241        const effect_uuid_t *type, bool suspend, int sessionId)
1242{
1243    sp<EffectChain> chain;
1244    chain = getEffectChain_l(sessionId);
1245    if (chain != 0) {
1246        if (type != NULL) {
1247            chain->setEffectSuspended_l(type, suspend);
1248        } else {
1249            chain->setEffectSuspendedAll_l(suspend);
1250        }
1251    }
1252
1253    updateSuspendedSessions_l(type, suspend, sessionId);
1254}
1255
1256void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1257{
1258    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1259    if (index < 0) {
1260        return;
1261    }
1262
1263    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1264            mSuspendedSessions.editValueAt(index);
1265
1266    for (size_t i = 0; i < sessionEffects.size(); i++) {
1267        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1268        for (int j = 0; j < desc->mRefCount; j++) {
1269            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1270                chain->setEffectSuspendedAll_l(true);
1271            } else {
1272                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1273                     desc->mType.timeLow);
1274                chain->setEffectSuspended_l(&desc->mType, true);
1275            }
1276        }
1277    }
1278}
1279
1280void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1281                                                         bool suspend,
1282                                                         int sessionId)
1283{
1284    int index = mSuspendedSessions.indexOfKey(sessionId);
1285
1286    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1287
1288    if (suspend) {
1289        if (index >= 0) {
1290            sessionEffects = mSuspendedSessions.editValueAt(index);
1291        } else {
1292            mSuspendedSessions.add(sessionId, sessionEffects);
1293        }
1294    } else {
1295        if (index < 0) {
1296            return;
1297        }
1298        sessionEffects = mSuspendedSessions.editValueAt(index);
1299    }
1300
1301
1302    int key = EffectChain::kKeyForSuspendAll;
1303    if (type != NULL) {
1304        key = type->timeLow;
1305    }
1306    index = sessionEffects.indexOfKey(key);
1307
1308    sp <SuspendedSessionDesc> desc;
1309    if (suspend) {
1310        if (index >= 0) {
1311            desc = sessionEffects.valueAt(index);
1312        } else {
1313            desc = new SuspendedSessionDesc();
1314            if (type != NULL) {
1315                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1316            }
1317            sessionEffects.add(key, desc);
1318            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1319        }
1320        desc->mRefCount++;
1321    } else {
1322        if (index < 0) {
1323            return;
1324        }
1325        desc = sessionEffects.valueAt(index);
1326        if (--desc->mRefCount == 0) {
1327            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1328            sessionEffects.removeItemsAt(index);
1329            if (sessionEffects.isEmpty()) {
1330                ALOGV("updateSuspendedSessions_l() restore removing session %d",
1331                                 sessionId);
1332                mSuspendedSessions.removeItem(sessionId);
1333            }
1334        }
1335    }
1336    if (!sessionEffects.isEmpty()) {
1337        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1338    }
1339}
1340
1341void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1342                                                            bool enabled,
1343                                                            int sessionId)
1344{
1345    Mutex::Autolock _l(mLock);
1346    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1347}
1348
1349void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1350                                                            bool enabled,
1351                                                            int sessionId)
1352{
1353    if (mType != RECORD) {
1354        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1355        // another session. This gives the priority to well behaved effect control panels
1356        // and applications not using global effects.
1357        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1358            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1359        }
1360    }
1361
1362    sp<EffectChain> chain = getEffectChain_l(sessionId);
1363    if (chain != 0) {
1364        chain->checkSuspendOnEffectEnabled(effect, enabled);
1365    }
1366}
1367
1368// ----------------------------------------------------------------------------
1369
1370AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1371                                             AudioStreamOut* output,
1372                                             int id,
1373                                             uint32_t device)
1374    :   ThreadBase(audioFlinger, id, device),
1375        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output),
1376        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1377{
1378    snprintf(mName, kNameLength, "AudioOut_%d", id);
1379
1380    readOutputParameters();
1381
1382    mMasterVolume = mAudioFlinger->masterVolume();
1383    mMasterMute = mAudioFlinger->masterMute();
1384
1385    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
1386    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
1387            stream = (audio_stream_type_t) (stream + 1)) {
1388        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1389        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1390        mStreamTypes[stream].valid = true;
1391    }
1392}
1393
1394AudioFlinger::PlaybackThread::~PlaybackThread()
1395{
1396    delete [] mMixBuffer;
1397}
1398
1399status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1400{
1401    dumpInternals(fd, args);
1402    dumpTracks(fd, args);
1403    dumpEffectChains(fd, args);
1404    return NO_ERROR;
1405}
1406
1407status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1408{
1409    const size_t SIZE = 256;
1410    char buffer[SIZE];
1411    String8 result;
1412
1413    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1414    result.append(buffer);
1415    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1416    for (size_t i = 0; i < mTracks.size(); ++i) {
1417        sp<Track> track = mTracks[i];
1418        if (track != 0) {
1419            track->dump(buffer, SIZE);
1420            result.append(buffer);
1421        }
1422    }
1423
1424    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1425    result.append(buffer);
1426    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1427    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1428        wp<Track> wTrack = mActiveTracks[i];
1429        if (wTrack != 0) {
1430            sp<Track> track = wTrack.promote();
1431            if (track != 0) {
1432                track->dump(buffer, SIZE);
1433                result.append(buffer);
1434            }
1435        }
1436    }
1437    write(fd, result.string(), result.size());
1438    return NO_ERROR;
1439}
1440
1441status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1442{
1443    const size_t SIZE = 256;
1444    char buffer[SIZE];
1445    String8 result;
1446
1447    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1448    result.append(buffer);
1449    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1450    result.append(buffer);
1451    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1452    result.append(buffer);
1453    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1454    result.append(buffer);
1455    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1456    result.append(buffer);
1457    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1458    result.append(buffer);
1459    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1460    result.append(buffer);
1461    write(fd, result.string(), result.size());
1462
1463    dumpBase(fd, args);
1464
1465    return NO_ERROR;
1466}
1467
1468// Thread virtuals
1469status_t AudioFlinger::PlaybackThread::readyToRun()
1470{
1471    status_t status = initCheck();
1472    if (status == NO_ERROR) {
1473        ALOGI("AudioFlinger's thread %p ready to run", this);
1474    } else {
1475        ALOGE("No working audio driver found.");
1476    }
1477    return status;
1478}
1479
1480void AudioFlinger::PlaybackThread::onFirstRef()
1481{
1482    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1483}
1484
1485// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1486sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1487        const sp<AudioFlinger::Client>& client,
1488        audio_stream_type_t streamType,
1489        uint32_t sampleRate,
1490        uint32_t format,
1491        uint32_t channelMask,
1492        int frameCount,
1493        const sp<IMemory>& sharedBuffer,
1494        int sessionId,
1495        status_t *status)
1496{
1497    sp<Track> track;
1498    status_t lStatus;
1499
1500    if (mType == DIRECT) {
1501        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1502            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1503                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1504                        "for output %p with format %d",
1505                        sampleRate, format, channelMask, mOutput, mFormat);
1506                lStatus = BAD_VALUE;
1507                goto Exit;
1508            }
1509        }
1510    } else {
1511        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1512        if (sampleRate > mSampleRate*2) {
1513            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1514            lStatus = BAD_VALUE;
1515            goto Exit;
1516        }
1517    }
1518
1519    lStatus = initCheck();
1520    if (lStatus != NO_ERROR) {
1521        ALOGE("Audio driver not initialized.");
1522        goto Exit;
1523    }
1524
1525    { // scope for mLock
1526        Mutex::Autolock _l(mLock);
1527
1528        // all tracks in same audio session must share the same routing strategy otherwise
1529        // conflicts will happen when tracks are moved from one output to another by audio policy
1530        // manager
1531        uint32_t strategy =
1532                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1533        for (size_t i = 0; i < mTracks.size(); ++i) {
1534            sp<Track> t = mTracks[i];
1535            if (t != 0) {
1536                uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
1537                if (sessionId == t->sessionId() && strategy != actual) {
1538                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1539                            strategy, actual);
1540                    lStatus = BAD_VALUE;
1541                    goto Exit;
1542                }
1543            }
1544        }
1545
1546        track = new Track(this, client, streamType, sampleRate, format,
1547                channelMask, frameCount, sharedBuffer, sessionId);
1548        if (track->getCblk() == NULL || track->name() < 0) {
1549            lStatus = NO_MEMORY;
1550            goto Exit;
1551        }
1552        mTracks.add(track);
1553
1554        sp<EffectChain> chain = getEffectChain_l(sessionId);
1555        if (chain != 0) {
1556            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1557            track->setMainBuffer(chain->inBuffer());
1558            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1559            chain->incTrackCnt();
1560        }
1561
1562        // invalidate track immediately if the stream type was moved to another thread since
1563        // createTrack() was called by the client process.
1564        if (!mStreamTypes[streamType].valid) {
1565            ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1566                 this, streamType);
1567            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1568        }
1569    }
1570    lStatus = NO_ERROR;
1571
1572Exit:
1573    if(status) {
1574        *status = lStatus;
1575    }
1576    return track;
1577}
1578
1579uint32_t AudioFlinger::PlaybackThread::latency() const
1580{
1581    Mutex::Autolock _l(mLock);
1582    if (initCheck() == NO_ERROR) {
1583        return mOutput->stream->get_latency(mOutput->stream);
1584    } else {
1585        return 0;
1586    }
1587}
1588
1589status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1590{
1591    mMasterVolume = value;
1592    return NO_ERROR;
1593}
1594
1595status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1596{
1597    mMasterMute = muted;
1598    return NO_ERROR;
1599}
1600
1601float AudioFlinger::PlaybackThread::masterVolume() const
1602{
1603    return mMasterVolume;
1604}
1605
1606bool AudioFlinger::PlaybackThread::masterMute() const
1607{
1608    return mMasterMute;
1609}
1610
1611status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
1612{
1613    mStreamTypes[stream].volume = value;
1614    return NO_ERROR;
1615}
1616
1617status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
1618{
1619    mStreamTypes[stream].mute = muted;
1620    return NO_ERROR;
1621}
1622
1623float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
1624{
1625    return mStreamTypes[stream].volume;
1626}
1627
1628bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const
1629{
1630    return mStreamTypes[stream].mute;
1631}
1632
1633// addTrack_l() must be called with ThreadBase::mLock held
1634status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1635{
1636    status_t status = ALREADY_EXISTS;
1637
1638    // set retry count for buffer fill
1639    track->mRetryCount = kMaxTrackStartupRetries;
1640    if (mActiveTracks.indexOf(track) < 0) {
1641        // the track is newly added, make sure it fills up all its
1642        // buffers before playing. This is to ensure the client will
1643        // effectively get the latency it requested.
1644        track->mFillingUpStatus = Track::FS_FILLING;
1645        track->mResetDone = false;
1646        mActiveTracks.add(track);
1647        if (track->mainBuffer() != mMixBuffer) {
1648            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1649            if (chain != 0) {
1650                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1651                chain->incActiveTrackCnt();
1652            }
1653        }
1654
1655        status = NO_ERROR;
1656    }
1657
1658    ALOGV("mWaitWorkCV.broadcast");
1659    mWaitWorkCV.broadcast();
1660
1661    return status;
1662}
1663
1664// destroyTrack_l() must be called with ThreadBase::mLock held
1665void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1666{
1667    track->mState = TrackBase::TERMINATED;
1668    if (mActiveTracks.indexOf(track) < 0) {
1669        removeTrack_l(track);
1670    }
1671}
1672
1673void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1674{
1675    mTracks.remove(track);
1676    deleteTrackName_l(track->name());
1677    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1678    if (chain != 0) {
1679        chain->decTrackCnt();
1680    }
1681}
1682
1683String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1684{
1685    String8 out_s8 = String8("");
1686    char *s;
1687
1688    Mutex::Autolock _l(mLock);
1689    if (initCheck() != NO_ERROR) {
1690        return out_s8;
1691    }
1692
1693    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1694    out_s8 = String8(s);
1695    free(s);
1696    return out_s8;
1697}
1698
1699// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1700void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1701    AudioSystem::OutputDescriptor desc;
1702    void *param2 = 0;
1703
1704    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1705
1706    switch (event) {
1707    case AudioSystem::OUTPUT_OPENED:
1708    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1709        desc.channels = mChannelMask;
1710        desc.samplingRate = mSampleRate;
1711        desc.format = mFormat;
1712        desc.frameCount = mFrameCount;
1713        desc.latency = latency();
1714        param2 = &desc;
1715        break;
1716
1717    case AudioSystem::STREAM_CONFIG_CHANGED:
1718        param2 = &param;
1719    case AudioSystem::OUTPUT_CLOSED:
1720    default:
1721        break;
1722    }
1723    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1724}
1725
1726void AudioFlinger::PlaybackThread::readOutputParameters()
1727{
1728    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1729    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1730    mChannelCount = (uint16_t)popcount(mChannelMask);
1731    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1732    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
1733    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1734
1735    // FIXME - Current mixer implementation only supports stereo output: Always
1736    // Allocate a stereo buffer even if HW output is mono.
1737    if (mMixBuffer != NULL) delete[] mMixBuffer;
1738    mMixBuffer = new int16_t[mFrameCount * 2];
1739    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1740
1741    // force reconfiguration of effect chains and engines to take new buffer size and audio
1742    // parameters into account
1743    // Note that mLock is not held when readOutputParameters() is called from the constructor
1744    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1745    // matter.
1746    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1747    Vector< sp<EffectChain> > effectChains = mEffectChains;
1748    for (size_t i = 0; i < effectChains.size(); i ++) {
1749        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1750    }
1751}
1752
1753status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1754{
1755    if (halFrames == 0 || dspFrames == 0) {
1756        return BAD_VALUE;
1757    }
1758    Mutex::Autolock _l(mLock);
1759    if (initCheck() != NO_ERROR) {
1760        return INVALID_OPERATION;
1761    }
1762    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1763
1764    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1765}
1766
1767uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1768{
1769    Mutex::Autolock _l(mLock);
1770    uint32_t result = 0;
1771    if (getEffectChain_l(sessionId) != 0) {
1772        result = EFFECT_SESSION;
1773    }
1774
1775    for (size_t i = 0; i < mTracks.size(); ++i) {
1776        sp<Track> track = mTracks[i];
1777        if (sessionId == track->sessionId() &&
1778                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1779            result |= TRACK_SESSION;
1780            break;
1781        }
1782    }
1783
1784    return result;
1785}
1786
1787uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1788{
1789    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1790    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1791    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1792        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1793    }
1794    for (size_t i = 0; i < mTracks.size(); i++) {
1795        sp<Track> track = mTracks[i];
1796        if (sessionId == track->sessionId() &&
1797                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1798            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1799        }
1800    }
1801    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1802}
1803
1804
1805AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1806{
1807    Mutex::Autolock _l(mLock);
1808    return mOutput;
1809}
1810
1811AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1812{
1813    Mutex::Autolock _l(mLock);
1814    AudioStreamOut *output = mOutput;
1815    mOutput = NULL;
1816    return output;
1817}
1818
1819// this method must always be called either with ThreadBase mLock held or inside the thread loop
1820audio_stream_t* AudioFlinger::PlaybackThread::stream()
1821{
1822    if (mOutput == NULL) {
1823        return NULL;
1824    }
1825    return &mOutput->stream->common;
1826}
1827
1828uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
1829{
1830    // A2DP output latency is not due only to buffering capacity. It also reflects encoding,
1831    // decoding and transfer time. So sleeping for half of the latency would likely cause
1832    // underruns
1833    if (audio_is_a2dp_device((audio_devices_t)mDevice)) {
1834        return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000);
1835    } else {
1836        return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
1837    }
1838}
1839
1840// ----------------------------------------------------------------------------
1841
1842AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1843    :   PlaybackThread(audioFlinger, output, id, device),
1844        mAudioMixer(NULL)
1845{
1846    mType = ThreadBase::MIXER;
1847    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1848
1849    // FIXME - Current mixer implementation only supports stereo output
1850    if (mChannelCount == 1) {
1851        ALOGE("Invalid audio hardware channel count");
1852    }
1853}
1854
1855AudioFlinger::MixerThread::~MixerThread()
1856{
1857    delete mAudioMixer;
1858}
1859
1860bool AudioFlinger::MixerThread::threadLoop()
1861{
1862    Vector< sp<Track> > tracksToRemove;
1863    uint32_t mixerStatus = MIXER_IDLE;
1864    nsecs_t standbyTime = systemTime();
1865    size_t mixBufferSize = mFrameCount * mFrameSize;
1866    // FIXME: Relaxed timing because of a certain device that can't meet latency
1867    // Should be reduced to 2x after the vendor fixes the driver issue
1868    // increase threshold again due to low power audio mode. The way this warning threshold is
1869    // calculated and its usefulness should be reconsidered anyway.
1870    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1871    nsecs_t lastWarning = 0;
1872    bool longStandbyExit = false;
1873    uint32_t activeSleepTime = activeSleepTimeUs();
1874    uint32_t idleSleepTime = idleSleepTimeUs();
1875    uint32_t sleepTime = idleSleepTime;
1876    uint32_t sleepTimeShift = 0;
1877    Vector< sp<EffectChain> > effectChains;
1878#ifdef DEBUG_CPU_USAGE
1879    ThreadCpuUsage cpu;
1880    const CentralTendencyStatistics& stats = cpu.statistics();
1881#endif
1882
1883    acquireWakeLock();
1884
1885    while (!exitPending())
1886    {
1887#ifdef DEBUG_CPU_USAGE
1888        cpu.sampleAndEnable();
1889        unsigned n = stats.n();
1890        // cpu.elapsed() is expensive, so don't call it every loop
1891        if ((n & 127) == 1) {
1892            long long elapsed = cpu.elapsed();
1893            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1894                double perLoop = elapsed / (double) n;
1895                double perLoop100 = perLoop * 0.01;
1896                double mean = stats.mean();
1897                double stddev = stats.stddev();
1898                double minimum = stats.minimum();
1899                double maximum = stats.maximum();
1900                cpu.resetStatistics();
1901                ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
1902                        elapsed * .000000001, n, perLoop * .000001,
1903                        mean * .001,
1904                        stddev * .001,
1905                        minimum * .001,
1906                        maximum * .001,
1907                        mean / perLoop100,
1908                        stddev / perLoop100,
1909                        minimum / perLoop100,
1910                        maximum / perLoop100);
1911            }
1912        }
1913#endif
1914        processConfigEvents();
1915
1916        mixerStatus = MIXER_IDLE;
1917        { // scope for mLock
1918
1919            Mutex::Autolock _l(mLock);
1920
1921            if (checkForNewParameters_l()) {
1922                mixBufferSize = mFrameCount * mFrameSize;
1923                // FIXME: Relaxed timing because of a certain device that can't meet latency
1924                // Should be reduced to 2x after the vendor fixes the driver issue
1925                // increase threshold again due to low power audio mode. The way this warning
1926                // threshold is calculated and its usefulness should be reconsidered anyway.
1927                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1928                activeSleepTime = activeSleepTimeUs();
1929                idleSleepTime = idleSleepTimeUs();
1930            }
1931
1932            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1933
1934            // put audio hardware into standby after short delay
1935            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1936                        mSuspended)) {
1937                if (!mStandby) {
1938                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1939                    mOutput->stream->common.standby(&mOutput->stream->common);
1940                    mStandby = true;
1941                    mBytesWritten = 0;
1942                }
1943
1944                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1945                    // we're about to wait, flush the binder command buffer
1946                    IPCThreadState::self()->flushCommands();
1947
1948                    if (exitPending()) break;
1949
1950                    releaseWakeLock_l();
1951                    // wait until we have something to do...
1952                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1953                    mWaitWorkCV.wait(mLock);
1954                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1955                    acquireWakeLock_l();
1956
1957                    if (mMasterMute == false) {
1958                        char value[PROPERTY_VALUE_MAX];
1959                        property_get("ro.audio.silent", value, "0");
1960                        if (atoi(value)) {
1961                            ALOGD("Silence is golden");
1962                            setMasterMute(true);
1963                        }
1964                    }
1965
1966                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1967                    sleepTime = idleSleepTime;
1968                    sleepTimeShift = 0;
1969                    continue;
1970                }
1971            }
1972
1973            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1974
1975            // prevent any changes in effect chain list and in each effect chain
1976            // during mixing and effect process as the audio buffers could be deleted
1977            // or modified if an effect is created or deleted
1978            lockEffectChains_l(effectChains);
1979        }
1980
1981        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1982            // mix buffers...
1983            mAudioMixer->process();
1984            sleepTime = 0;
1985            // increase sleep time progressively when application underrun condition clears
1986            if (sleepTimeShift > 0) {
1987                sleepTimeShift--;
1988            }
1989            standbyTime = systemTime() + kStandbyTimeInNsecs;
1990            //TODO: delay standby when effects have a tail
1991        } else {
1992            // If no tracks are ready, sleep once for the duration of an output
1993            // buffer size, then write 0s to the output
1994            if (sleepTime == 0) {
1995                if (mixerStatus == MIXER_TRACKS_ENABLED) {
1996                    sleepTime = activeSleepTime >> sleepTimeShift;
1997                    if (sleepTime < kMinThreadSleepTimeUs) {
1998                        sleepTime = kMinThreadSleepTimeUs;
1999                    }
2000                    // reduce sleep time in case of consecutive application underruns to avoid
2001                    // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
2002                    // duration we would end up writing less data than needed by the audio HAL if
2003                    // the condition persists.
2004                    if (sleepTimeShift < kMaxThreadSleepTimeShift) {
2005                        sleepTimeShift++;
2006                    }
2007                } else {
2008                    sleepTime = idleSleepTime;
2009                }
2010            } else if (mBytesWritten != 0 ||
2011                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
2012                memset (mMixBuffer, 0, mixBufferSize);
2013                sleepTime = 0;
2014                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
2015            }
2016            // TODO add standby time extension fct of effect tail
2017        }
2018
2019        if (mSuspended) {
2020            sleepTime = suspendSleepTimeUs();
2021        }
2022        // sleepTime == 0 means we must write to audio hardware
2023        if (sleepTime == 0) {
2024            for (size_t i = 0; i < effectChains.size(); i ++) {
2025                effectChains[i]->process_l();
2026            }
2027            // enable changes in effect chain
2028            unlockEffectChains(effectChains);
2029            mLastWriteTime = systemTime();
2030            mInWrite = true;
2031            mBytesWritten += mixBufferSize;
2032
2033            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2034            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2035            mNumWrites++;
2036            mInWrite = false;
2037            nsecs_t now = systemTime();
2038            nsecs_t delta = now - mLastWriteTime;
2039            if (!mStandby && delta > maxPeriod) {
2040                mNumDelayedWrites++;
2041                if ((now - lastWarning) > kWarningThrottleNs) {
2042                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2043                            ns2ms(delta), mNumDelayedWrites, this);
2044                    lastWarning = now;
2045                }
2046                if (mStandby) {
2047                    longStandbyExit = true;
2048                }
2049            }
2050            mStandby = false;
2051        } else {
2052            // enable changes in effect chain
2053            unlockEffectChains(effectChains);
2054            usleep(sleepTime);
2055        }
2056
2057        // finally let go of all our tracks, without the lock held
2058        // since we can't guarantee the destructors won't acquire that
2059        // same lock.
2060        tracksToRemove.clear();
2061
2062        // Effect chains will be actually deleted here if they were removed from
2063        // mEffectChains list during mixing or effects processing
2064        effectChains.clear();
2065    }
2066
2067    if (!mStandby) {
2068        mOutput->stream->common.standby(&mOutput->stream->common);
2069    }
2070
2071    releaseWakeLock();
2072
2073    ALOGV("MixerThread %p exiting", this);
2074    return false;
2075}
2076
2077// prepareTracks_l() must be called with ThreadBase::mLock held
2078uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2079{
2080
2081    uint32_t mixerStatus = MIXER_IDLE;
2082    // find out which tracks need to be processed
2083    size_t count = activeTracks.size();
2084    size_t mixedTracks = 0;
2085    size_t tracksWithEffect = 0;
2086
2087    float masterVolume = mMasterVolume;
2088    bool  masterMute = mMasterMute;
2089
2090    if (masterMute) {
2091        masterVolume = 0;
2092    }
2093    // Delegate master volume control to effect in output mix effect chain if needed
2094    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2095    if (chain != 0) {
2096        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2097        chain->setVolume_l(&v, &v);
2098        masterVolume = (float)((v + (1 << 23)) >> 24);
2099        chain.clear();
2100    }
2101
2102    for (size_t i=0 ; i<count ; i++) {
2103        sp<Track> t = activeTracks[i].promote();
2104        if (t == 0) continue;
2105
2106        // this const just means the local variable doesn't change
2107        Track* const track = t.get();
2108        audio_track_cblk_t* cblk = track->cblk();
2109
2110        // The first time a track is added we wait
2111        // for all its buffers to be filled before processing it
2112        int name = track->name();
2113        // make sure that we have enough frames to mix one full buffer.
2114        // enforce this condition only once to enable draining the buffer in case the client
2115        // app does not call stop() and relies on underrun to stop:
2116        // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed
2117        // during last round
2118        uint32_t minFrames = 1;
2119        if (!track->isStopped() && !track->isPausing() &&
2120                (track->mRetryCount >= kMaxTrackRetries)) {
2121            if (t->sampleRate() == (int)mSampleRate) {
2122                minFrames = mFrameCount;
2123            } else {
2124                // +1 for rounding and +1 for additional sample needed for interpolation
2125                minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
2126                // add frames already consumed but not yet released by the resampler
2127                // because cblk->framesReady() will  include these frames
2128                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
2129                // the minimum track buffer size is normally twice the number of frames necessary
2130                // to fill one buffer and the resampler should not leave more than one buffer worth
2131                // of unreleased frames after each pass, but just in case...
2132                ALOG_ASSERT(minFrames <= cblk->frameCount);
2133            }
2134        }
2135        if ((cblk->framesReady() >= minFrames) && track->isReady() &&
2136                !track->isPaused() && !track->isTerminated())
2137        {
2138            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
2139
2140            mixedTracks++;
2141
2142            // track->mainBuffer() != mMixBuffer means there is an effect chain
2143            // connected to the track
2144            chain.clear();
2145            if (track->mainBuffer() != mMixBuffer) {
2146                chain = getEffectChain_l(track->sessionId());
2147                // Delegate volume control to effect in track effect chain if needed
2148                if (chain != 0) {
2149                    tracksWithEffect++;
2150                } else {
2151                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
2152                            name, track->sessionId());
2153                }
2154            }
2155
2156
2157            int param = AudioMixer::VOLUME;
2158            if (track->mFillingUpStatus == Track::FS_FILLED) {
2159                // no ramp for the first volume setting
2160                track->mFillingUpStatus = Track::FS_ACTIVE;
2161                if (track->mState == TrackBase::RESUMING) {
2162                    track->mState = TrackBase::ACTIVE;
2163                    param = AudioMixer::RAMP_VOLUME;
2164                }
2165                mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2166            } else if (cblk->server != 0) {
2167                // If the track is stopped before the first frame was mixed,
2168                // do not apply ramp
2169                param = AudioMixer::RAMP_VOLUME;
2170            }
2171
2172            // compute volume for this track
2173            uint32_t vl, vr, va;
2174            if (track->isMuted() || track->isPausing() ||
2175                mStreamTypes[track->type()].mute) {
2176                vl = vr = va = 0;
2177                if (track->isPausing()) {
2178                    track->setPaused();
2179                }
2180            } else {
2181
2182                // read original volumes with volume control
2183                float typeVolume = mStreamTypes[track->type()].volume;
2184                float v = masterVolume * typeVolume;
2185                vl = (uint32_t)(v * cblk->volume[0]) << 12;
2186                vr = (uint32_t)(v * cblk->volume[1]) << 12;
2187
2188                uint16_t sendLevel = cblk->getSendLevel_U4_12();
2189                // send level comes from shared memory and so may be corrupt
2190                if (sendLevel >= 0x1000) {
2191                    ALOGV("Track send level out of range: %04X", sendLevel);
2192                    sendLevel = 0x1000;
2193                }
2194                va = (uint32_t)(v * sendLevel);
2195            }
2196            // Delegate volume control to effect in track effect chain if needed
2197            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2198                // Do not ramp volume if volume is controlled by effect
2199                param = AudioMixer::VOLUME;
2200                track->mHasVolumeController = true;
2201            } else {
2202                // force no volume ramp when volume controller was just disabled or removed
2203                // from effect chain to avoid volume spike
2204                if (track->mHasVolumeController) {
2205                    param = AudioMixer::VOLUME;
2206                }
2207                track->mHasVolumeController = false;
2208            }
2209
2210            // Convert volumes from 8.24 to 4.12 format
2211            int16_t left, right, aux;
2212            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2213            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2214            left = int16_t(v_clamped);
2215            v_clamped = (vr + (1 << 11)) >> 12;
2216            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2217            right = int16_t(v_clamped);
2218
2219            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2220            aux = int16_t(va);
2221
2222            // XXX: these things DON'T need to be done each time
2223            mAudioMixer->setBufferProvider(name, track);
2224            mAudioMixer->enable(name);
2225
2226            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left);
2227            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right);
2228            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux);
2229            mAudioMixer->setParameter(
2230                name,
2231                AudioMixer::TRACK,
2232                AudioMixer::FORMAT, (void *)track->format());
2233            mAudioMixer->setParameter(
2234                name,
2235                AudioMixer::TRACK,
2236                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2237            mAudioMixer->setParameter(
2238                name,
2239                AudioMixer::RESAMPLE,
2240                AudioMixer::SAMPLE_RATE,
2241                (void *)(cblk->sampleRate));
2242            mAudioMixer->setParameter(
2243                name,
2244                AudioMixer::TRACK,
2245                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2246            mAudioMixer->setParameter(
2247                name,
2248                AudioMixer::TRACK,
2249                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2250
2251            // reset retry count
2252            track->mRetryCount = kMaxTrackRetries;
2253            mixerStatus = MIXER_TRACKS_READY;
2254        } else {
2255            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
2256            if (track->isStopped()) {
2257                track->reset();
2258            }
2259            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2260                // We have consumed all the buffers of this track.
2261                // Remove it from the list of active tracks.
2262                tracksToRemove->add(track);
2263            } else {
2264                // No buffers for this track. Give it a few chances to
2265                // fill a buffer, then remove it from active list.
2266                if (--(track->mRetryCount) <= 0) {
2267                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
2268                    tracksToRemove->add(track);
2269                    // indicate to client process that the track was disabled because of underrun
2270                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2271                } else if (mixerStatus != MIXER_TRACKS_READY) {
2272                    mixerStatus = MIXER_TRACKS_ENABLED;
2273                }
2274            }
2275            mAudioMixer->disable(name);
2276        }
2277    }
2278
2279    // remove all the tracks that need to be...
2280    count = tracksToRemove->size();
2281    if (CC_UNLIKELY(count)) {
2282        for (size_t i=0 ; i<count ; i++) {
2283            const sp<Track>& track = tracksToRemove->itemAt(i);
2284            mActiveTracks.remove(track);
2285            if (track->mainBuffer() != mMixBuffer) {
2286                chain = getEffectChain_l(track->sessionId());
2287                if (chain != 0) {
2288                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2289                    chain->decActiveTrackCnt();
2290                }
2291            }
2292            if (track->isTerminated()) {
2293                removeTrack_l(track);
2294            }
2295        }
2296    }
2297
2298    // mix buffer must be cleared if all tracks are connected to an
2299    // effect chain as in this case the mixer will not write to
2300    // mix buffer and track effects will accumulate into it
2301    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2302        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2303    }
2304
2305    return mixerStatus;
2306}
2307
2308void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType)
2309{
2310    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2311            this,  streamType, mTracks.size());
2312    Mutex::Autolock _l(mLock);
2313
2314    size_t size = mTracks.size();
2315    for (size_t i = 0; i < size; i++) {
2316        sp<Track> t = mTracks[i];
2317        if (t->type() == streamType) {
2318            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2319            t->mCblk->cv.signal();
2320        }
2321    }
2322}
2323
2324void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid)
2325{
2326    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2327            this,  streamType, valid);
2328    Mutex::Autolock _l(mLock);
2329
2330    mStreamTypes[streamType].valid = valid;
2331}
2332
2333// getTrackName_l() must be called with ThreadBase::mLock held
2334int AudioFlinger::MixerThread::getTrackName_l()
2335{
2336    return mAudioMixer->getTrackName();
2337}
2338
2339// deleteTrackName_l() must be called with ThreadBase::mLock held
2340void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2341{
2342    ALOGV("remove track (%d) and delete from mixer", name);
2343    mAudioMixer->deleteTrackName(name);
2344}
2345
2346// checkForNewParameters_l() must be called with ThreadBase::mLock held
2347bool AudioFlinger::MixerThread::checkForNewParameters_l()
2348{
2349    bool reconfig = false;
2350
2351    while (!mNewParameters.isEmpty()) {
2352        status_t status = NO_ERROR;
2353        String8 keyValuePair = mNewParameters[0];
2354        AudioParameter param = AudioParameter(keyValuePair);
2355        int value;
2356
2357        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2358            reconfig = true;
2359        }
2360        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2361            if (value != AUDIO_FORMAT_PCM_16_BIT) {
2362                status = BAD_VALUE;
2363            } else {
2364                reconfig = true;
2365            }
2366        }
2367        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2368            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2369                status = BAD_VALUE;
2370            } else {
2371                reconfig = true;
2372            }
2373        }
2374        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2375            // do not accept frame count changes if tracks are open as the track buffer
2376            // size depends on frame count and correct behavior would not be guaranteed
2377            // if frame count is changed after track creation
2378            if (!mTracks.isEmpty()) {
2379                status = INVALID_OPERATION;
2380            } else {
2381                reconfig = true;
2382            }
2383        }
2384        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2385            // when changing the audio output device, call addBatteryData to notify
2386            // the change
2387            if ((int)mDevice != value) {
2388                uint32_t params = 0;
2389                // check whether speaker is on
2390                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2391                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2392                }
2393
2394                int deviceWithoutSpeaker
2395                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2396                // check if any other device (except speaker) is on
2397                if (value & deviceWithoutSpeaker ) {
2398                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2399                }
2400
2401                if (params != 0) {
2402                    addBatteryData(params);
2403                }
2404            }
2405
2406            // forward device change to effects that have requested to be
2407            // aware of attached audio device.
2408            mDevice = (uint32_t)value;
2409            for (size_t i = 0; i < mEffectChains.size(); i++) {
2410                mEffectChains[i]->setDevice_l(mDevice);
2411            }
2412        }
2413
2414        if (status == NO_ERROR) {
2415            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2416                                                    keyValuePair.string());
2417            if (!mStandby && status == INVALID_OPERATION) {
2418               mOutput->stream->common.standby(&mOutput->stream->common);
2419               mStandby = true;
2420               mBytesWritten = 0;
2421               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2422                                                       keyValuePair.string());
2423            }
2424            if (status == NO_ERROR && reconfig) {
2425                delete mAudioMixer;
2426                readOutputParameters();
2427                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2428                for (size_t i = 0; i < mTracks.size() ; i++) {
2429                    int name = getTrackName_l();
2430                    if (name < 0) break;
2431                    mTracks[i]->mName = name;
2432                    // limit track sample rate to 2 x new output sample rate
2433                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2434                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2435                    }
2436                }
2437                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2438            }
2439        }
2440
2441        mNewParameters.removeAt(0);
2442
2443        mParamStatus = status;
2444        mParamCond.signal();
2445        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2446        // already timed out waiting for the status and will never signal the condition.
2447        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2448    }
2449    return reconfig;
2450}
2451
2452status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2453{
2454    const size_t SIZE = 256;
2455    char buffer[SIZE];
2456    String8 result;
2457
2458    PlaybackThread::dumpInternals(fd, args);
2459
2460    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2461    result.append(buffer);
2462    write(fd, result.string(), result.size());
2463    return NO_ERROR;
2464}
2465
2466uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2467{
2468    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2469}
2470
2471uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2472{
2473    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2474}
2475
2476// ----------------------------------------------------------------------------
2477AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2478    :   PlaybackThread(audioFlinger, output, id, device)
2479{
2480    mType = ThreadBase::DIRECT;
2481}
2482
2483AudioFlinger::DirectOutputThread::~DirectOutputThread()
2484{
2485}
2486
2487static inline
2488int32_t mul(int16_t in, int16_t v)
2489{
2490#if defined(__arm__) && !defined(__thumb__)
2491    int32_t out;
2492    asm( "smulbb %[out], %[in], %[v] \n"
2493         : [out]"=r"(out)
2494         : [in]"%r"(in), [v]"r"(v)
2495         : );
2496    return out;
2497#else
2498    return in * int32_t(v);
2499#endif
2500}
2501
2502void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2503{
2504    // Do not apply volume on compressed audio
2505    if (!audio_is_linear_pcm(mFormat)) {
2506        return;
2507    }
2508
2509    // convert to signed 16 bit before volume calculation
2510    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2511        size_t count = mFrameCount * mChannelCount;
2512        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2513        int16_t *dst = mMixBuffer + count-1;
2514        while(count--) {
2515            *dst-- = (int16_t)(*src--^0x80) << 8;
2516        }
2517    }
2518
2519    size_t frameCount = mFrameCount;
2520    int16_t *out = mMixBuffer;
2521    if (ramp) {
2522        if (mChannelCount == 1) {
2523            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2524            int32_t vlInc = d / (int32_t)frameCount;
2525            int32_t vl = ((int32_t)mLeftVolShort << 16);
2526            do {
2527                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2528                out++;
2529                vl += vlInc;
2530            } while (--frameCount);
2531
2532        } else {
2533            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2534            int32_t vlInc = d / (int32_t)frameCount;
2535            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2536            int32_t vrInc = d / (int32_t)frameCount;
2537            int32_t vl = ((int32_t)mLeftVolShort << 16);
2538            int32_t vr = ((int32_t)mRightVolShort << 16);
2539            do {
2540                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2541                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2542                out += 2;
2543                vl += vlInc;
2544                vr += vrInc;
2545            } while (--frameCount);
2546        }
2547    } else {
2548        if (mChannelCount == 1) {
2549            do {
2550                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2551                out++;
2552            } while (--frameCount);
2553        } else {
2554            do {
2555                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2556                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2557                out += 2;
2558            } while (--frameCount);
2559        }
2560    }
2561
2562    // convert back to unsigned 8 bit after volume calculation
2563    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2564        size_t count = mFrameCount * mChannelCount;
2565        int16_t *src = mMixBuffer;
2566        uint8_t *dst = (uint8_t *)mMixBuffer;
2567        while(count--) {
2568            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2569        }
2570    }
2571
2572    mLeftVolShort = leftVol;
2573    mRightVolShort = rightVol;
2574}
2575
2576bool AudioFlinger::DirectOutputThread::threadLoop()
2577{
2578    uint32_t mixerStatus = MIXER_IDLE;
2579    sp<Track> trackToRemove;
2580    sp<Track> activeTrack;
2581    nsecs_t standbyTime = systemTime();
2582    int8_t *curBuf;
2583    size_t mixBufferSize = mFrameCount*mFrameSize;
2584    uint32_t activeSleepTime = activeSleepTimeUs();
2585    uint32_t idleSleepTime = idleSleepTimeUs();
2586    uint32_t sleepTime = idleSleepTime;
2587    // use shorter standby delay as on normal output to release
2588    // hardware resources as soon as possible
2589    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2590
2591    acquireWakeLock();
2592
2593    while (!exitPending())
2594    {
2595        bool rampVolume;
2596        uint16_t leftVol;
2597        uint16_t rightVol;
2598        Vector< sp<EffectChain> > effectChains;
2599
2600        processConfigEvents();
2601
2602        mixerStatus = MIXER_IDLE;
2603
2604        { // scope for the mLock
2605
2606            Mutex::Autolock _l(mLock);
2607
2608            if (checkForNewParameters_l()) {
2609                mixBufferSize = mFrameCount*mFrameSize;
2610                activeSleepTime = activeSleepTimeUs();
2611                idleSleepTime = idleSleepTimeUs();
2612                standbyDelay = microseconds(activeSleepTime*2);
2613            }
2614
2615            // put audio hardware into standby after short delay
2616            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2617                        mSuspended)) {
2618                // wait until we have something to do...
2619                if (!mStandby) {
2620                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2621                    mOutput->stream->common.standby(&mOutput->stream->common);
2622                    mStandby = true;
2623                    mBytesWritten = 0;
2624                }
2625
2626                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2627                    // we're about to wait, flush the binder command buffer
2628                    IPCThreadState::self()->flushCommands();
2629
2630                    if (exitPending()) break;
2631
2632                    releaseWakeLock_l();
2633                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2634                    mWaitWorkCV.wait(mLock);
2635                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2636                    acquireWakeLock_l();
2637
2638                    if (mMasterMute == false) {
2639                        char value[PROPERTY_VALUE_MAX];
2640                        property_get("ro.audio.silent", value, "0");
2641                        if (atoi(value)) {
2642                            ALOGD("Silence is golden");
2643                            setMasterMute(true);
2644                        }
2645                    }
2646
2647                    standbyTime = systemTime() + standbyDelay;
2648                    sleepTime = idleSleepTime;
2649                    continue;
2650                }
2651            }
2652
2653            effectChains = mEffectChains;
2654
2655            // find out which tracks need to be processed
2656            if (mActiveTracks.size() != 0) {
2657                sp<Track> t = mActiveTracks[0].promote();
2658                if (t == 0) continue;
2659
2660                Track* const track = t.get();
2661                audio_track_cblk_t* cblk = track->cblk();
2662
2663                // The first time a track is added we wait
2664                // for all its buffers to be filled before processing it
2665                if (cblk->framesReady() && track->isReady() &&
2666                        !track->isPaused() && !track->isTerminated())
2667                {
2668                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2669
2670                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2671                        track->mFillingUpStatus = Track::FS_ACTIVE;
2672                        mLeftVolFloat = mRightVolFloat = 0;
2673                        mLeftVolShort = mRightVolShort = 0;
2674                        if (track->mState == TrackBase::RESUMING) {
2675                            track->mState = TrackBase::ACTIVE;
2676                            rampVolume = true;
2677                        }
2678                    } else if (cblk->server != 0) {
2679                        // If the track is stopped before the first frame was mixed,
2680                        // do not apply ramp
2681                        rampVolume = true;
2682                    }
2683                    // compute volume for this track
2684                    float left, right;
2685                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2686                        mStreamTypes[track->type()].mute) {
2687                        left = right = 0;
2688                        if (track->isPausing()) {
2689                            track->setPaused();
2690                        }
2691                    } else {
2692                        float typeVolume = mStreamTypes[track->type()].volume;
2693                        float v = mMasterVolume * typeVolume;
2694                        float v_clamped = v * cblk->volume[0];
2695                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2696                        left = v_clamped/MAX_GAIN;
2697                        v_clamped = v * cblk->volume[1];
2698                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2699                        right = v_clamped/MAX_GAIN;
2700                    }
2701
2702                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2703                        mLeftVolFloat = left;
2704                        mRightVolFloat = right;
2705
2706                        // If audio HAL implements volume control,
2707                        // force software volume to nominal value
2708                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2709                            left = 1.0f;
2710                            right = 1.0f;
2711                        }
2712
2713                        // Convert volumes from float to 8.24
2714                        uint32_t vl = (uint32_t)(left * (1 << 24));
2715                        uint32_t vr = (uint32_t)(right * (1 << 24));
2716
2717                        // Delegate volume control to effect in track effect chain if needed
2718                        // only one effect chain can be present on DirectOutputThread, so if
2719                        // there is one, the track is connected to it
2720                        if (!effectChains.isEmpty()) {
2721                            // Do not ramp volume if volume is controlled by effect
2722                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2723                                rampVolume = false;
2724                            }
2725                        }
2726
2727                        // Convert volumes from 8.24 to 4.12 format
2728                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2729                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2730                        leftVol = (uint16_t)v_clamped;
2731                        v_clamped = (vr + (1 << 11)) >> 12;
2732                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2733                        rightVol = (uint16_t)v_clamped;
2734                    } else {
2735                        leftVol = mLeftVolShort;
2736                        rightVol = mRightVolShort;
2737                        rampVolume = false;
2738                    }
2739
2740                    // reset retry count
2741                    track->mRetryCount = kMaxTrackRetriesDirect;
2742                    activeTrack = t;
2743                    mixerStatus = MIXER_TRACKS_READY;
2744                } else {
2745                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2746                    if (track->isStopped()) {
2747                        track->reset();
2748                    }
2749                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2750                        // We have consumed all the buffers of this track.
2751                        // Remove it from the list of active tracks.
2752                        trackToRemove = track;
2753                    } else {
2754                        // No buffers for this track. Give it a few chances to
2755                        // fill a buffer, then remove it from active list.
2756                        if (--(track->mRetryCount) <= 0) {
2757                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2758                            trackToRemove = track;
2759                        } else {
2760                            mixerStatus = MIXER_TRACKS_ENABLED;
2761                        }
2762                    }
2763                }
2764            }
2765
2766            // remove all the tracks that need to be...
2767            if (CC_UNLIKELY(trackToRemove != 0)) {
2768                mActiveTracks.remove(trackToRemove);
2769                if (!effectChains.isEmpty()) {
2770                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2771                            trackToRemove->sessionId());
2772                    effectChains[0]->decActiveTrackCnt();
2773                }
2774                if (trackToRemove->isTerminated()) {
2775                    removeTrack_l(trackToRemove);
2776                }
2777            }
2778
2779            lockEffectChains_l(effectChains);
2780       }
2781
2782        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2783            AudioBufferProvider::Buffer buffer;
2784            size_t frameCount = mFrameCount;
2785            curBuf = (int8_t *)mMixBuffer;
2786            // output audio to hardware
2787            while (frameCount) {
2788                buffer.frameCount = frameCount;
2789                activeTrack->getNextBuffer(&buffer);
2790                if (CC_UNLIKELY(buffer.raw == NULL)) {
2791                    memset(curBuf, 0, frameCount * mFrameSize);
2792                    break;
2793                }
2794                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2795                frameCount -= buffer.frameCount;
2796                curBuf += buffer.frameCount * mFrameSize;
2797                activeTrack->releaseBuffer(&buffer);
2798            }
2799            sleepTime = 0;
2800            standbyTime = systemTime() + standbyDelay;
2801        } else {
2802            if (sleepTime == 0) {
2803                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2804                    sleepTime = activeSleepTime;
2805                } else {
2806                    sleepTime = idleSleepTime;
2807                }
2808            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2809                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2810                sleepTime = 0;
2811            }
2812        }
2813
2814        if (mSuspended) {
2815            sleepTime = suspendSleepTimeUs();
2816        }
2817        // sleepTime == 0 means we must write to audio hardware
2818        if (sleepTime == 0) {
2819            if (mixerStatus == MIXER_TRACKS_READY) {
2820                applyVolume(leftVol, rightVol, rampVolume);
2821            }
2822            for (size_t i = 0; i < effectChains.size(); i ++) {
2823                effectChains[i]->process_l();
2824            }
2825            unlockEffectChains(effectChains);
2826
2827            mLastWriteTime = systemTime();
2828            mInWrite = true;
2829            mBytesWritten += mixBufferSize;
2830            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2831            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2832            mNumWrites++;
2833            mInWrite = false;
2834            mStandby = false;
2835        } else {
2836            unlockEffectChains(effectChains);
2837            usleep(sleepTime);
2838        }
2839
2840        // finally let go of removed track, without the lock held
2841        // since we can't guarantee the destructors won't acquire that
2842        // same lock.
2843        trackToRemove.clear();
2844        activeTrack.clear();
2845
2846        // Effect chains will be actually deleted here if they were removed from
2847        // mEffectChains list during mixing or effects processing
2848        effectChains.clear();
2849    }
2850
2851    if (!mStandby) {
2852        mOutput->stream->common.standby(&mOutput->stream->common);
2853    }
2854
2855    releaseWakeLock();
2856
2857    ALOGV("DirectOutputThread %p exiting", this);
2858    return false;
2859}
2860
2861// getTrackName_l() must be called with ThreadBase::mLock held
2862int AudioFlinger::DirectOutputThread::getTrackName_l()
2863{
2864    return 0;
2865}
2866
2867// deleteTrackName_l() must be called with ThreadBase::mLock held
2868void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2869{
2870}
2871
2872// checkForNewParameters_l() must be called with ThreadBase::mLock held
2873bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2874{
2875    bool reconfig = false;
2876
2877    while (!mNewParameters.isEmpty()) {
2878        status_t status = NO_ERROR;
2879        String8 keyValuePair = mNewParameters[0];
2880        AudioParameter param = AudioParameter(keyValuePair);
2881        int value;
2882
2883        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2884            // do not accept frame count changes if tracks are open as the track buffer
2885            // size depends on frame count and correct behavior would not be garantied
2886            // if frame count is changed after track creation
2887            if (!mTracks.isEmpty()) {
2888                status = INVALID_OPERATION;
2889            } else {
2890                reconfig = true;
2891            }
2892        }
2893        if (status == NO_ERROR) {
2894            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2895                                                    keyValuePair.string());
2896            if (!mStandby && status == INVALID_OPERATION) {
2897               mOutput->stream->common.standby(&mOutput->stream->common);
2898               mStandby = true;
2899               mBytesWritten = 0;
2900               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2901                                                       keyValuePair.string());
2902            }
2903            if (status == NO_ERROR && reconfig) {
2904                readOutputParameters();
2905                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2906            }
2907        }
2908
2909        mNewParameters.removeAt(0);
2910
2911        mParamStatus = status;
2912        mParamCond.signal();
2913        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2914        // already timed out waiting for the status and will never signal the condition.
2915        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2916    }
2917    return reconfig;
2918}
2919
2920uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2921{
2922    uint32_t time;
2923    if (audio_is_linear_pcm(mFormat)) {
2924        time = PlaybackThread::activeSleepTimeUs();
2925    } else {
2926        time = 10000;
2927    }
2928    return time;
2929}
2930
2931uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2932{
2933    uint32_t time;
2934    if (audio_is_linear_pcm(mFormat)) {
2935        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2936    } else {
2937        time = 10000;
2938    }
2939    return time;
2940}
2941
2942uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2943{
2944    uint32_t time;
2945    if (audio_is_linear_pcm(mFormat)) {
2946        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2947    } else {
2948        time = 10000;
2949    }
2950    return time;
2951}
2952
2953
2954// ----------------------------------------------------------------------------
2955
2956AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2957    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2958{
2959    mType = ThreadBase::DUPLICATING;
2960    addOutputTrack(mainThread);
2961}
2962
2963AudioFlinger::DuplicatingThread::~DuplicatingThread()
2964{
2965    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2966        mOutputTracks[i]->destroy();
2967    }
2968    mOutputTracks.clear();
2969}
2970
2971bool AudioFlinger::DuplicatingThread::threadLoop()
2972{
2973    Vector< sp<Track> > tracksToRemove;
2974    uint32_t mixerStatus = MIXER_IDLE;
2975    nsecs_t standbyTime = systemTime();
2976    size_t mixBufferSize = mFrameCount*mFrameSize;
2977    SortedVector< sp<OutputTrack> > outputTracks;
2978    uint32_t writeFrames = 0;
2979    uint32_t activeSleepTime = activeSleepTimeUs();
2980    uint32_t idleSleepTime = idleSleepTimeUs();
2981    uint32_t sleepTime = idleSleepTime;
2982    Vector< sp<EffectChain> > effectChains;
2983
2984    acquireWakeLock();
2985
2986    while (!exitPending())
2987    {
2988        processConfigEvents();
2989
2990        mixerStatus = MIXER_IDLE;
2991        { // scope for the mLock
2992
2993            Mutex::Autolock _l(mLock);
2994
2995            if (checkForNewParameters_l()) {
2996                mixBufferSize = mFrameCount*mFrameSize;
2997                updateWaitTime();
2998                activeSleepTime = activeSleepTimeUs();
2999                idleSleepTime = idleSleepTimeUs();
3000            }
3001
3002            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
3003
3004            for (size_t i = 0; i < mOutputTracks.size(); i++) {
3005                outputTracks.add(mOutputTracks[i]);
3006            }
3007
3008            // put audio hardware into standby after short delay
3009            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
3010                         mSuspended)) {
3011                if (!mStandby) {
3012                    for (size_t i = 0; i < outputTracks.size(); i++) {
3013                        outputTracks[i]->stop();
3014                    }
3015                    mStandby = true;
3016                    mBytesWritten = 0;
3017                }
3018
3019                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
3020                    // we're about to wait, flush the binder command buffer
3021                    IPCThreadState::self()->flushCommands();
3022                    outputTracks.clear();
3023
3024                    if (exitPending()) break;
3025
3026                    releaseWakeLock_l();
3027                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3028                    mWaitWorkCV.wait(mLock);
3029                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3030                    acquireWakeLock_l();
3031
3032                    if (mMasterMute == false) {
3033                        char value[PROPERTY_VALUE_MAX];
3034                        property_get("ro.audio.silent", value, "0");
3035                        if (atoi(value)) {
3036                            ALOGD("Silence is golden");
3037                            setMasterMute(true);
3038                        }
3039                    }
3040
3041                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3042                    sleepTime = idleSleepTime;
3043                    continue;
3044                }
3045            }
3046
3047            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3048
3049            // prevent any changes in effect chain list and in each effect chain
3050            // during mixing and effect process as the audio buffers could be deleted
3051            // or modified if an effect is created or deleted
3052            lockEffectChains_l(effectChains);
3053        }
3054
3055        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3056            // mix buffers...
3057            if (outputsReady(outputTracks)) {
3058                mAudioMixer->process();
3059            } else {
3060                memset(mMixBuffer, 0, mixBufferSize);
3061            }
3062            sleepTime = 0;
3063            writeFrames = mFrameCount;
3064        } else {
3065            if (sleepTime == 0) {
3066                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3067                    sleepTime = activeSleepTime;
3068                } else {
3069                    sleepTime = idleSleepTime;
3070                }
3071            } else if (mBytesWritten != 0) {
3072                // flush remaining overflow buffers in output tracks
3073                for (size_t i = 0; i < outputTracks.size(); i++) {
3074                    if (outputTracks[i]->isActive()) {
3075                        sleepTime = 0;
3076                        writeFrames = 0;
3077                        memset(mMixBuffer, 0, mixBufferSize);
3078                        break;
3079                    }
3080                }
3081            }
3082        }
3083
3084        if (mSuspended) {
3085            sleepTime = suspendSleepTimeUs();
3086        }
3087        // sleepTime == 0 means we must write to audio hardware
3088        if (sleepTime == 0) {
3089            for (size_t i = 0; i < effectChains.size(); i ++) {
3090                effectChains[i]->process_l();
3091            }
3092            // enable changes in effect chain
3093            unlockEffectChains(effectChains);
3094
3095            standbyTime = systemTime() + kStandbyTimeInNsecs;
3096            for (size_t i = 0; i < outputTracks.size(); i++) {
3097                outputTracks[i]->write(mMixBuffer, writeFrames);
3098            }
3099            mStandby = false;
3100            mBytesWritten += mixBufferSize;
3101        } else {
3102            // enable changes in effect chain
3103            unlockEffectChains(effectChains);
3104            usleep(sleepTime);
3105        }
3106
3107        // finally let go of all our tracks, without the lock held
3108        // since we can't guarantee the destructors won't acquire that
3109        // same lock.
3110        tracksToRemove.clear();
3111        outputTracks.clear();
3112
3113        // Effect chains will be actually deleted here if they were removed from
3114        // mEffectChains list during mixing or effects processing
3115        effectChains.clear();
3116    }
3117
3118    releaseWakeLock();
3119
3120    return false;
3121}
3122
3123void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3124{
3125    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3126    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3127                                            this,
3128                                            mSampleRate,
3129                                            mFormat,
3130                                            mChannelMask,
3131                                            frameCount);
3132    if (outputTrack->cblk() != NULL) {
3133        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3134        mOutputTracks.add(outputTrack);
3135        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3136        updateWaitTime();
3137    }
3138}
3139
3140void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3141{
3142    Mutex::Autolock _l(mLock);
3143    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3144        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3145            mOutputTracks[i]->destroy();
3146            mOutputTracks.removeAt(i);
3147            updateWaitTime();
3148            return;
3149        }
3150    }
3151    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3152}
3153
3154void AudioFlinger::DuplicatingThread::updateWaitTime()
3155{
3156    mWaitTimeMs = UINT_MAX;
3157    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3158        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3159        if (strong != NULL) {
3160            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3161            if (waitTimeMs < mWaitTimeMs) {
3162                mWaitTimeMs = waitTimeMs;
3163            }
3164        }
3165    }
3166}
3167
3168
3169bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3170{
3171    for (size_t i = 0; i < outputTracks.size(); i++) {
3172        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3173        if (thread == 0) {
3174            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3175            return false;
3176        }
3177        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3178        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3179            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3180            return false;
3181        }
3182    }
3183    return true;
3184}
3185
3186uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3187{
3188    return (mWaitTimeMs * 1000) / 2;
3189}
3190
3191// ----------------------------------------------------------------------------
3192
3193// TrackBase constructor must be called with AudioFlinger::mLock held
3194AudioFlinger::ThreadBase::TrackBase::TrackBase(
3195            const wp<ThreadBase>& thread,
3196            const sp<Client>& client,
3197            uint32_t sampleRate,
3198            uint32_t format,
3199            uint32_t channelMask,
3200            int frameCount,
3201            uint32_t flags,
3202            const sp<IMemory>& sharedBuffer,
3203            int sessionId)
3204    :   RefBase(),
3205        mThread(thread),
3206        mClient(client),
3207        mCblk(0),
3208        mFrameCount(0),
3209        mState(IDLE),
3210        mClientTid(-1),
3211        mFormat(format),
3212        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3213        mSessionId(sessionId)
3214{
3215    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3216
3217    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3218   size_t size = sizeof(audio_track_cblk_t);
3219   uint8_t channelCount = popcount(channelMask);
3220   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3221   if (sharedBuffer == 0) {
3222       size += bufferSize;
3223   }
3224
3225   if (client != NULL) {
3226        mCblkMemory = client->heap()->allocate(size);
3227        if (mCblkMemory != 0) {
3228            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3229            if (mCblk) { // construct the shared structure in-place.
3230                new(mCblk) audio_track_cblk_t();
3231                // clear all buffers
3232                mCblk->frameCount = frameCount;
3233                mCblk->sampleRate = sampleRate;
3234                mChannelCount = channelCount;
3235                mChannelMask = channelMask;
3236                if (sharedBuffer == 0) {
3237                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3238                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3239                    // Force underrun condition to avoid false underrun callback until first data is
3240                    // written to buffer (other flags are cleared)
3241                    mCblk->flags = CBLK_UNDERRUN_ON;
3242                } else {
3243                    mBuffer = sharedBuffer->pointer();
3244                }
3245                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3246            }
3247        } else {
3248            ALOGE("not enough memory for AudioTrack size=%u", size);
3249            client->heap()->dump("AudioTrack");
3250            return;
3251        }
3252   } else {
3253       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3254           // construct the shared structure in-place.
3255           new(mCblk) audio_track_cblk_t();
3256           // clear all buffers
3257           mCblk->frameCount = frameCount;
3258           mCblk->sampleRate = sampleRate;
3259           mChannelCount = channelCount;
3260           mChannelMask = channelMask;
3261           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3262           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3263           // Force underrun condition to avoid false underrun callback until first data is
3264           // written to buffer (other flags are cleared)
3265           mCblk->flags = CBLK_UNDERRUN_ON;
3266           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3267   }
3268}
3269
3270AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3271{
3272    if (mCblk) {
3273        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3274        if (mClient == NULL) {
3275            delete mCblk;
3276        }
3277    }
3278    mCblkMemory.clear();            // and free the shared memory
3279    if (mClient != NULL) {
3280        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3281        mClient.clear();
3282    }
3283}
3284
3285void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3286{
3287    buffer->raw = NULL;
3288    mFrameCount = buffer->frameCount;
3289    step();
3290    buffer->frameCount = 0;
3291}
3292
3293bool AudioFlinger::ThreadBase::TrackBase::step() {
3294    bool result;
3295    audio_track_cblk_t* cblk = this->cblk();
3296
3297    result = cblk->stepServer(mFrameCount);
3298    if (!result) {
3299        ALOGV("stepServer failed acquiring cblk mutex");
3300        mFlags |= STEPSERVER_FAILED;
3301    }
3302    return result;
3303}
3304
3305void AudioFlinger::ThreadBase::TrackBase::reset() {
3306    audio_track_cblk_t* cblk = this->cblk();
3307
3308    cblk->user = 0;
3309    cblk->server = 0;
3310    cblk->userBase = 0;
3311    cblk->serverBase = 0;
3312    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3313    ALOGV("TrackBase::reset");
3314}
3315
3316sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3317{
3318    return mCblkMemory;
3319}
3320
3321int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3322    return (int)mCblk->sampleRate;
3323}
3324
3325int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3326    return (const int)mChannelCount;
3327}
3328
3329uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3330    return mChannelMask;
3331}
3332
3333void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3334    audio_track_cblk_t* cblk = this->cblk();
3335    size_t frameSize = cblk->frameSize;
3336    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
3337    int8_t *bufferEnd = bufferStart + frames * frameSize;
3338
3339    // Check validity of returned pointer in case the track control block would have been corrupted.
3340    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3341        ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) {
3342        ALOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3343                server %d, serverBase %d, user %d, userBase %d",
3344                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3345                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3346        return 0;
3347    }
3348
3349    return bufferStart;
3350}
3351
3352// ----------------------------------------------------------------------------
3353
3354// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3355AudioFlinger::PlaybackThread::Track::Track(
3356            const wp<ThreadBase>& thread,
3357            const sp<Client>& client,
3358            audio_stream_type_t streamType,
3359            uint32_t sampleRate,
3360            uint32_t format,
3361            uint32_t channelMask,
3362            int frameCount,
3363            const sp<IMemory>& sharedBuffer,
3364            int sessionId)
3365    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3366    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3367    mAuxEffectId(0), mHasVolumeController(false)
3368{
3369    if (mCblk != NULL) {
3370        sp<ThreadBase> baseThread = thread.promote();
3371        if (baseThread != 0) {
3372            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3373            mName = playbackThread->getTrackName_l();
3374            mMainBuffer = playbackThread->mixBuffer();
3375        }
3376        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3377        if (mName < 0) {
3378            ALOGE("no more track names available");
3379        }
3380        mVolume[0] = 1.0f;
3381        mVolume[1] = 1.0f;
3382        mStreamType = streamType;
3383        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3384        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3385        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3386    }
3387}
3388
3389AudioFlinger::PlaybackThread::Track::~Track()
3390{
3391    ALOGV("PlaybackThread::Track destructor");
3392    sp<ThreadBase> thread = mThread.promote();
3393    if (thread != 0) {
3394        Mutex::Autolock _l(thread->mLock);
3395        mState = TERMINATED;
3396    }
3397}
3398
3399void AudioFlinger::PlaybackThread::Track::destroy()
3400{
3401    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3402    // by removing it from mTracks vector, so there is a risk that this Tracks's
3403    // desctructor is called. As the destructor needs to lock mLock,
3404    // we must acquire a strong reference on this Track before locking mLock
3405    // here so that the destructor is called only when exiting this function.
3406    // On the other hand, as long as Track::destroy() is only called by
3407    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3408    // this Track with its member mTrack.
3409    sp<Track> keep(this);
3410    { // scope for mLock
3411        sp<ThreadBase> thread = mThread.promote();
3412        if (thread != 0) {
3413            if (!isOutputTrack()) {
3414                if (mState == ACTIVE || mState == RESUMING) {
3415                    AudioSystem::stopOutput(thread->id(),
3416                                            (audio_stream_type_t)mStreamType,
3417                                            mSessionId);
3418
3419                    // to track the speaker usage
3420                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3421                }
3422                AudioSystem::releaseOutput(thread->id());
3423            }
3424            Mutex::Autolock _l(thread->mLock);
3425            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3426            playbackThread->destroyTrack_l(this);
3427        }
3428    }
3429}
3430
3431void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3432{
3433    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",
3434            mName - AudioMixer::TRACK0,
3435            (mClient == NULL) ? getpid() : mClient->pid(),
3436            mStreamType,
3437            mFormat,
3438            mChannelMask,
3439            mSessionId,
3440            mFrameCount,
3441            mState,
3442            mMute,
3443            mFillingUpStatus,
3444            mCblk->sampleRate,
3445            mCblk->volume[0],
3446            mCblk->volume[1],
3447            mCblk->server,
3448            mCblk->user,
3449            (int)mMainBuffer,
3450            (int)mAuxBuffer);
3451}
3452
3453status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3454{
3455     audio_track_cblk_t* cblk = this->cblk();
3456     uint32_t framesReady;
3457     uint32_t framesReq = buffer->frameCount;
3458
3459     // Check if last stepServer failed, try to step now
3460     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3461         if (!step())  goto getNextBuffer_exit;
3462         ALOGV("stepServer recovered");
3463         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3464     }
3465
3466     framesReady = cblk->framesReady();
3467
3468     if (CC_LIKELY(framesReady)) {
3469        uint32_t s = cblk->server;
3470        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3471
3472        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3473        if (framesReq > framesReady) {
3474            framesReq = framesReady;
3475        }
3476        if (s + framesReq > bufferEnd) {
3477            framesReq = bufferEnd - s;
3478        }
3479
3480         buffer->raw = getBuffer(s, framesReq);
3481         if (buffer->raw == NULL) goto getNextBuffer_exit;
3482
3483         buffer->frameCount = framesReq;
3484        return NO_ERROR;
3485     }
3486
3487getNextBuffer_exit:
3488     buffer->raw = NULL;
3489     buffer->frameCount = 0;
3490     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3491     return NOT_ENOUGH_DATA;
3492}
3493
3494bool AudioFlinger::PlaybackThread::Track::isReady() const {
3495    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3496
3497    if (mCblk->framesReady() >= mCblk->frameCount ||
3498            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3499        mFillingUpStatus = FS_FILLED;
3500        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3501        return true;
3502    }
3503    return false;
3504}
3505
3506status_t AudioFlinger::PlaybackThread::Track::start()
3507{
3508    status_t status = NO_ERROR;
3509    ALOGV("start(%d), calling thread %d session %d",
3510            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3511    sp<ThreadBase> thread = mThread.promote();
3512    if (thread != 0) {
3513        Mutex::Autolock _l(thread->mLock);
3514        int state = mState;
3515        // here the track could be either new, or restarted
3516        // in both cases "unstop" the track
3517        if (mState == PAUSED) {
3518            mState = TrackBase::RESUMING;
3519            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3520        } else {
3521            mState = TrackBase::ACTIVE;
3522            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3523        }
3524
3525        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3526            thread->mLock.unlock();
3527            status = AudioSystem::startOutput(thread->id(),
3528                                              (audio_stream_type_t)mStreamType,
3529                                              mSessionId);
3530            thread->mLock.lock();
3531
3532            // to track the speaker usage
3533            if (status == NO_ERROR) {
3534                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3535            }
3536        }
3537        if (status == NO_ERROR) {
3538            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3539            playbackThread->addTrack_l(this);
3540        } else {
3541            mState = state;
3542        }
3543    } else {
3544        status = BAD_VALUE;
3545    }
3546    return status;
3547}
3548
3549void AudioFlinger::PlaybackThread::Track::stop()
3550{
3551    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3552    sp<ThreadBase> thread = mThread.promote();
3553    if (thread != 0) {
3554        Mutex::Autolock _l(thread->mLock);
3555        int state = mState;
3556        if (mState > STOPPED) {
3557            mState = STOPPED;
3558            // If the track is not active (PAUSED and buffers full), flush buffers
3559            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3560            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3561                reset();
3562            }
3563            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3564        }
3565        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
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
3578void AudioFlinger::PlaybackThread::Track::pause()
3579{
3580    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3581    sp<ThreadBase> thread = mThread.promote();
3582    if (thread != 0) {
3583        Mutex::Autolock _l(thread->mLock);
3584        if (mState == ACTIVE || mState == RESUMING) {
3585            mState = PAUSING;
3586            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3587            if (!isOutputTrack()) {
3588                thread->mLock.unlock();
3589                AudioSystem::stopOutput(thread->id(),
3590                                        (audio_stream_type_t)mStreamType,
3591                                        mSessionId);
3592                thread->mLock.lock();
3593
3594                // to track the speaker usage
3595                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3596            }
3597        }
3598    }
3599}
3600
3601void AudioFlinger::PlaybackThread::Track::flush()
3602{
3603    ALOGV("flush(%d)", mName);
3604    sp<ThreadBase> thread = mThread.promote();
3605    if (thread != 0) {
3606        Mutex::Autolock _l(thread->mLock);
3607        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3608            return;
3609        }
3610        // No point remaining in PAUSED state after a flush => go to
3611        // STOPPED state
3612        mState = STOPPED;
3613
3614        // do not reset the track if it is still in the process of being stopped or paused.
3615        // this will be done by prepareTracks_l() when the track is stopped.
3616        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3617        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3618            reset();
3619        }
3620    }
3621}
3622
3623void AudioFlinger::PlaybackThread::Track::reset()
3624{
3625    // Do not reset twice to avoid discarding data written just after a flush and before
3626    // the audioflinger thread detects the track is stopped.
3627    if (!mResetDone) {
3628        TrackBase::reset();
3629        // Force underrun condition to avoid false underrun callback until first data is
3630        // written to buffer
3631        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3632        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3633        mFillingUpStatus = FS_FILLING;
3634        mResetDone = true;
3635    }
3636}
3637
3638void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3639{
3640    mMute = muted;
3641}
3642
3643void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3644{
3645    mVolume[0] = left;
3646    mVolume[1] = right;
3647}
3648
3649status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3650{
3651    status_t status = DEAD_OBJECT;
3652    sp<ThreadBase> thread = mThread.promote();
3653    if (thread != 0) {
3654       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3655       status = playbackThread->attachAuxEffect(this, EffectId);
3656    }
3657    return status;
3658}
3659
3660void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3661{
3662    mAuxEffectId = EffectId;
3663    mAuxBuffer = buffer;
3664}
3665
3666// ----------------------------------------------------------------------------
3667
3668// RecordTrack constructor must be called with AudioFlinger::mLock held
3669AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3670            const wp<ThreadBase>& thread,
3671            const sp<Client>& client,
3672            uint32_t sampleRate,
3673            uint32_t format,
3674            uint32_t channelMask,
3675            int frameCount,
3676            uint32_t flags,
3677            int sessionId)
3678    :   TrackBase(thread, client, sampleRate, format,
3679                  channelMask, frameCount, flags, 0, sessionId),
3680        mOverflow(false)
3681{
3682    if (mCblk != NULL) {
3683       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3684       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3685           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3686       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3687           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3688       } else {
3689           mCblk->frameSize = sizeof(int8_t);
3690       }
3691    }
3692}
3693
3694AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3695{
3696    sp<ThreadBase> thread = mThread.promote();
3697    if (thread != 0) {
3698        AudioSystem::releaseInput(thread->id());
3699    }
3700}
3701
3702status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3703{
3704    audio_track_cblk_t* cblk = this->cblk();
3705    uint32_t framesAvail;
3706    uint32_t framesReq = buffer->frameCount;
3707
3708     // Check if last stepServer failed, try to step now
3709    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3710        if (!step()) goto getNextBuffer_exit;
3711        ALOGV("stepServer recovered");
3712        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3713    }
3714
3715    framesAvail = cblk->framesAvailable_l();
3716
3717    if (CC_LIKELY(framesAvail)) {
3718        uint32_t s = cblk->server;
3719        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3720
3721        if (framesReq > framesAvail) {
3722            framesReq = framesAvail;
3723        }
3724        if (s + framesReq > bufferEnd) {
3725            framesReq = bufferEnd - s;
3726        }
3727
3728        buffer->raw = getBuffer(s, framesReq);
3729        if (buffer->raw == NULL) goto getNextBuffer_exit;
3730
3731        buffer->frameCount = framesReq;
3732        return NO_ERROR;
3733    }
3734
3735getNextBuffer_exit:
3736    buffer->raw = NULL;
3737    buffer->frameCount = 0;
3738    return NOT_ENOUGH_DATA;
3739}
3740
3741status_t AudioFlinger::RecordThread::RecordTrack::start()
3742{
3743    sp<ThreadBase> thread = mThread.promote();
3744    if (thread != 0) {
3745        RecordThread *recordThread = (RecordThread *)thread.get();
3746        return recordThread->start(this);
3747    } else {
3748        return BAD_VALUE;
3749    }
3750}
3751
3752void AudioFlinger::RecordThread::RecordTrack::stop()
3753{
3754    sp<ThreadBase> thread = mThread.promote();
3755    if (thread != 0) {
3756        RecordThread *recordThread = (RecordThread *)thread.get();
3757        recordThread->stop(this);
3758        TrackBase::reset();
3759        // Force overerrun condition to avoid false overrun callback until first data is
3760        // read from buffer
3761        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3762    }
3763}
3764
3765void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3766{
3767    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3768            (mClient == NULL) ? getpid() : mClient->pid(),
3769            mFormat,
3770            mChannelMask,
3771            mSessionId,
3772            mFrameCount,
3773            mState,
3774            mCblk->sampleRate,
3775            mCblk->server,
3776            mCblk->user);
3777}
3778
3779
3780// ----------------------------------------------------------------------------
3781
3782AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3783            const wp<ThreadBase>& thread,
3784            DuplicatingThread *sourceThread,
3785            uint32_t sampleRate,
3786            uint32_t format,
3787            uint32_t channelMask,
3788            int frameCount)
3789    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3790    mActive(false), mSourceThread(sourceThread)
3791{
3792
3793    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3794    if (mCblk != NULL) {
3795        mCblk->flags |= CBLK_DIRECTION_OUT;
3796        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3797        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
3798        mOutBuffer.frameCount = 0;
3799        playbackThread->mTracks.add(this);
3800        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3801                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3802                mCblk, mBuffer, mCblk->buffers,
3803                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3804    } else {
3805        ALOGW("Error creating output track on thread %p", playbackThread);
3806    }
3807}
3808
3809AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3810{
3811    clearBufferQueue();
3812}
3813
3814status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3815{
3816    status_t status = Track::start();
3817    if (status != NO_ERROR) {
3818        return status;
3819    }
3820
3821    mActive = true;
3822    mRetryCount = 127;
3823    return status;
3824}
3825
3826void AudioFlinger::PlaybackThread::OutputTrack::stop()
3827{
3828    Track::stop();
3829    clearBufferQueue();
3830    mOutBuffer.frameCount = 0;
3831    mActive = false;
3832}
3833
3834bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3835{
3836    Buffer *pInBuffer;
3837    Buffer inBuffer;
3838    uint32_t channelCount = mChannelCount;
3839    bool outputBufferFull = false;
3840    inBuffer.frameCount = frames;
3841    inBuffer.i16 = data;
3842
3843    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3844
3845    if (!mActive && frames != 0) {
3846        start();
3847        sp<ThreadBase> thread = mThread.promote();
3848        if (thread != 0) {
3849            MixerThread *mixerThread = (MixerThread *)thread.get();
3850            if (mCblk->frameCount > frames){
3851                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3852                    uint32_t startFrames = (mCblk->frameCount - frames);
3853                    pInBuffer = new Buffer;
3854                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3855                    pInBuffer->frameCount = startFrames;
3856                    pInBuffer->i16 = pInBuffer->mBuffer;
3857                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3858                    mBufferQueue.add(pInBuffer);
3859                } else {
3860                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
3861                }
3862            }
3863        }
3864    }
3865
3866    while (waitTimeLeftMs) {
3867        // First write pending buffers, then new data
3868        if (mBufferQueue.size()) {
3869            pInBuffer = mBufferQueue.itemAt(0);
3870        } else {
3871            pInBuffer = &inBuffer;
3872        }
3873
3874        if (pInBuffer->frameCount == 0) {
3875            break;
3876        }
3877
3878        if (mOutBuffer.frameCount == 0) {
3879            mOutBuffer.frameCount = pInBuffer->frameCount;
3880            nsecs_t startTime = systemTime();
3881            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
3882                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3883                outputBufferFull = true;
3884                break;
3885            }
3886            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3887            if (waitTimeLeftMs >= waitTimeMs) {
3888                waitTimeLeftMs -= waitTimeMs;
3889            } else {
3890                waitTimeLeftMs = 0;
3891            }
3892        }
3893
3894        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3895        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3896        mCblk->stepUser(outFrames);
3897        pInBuffer->frameCount -= outFrames;
3898        pInBuffer->i16 += outFrames * channelCount;
3899        mOutBuffer.frameCount -= outFrames;
3900        mOutBuffer.i16 += outFrames * channelCount;
3901
3902        if (pInBuffer->frameCount == 0) {
3903            if (mBufferQueue.size()) {
3904                mBufferQueue.removeAt(0);
3905                delete [] pInBuffer->mBuffer;
3906                delete pInBuffer;
3907                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3908            } else {
3909                break;
3910            }
3911        }
3912    }
3913
3914    // If we could not write all frames, allocate a buffer and queue it for next time.
3915    if (inBuffer.frameCount) {
3916        sp<ThreadBase> thread = mThread.promote();
3917        if (thread != 0 && !thread->standby()) {
3918            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3919                pInBuffer = new Buffer;
3920                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3921                pInBuffer->frameCount = inBuffer.frameCount;
3922                pInBuffer->i16 = pInBuffer->mBuffer;
3923                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3924                mBufferQueue.add(pInBuffer);
3925                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3926            } else {
3927                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3928            }
3929        }
3930    }
3931
3932    // Calling write() with a 0 length buffer, means that no more data will be written:
3933    // If no more buffers are pending, fill output track buffer to make sure it is started
3934    // by output mixer.
3935    if (frames == 0 && mBufferQueue.size() == 0) {
3936        if (mCblk->user < mCblk->frameCount) {
3937            frames = mCblk->frameCount - mCblk->user;
3938            pInBuffer = new Buffer;
3939            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3940            pInBuffer->frameCount = frames;
3941            pInBuffer->i16 = pInBuffer->mBuffer;
3942            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3943            mBufferQueue.add(pInBuffer);
3944        } else if (mActive) {
3945            stop();
3946        }
3947    }
3948
3949    return outputBufferFull;
3950}
3951
3952status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3953{
3954    int active;
3955    status_t result;
3956    audio_track_cblk_t* cblk = mCblk;
3957    uint32_t framesReq = buffer->frameCount;
3958
3959//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3960    buffer->frameCount  = 0;
3961
3962    uint32_t framesAvail = cblk->framesAvailable();
3963
3964
3965    if (framesAvail == 0) {
3966        Mutex::Autolock _l(cblk->lock);
3967        goto start_loop_here;
3968        while (framesAvail == 0) {
3969            active = mActive;
3970            if (CC_UNLIKELY(!active)) {
3971                ALOGV("Not active and NO_MORE_BUFFERS");
3972                return AudioTrack::NO_MORE_BUFFERS;
3973            }
3974            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
3975            if (result != NO_ERROR) {
3976                return AudioTrack::NO_MORE_BUFFERS;
3977            }
3978            // read the server count again
3979        start_loop_here:
3980            framesAvail = cblk->framesAvailable_l();
3981        }
3982    }
3983
3984//    if (framesAvail < framesReq) {
3985//        return AudioTrack::NO_MORE_BUFFERS;
3986//    }
3987
3988    if (framesReq > framesAvail) {
3989        framesReq = framesAvail;
3990    }
3991
3992    uint32_t u = cblk->user;
3993    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
3994
3995    if (u + framesReq > bufferEnd) {
3996        framesReq = bufferEnd - u;
3997    }
3998
3999    buffer->frameCount  = framesReq;
4000    buffer->raw         = (void *)cblk->buffer(u);
4001    return NO_ERROR;
4002}
4003
4004
4005void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
4006{
4007    size_t size = mBufferQueue.size();
4008    Buffer *pBuffer;
4009
4010    for (size_t i = 0; i < size; i++) {
4011        pBuffer = mBufferQueue.itemAt(i);
4012        delete [] pBuffer->mBuffer;
4013        delete pBuffer;
4014    }
4015    mBufferQueue.clear();
4016}
4017
4018// ----------------------------------------------------------------------------
4019
4020AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
4021    :   RefBase(),
4022        mAudioFlinger(audioFlinger),
4023        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
4024        mPid(pid)
4025{
4026    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
4027}
4028
4029// Client destructor must be called with AudioFlinger::mLock held
4030AudioFlinger::Client::~Client()
4031{
4032    mAudioFlinger->removeClient_l(mPid);
4033}
4034
4035const sp<MemoryDealer>& AudioFlinger::Client::heap() const
4036{
4037    return mMemoryDealer;
4038}
4039
4040// ----------------------------------------------------------------------------
4041
4042AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4043                                                     const sp<IAudioFlingerClient>& client,
4044                                                     pid_t pid)
4045    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
4046{
4047}
4048
4049AudioFlinger::NotificationClient::~NotificationClient()
4050{
4051    mClient.clear();
4052}
4053
4054void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4055{
4056    sp<NotificationClient> keep(this);
4057    {
4058        mAudioFlinger->removeNotificationClient(mPid);
4059    }
4060}
4061
4062// ----------------------------------------------------------------------------
4063
4064AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4065    : BnAudioTrack(),
4066      mTrack(track)
4067{
4068}
4069
4070AudioFlinger::TrackHandle::~TrackHandle() {
4071    // just stop the track on deletion, associated resources
4072    // will be freed from the main thread once all pending buffers have
4073    // been played. Unless it's not in the active track list, in which
4074    // case we free everything now...
4075    mTrack->destroy();
4076}
4077
4078status_t AudioFlinger::TrackHandle::start() {
4079    return mTrack->start();
4080}
4081
4082void AudioFlinger::TrackHandle::stop() {
4083    mTrack->stop();
4084}
4085
4086void AudioFlinger::TrackHandle::flush() {
4087    mTrack->flush();
4088}
4089
4090void AudioFlinger::TrackHandle::mute(bool e) {
4091    mTrack->mute(e);
4092}
4093
4094void AudioFlinger::TrackHandle::pause() {
4095    mTrack->pause();
4096}
4097
4098void AudioFlinger::TrackHandle::setVolume(float left, float right) {
4099    mTrack->setVolume(left, right);
4100}
4101
4102sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4103    return mTrack->getCblk();
4104}
4105
4106status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4107{
4108    return mTrack->attachAuxEffect(EffectId);
4109}
4110
4111status_t AudioFlinger::TrackHandle::onTransact(
4112    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4113{
4114    return BnAudioTrack::onTransact(code, data, reply, flags);
4115}
4116
4117// ----------------------------------------------------------------------------
4118
4119sp<IAudioRecord> AudioFlinger::openRecord(
4120        pid_t pid,
4121        int input,
4122        uint32_t sampleRate,
4123        uint32_t format,
4124        uint32_t channelMask,
4125        int frameCount,
4126        uint32_t flags,
4127        int *sessionId,
4128        status_t *status)
4129{
4130    sp<RecordThread::RecordTrack> recordTrack;
4131    sp<RecordHandle> recordHandle;
4132    sp<Client> client;
4133    wp<Client> wclient;
4134    status_t lStatus;
4135    RecordThread *thread;
4136    size_t inFrameCount;
4137    int lSessionId;
4138
4139    // check calling permissions
4140    if (!recordingAllowed()) {
4141        lStatus = PERMISSION_DENIED;
4142        goto Exit;
4143    }
4144
4145    // add client to list
4146    { // scope for mLock
4147        Mutex::Autolock _l(mLock);
4148        thread = checkRecordThread_l(input);
4149        if (thread == NULL) {
4150            lStatus = BAD_VALUE;
4151            goto Exit;
4152        }
4153
4154        wclient = mClients.valueFor(pid);
4155        if (wclient != NULL) {
4156            client = wclient.promote();
4157        } else {
4158            client = new Client(this, pid);
4159            mClients.add(pid, client);
4160        }
4161
4162        // If no audio session id is provided, create one here
4163        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4164            lSessionId = *sessionId;
4165        } else {
4166            lSessionId = nextUniqueId();
4167            if (sessionId != NULL) {
4168                *sessionId = lSessionId;
4169            }
4170        }
4171        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4172        recordTrack = thread->createRecordTrack_l(client,
4173                                                sampleRate,
4174                                                format,
4175                                                channelMask,
4176                                                frameCount,
4177                                                flags,
4178                                                lSessionId,
4179                                                &lStatus);
4180    }
4181    if (lStatus != NO_ERROR) {
4182        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4183        // destructor is called by the TrackBase destructor with mLock held
4184        client.clear();
4185        recordTrack.clear();
4186        goto Exit;
4187    }
4188
4189    // return to handle to client
4190    recordHandle = new RecordHandle(recordTrack);
4191    lStatus = NO_ERROR;
4192
4193Exit:
4194    if (status) {
4195        *status = lStatus;
4196    }
4197    return recordHandle;
4198}
4199
4200// ----------------------------------------------------------------------------
4201
4202AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4203    : BnAudioRecord(),
4204    mRecordTrack(recordTrack)
4205{
4206}
4207
4208AudioFlinger::RecordHandle::~RecordHandle() {
4209    stop();
4210}
4211
4212status_t AudioFlinger::RecordHandle::start() {
4213    ALOGV("RecordHandle::start()");
4214    return mRecordTrack->start();
4215}
4216
4217void AudioFlinger::RecordHandle::stop() {
4218    ALOGV("RecordHandle::stop()");
4219    mRecordTrack->stop();
4220}
4221
4222sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4223    return mRecordTrack->getCblk();
4224}
4225
4226status_t AudioFlinger::RecordHandle::onTransact(
4227    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4228{
4229    return BnAudioRecord::onTransact(code, data, reply, flags);
4230}
4231
4232// ----------------------------------------------------------------------------
4233
4234AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4235                                         AudioStreamIn *input,
4236                                         uint32_t sampleRate,
4237                                         uint32_t channels,
4238                                         int id,
4239                                         uint32_t device) :
4240    ThreadBase(audioFlinger, id, device),
4241    mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
4242{
4243    mType = ThreadBase::RECORD;
4244
4245    snprintf(mName, kNameLength, "AudioIn_%d", id);
4246
4247    mReqChannelCount = popcount(channels);
4248    mReqSampleRate = sampleRate;
4249    readInputParameters();
4250}
4251
4252
4253AudioFlinger::RecordThread::~RecordThread()
4254{
4255    delete[] mRsmpInBuffer;
4256    if (mResampler != NULL) {
4257        delete mResampler;
4258        delete[] mRsmpOutBuffer;
4259    }
4260}
4261
4262void AudioFlinger::RecordThread::onFirstRef()
4263{
4264    run(mName, PRIORITY_URGENT_AUDIO);
4265}
4266
4267status_t AudioFlinger::RecordThread::readyToRun()
4268{
4269    status_t status = initCheck();
4270    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4271    return status;
4272}
4273
4274bool AudioFlinger::RecordThread::threadLoop()
4275{
4276    AudioBufferProvider::Buffer buffer;
4277    sp<RecordTrack> activeTrack;
4278    Vector< sp<EffectChain> > effectChains;
4279
4280    nsecs_t lastWarning = 0;
4281
4282    acquireWakeLock();
4283
4284    // start recording
4285    while (!exitPending()) {
4286
4287        processConfigEvents();
4288
4289        { // scope for mLock
4290            Mutex::Autolock _l(mLock);
4291            checkForNewParameters_l();
4292            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4293                if (!mStandby) {
4294                    mInput->stream->common.standby(&mInput->stream->common);
4295                    mStandby = true;
4296                }
4297
4298                if (exitPending()) break;
4299
4300                releaseWakeLock_l();
4301                ALOGV("RecordThread: loop stopping");
4302                // go to sleep
4303                mWaitWorkCV.wait(mLock);
4304                ALOGV("RecordThread: loop starting");
4305                acquireWakeLock_l();
4306                continue;
4307            }
4308            if (mActiveTrack != 0) {
4309                if (mActiveTrack->mState == TrackBase::PAUSING) {
4310                    if (!mStandby) {
4311                        mInput->stream->common.standby(&mInput->stream->common);
4312                        mStandby = true;
4313                    }
4314                    mActiveTrack.clear();
4315                    mStartStopCond.broadcast();
4316                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4317                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4318                        mActiveTrack.clear();
4319                        mStartStopCond.broadcast();
4320                    } else if (mBytesRead != 0) {
4321                        // record start succeeds only if first read from audio input
4322                        // succeeds
4323                        if (mBytesRead > 0) {
4324                            mActiveTrack->mState = TrackBase::ACTIVE;
4325                        } else {
4326                            mActiveTrack.clear();
4327                        }
4328                        mStartStopCond.broadcast();
4329                    }
4330                    mStandby = false;
4331                }
4332            }
4333            lockEffectChains_l(effectChains);
4334        }
4335
4336        if (mActiveTrack != 0) {
4337            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4338                mActiveTrack->mState != TrackBase::RESUMING) {
4339                unlockEffectChains(effectChains);
4340                usleep(kRecordThreadSleepUs);
4341                continue;
4342            }
4343            for (size_t i = 0; i < effectChains.size(); i ++) {
4344                effectChains[i]->process_l();
4345            }
4346
4347            buffer.frameCount = mFrameCount;
4348            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4349                size_t framesOut = buffer.frameCount;
4350                if (mResampler == NULL) {
4351                    // no resampling
4352                    while (framesOut) {
4353                        size_t framesIn = mFrameCount - mRsmpInIndex;
4354                        if (framesIn) {
4355                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4356                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4357                            if (framesIn > framesOut)
4358                                framesIn = framesOut;
4359                            mRsmpInIndex += framesIn;
4360                            framesOut -= framesIn;
4361                            if ((int)mChannelCount == mReqChannelCount ||
4362                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4363                                memcpy(dst, src, framesIn * mFrameSize);
4364                            } else {
4365                                int16_t *src16 = (int16_t *)src;
4366                                int16_t *dst16 = (int16_t *)dst;
4367                                if (mChannelCount == 1) {
4368                                    while (framesIn--) {
4369                                        *dst16++ = *src16;
4370                                        *dst16++ = *src16++;
4371                                    }
4372                                } else {
4373                                    while (framesIn--) {
4374                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4375                                        src16 += 2;
4376                                    }
4377                                }
4378                            }
4379                        }
4380                        if (framesOut && mFrameCount == mRsmpInIndex) {
4381                            if (framesOut == mFrameCount &&
4382                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4383                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4384                                framesOut = 0;
4385                            } else {
4386                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4387                                mRsmpInIndex = 0;
4388                            }
4389                            if (mBytesRead < 0) {
4390                                ALOGE("Error reading audio input");
4391                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4392                                    // Force input into standby so that it tries to
4393                                    // recover at next read attempt
4394                                    mInput->stream->common.standby(&mInput->stream->common);
4395                                    usleep(kRecordThreadSleepUs);
4396                                }
4397                                mRsmpInIndex = mFrameCount;
4398                                framesOut = 0;
4399                                buffer.frameCount = 0;
4400                            }
4401                        }
4402                    }
4403                } else {
4404                    // resampling
4405
4406                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4407                    // alter output frame count as if we were expecting stereo samples
4408                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4409                        framesOut >>= 1;
4410                    }
4411                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4412                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4413                    // are 32 bit aligned which should be always true.
4414                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4415                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4416                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4417                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4418                        int16_t *dst = buffer.i16;
4419                        while (framesOut--) {
4420                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4421                            src += 2;
4422                        }
4423                    } else {
4424                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4425                    }
4426
4427                }
4428                mActiveTrack->releaseBuffer(&buffer);
4429                mActiveTrack->overflow();
4430            }
4431            // client isn't retrieving buffers fast enough
4432            else {
4433                if (!mActiveTrack->setOverflow()) {
4434                    nsecs_t now = systemTime();
4435                    if ((now - lastWarning) > kWarningThrottleNs) {
4436                        ALOGW("RecordThread: buffer overflow");
4437                        lastWarning = now;
4438                    }
4439                }
4440                // Release the processor for a while before asking for a new buffer.
4441                // This will give the application more chance to read from the buffer and
4442                // clear the overflow.
4443                usleep(kRecordThreadSleepUs);
4444            }
4445        }
4446        // enable changes in effect chain
4447        unlockEffectChains(effectChains);
4448        effectChains.clear();
4449    }
4450
4451    if (!mStandby) {
4452        mInput->stream->common.standby(&mInput->stream->common);
4453    }
4454    mActiveTrack.clear();
4455
4456    mStartStopCond.broadcast();
4457
4458    releaseWakeLock();
4459
4460    ALOGV("RecordThread %p exiting", this);
4461    return false;
4462}
4463
4464
4465sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4466        const sp<AudioFlinger::Client>& client,
4467        uint32_t sampleRate,
4468        int format,
4469        int channelMask,
4470        int frameCount,
4471        uint32_t flags,
4472        int sessionId,
4473        status_t *status)
4474{
4475    sp<RecordTrack> track;
4476    status_t lStatus;
4477
4478    lStatus = initCheck();
4479    if (lStatus != NO_ERROR) {
4480        ALOGE("Audio driver not initialized.");
4481        goto Exit;
4482    }
4483
4484    { // scope for mLock
4485        Mutex::Autolock _l(mLock);
4486
4487        track = new RecordTrack(this, client, sampleRate,
4488                      format, channelMask, frameCount, flags, sessionId);
4489
4490        if (track->getCblk() == NULL) {
4491            lStatus = NO_MEMORY;
4492            goto Exit;
4493        }
4494
4495        mTrack = track.get();
4496        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4497        bool suspend = audio_is_bluetooth_sco_device(
4498                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4499        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4500        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4501    }
4502    lStatus = NO_ERROR;
4503
4504Exit:
4505    if (status) {
4506        *status = lStatus;
4507    }
4508    return track;
4509}
4510
4511status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4512{
4513    ALOGV("RecordThread::start");
4514    sp <ThreadBase> strongMe = this;
4515    status_t status = NO_ERROR;
4516    {
4517        AutoMutex lock(mLock);
4518        if (mActiveTrack != 0) {
4519            if (recordTrack != mActiveTrack.get()) {
4520                status = -EBUSY;
4521            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4522                mActiveTrack->mState = TrackBase::ACTIVE;
4523            }
4524            return status;
4525        }
4526
4527        recordTrack->mState = TrackBase::IDLE;
4528        mActiveTrack = recordTrack;
4529        mLock.unlock();
4530        status_t status = AudioSystem::startInput(mId);
4531        mLock.lock();
4532        if (status != NO_ERROR) {
4533            mActiveTrack.clear();
4534            return status;
4535        }
4536        mRsmpInIndex = mFrameCount;
4537        mBytesRead = 0;
4538        if (mResampler != NULL) {
4539            mResampler->reset();
4540        }
4541        mActiveTrack->mState = TrackBase::RESUMING;
4542        // signal thread to start
4543        ALOGV("Signal record thread");
4544        mWaitWorkCV.signal();
4545        // do not wait for mStartStopCond if exiting
4546        if (mExiting) {
4547            mActiveTrack.clear();
4548            status = INVALID_OPERATION;
4549            goto startError;
4550        }
4551        mStartStopCond.wait(mLock);
4552        if (mActiveTrack == 0) {
4553            ALOGV("Record failed to start");
4554            status = BAD_VALUE;
4555            goto startError;
4556        }
4557        ALOGV("Record started OK");
4558        return status;
4559    }
4560startError:
4561    AudioSystem::stopInput(mId);
4562    return status;
4563}
4564
4565void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4566    ALOGV("RecordThread::stop");
4567    sp <ThreadBase> strongMe = this;
4568    {
4569        AutoMutex lock(mLock);
4570        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4571            mActiveTrack->mState = TrackBase::PAUSING;
4572            // do not wait for mStartStopCond if exiting
4573            if (mExiting) {
4574                return;
4575            }
4576            mStartStopCond.wait(mLock);
4577            // if we have been restarted, recordTrack == mActiveTrack.get() here
4578            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4579                mLock.unlock();
4580                AudioSystem::stopInput(mId);
4581                mLock.lock();
4582                ALOGV("Record stopped OK");
4583            }
4584        }
4585    }
4586}
4587
4588status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4589{
4590    const size_t SIZE = 256;
4591    char buffer[SIZE];
4592    String8 result;
4593    pid_t pid = 0;
4594
4595    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4596    result.append(buffer);
4597
4598    if (mActiveTrack != 0) {
4599        result.append("Active Track:\n");
4600        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4601        mActiveTrack->dump(buffer, SIZE);
4602        result.append(buffer);
4603
4604        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4605        result.append(buffer);
4606        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4607        result.append(buffer);
4608        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4609        result.append(buffer);
4610        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4611        result.append(buffer);
4612        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4613        result.append(buffer);
4614
4615
4616    } else {
4617        result.append("No record client\n");
4618    }
4619    write(fd, result.string(), result.size());
4620
4621    dumpBase(fd, args);
4622    dumpEffectChains(fd, args);
4623
4624    return NO_ERROR;
4625}
4626
4627status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4628{
4629    size_t framesReq = buffer->frameCount;
4630    size_t framesReady = mFrameCount - mRsmpInIndex;
4631    int channelCount;
4632
4633    if (framesReady == 0) {
4634        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4635        if (mBytesRead < 0) {
4636            ALOGE("RecordThread::getNextBuffer() Error reading audio input");
4637            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4638                // Force input into standby so that it tries to
4639                // recover at next read attempt
4640                mInput->stream->common.standby(&mInput->stream->common);
4641                usleep(kRecordThreadSleepUs);
4642            }
4643            buffer->raw = NULL;
4644            buffer->frameCount = 0;
4645            return NOT_ENOUGH_DATA;
4646        }
4647        mRsmpInIndex = 0;
4648        framesReady = mFrameCount;
4649    }
4650
4651    if (framesReq > framesReady) {
4652        framesReq = framesReady;
4653    }
4654
4655    if (mChannelCount == 1 && mReqChannelCount == 2) {
4656        channelCount = 1;
4657    } else {
4658        channelCount = 2;
4659    }
4660    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4661    buffer->frameCount = framesReq;
4662    return NO_ERROR;
4663}
4664
4665void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4666{
4667    mRsmpInIndex += buffer->frameCount;
4668    buffer->frameCount = 0;
4669}
4670
4671bool AudioFlinger::RecordThread::checkForNewParameters_l()
4672{
4673    bool reconfig = false;
4674
4675    while (!mNewParameters.isEmpty()) {
4676        status_t status = NO_ERROR;
4677        String8 keyValuePair = mNewParameters[0];
4678        AudioParameter param = AudioParameter(keyValuePair);
4679        int value;
4680        int reqFormat = mFormat;
4681        int reqSamplingRate = mReqSampleRate;
4682        int reqChannelCount = mReqChannelCount;
4683
4684        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4685            reqSamplingRate = value;
4686            reconfig = true;
4687        }
4688        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4689            reqFormat = value;
4690            reconfig = true;
4691        }
4692        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4693            reqChannelCount = popcount(value);
4694            reconfig = true;
4695        }
4696        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4697            // do not accept frame count changes if tracks are open as the track buffer
4698            // size depends on frame count and correct behavior would not be garantied
4699            // if frame count is changed after track creation
4700            if (mActiveTrack != 0) {
4701                status = INVALID_OPERATION;
4702            } else {
4703                reconfig = true;
4704            }
4705        }
4706        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4707            // forward device change to effects that have requested to be
4708            // aware of attached audio device.
4709            for (size_t i = 0; i < mEffectChains.size(); i++) {
4710                mEffectChains[i]->setDevice_l(value);
4711            }
4712            // store input device and output device but do not forward output device to audio HAL.
4713            // Note that status is ignored by the caller for output device
4714            // (see AudioFlinger::setParameters()
4715            if (value & AUDIO_DEVICE_OUT_ALL) {
4716                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4717                status = BAD_VALUE;
4718            } else {
4719                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4720                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4721                if (mTrack != NULL) {
4722                    bool suspend = audio_is_bluetooth_sco_device(
4723                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4724                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4725                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4726                }
4727            }
4728            mDevice |= (uint32_t)value;
4729        }
4730        if (status == NO_ERROR) {
4731            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4732            if (status == INVALID_OPERATION) {
4733               mInput->stream->common.standby(&mInput->stream->common);
4734               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4735            }
4736            if (reconfig) {
4737                if (status == BAD_VALUE &&
4738                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4739                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4740                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4741                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4742                    (reqChannelCount < 3)) {
4743                    status = NO_ERROR;
4744                }
4745                if (status == NO_ERROR) {
4746                    readInputParameters();
4747                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4748                }
4749            }
4750        }
4751
4752        mNewParameters.removeAt(0);
4753
4754        mParamStatus = status;
4755        mParamCond.signal();
4756        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4757        // already timed out waiting for the status and will never signal the condition.
4758        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4759    }
4760    return reconfig;
4761}
4762
4763String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4764{
4765    char *s;
4766    String8 out_s8 = String8();
4767
4768    Mutex::Autolock _l(mLock);
4769    if (initCheck() != NO_ERROR) {
4770        return out_s8;
4771    }
4772
4773    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4774    out_s8 = String8(s);
4775    free(s);
4776    return out_s8;
4777}
4778
4779void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4780    AudioSystem::OutputDescriptor desc;
4781    void *param2 = 0;
4782
4783    switch (event) {
4784    case AudioSystem::INPUT_OPENED:
4785    case AudioSystem::INPUT_CONFIG_CHANGED:
4786        desc.channels = mChannelMask;
4787        desc.samplingRate = mSampleRate;
4788        desc.format = mFormat;
4789        desc.frameCount = mFrameCount;
4790        desc.latency = 0;
4791        param2 = &desc;
4792        break;
4793
4794    case AudioSystem::INPUT_CLOSED:
4795    default:
4796        break;
4797    }
4798    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4799}
4800
4801void AudioFlinger::RecordThread::readInputParameters()
4802{
4803    if (mRsmpInBuffer) delete mRsmpInBuffer;
4804    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4805    if (mResampler) delete mResampler;
4806    mResampler = NULL;
4807
4808    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4809    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4810    mChannelCount = (uint16_t)popcount(mChannelMask);
4811    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4812    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
4813    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4814    mFrameCount = mInputBytes / mFrameSize;
4815    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4816
4817    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4818    {
4819        int channelCount;
4820         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4821         // stereo to mono post process as the resampler always outputs stereo.
4822        if (mChannelCount == 1 && mReqChannelCount == 2) {
4823            channelCount = 1;
4824        } else {
4825            channelCount = 2;
4826        }
4827        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4828        mResampler->setSampleRate(mSampleRate);
4829        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4830        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4831
4832        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4833        if (mChannelCount == 1 && mReqChannelCount == 1) {
4834            mFrameCount >>= 1;
4835        }
4836
4837    }
4838    mRsmpInIndex = mFrameCount;
4839}
4840
4841unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4842{
4843    Mutex::Autolock _l(mLock);
4844    if (initCheck() != NO_ERROR) {
4845        return 0;
4846    }
4847
4848    return mInput->stream->get_input_frames_lost(mInput->stream);
4849}
4850
4851uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4852{
4853    Mutex::Autolock _l(mLock);
4854    uint32_t result = 0;
4855    if (getEffectChain_l(sessionId) != 0) {
4856        result = EFFECT_SESSION;
4857    }
4858
4859    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4860        result |= TRACK_SESSION;
4861    }
4862
4863    return result;
4864}
4865
4866AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4867{
4868    Mutex::Autolock _l(mLock);
4869    return mTrack;
4870}
4871
4872AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
4873{
4874    Mutex::Autolock _l(mLock);
4875    return mInput;
4876}
4877
4878AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4879{
4880    Mutex::Autolock _l(mLock);
4881    AudioStreamIn *input = mInput;
4882    mInput = NULL;
4883    return input;
4884}
4885
4886// this method must always be called either with ThreadBase mLock held or inside the thread loop
4887audio_stream_t* AudioFlinger::RecordThread::stream()
4888{
4889    if (mInput == NULL) {
4890        return NULL;
4891    }
4892    return &mInput->stream->common;
4893}
4894
4895
4896// ----------------------------------------------------------------------------
4897
4898int AudioFlinger::openOutput(uint32_t *pDevices,
4899                                uint32_t *pSamplingRate,
4900                                uint32_t *pFormat,
4901                                uint32_t *pChannels,
4902                                uint32_t *pLatencyMs,
4903                                uint32_t flags)
4904{
4905    status_t status;
4906    PlaybackThread *thread = NULL;
4907    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4908    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4909    uint32_t format = pFormat ? *pFormat : 0;
4910    uint32_t channels = pChannels ? *pChannels : 0;
4911    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4912    audio_stream_out_t *outStream;
4913    audio_hw_device_t *outHwDev;
4914
4915    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4916            pDevices ? *pDevices : 0,
4917            samplingRate,
4918            format,
4919            channels,
4920            flags);
4921
4922    if (pDevices == NULL || *pDevices == 0) {
4923        return 0;
4924    }
4925
4926    Mutex::Autolock _l(mLock);
4927
4928    outHwDev = findSuitableHwDev_l(*pDevices);
4929    if (outHwDev == NULL)
4930        return 0;
4931
4932    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
4933                                          &channels, &samplingRate, &outStream);
4934    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4935            outStream,
4936            samplingRate,
4937            format,
4938            channels,
4939            status);
4940
4941    mHardwareStatus = AUDIO_HW_IDLE;
4942    if (outStream != NULL) {
4943        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4944        int id = nextUniqueId();
4945
4946        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4947            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4948            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4949            thread = new DirectOutputThread(this, output, id, *pDevices);
4950            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4951        } else {
4952            thread = new MixerThread(this, output, id, *pDevices);
4953            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4954        }
4955        mPlaybackThreads.add(id, thread);
4956
4957        if (pSamplingRate) *pSamplingRate = samplingRate;
4958        if (pFormat) *pFormat = format;
4959        if (pChannels) *pChannels = channels;
4960        if (pLatencyMs) *pLatencyMs = thread->latency();
4961
4962        // notify client processes of the new output creation
4963        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4964        return id;
4965    }
4966
4967    return 0;
4968}
4969
4970int AudioFlinger::openDuplicateOutput(int output1, int output2)
4971{
4972    Mutex::Autolock _l(mLock);
4973    MixerThread *thread1 = checkMixerThread_l(output1);
4974    MixerThread *thread2 = checkMixerThread_l(output2);
4975
4976    if (thread1 == NULL || thread2 == NULL) {
4977        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
4978        return 0;
4979    }
4980
4981    int id = nextUniqueId();
4982    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
4983    thread->addOutputTrack(thread2);
4984    mPlaybackThreads.add(id, thread);
4985    // notify client processes of the new output creation
4986    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4987    return id;
4988}
4989
4990status_t AudioFlinger::closeOutput(int output)
4991{
4992    // keep strong reference on the playback thread so that
4993    // it is not destroyed while exit() is executed
4994    sp <PlaybackThread> thread;
4995    {
4996        Mutex::Autolock _l(mLock);
4997        thread = checkPlaybackThread_l(output);
4998        if (thread == NULL) {
4999            return BAD_VALUE;
5000        }
5001
5002        ALOGV("closeOutput() %d", output);
5003
5004        if (thread->type() == ThreadBase::MIXER) {
5005            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5006                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
5007                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
5008                    dupThread->removeOutputTrack((MixerThread *)thread.get());
5009                }
5010            }
5011        }
5012        void *param2 = 0;
5013        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
5014        mPlaybackThreads.removeItem(output);
5015    }
5016    thread->exit();
5017
5018    if (thread->type() != ThreadBase::DUPLICATING) {
5019        AudioStreamOut *out = thread->clearOutput();
5020        // from now on thread->mOutput is NULL
5021        out->hwDev->close_output_stream(out->hwDev, out->stream);
5022        delete out;
5023    }
5024    return NO_ERROR;
5025}
5026
5027status_t AudioFlinger::suspendOutput(int output)
5028{
5029    Mutex::Autolock _l(mLock);
5030    PlaybackThread *thread = checkPlaybackThread_l(output);
5031
5032    if (thread == NULL) {
5033        return BAD_VALUE;
5034    }
5035
5036    ALOGV("suspendOutput() %d", output);
5037    thread->suspend();
5038
5039    return NO_ERROR;
5040}
5041
5042status_t AudioFlinger::restoreOutput(int output)
5043{
5044    Mutex::Autolock _l(mLock);
5045    PlaybackThread *thread = checkPlaybackThread_l(output);
5046
5047    if (thread == NULL) {
5048        return BAD_VALUE;
5049    }
5050
5051    ALOGV("restoreOutput() %d", output);
5052
5053    thread->restore();
5054
5055    return NO_ERROR;
5056}
5057
5058int AudioFlinger::openInput(uint32_t *pDevices,
5059                                uint32_t *pSamplingRate,
5060                                uint32_t *pFormat,
5061                                uint32_t *pChannels,
5062                                uint32_t acoustics)
5063{
5064    status_t status;
5065    RecordThread *thread = NULL;
5066    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5067    uint32_t format = pFormat ? *pFormat : 0;
5068    uint32_t channels = pChannels ? *pChannels : 0;
5069    uint32_t reqSamplingRate = samplingRate;
5070    uint32_t reqFormat = format;
5071    uint32_t reqChannels = channels;
5072    audio_stream_in_t *inStream;
5073    audio_hw_device_t *inHwDev;
5074
5075    if (pDevices == NULL || *pDevices == 0) {
5076        return 0;
5077    }
5078
5079    Mutex::Autolock _l(mLock);
5080
5081    inHwDev = findSuitableHwDev_l(*pDevices);
5082    if (inHwDev == NULL)
5083        return 0;
5084
5085    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5086                                        &channels, &samplingRate,
5087                                        (audio_in_acoustics_t)acoustics,
5088                                        &inStream);
5089    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5090            inStream,
5091            samplingRate,
5092            format,
5093            channels,
5094            acoustics,
5095            status);
5096
5097    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5098    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5099    // or stereo to mono conversions on 16 bit PCM inputs.
5100    if (inStream == NULL && status == BAD_VALUE &&
5101        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5102        (samplingRate <= 2 * reqSamplingRate) &&
5103        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5104        ALOGV("openInput() reopening with proposed sampling rate and channels");
5105        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5106                                            &channels, &samplingRate,
5107                                            (audio_in_acoustics_t)acoustics,
5108                                            &inStream);
5109    }
5110
5111    if (inStream != NULL) {
5112        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5113
5114        int id = nextUniqueId();
5115        // Start record thread
5116        // RecorThread require both input and output device indication to forward to audio
5117        // pre processing modules
5118        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5119        thread = new RecordThread(this,
5120                                  input,
5121                                  reqSamplingRate,
5122                                  reqChannels,
5123                                  id,
5124                                  device);
5125        mRecordThreads.add(id, thread);
5126        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5127        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5128        if (pFormat) *pFormat = format;
5129        if (pChannels) *pChannels = reqChannels;
5130
5131        input->stream->common.standby(&input->stream->common);
5132
5133        // notify client processes of the new input creation
5134        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5135        return id;
5136    }
5137
5138    return 0;
5139}
5140
5141status_t AudioFlinger::closeInput(int input)
5142{
5143    // keep strong reference on the record thread so that
5144    // it is not destroyed while exit() is executed
5145    sp <RecordThread> thread;
5146    {
5147        Mutex::Autolock _l(mLock);
5148        thread = checkRecordThread_l(input);
5149        if (thread == NULL) {
5150            return BAD_VALUE;
5151        }
5152
5153        ALOGV("closeInput() %d", input);
5154        void *param2 = 0;
5155        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5156        mRecordThreads.removeItem(input);
5157    }
5158    thread->exit();
5159
5160    AudioStreamIn *in = thread->clearInput();
5161    // from now on thread->mInput is NULL
5162    in->hwDev->close_input_stream(in->hwDev, in->stream);
5163    delete in;
5164
5165    return NO_ERROR;
5166}
5167
5168status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output)
5169{
5170    Mutex::Autolock _l(mLock);
5171    MixerThread *dstThread = checkMixerThread_l(output);
5172    if (dstThread == NULL) {
5173        ALOGW("setStreamOutput() bad output id %d", output);
5174        return BAD_VALUE;
5175    }
5176
5177    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5178    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5179
5180    dstThread->setStreamValid(stream, true);
5181
5182    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5183        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5184        if (thread != dstThread &&
5185            thread->type() != ThreadBase::DIRECT) {
5186            MixerThread *srcThread = (MixerThread *)thread;
5187            srcThread->setStreamValid(stream, false);
5188            srcThread->invalidateTracks(stream);
5189        }
5190    }
5191
5192    return NO_ERROR;
5193}
5194
5195
5196int AudioFlinger::newAudioSessionId()
5197{
5198    return nextUniqueId();
5199}
5200
5201void AudioFlinger::acquireAudioSessionId(int audioSession)
5202{
5203    Mutex::Autolock _l(mLock);
5204    int caller = IPCThreadState::self()->getCallingPid();
5205    ALOGV("acquiring %d from %d", audioSession, caller);
5206    int num = mAudioSessionRefs.size();
5207    for (int i = 0; i< num; i++) {
5208        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5209        if (ref->sessionid == audioSession && ref->pid == caller) {
5210            ref->cnt++;
5211            ALOGV(" incremented refcount to %d", ref->cnt);
5212            return;
5213        }
5214    }
5215    AudioSessionRef *ref = new AudioSessionRef();
5216    ref->sessionid = audioSession;
5217    ref->pid = caller;
5218    ref->cnt = 1;
5219    mAudioSessionRefs.push(ref);
5220    ALOGV(" added new entry for %d", ref->sessionid);
5221}
5222
5223void AudioFlinger::releaseAudioSessionId(int audioSession)
5224{
5225    Mutex::Autolock _l(mLock);
5226    int caller = IPCThreadState::self()->getCallingPid();
5227    ALOGV("releasing %d from %d", audioSession, caller);
5228    int num = mAudioSessionRefs.size();
5229    for (int i = 0; i< num; i++) {
5230        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5231        if (ref->sessionid == audioSession && ref->pid == caller) {
5232            ref->cnt--;
5233            ALOGV(" decremented refcount to %d", ref->cnt);
5234            if (ref->cnt == 0) {
5235                mAudioSessionRefs.removeAt(i);
5236                delete ref;
5237                purgeStaleEffects_l();
5238            }
5239            return;
5240        }
5241    }
5242    ALOGW("session id %d not found for pid %d", audioSession, caller);
5243}
5244
5245void AudioFlinger::purgeStaleEffects_l() {
5246
5247    ALOGV("purging stale effects");
5248
5249    Vector< sp<EffectChain> > chains;
5250
5251    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5252        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5253        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5254            sp<EffectChain> ec = t->mEffectChains[j];
5255            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5256                chains.push(ec);
5257            }
5258        }
5259    }
5260    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5261        sp<RecordThread> t = mRecordThreads.valueAt(i);
5262        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5263            sp<EffectChain> ec = t->mEffectChains[j];
5264            chains.push(ec);
5265        }
5266    }
5267
5268    for (size_t i = 0; i < chains.size(); i++) {
5269        sp<EffectChain> ec = chains[i];
5270        int sessionid = ec->sessionId();
5271        sp<ThreadBase> t = ec->mThread.promote();
5272        if (t == 0) {
5273            continue;
5274        }
5275        size_t numsessionrefs = mAudioSessionRefs.size();
5276        bool found = false;
5277        for (size_t k = 0; k < numsessionrefs; k++) {
5278            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5279            if (ref->sessionid == sessionid) {
5280                ALOGV(" session %d still exists for %d with %d refs",
5281                     sessionid, ref->pid, ref->cnt);
5282                found = true;
5283                break;
5284            }
5285        }
5286        if (!found) {
5287            // remove all effects from the chain
5288            while (ec->mEffects.size()) {
5289                sp<EffectModule> effect = ec->mEffects[0];
5290                effect->unPin();
5291                Mutex::Autolock _l (t->mLock);
5292                t->removeEffect_l(effect);
5293                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5294                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5295                    if (handle != 0) {
5296                        handle->mEffect.clear();
5297                        if (handle->mHasControl && handle->mEnabled) {
5298                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5299                        }
5300                    }
5301                }
5302                AudioSystem::unregisterEffect(effect->id());
5303            }
5304        }
5305    }
5306    return;
5307}
5308
5309// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5310AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5311{
5312    PlaybackThread *thread = NULL;
5313    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5314        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5315    }
5316    return thread;
5317}
5318
5319// checkMixerThread_l() must be called with AudioFlinger::mLock held
5320AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5321{
5322    PlaybackThread *thread = checkPlaybackThread_l(output);
5323    if (thread != NULL) {
5324        if (thread->type() == ThreadBase::DIRECT) {
5325            thread = NULL;
5326        }
5327    }
5328    return (MixerThread *)thread;
5329}
5330
5331// checkRecordThread_l() must be called with AudioFlinger::mLock held
5332AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5333{
5334    RecordThread *thread = NULL;
5335    if (mRecordThreads.indexOfKey(input) >= 0) {
5336        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5337    }
5338    return thread;
5339}
5340
5341uint32_t AudioFlinger::nextUniqueId()
5342{
5343    return android_atomic_inc(&mNextUniqueId);
5344}
5345
5346AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5347{
5348    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5349        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5350        AudioStreamOut *output = thread->getOutput();
5351        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5352            return thread;
5353        }
5354    }
5355    return NULL;
5356}
5357
5358uint32_t AudioFlinger::primaryOutputDevice_l()
5359{
5360    PlaybackThread *thread = primaryPlaybackThread_l();
5361
5362    if (thread == NULL) {
5363        return 0;
5364    }
5365
5366    return thread->device();
5367}
5368
5369
5370// ----------------------------------------------------------------------------
5371//  Effect management
5372// ----------------------------------------------------------------------------
5373
5374
5375status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5376{
5377    Mutex::Autolock _l(mLock);
5378    return EffectQueryNumberEffects(numEffects);
5379}
5380
5381status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5382{
5383    Mutex::Autolock _l(mLock);
5384    return EffectQueryEffect(index, descriptor);
5385}
5386
5387status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5388{
5389    Mutex::Autolock _l(mLock);
5390    return EffectGetDescriptor(pUuid, descriptor);
5391}
5392
5393
5394sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5395        effect_descriptor_t *pDesc,
5396        const sp<IEffectClient>& effectClient,
5397        int32_t priority,
5398        int io,
5399        int sessionId,
5400        status_t *status,
5401        int *id,
5402        int *enabled)
5403{
5404    status_t lStatus = NO_ERROR;
5405    sp<EffectHandle> handle;
5406    effect_descriptor_t desc;
5407    sp<Client> client;
5408    wp<Client> wclient;
5409
5410    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5411            pid, effectClient.get(), priority, sessionId, io);
5412
5413    if (pDesc == NULL) {
5414        lStatus = BAD_VALUE;
5415        goto Exit;
5416    }
5417
5418    // check audio settings permission for global effects
5419    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5420        lStatus = PERMISSION_DENIED;
5421        goto Exit;
5422    }
5423
5424    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5425    // that can only be created by audio policy manager (running in same process)
5426    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5427        lStatus = PERMISSION_DENIED;
5428        goto Exit;
5429    }
5430
5431    if (io == 0) {
5432        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5433            // output must be specified by AudioPolicyManager when using session
5434            // AUDIO_SESSION_OUTPUT_STAGE
5435            lStatus = BAD_VALUE;
5436            goto Exit;
5437        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5438            // if the output returned by getOutputForEffect() is removed before we lock the
5439            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5440            // and we will exit safely
5441            io = AudioSystem::getOutputForEffect(&desc);
5442        }
5443    }
5444
5445    {
5446        Mutex::Autolock _l(mLock);
5447
5448
5449        if (!EffectIsNullUuid(&pDesc->uuid)) {
5450            // if uuid is specified, request effect descriptor
5451            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5452            if (lStatus < 0) {
5453                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5454                goto Exit;
5455            }
5456        } else {
5457            // if uuid is not specified, look for an available implementation
5458            // of the required type in effect factory
5459            if (EffectIsNullUuid(&pDesc->type)) {
5460                ALOGW("createEffect() no effect type");
5461                lStatus = BAD_VALUE;
5462                goto Exit;
5463            }
5464            uint32_t numEffects = 0;
5465            effect_descriptor_t d;
5466            d.flags = 0; // prevent compiler warning
5467            bool found = false;
5468
5469            lStatus = EffectQueryNumberEffects(&numEffects);
5470            if (lStatus < 0) {
5471                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5472                goto Exit;
5473            }
5474            for (uint32_t i = 0; i < numEffects; i++) {
5475                lStatus = EffectQueryEffect(i, &desc);
5476                if (lStatus < 0) {
5477                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5478                    continue;
5479                }
5480                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5481                    // If matching type found save effect descriptor. If the session is
5482                    // 0 and the effect is not auxiliary, continue enumeration in case
5483                    // an auxiliary version of this effect type is available
5484                    found = true;
5485                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5486                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5487                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5488                        break;
5489                    }
5490                }
5491            }
5492            if (!found) {
5493                lStatus = BAD_VALUE;
5494                ALOGW("createEffect() effect not found");
5495                goto Exit;
5496            }
5497            // For same effect type, chose auxiliary version over insert version if
5498            // connect to output mix (Compliance to OpenSL ES)
5499            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5500                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5501                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5502            }
5503        }
5504
5505        // Do not allow auxiliary effects on a session different from 0 (output mix)
5506        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5507             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5508            lStatus = INVALID_OPERATION;
5509            goto Exit;
5510        }
5511
5512        // check recording permission for visualizer
5513        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5514            !recordingAllowed()) {
5515            lStatus = PERMISSION_DENIED;
5516            goto Exit;
5517        }
5518
5519        // return effect descriptor
5520        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5521
5522        // If output is not specified try to find a matching audio session ID in one of the
5523        // output threads.
5524        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5525        // because of code checking output when entering the function.
5526        // Note: io is never 0 when creating an effect on an input
5527        if (io == 0) {
5528             // look for the thread where the specified audio session is present
5529            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5530                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5531                    io = mPlaybackThreads.keyAt(i);
5532                    break;
5533                }
5534            }
5535            if (io == 0) {
5536               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5537                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5538                       io = mRecordThreads.keyAt(i);
5539                       break;
5540                   }
5541               }
5542            }
5543            // If no output thread contains the requested session ID, default to
5544            // first output. The effect chain will be moved to the correct output
5545            // thread when a track with the same session ID is created
5546            if (io == 0 && mPlaybackThreads.size()) {
5547                io = mPlaybackThreads.keyAt(0);
5548            }
5549            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5550        }
5551        ThreadBase *thread = checkRecordThread_l(io);
5552        if (thread == NULL) {
5553            thread = checkPlaybackThread_l(io);
5554            if (thread == NULL) {
5555                ALOGE("createEffect() unknown output thread");
5556                lStatus = BAD_VALUE;
5557                goto Exit;
5558            }
5559        }
5560
5561        wclient = mClients.valueFor(pid);
5562
5563        if (wclient != NULL) {
5564            client = wclient.promote();
5565        } else {
5566            client = new Client(this, pid);
5567            mClients.add(pid, client);
5568        }
5569
5570        // create effect on selected output thread
5571        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5572                &desc, enabled, &lStatus);
5573        if (handle != 0 && id != NULL) {
5574            *id = handle->id();
5575        }
5576    }
5577
5578Exit:
5579    if(status) {
5580        *status = lStatus;
5581    }
5582    return handle;
5583}
5584
5585status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5586{
5587    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5588            sessionId, srcOutput, dstOutput);
5589    Mutex::Autolock _l(mLock);
5590    if (srcOutput == dstOutput) {
5591        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
5592        return NO_ERROR;
5593    }
5594    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5595    if (srcThread == NULL) {
5596        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
5597        return BAD_VALUE;
5598    }
5599    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5600    if (dstThread == NULL) {
5601        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
5602        return BAD_VALUE;
5603    }
5604
5605    Mutex::Autolock _dl(dstThread->mLock);
5606    Mutex::Autolock _sl(srcThread->mLock);
5607    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5608
5609    return NO_ERROR;
5610}
5611
5612// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5613status_t AudioFlinger::moveEffectChain_l(int sessionId,
5614                                   AudioFlinger::PlaybackThread *srcThread,
5615                                   AudioFlinger::PlaybackThread *dstThread,
5616                                   bool reRegister)
5617{
5618    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5619            sessionId, srcThread, dstThread);
5620
5621    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5622    if (chain == 0) {
5623        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5624                sessionId, srcThread);
5625        return INVALID_OPERATION;
5626    }
5627
5628    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5629    // so that a new chain is created with correct parameters when first effect is added. This is
5630    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5631    // removed.
5632    srcThread->removeEffectChain_l(chain);
5633
5634    // transfer all effects one by one so that new effect chain is created on new thread with
5635    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5636    int dstOutput = dstThread->id();
5637    sp<EffectChain> dstChain;
5638    uint32_t strategy = 0; // prevent compiler warning
5639    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5640    while (effect != 0) {
5641        srcThread->removeEffect_l(effect);
5642        dstThread->addEffect_l(effect);
5643        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5644        if (effect->state() == EffectModule::ACTIVE ||
5645                effect->state() == EffectModule::STOPPING) {
5646            effect->start();
5647        }
5648        // if the move request is not received from audio policy manager, the effect must be
5649        // re-registered with the new strategy and output
5650        if (dstChain == 0) {
5651            dstChain = effect->chain().promote();
5652            if (dstChain == 0) {
5653                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5654                srcThread->addEffect_l(effect);
5655                return NO_INIT;
5656            }
5657            strategy = dstChain->strategy();
5658        }
5659        if (reRegister) {
5660            AudioSystem::unregisterEffect(effect->id());
5661            AudioSystem::registerEffect(&effect->desc(),
5662                                        dstOutput,
5663                                        strategy,
5664                                        sessionId,
5665                                        effect->id());
5666        }
5667        effect = chain->getEffectFromId_l(0);
5668    }
5669
5670    return NO_ERROR;
5671}
5672
5673
5674// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5675sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5676        const sp<AudioFlinger::Client>& client,
5677        const sp<IEffectClient>& effectClient,
5678        int32_t priority,
5679        int sessionId,
5680        effect_descriptor_t *desc,
5681        int *enabled,
5682        status_t *status
5683        )
5684{
5685    sp<EffectModule> effect;
5686    sp<EffectHandle> handle;
5687    status_t lStatus;
5688    sp<EffectChain> chain;
5689    bool chainCreated = false;
5690    bool effectCreated = false;
5691    bool effectRegistered = false;
5692
5693    lStatus = initCheck();
5694    if (lStatus != NO_ERROR) {
5695        ALOGW("createEffect_l() Audio driver not initialized.");
5696        goto Exit;
5697    }
5698
5699    // Do not allow effects with session ID 0 on direct output or duplicating threads
5700    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5701    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5702        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5703                desc->name, sessionId);
5704        lStatus = BAD_VALUE;
5705        goto Exit;
5706    }
5707    // Only Pre processor effects are allowed on input threads and only on input threads
5708    if ((mType == RECORD &&
5709            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5710            (mType != RECORD &&
5711                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5712        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5713                desc->name, desc->flags, mType);
5714        lStatus = BAD_VALUE;
5715        goto Exit;
5716    }
5717
5718    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5719
5720    { // scope for mLock
5721        Mutex::Autolock _l(mLock);
5722
5723        // check for existing effect chain with the requested audio session
5724        chain = getEffectChain_l(sessionId);
5725        if (chain == 0) {
5726            // create a new chain for this session
5727            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5728            chain = new EffectChain(this, sessionId);
5729            addEffectChain_l(chain);
5730            chain->setStrategy(getStrategyForSession_l(sessionId));
5731            chainCreated = true;
5732        } else {
5733            effect = chain->getEffectFromDesc_l(desc);
5734        }
5735
5736        ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
5737
5738        if (effect == 0) {
5739            int id = mAudioFlinger->nextUniqueId();
5740            // Check CPU and memory usage
5741            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5742            if (lStatus != NO_ERROR) {
5743                goto Exit;
5744            }
5745            effectRegistered = true;
5746            // create a new effect module if none present in the chain
5747            effect = new EffectModule(this, chain, desc, id, sessionId);
5748            lStatus = effect->status();
5749            if (lStatus != NO_ERROR) {
5750                goto Exit;
5751            }
5752            lStatus = chain->addEffect_l(effect);
5753            if (lStatus != NO_ERROR) {
5754                goto Exit;
5755            }
5756            effectCreated = true;
5757
5758            effect->setDevice(mDevice);
5759            effect->setMode(mAudioFlinger->getMode());
5760        }
5761        // create effect handle and connect it to effect module
5762        handle = new EffectHandle(effect, client, effectClient, priority);
5763        lStatus = effect->addHandle(handle);
5764        if (enabled) {
5765            *enabled = (int)effect->isEnabled();
5766        }
5767    }
5768
5769Exit:
5770    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5771        Mutex::Autolock _l(mLock);
5772        if (effectCreated) {
5773            chain->removeEffect_l(effect);
5774        }
5775        if (effectRegistered) {
5776            AudioSystem::unregisterEffect(effect->id());
5777        }
5778        if (chainCreated) {
5779            removeEffectChain_l(chain);
5780        }
5781        handle.clear();
5782    }
5783
5784    if(status) {
5785        *status = lStatus;
5786    }
5787    return handle;
5788}
5789
5790sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5791{
5792    sp<EffectModule> effect;
5793
5794    sp<EffectChain> chain = getEffectChain_l(sessionId);
5795    if (chain != 0) {
5796        effect = chain->getEffectFromId_l(effectId);
5797    }
5798    return effect;
5799}
5800
5801// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5802// PlaybackThread::mLock held
5803status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5804{
5805    // check for existing effect chain with the requested audio session
5806    int sessionId = effect->sessionId();
5807    sp<EffectChain> chain = getEffectChain_l(sessionId);
5808    bool chainCreated = false;
5809
5810    if (chain == 0) {
5811        // create a new chain for this session
5812        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5813        chain = new EffectChain(this, sessionId);
5814        addEffectChain_l(chain);
5815        chain->setStrategy(getStrategyForSession_l(sessionId));
5816        chainCreated = true;
5817    }
5818    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5819
5820    if (chain->getEffectFromId_l(effect->id()) != 0) {
5821        ALOGW("addEffect_l() %p effect %s already present in chain %p",
5822                this, effect->desc().name, chain.get());
5823        return BAD_VALUE;
5824    }
5825
5826    status_t status = chain->addEffect_l(effect);
5827    if (status != NO_ERROR) {
5828        if (chainCreated) {
5829            removeEffectChain_l(chain);
5830        }
5831        return status;
5832    }
5833
5834    effect->setDevice(mDevice);
5835    effect->setMode(mAudioFlinger->getMode());
5836    return NO_ERROR;
5837}
5838
5839void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5840
5841    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5842    effect_descriptor_t desc = effect->desc();
5843    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5844        detachAuxEffect_l(effect->id());
5845    }
5846
5847    sp<EffectChain> chain = effect->chain().promote();
5848    if (chain != 0) {
5849        // remove effect chain if removing last effect
5850        if (chain->removeEffect_l(effect) == 0) {
5851            removeEffectChain_l(chain);
5852        }
5853    } else {
5854        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5855    }
5856}
5857
5858void AudioFlinger::ThreadBase::lockEffectChains_l(
5859        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5860{
5861    effectChains = mEffectChains;
5862    for (size_t i = 0; i < mEffectChains.size(); i++) {
5863        mEffectChains[i]->lock();
5864    }
5865}
5866
5867void AudioFlinger::ThreadBase::unlockEffectChains(
5868        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5869{
5870    for (size_t i = 0; i < effectChains.size(); i++) {
5871        effectChains[i]->unlock();
5872    }
5873}
5874
5875sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5876{
5877    Mutex::Autolock _l(mLock);
5878    return getEffectChain_l(sessionId);
5879}
5880
5881sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5882{
5883    sp<EffectChain> chain;
5884
5885    size_t size = mEffectChains.size();
5886    for (size_t i = 0; i < size; i++) {
5887        if (mEffectChains[i]->sessionId() == sessionId) {
5888            chain = mEffectChains[i];
5889            break;
5890        }
5891    }
5892    return chain;
5893}
5894
5895void AudioFlinger::ThreadBase::setMode(uint32_t mode)
5896{
5897    Mutex::Autolock _l(mLock);
5898    size_t size = mEffectChains.size();
5899    for (size_t i = 0; i < size; i++) {
5900        mEffectChains[i]->setMode_l(mode);
5901    }
5902}
5903
5904void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5905                                                    const wp<EffectHandle>& handle,
5906                                                    bool unpiniflast) {
5907
5908    Mutex::Autolock _l(mLock);
5909    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5910    // delete the effect module if removing last handle on it
5911    if (effect->removeHandle(handle) == 0) {
5912        if (!effect->isPinned() || unpiniflast) {
5913            removeEffect_l(effect);
5914            AudioSystem::unregisterEffect(effect->id());
5915        }
5916    }
5917}
5918
5919status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5920{
5921    int session = chain->sessionId();
5922    int16_t *buffer = mMixBuffer;
5923    bool ownsBuffer = false;
5924
5925    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5926    if (session > 0) {
5927        // Only one effect chain can be present in direct output thread and it uses
5928        // the mix buffer as input
5929        if (mType != DIRECT) {
5930            size_t numSamples = mFrameCount * mChannelCount;
5931            buffer = new int16_t[numSamples];
5932            memset(buffer, 0, numSamples * sizeof(int16_t));
5933            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5934            ownsBuffer = true;
5935        }
5936
5937        // Attach all tracks with same session ID to this chain.
5938        for (size_t i = 0; i < mTracks.size(); ++i) {
5939            sp<Track> track = mTracks[i];
5940            if (session == track->sessionId()) {
5941                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5942                track->setMainBuffer(buffer);
5943                chain->incTrackCnt();
5944            }
5945        }
5946
5947        // indicate all active tracks in the chain
5948        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5949            sp<Track> track = mActiveTracks[i].promote();
5950            if (track == 0) continue;
5951            if (session == track->sessionId()) {
5952                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5953                chain->incActiveTrackCnt();
5954            }
5955        }
5956    }
5957
5958    chain->setInBuffer(buffer, ownsBuffer);
5959    chain->setOutBuffer(mMixBuffer);
5960    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5961    // chains list in order to be processed last as it contains output stage effects
5962    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5963    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5964    // after track specific effects and before output stage
5965    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5966    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5967    // Effect chain for other sessions are inserted at beginning of effect
5968    // chains list to be processed before output mix effects. Relative order between other
5969    // sessions is not important
5970    size_t size = mEffectChains.size();
5971    size_t i = 0;
5972    for (i = 0; i < size; i++) {
5973        if (mEffectChains[i]->sessionId() < session) break;
5974    }
5975    mEffectChains.insertAt(chain, i);
5976    checkSuspendOnAddEffectChain_l(chain);
5977
5978    return NO_ERROR;
5979}
5980
5981size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
5982{
5983    int session = chain->sessionId();
5984
5985    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
5986
5987    for (size_t i = 0; i < mEffectChains.size(); i++) {
5988        if (chain == mEffectChains[i]) {
5989            mEffectChains.removeAt(i);
5990            // detach all active tracks from the chain
5991            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5992                sp<Track> track = mActiveTracks[i].promote();
5993                if (track == 0) continue;
5994                if (session == track->sessionId()) {
5995                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
5996                            chain.get(), session);
5997                    chain->decActiveTrackCnt();
5998                }
5999            }
6000
6001            // detach all tracks with same session ID from this chain
6002            for (size_t i = 0; i < mTracks.size(); ++i) {
6003                sp<Track> track = mTracks[i];
6004                if (session == track->sessionId()) {
6005                    track->setMainBuffer(mMixBuffer);
6006                    chain->decTrackCnt();
6007                }
6008            }
6009            break;
6010        }
6011    }
6012    return mEffectChains.size();
6013}
6014
6015status_t AudioFlinger::PlaybackThread::attachAuxEffect(
6016        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6017{
6018    Mutex::Autolock _l(mLock);
6019    return attachAuxEffect_l(track, EffectId);
6020}
6021
6022status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6023        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6024{
6025    status_t status = NO_ERROR;
6026
6027    if (EffectId == 0) {
6028        track->setAuxBuffer(0, NULL);
6029    } else {
6030        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6031        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6032        if (effect != 0) {
6033            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6034                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6035            } else {
6036                status = INVALID_OPERATION;
6037            }
6038        } else {
6039            status = BAD_VALUE;
6040        }
6041    }
6042    return status;
6043}
6044
6045void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6046{
6047     for (size_t i = 0; i < mTracks.size(); ++i) {
6048        sp<Track> track = mTracks[i];
6049        if (track->auxEffectId() == effectId) {
6050            attachAuxEffect_l(track, 0);
6051        }
6052    }
6053}
6054
6055status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6056{
6057    // only one chain per input thread
6058    if (mEffectChains.size() != 0) {
6059        return INVALID_OPERATION;
6060    }
6061    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6062
6063    chain->setInBuffer(NULL);
6064    chain->setOutBuffer(NULL);
6065
6066    checkSuspendOnAddEffectChain_l(chain);
6067
6068    mEffectChains.add(chain);
6069
6070    return NO_ERROR;
6071}
6072
6073size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6074{
6075    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6076    ALOGW_IF(mEffectChains.size() != 1,
6077            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6078            chain.get(), mEffectChains.size(), this);
6079    if (mEffectChains.size() == 1) {
6080        mEffectChains.removeAt(0);
6081    }
6082    return 0;
6083}
6084
6085// ----------------------------------------------------------------------------
6086//  EffectModule implementation
6087// ----------------------------------------------------------------------------
6088
6089#undef LOG_TAG
6090#define LOG_TAG "AudioFlinger::EffectModule"
6091
6092AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6093                                        const wp<AudioFlinger::EffectChain>& chain,
6094                                        effect_descriptor_t *desc,
6095                                        int id,
6096                                        int sessionId)
6097    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6098      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6099{
6100    ALOGV("Constructor %p", this);
6101    int lStatus;
6102    sp<ThreadBase> thread = mThread.promote();
6103    if (thread == 0) {
6104        return;
6105    }
6106
6107    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6108
6109    // create effect engine from effect factory
6110    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6111
6112    if (mStatus != NO_ERROR) {
6113        return;
6114    }
6115    lStatus = init();
6116    if (lStatus < 0) {
6117        mStatus = lStatus;
6118        goto Error;
6119    }
6120
6121    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6122        mPinned = true;
6123    }
6124    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6125    return;
6126Error:
6127    EffectRelease(mEffectInterface);
6128    mEffectInterface = NULL;
6129    ALOGV("Constructor Error %d", mStatus);
6130}
6131
6132AudioFlinger::EffectModule::~EffectModule()
6133{
6134    ALOGV("Destructor %p", this);
6135    if (mEffectInterface != NULL) {
6136        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6137                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6138            sp<ThreadBase> thread = mThread.promote();
6139            if (thread != 0) {
6140                audio_stream_t *stream = thread->stream();
6141                if (stream != NULL) {
6142                    stream->remove_audio_effect(stream, mEffectInterface);
6143                }
6144            }
6145        }
6146        // release effect engine
6147        EffectRelease(mEffectInterface);
6148    }
6149}
6150
6151status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6152{
6153    status_t status;
6154
6155    Mutex::Autolock _l(mLock);
6156    // First handle in mHandles has highest priority and controls the effect module
6157    int priority = handle->priority();
6158    size_t size = mHandles.size();
6159    sp<EffectHandle> h;
6160    size_t i;
6161    for (i = 0; i < size; i++) {
6162        h = mHandles[i].promote();
6163        if (h == 0) continue;
6164        if (h->priority() <= priority) break;
6165    }
6166    // if inserted in first place, move effect control from previous owner to this handle
6167    if (i == 0) {
6168        bool enabled = false;
6169        if (h != 0) {
6170            enabled = h->enabled();
6171            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6172        }
6173        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6174        status = NO_ERROR;
6175    } else {
6176        status = ALREADY_EXISTS;
6177    }
6178    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6179    mHandles.insertAt(handle, i);
6180    return status;
6181}
6182
6183size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6184{
6185    Mutex::Autolock _l(mLock);
6186    size_t size = mHandles.size();
6187    size_t i;
6188    for (i = 0; i < size; i++) {
6189        if (mHandles[i] == handle) break;
6190    }
6191    if (i == size) {
6192        return size;
6193    }
6194    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6195
6196    bool enabled = false;
6197    EffectHandle *hdl = handle.unsafe_get();
6198    if (hdl) {
6199        ALOGV("removeHandle() unsafe_get OK");
6200        enabled = hdl->enabled();
6201    }
6202    mHandles.removeAt(i);
6203    size = mHandles.size();
6204    // if removed from first place, move effect control from this handle to next in line
6205    if (i == 0 && size != 0) {
6206        sp<EffectHandle> h = mHandles[0].promote();
6207        if (h != 0) {
6208            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6209        }
6210    }
6211
6212    // Prevent calls to process() and other functions on effect interface from now on.
6213    // The effect engine will be released by the destructor when the last strong reference on
6214    // this object is released which can happen after next process is called.
6215    if (size == 0 && !mPinned) {
6216        mState = DESTROYED;
6217    }
6218
6219    return size;
6220}
6221
6222sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6223{
6224    Mutex::Autolock _l(mLock);
6225    sp<EffectHandle> handle;
6226    if (mHandles.size() != 0) {
6227        handle = mHandles[0].promote();
6228    }
6229    return handle;
6230}
6231
6232void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6233{
6234    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6235    // keep a strong reference on this EffectModule to avoid calling the
6236    // destructor before we exit
6237    sp<EffectModule> keep(this);
6238    {
6239        sp<ThreadBase> thread = mThread.promote();
6240        if (thread != 0) {
6241            thread->disconnectEffect(keep, handle, unpiniflast);
6242        }
6243    }
6244}
6245
6246void AudioFlinger::EffectModule::updateState() {
6247    Mutex::Autolock _l(mLock);
6248
6249    switch (mState) {
6250    case RESTART:
6251        reset_l();
6252        // FALL THROUGH
6253
6254    case STARTING:
6255        // clear auxiliary effect input buffer for next accumulation
6256        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6257            memset(mConfig.inputCfg.buffer.raw,
6258                   0,
6259                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6260        }
6261        start_l();
6262        mState = ACTIVE;
6263        break;
6264    case STOPPING:
6265        stop_l();
6266        mDisableWaitCnt = mMaxDisableWaitCnt;
6267        mState = STOPPED;
6268        break;
6269    case STOPPED:
6270        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6271        // turn off sequence.
6272        if (--mDisableWaitCnt == 0) {
6273            reset_l();
6274            mState = IDLE;
6275        }
6276        break;
6277    default: //IDLE , ACTIVE, DESTROYED
6278        break;
6279    }
6280}
6281
6282void AudioFlinger::EffectModule::process()
6283{
6284    Mutex::Autolock _l(mLock);
6285
6286    if (mState == DESTROYED || mEffectInterface == NULL ||
6287            mConfig.inputCfg.buffer.raw == NULL ||
6288            mConfig.outputCfg.buffer.raw == NULL) {
6289        return;
6290    }
6291
6292    if (isProcessEnabled()) {
6293        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6294        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6295            ditherAndClamp(mConfig.inputCfg.buffer.s32,
6296                                        mConfig.inputCfg.buffer.s32,
6297                                        mConfig.inputCfg.buffer.frameCount/2);
6298        }
6299
6300        // do the actual processing in the effect engine
6301        int ret = (*mEffectInterface)->process(mEffectInterface,
6302                                               &mConfig.inputCfg.buffer,
6303                                               &mConfig.outputCfg.buffer);
6304
6305        // force transition to IDLE state when engine is ready
6306        if (mState == STOPPED && ret == -ENODATA) {
6307            mDisableWaitCnt = 1;
6308        }
6309
6310        // clear auxiliary effect input buffer for next accumulation
6311        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6312            memset(mConfig.inputCfg.buffer.raw, 0,
6313                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6314        }
6315    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6316                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6317        // If an insert effect is idle and input buffer is different from output buffer,
6318        // accumulate input onto output
6319        sp<EffectChain> chain = mChain.promote();
6320        if (chain != 0 && chain->activeTrackCnt() != 0) {
6321            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6322            int16_t *in = mConfig.inputCfg.buffer.s16;
6323            int16_t *out = mConfig.outputCfg.buffer.s16;
6324            for (size_t i = 0; i < frameCnt; i++) {
6325                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6326            }
6327        }
6328    }
6329}
6330
6331void AudioFlinger::EffectModule::reset_l()
6332{
6333    if (mEffectInterface == NULL) {
6334        return;
6335    }
6336    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6337}
6338
6339status_t AudioFlinger::EffectModule::configure()
6340{
6341    uint32_t channels;
6342    if (mEffectInterface == NULL) {
6343        return NO_INIT;
6344    }
6345
6346    sp<ThreadBase> thread = mThread.promote();
6347    if (thread == 0) {
6348        return DEAD_OBJECT;
6349    }
6350
6351    // TODO: handle configuration of effects replacing track process
6352    if (thread->channelCount() == 1) {
6353        channels = AUDIO_CHANNEL_OUT_MONO;
6354    } else {
6355        channels = AUDIO_CHANNEL_OUT_STEREO;
6356    }
6357
6358    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6359        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6360    } else {
6361        mConfig.inputCfg.channels = channels;
6362    }
6363    mConfig.outputCfg.channels = channels;
6364    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6365    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6366    mConfig.inputCfg.samplingRate = thread->sampleRate();
6367    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6368    mConfig.inputCfg.bufferProvider.cookie = NULL;
6369    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6370    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6371    mConfig.outputCfg.bufferProvider.cookie = NULL;
6372    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6373    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6374    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6375    // Insert effect:
6376    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6377    // always overwrites output buffer: input buffer == output buffer
6378    // - in other sessions:
6379    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6380    //      other effect: overwrites output buffer: input buffer == output buffer
6381    // Auxiliary effect:
6382    //      accumulates in output buffer: input buffer != output buffer
6383    // Therefore: accumulate <=> input buffer != output buffer
6384    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6385        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6386    } else {
6387        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6388    }
6389    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6390    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6391    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6392    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6393
6394    ALOGV("configure() %p thread %p buffer %p framecount %d",
6395            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6396
6397    status_t cmdStatus;
6398    uint32_t size = sizeof(int);
6399    status_t status = (*mEffectInterface)->command(mEffectInterface,
6400                                                   EFFECT_CMD_SET_CONFIG,
6401                                                   sizeof(effect_config_t),
6402                                                   &mConfig,
6403                                                   &size,
6404                                                   &cmdStatus);
6405    if (status == 0) {
6406        status = cmdStatus;
6407    }
6408
6409    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6410            (1000 * mConfig.outputCfg.buffer.frameCount);
6411
6412    return status;
6413}
6414
6415status_t AudioFlinger::EffectModule::init()
6416{
6417    Mutex::Autolock _l(mLock);
6418    if (mEffectInterface == NULL) {
6419        return NO_INIT;
6420    }
6421    status_t cmdStatus;
6422    uint32_t size = sizeof(status_t);
6423    status_t status = (*mEffectInterface)->command(mEffectInterface,
6424                                                   EFFECT_CMD_INIT,
6425                                                   0,
6426                                                   NULL,
6427                                                   &size,
6428                                                   &cmdStatus);
6429    if (status == 0) {
6430        status = cmdStatus;
6431    }
6432    return status;
6433}
6434
6435status_t AudioFlinger::EffectModule::start()
6436{
6437    Mutex::Autolock _l(mLock);
6438    return start_l();
6439}
6440
6441status_t AudioFlinger::EffectModule::start_l()
6442{
6443    if (mEffectInterface == NULL) {
6444        return NO_INIT;
6445    }
6446    status_t cmdStatus;
6447    uint32_t size = sizeof(status_t);
6448    status_t status = (*mEffectInterface)->command(mEffectInterface,
6449                                                   EFFECT_CMD_ENABLE,
6450                                                   0,
6451                                                   NULL,
6452                                                   &size,
6453                                                   &cmdStatus);
6454    if (status == 0) {
6455        status = cmdStatus;
6456    }
6457    if (status == 0 &&
6458            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6459             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6460        sp<ThreadBase> thread = mThread.promote();
6461        if (thread != 0) {
6462            audio_stream_t *stream = thread->stream();
6463            if (stream != NULL) {
6464                stream->add_audio_effect(stream, mEffectInterface);
6465            }
6466        }
6467    }
6468    return status;
6469}
6470
6471status_t AudioFlinger::EffectModule::stop()
6472{
6473    Mutex::Autolock _l(mLock);
6474    return stop_l();
6475}
6476
6477status_t AudioFlinger::EffectModule::stop_l()
6478{
6479    if (mEffectInterface == NULL) {
6480        return NO_INIT;
6481    }
6482    status_t cmdStatus;
6483    uint32_t size = sizeof(status_t);
6484    status_t status = (*mEffectInterface)->command(mEffectInterface,
6485                                                   EFFECT_CMD_DISABLE,
6486                                                   0,
6487                                                   NULL,
6488                                                   &size,
6489                                                   &cmdStatus);
6490    if (status == 0) {
6491        status = cmdStatus;
6492    }
6493    if (status == 0 &&
6494            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6495             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6496        sp<ThreadBase> thread = mThread.promote();
6497        if (thread != 0) {
6498            audio_stream_t *stream = thread->stream();
6499            if (stream != NULL) {
6500                stream->remove_audio_effect(stream, mEffectInterface);
6501            }
6502        }
6503    }
6504    return status;
6505}
6506
6507status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6508                                             uint32_t cmdSize,
6509                                             void *pCmdData,
6510                                             uint32_t *replySize,
6511                                             void *pReplyData)
6512{
6513    Mutex::Autolock _l(mLock);
6514//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6515
6516    if (mState == DESTROYED || mEffectInterface == NULL) {
6517        return NO_INIT;
6518    }
6519    status_t status = (*mEffectInterface)->command(mEffectInterface,
6520                                                   cmdCode,
6521                                                   cmdSize,
6522                                                   pCmdData,
6523                                                   replySize,
6524                                                   pReplyData);
6525    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6526        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6527        for (size_t i = 1; i < mHandles.size(); i++) {
6528            sp<EffectHandle> h = mHandles[i].promote();
6529            if (h != 0) {
6530                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6531            }
6532        }
6533    }
6534    return status;
6535}
6536
6537status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6538{
6539
6540    Mutex::Autolock _l(mLock);
6541    ALOGV("setEnabled %p enabled %d", this, enabled);
6542
6543    if (enabled != isEnabled()) {
6544        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6545        if (enabled && status != NO_ERROR) {
6546            return status;
6547        }
6548
6549        switch (mState) {
6550        // going from disabled to enabled
6551        case IDLE:
6552            mState = STARTING;
6553            break;
6554        case STOPPED:
6555            mState = RESTART;
6556            break;
6557        case STOPPING:
6558            mState = ACTIVE;
6559            break;
6560
6561        // going from enabled to disabled
6562        case RESTART:
6563            mState = STOPPED;
6564            break;
6565        case STARTING:
6566            mState = IDLE;
6567            break;
6568        case ACTIVE:
6569            mState = STOPPING;
6570            break;
6571        case DESTROYED:
6572            return NO_ERROR; // simply ignore as we are being destroyed
6573        }
6574        for (size_t i = 1; i < mHandles.size(); i++) {
6575            sp<EffectHandle> h = mHandles[i].promote();
6576            if (h != 0) {
6577                h->setEnabled(enabled);
6578            }
6579        }
6580    }
6581    return NO_ERROR;
6582}
6583
6584bool AudioFlinger::EffectModule::isEnabled()
6585{
6586    switch (mState) {
6587    case RESTART:
6588    case STARTING:
6589    case ACTIVE:
6590        return true;
6591    case IDLE:
6592    case STOPPING:
6593    case STOPPED:
6594    case DESTROYED:
6595    default:
6596        return false;
6597    }
6598}
6599
6600bool AudioFlinger::EffectModule::isProcessEnabled()
6601{
6602    switch (mState) {
6603    case RESTART:
6604    case ACTIVE:
6605    case STOPPING:
6606    case STOPPED:
6607        return true;
6608    case IDLE:
6609    case STARTING:
6610    case DESTROYED:
6611    default:
6612        return false;
6613    }
6614}
6615
6616status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6617{
6618    Mutex::Autolock _l(mLock);
6619    status_t status = NO_ERROR;
6620
6621    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6622    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6623    if (isProcessEnabled() &&
6624            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6625            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6626        status_t cmdStatus;
6627        uint32_t volume[2];
6628        uint32_t *pVolume = NULL;
6629        uint32_t size = sizeof(volume);
6630        volume[0] = *left;
6631        volume[1] = *right;
6632        if (controller) {
6633            pVolume = volume;
6634        }
6635        status = (*mEffectInterface)->command(mEffectInterface,
6636                                              EFFECT_CMD_SET_VOLUME,
6637                                              size,
6638                                              volume,
6639                                              &size,
6640                                              pVolume);
6641        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6642            *left = volume[0];
6643            *right = volume[1];
6644        }
6645    }
6646    return status;
6647}
6648
6649status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6650{
6651    Mutex::Autolock _l(mLock);
6652    status_t status = NO_ERROR;
6653    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6654        // audio pre processing modules on RecordThread can receive both output and
6655        // input device indication in the same call
6656        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6657        if (dev) {
6658            status_t cmdStatus;
6659            uint32_t size = sizeof(status_t);
6660
6661            status = (*mEffectInterface)->command(mEffectInterface,
6662                                                  EFFECT_CMD_SET_DEVICE,
6663                                                  sizeof(uint32_t),
6664                                                  &dev,
6665                                                  &size,
6666                                                  &cmdStatus);
6667            if (status == NO_ERROR) {
6668                status = cmdStatus;
6669            }
6670        }
6671        dev = device & AUDIO_DEVICE_IN_ALL;
6672        if (dev) {
6673            status_t cmdStatus;
6674            uint32_t size = sizeof(status_t);
6675
6676            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6677                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6678                                                  sizeof(uint32_t),
6679                                                  &dev,
6680                                                  &size,
6681                                                  &cmdStatus);
6682            if (status2 == NO_ERROR) {
6683                status2 = cmdStatus;
6684            }
6685            if (status == NO_ERROR) {
6686                status = status2;
6687            }
6688        }
6689    }
6690    return status;
6691}
6692
6693status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
6694{
6695    Mutex::Autolock _l(mLock);
6696    status_t status = NO_ERROR;
6697    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6698        status_t cmdStatus;
6699        uint32_t size = sizeof(status_t);
6700        status = (*mEffectInterface)->command(mEffectInterface,
6701                                              EFFECT_CMD_SET_AUDIO_MODE,
6702                                              sizeof(int),
6703                                              &mode,
6704                                              &size,
6705                                              &cmdStatus);
6706        if (status == NO_ERROR) {
6707            status = cmdStatus;
6708        }
6709    }
6710    return status;
6711}
6712
6713void AudioFlinger::EffectModule::setSuspended(bool suspended)
6714{
6715    Mutex::Autolock _l(mLock);
6716    mSuspended = suspended;
6717}
6718
6719bool AudioFlinger::EffectModule::suspended() const
6720{
6721    Mutex::Autolock _l(mLock);
6722    return mSuspended;
6723}
6724
6725status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6726{
6727    const size_t SIZE = 256;
6728    char buffer[SIZE];
6729    String8 result;
6730
6731    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6732    result.append(buffer);
6733
6734    bool locked = tryLock(mLock);
6735    // failed to lock - AudioFlinger is probably deadlocked
6736    if (!locked) {
6737        result.append("\t\tCould not lock Fx mutex:\n");
6738    }
6739
6740    result.append("\t\tSession Status State Engine:\n");
6741    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6742            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6743    result.append(buffer);
6744
6745    result.append("\t\tDescriptor:\n");
6746    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6747            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6748            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6749            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6750    result.append(buffer);
6751    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6752                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6753                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6754                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6755    result.append(buffer);
6756    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6757            mDescriptor.apiVersion,
6758            mDescriptor.flags);
6759    result.append(buffer);
6760    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6761            mDescriptor.name);
6762    result.append(buffer);
6763    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6764            mDescriptor.implementor);
6765    result.append(buffer);
6766
6767    result.append("\t\t- Input configuration:\n");
6768    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6769    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6770            (uint32_t)mConfig.inputCfg.buffer.raw,
6771            mConfig.inputCfg.buffer.frameCount,
6772            mConfig.inputCfg.samplingRate,
6773            mConfig.inputCfg.channels,
6774            mConfig.inputCfg.format);
6775    result.append(buffer);
6776
6777    result.append("\t\t- Output configuration:\n");
6778    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6779    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6780            (uint32_t)mConfig.outputCfg.buffer.raw,
6781            mConfig.outputCfg.buffer.frameCount,
6782            mConfig.outputCfg.samplingRate,
6783            mConfig.outputCfg.channels,
6784            mConfig.outputCfg.format);
6785    result.append(buffer);
6786
6787    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6788    result.append(buffer);
6789    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6790    for (size_t i = 0; i < mHandles.size(); ++i) {
6791        sp<EffectHandle> handle = mHandles[i].promote();
6792        if (handle != 0) {
6793            handle->dump(buffer, SIZE);
6794            result.append(buffer);
6795        }
6796    }
6797
6798    result.append("\n");
6799
6800    write(fd, result.string(), result.length());
6801
6802    if (locked) {
6803        mLock.unlock();
6804    }
6805
6806    return NO_ERROR;
6807}
6808
6809// ----------------------------------------------------------------------------
6810//  EffectHandle implementation
6811// ----------------------------------------------------------------------------
6812
6813#undef LOG_TAG
6814#define LOG_TAG "AudioFlinger::EffectHandle"
6815
6816AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6817                                        const sp<AudioFlinger::Client>& client,
6818                                        const sp<IEffectClient>& effectClient,
6819                                        int32_t priority)
6820    : BnEffect(),
6821    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6822    mPriority(priority), mHasControl(false), mEnabled(false)
6823{
6824    ALOGV("constructor %p", this);
6825
6826    if (client == 0) {
6827        return;
6828    }
6829    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6830    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6831    if (mCblkMemory != 0) {
6832        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6833
6834        if (mCblk) {
6835            new(mCblk) effect_param_cblk_t();
6836            mBuffer = (uint8_t *)mCblk + bufOffset;
6837         }
6838    } else {
6839        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6840        return;
6841    }
6842}
6843
6844AudioFlinger::EffectHandle::~EffectHandle()
6845{
6846    ALOGV("Destructor %p", this);
6847    disconnect(false);
6848    ALOGV("Destructor DONE %p", this);
6849}
6850
6851status_t AudioFlinger::EffectHandle::enable()
6852{
6853    ALOGV("enable %p", this);
6854    if (!mHasControl) return INVALID_OPERATION;
6855    if (mEffect == 0) return DEAD_OBJECT;
6856
6857    if (mEnabled) {
6858        return NO_ERROR;
6859    }
6860
6861    mEnabled = true;
6862
6863    sp<ThreadBase> thread = mEffect->thread().promote();
6864    if (thread != 0) {
6865        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6866    }
6867
6868    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6869    if (mEffect->suspended()) {
6870        return NO_ERROR;
6871    }
6872
6873    status_t status = mEffect->setEnabled(true);
6874    if (status != NO_ERROR) {
6875        if (thread != 0) {
6876            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6877        }
6878        mEnabled = false;
6879    }
6880    return status;
6881}
6882
6883status_t AudioFlinger::EffectHandle::disable()
6884{
6885    ALOGV("disable %p", this);
6886    if (!mHasControl) return INVALID_OPERATION;
6887    if (mEffect == 0) return DEAD_OBJECT;
6888
6889    if (!mEnabled) {
6890        return NO_ERROR;
6891    }
6892    mEnabled = false;
6893
6894    if (mEffect->suspended()) {
6895        return NO_ERROR;
6896    }
6897
6898    status_t status = mEffect->setEnabled(false);
6899
6900    sp<ThreadBase> thread = mEffect->thread().promote();
6901    if (thread != 0) {
6902        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6903    }
6904
6905    return status;
6906}
6907
6908void AudioFlinger::EffectHandle::disconnect()
6909{
6910    disconnect(true);
6911}
6912
6913void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6914{
6915    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6916    if (mEffect == 0) {
6917        return;
6918    }
6919    mEffect->disconnect(this, unpiniflast);
6920
6921    if (mHasControl && mEnabled) {
6922        sp<ThreadBase> thread = mEffect->thread().promote();
6923        if (thread != 0) {
6924            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6925        }
6926    }
6927
6928    // release sp on module => module destructor can be called now
6929    mEffect.clear();
6930    if (mClient != 0) {
6931        if (mCblk) {
6932            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6933        }
6934        mCblkMemory.clear();            // and free the shared memory
6935        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6936        mClient.clear();
6937    }
6938}
6939
6940status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6941                                             uint32_t cmdSize,
6942                                             void *pCmdData,
6943                                             uint32_t *replySize,
6944                                             void *pReplyData)
6945{
6946//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6947//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6948
6949    // only get parameter command is permitted for applications not controlling the effect
6950    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6951        return INVALID_OPERATION;
6952    }
6953    if (mEffect == 0) return DEAD_OBJECT;
6954    if (mClient == 0) return INVALID_OPERATION;
6955
6956    // handle commands that are not forwarded transparently to effect engine
6957    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6958        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6959        // no risk to block the whole media server process or mixer threads is we are stuck here
6960        Mutex::Autolock _l(mCblk->lock);
6961        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6962            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6963            mCblk->serverIndex = 0;
6964            mCblk->clientIndex = 0;
6965            return BAD_VALUE;
6966        }
6967        status_t status = NO_ERROR;
6968        while (mCblk->serverIndex < mCblk->clientIndex) {
6969            int reply;
6970            uint32_t rsize = sizeof(int);
6971            int *p = (int *)(mBuffer + mCblk->serverIndex);
6972            int size = *p++;
6973            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
6974                ALOGW("command(): invalid parameter block size");
6975                break;
6976            }
6977            effect_param_t *param = (effect_param_t *)p;
6978            if (param->psize == 0 || param->vsize == 0) {
6979                ALOGW("command(): null parameter or value size");
6980                mCblk->serverIndex += size;
6981                continue;
6982            }
6983            uint32_t psize = sizeof(effect_param_t) +
6984                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
6985                             param->vsize;
6986            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
6987                                            psize,
6988                                            p,
6989                                            &rsize,
6990                                            &reply);
6991            // stop at first error encountered
6992            if (ret != NO_ERROR) {
6993                status = ret;
6994                *(int *)pReplyData = reply;
6995                break;
6996            } else if (reply != NO_ERROR) {
6997                *(int *)pReplyData = reply;
6998                break;
6999            }
7000            mCblk->serverIndex += size;
7001        }
7002        mCblk->serverIndex = 0;
7003        mCblk->clientIndex = 0;
7004        return status;
7005    } else if (cmdCode == EFFECT_CMD_ENABLE) {
7006        *(int *)pReplyData = NO_ERROR;
7007        return enable();
7008    } else if (cmdCode == EFFECT_CMD_DISABLE) {
7009        *(int *)pReplyData = NO_ERROR;
7010        return disable();
7011    }
7012
7013    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7014}
7015
7016sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
7017    return mCblkMemory;
7018}
7019
7020void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
7021{
7022    ALOGV("setControl %p control %d", this, hasControl);
7023
7024    mHasControl = hasControl;
7025    mEnabled = enabled;
7026
7027    if (signal && mEffectClient != 0) {
7028        mEffectClient->controlStatusChanged(hasControl);
7029    }
7030}
7031
7032void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7033                                                 uint32_t cmdSize,
7034                                                 void *pCmdData,
7035                                                 uint32_t replySize,
7036                                                 void *pReplyData)
7037{
7038    if (mEffectClient != 0) {
7039        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7040    }
7041}
7042
7043
7044
7045void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7046{
7047    if (mEffectClient != 0) {
7048        mEffectClient->enableStatusChanged(enabled);
7049    }
7050}
7051
7052status_t AudioFlinger::EffectHandle::onTransact(
7053    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7054{
7055    return BnEffect::onTransact(code, data, reply, flags);
7056}
7057
7058
7059void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7060{
7061    bool locked = mCblk ? tryLock(mCblk->lock) : false;
7062
7063    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7064            (mClient == NULL) ? getpid() : mClient->pid(),
7065            mPriority,
7066            mHasControl,
7067            !locked,
7068            mCblk ? mCblk->clientIndex : 0,
7069            mCblk ? mCblk->serverIndex : 0
7070            );
7071
7072    if (locked) {
7073        mCblk->lock.unlock();
7074    }
7075}
7076
7077#undef LOG_TAG
7078#define LOG_TAG "AudioFlinger::EffectChain"
7079
7080AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7081                                        int sessionId)
7082    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7083      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7084      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7085{
7086    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7087    sp<ThreadBase> thread = mThread.promote();
7088    if (thread == 0) {
7089        return;
7090    }
7091    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7092                                    thread->frameCount();
7093}
7094
7095AudioFlinger::EffectChain::~EffectChain()
7096{
7097    if (mOwnInBuffer) {
7098        delete mInBuffer;
7099    }
7100
7101}
7102
7103// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7104sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7105{
7106    sp<EffectModule> effect;
7107    size_t size = mEffects.size();
7108
7109    for (size_t i = 0; i < size; i++) {
7110        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7111            effect = mEffects[i];
7112            break;
7113        }
7114    }
7115    return effect;
7116}
7117
7118// getEffectFromId_l() must be called with ThreadBase::mLock held
7119sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7120{
7121    sp<EffectModule> effect;
7122    size_t size = mEffects.size();
7123
7124    for (size_t i = 0; i < size; i++) {
7125        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7126        if (id == 0 || mEffects[i]->id() == id) {
7127            effect = mEffects[i];
7128            break;
7129        }
7130    }
7131    return effect;
7132}
7133
7134// getEffectFromType_l() must be called with ThreadBase::mLock held
7135sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7136        const effect_uuid_t *type)
7137{
7138    sp<EffectModule> effect;
7139    size_t size = mEffects.size();
7140
7141    for (size_t i = 0; i < size; i++) {
7142        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7143            effect = mEffects[i];
7144            break;
7145        }
7146    }
7147    return effect;
7148}
7149
7150// Must be called with EffectChain::mLock locked
7151void AudioFlinger::EffectChain::process_l()
7152{
7153    sp<ThreadBase> thread = mThread.promote();
7154    if (thread == 0) {
7155        ALOGW("process_l(): cannot promote mixer thread");
7156        return;
7157    }
7158    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7159            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7160    // always process effects unless no more tracks are on the session and the effect tail
7161    // has been rendered
7162    bool doProcess = true;
7163    if (!isGlobalSession) {
7164        bool tracksOnSession = (trackCnt() != 0);
7165
7166        if (!tracksOnSession && mTailBufferCount == 0) {
7167            doProcess = false;
7168        }
7169
7170        if (activeTrackCnt() == 0) {
7171            // if no track is active and the effect tail has not been rendered,
7172            // the input buffer must be cleared here as the mixer process will not do it
7173            if (tracksOnSession || mTailBufferCount > 0) {
7174                size_t numSamples = thread->frameCount() * thread->channelCount();
7175                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7176                if (mTailBufferCount > 0) {
7177                    mTailBufferCount--;
7178                }
7179            }
7180        }
7181    }
7182
7183    size_t size = mEffects.size();
7184    if (doProcess) {
7185        for (size_t i = 0; i < size; i++) {
7186            mEffects[i]->process();
7187        }
7188    }
7189    for (size_t i = 0; i < size; i++) {
7190        mEffects[i]->updateState();
7191    }
7192}
7193
7194// addEffect_l() must be called with PlaybackThread::mLock held
7195status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7196{
7197    effect_descriptor_t desc = effect->desc();
7198    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7199
7200    Mutex::Autolock _l(mLock);
7201    effect->setChain(this);
7202    sp<ThreadBase> thread = mThread.promote();
7203    if (thread == 0) {
7204        return NO_INIT;
7205    }
7206    effect->setThread(thread);
7207
7208    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7209        // Auxiliary effects are inserted at the beginning of mEffects vector as
7210        // they are processed first and accumulated in chain input buffer
7211        mEffects.insertAt(effect, 0);
7212
7213        // the input buffer for auxiliary effect contains mono samples in
7214        // 32 bit format. This is to avoid saturation in AudoMixer
7215        // accumulation stage. Saturation is done in EffectModule::process() before
7216        // calling the process in effect engine
7217        size_t numSamples = thread->frameCount();
7218        int32_t *buffer = new int32_t[numSamples];
7219        memset(buffer, 0, numSamples * sizeof(int32_t));
7220        effect->setInBuffer((int16_t *)buffer);
7221        // auxiliary effects output samples to chain input buffer for further processing
7222        // by insert effects
7223        effect->setOutBuffer(mInBuffer);
7224    } else {
7225        // Insert effects are inserted at the end of mEffects vector as they are processed
7226        //  after track and auxiliary effects.
7227        // Insert effect order as a function of indicated preference:
7228        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7229        //  another effect is present
7230        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7231        //  last effect claiming first position
7232        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7233        //  first effect claiming last position
7234        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7235        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7236        // already present
7237
7238        int size = (int)mEffects.size();
7239        int idx_insert = size;
7240        int idx_insert_first = -1;
7241        int idx_insert_last = -1;
7242
7243        for (int i = 0; i < size; i++) {
7244            effect_descriptor_t d = mEffects[i]->desc();
7245            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7246            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7247            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7248                // check invalid effect chaining combinations
7249                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7250                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7251                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7252                    return INVALID_OPERATION;
7253                }
7254                // remember position of first insert effect and by default
7255                // select this as insert position for new effect
7256                if (idx_insert == size) {
7257                    idx_insert = i;
7258                }
7259                // remember position of last insert effect claiming
7260                // first position
7261                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7262                    idx_insert_first = i;
7263                }
7264                // remember position of first insert effect claiming
7265                // last position
7266                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7267                    idx_insert_last == -1) {
7268                    idx_insert_last = i;
7269                }
7270            }
7271        }
7272
7273        // modify idx_insert from first position if needed
7274        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7275            if (idx_insert_last != -1) {
7276                idx_insert = idx_insert_last;
7277            } else {
7278                idx_insert = size;
7279            }
7280        } else {
7281            if (idx_insert_first != -1) {
7282                idx_insert = idx_insert_first + 1;
7283            }
7284        }
7285
7286        // always read samples from chain input buffer
7287        effect->setInBuffer(mInBuffer);
7288
7289        // if last effect in the chain, output samples to chain
7290        // output buffer, otherwise to chain input buffer
7291        if (idx_insert == size) {
7292            if (idx_insert != 0) {
7293                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7294                mEffects[idx_insert-1]->configure();
7295            }
7296            effect->setOutBuffer(mOutBuffer);
7297        } else {
7298            effect->setOutBuffer(mInBuffer);
7299        }
7300        mEffects.insertAt(effect, idx_insert);
7301
7302        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7303    }
7304    effect->configure();
7305    return NO_ERROR;
7306}
7307
7308// removeEffect_l() must be called with PlaybackThread::mLock held
7309size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7310{
7311    Mutex::Autolock _l(mLock);
7312    int size = (int)mEffects.size();
7313    int i;
7314    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7315
7316    for (i = 0; i < size; i++) {
7317        if (effect == mEffects[i]) {
7318            // calling stop here will remove pre-processing effect from the audio HAL.
7319            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7320            // the middle of a read from audio HAL
7321            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7322                    mEffects[i]->state() == EffectModule::STOPPING) {
7323                mEffects[i]->stop();
7324            }
7325            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7326                delete[] effect->inBuffer();
7327            } else {
7328                if (i == size - 1 && i != 0) {
7329                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7330                    mEffects[i - 1]->configure();
7331                }
7332            }
7333            mEffects.removeAt(i);
7334            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7335            break;
7336        }
7337    }
7338
7339    return mEffects.size();
7340}
7341
7342// setDevice_l() must be called with PlaybackThread::mLock held
7343void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7344{
7345    size_t size = mEffects.size();
7346    for (size_t i = 0; i < size; i++) {
7347        mEffects[i]->setDevice(device);
7348    }
7349}
7350
7351// setMode_l() must be called with PlaybackThread::mLock held
7352void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
7353{
7354    size_t size = mEffects.size();
7355    for (size_t i = 0; i < size; i++) {
7356        mEffects[i]->setMode(mode);
7357    }
7358}
7359
7360// setVolume_l() must be called with PlaybackThread::mLock held
7361bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7362{
7363    uint32_t newLeft = *left;
7364    uint32_t newRight = *right;
7365    bool hasControl = false;
7366    int ctrlIdx = -1;
7367    size_t size = mEffects.size();
7368
7369    // first update volume controller
7370    for (size_t i = size; i > 0; i--) {
7371        if (mEffects[i - 1]->isProcessEnabled() &&
7372            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7373            ctrlIdx = i - 1;
7374            hasControl = true;
7375            break;
7376        }
7377    }
7378
7379    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7380        if (hasControl) {
7381            *left = mNewLeftVolume;
7382            *right = mNewRightVolume;
7383        }
7384        return hasControl;
7385    }
7386
7387    mVolumeCtrlIdx = ctrlIdx;
7388    mLeftVolume = newLeft;
7389    mRightVolume = newRight;
7390
7391    // second get volume update from volume controller
7392    if (ctrlIdx >= 0) {
7393        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7394        mNewLeftVolume = newLeft;
7395        mNewRightVolume = newRight;
7396    }
7397    // then indicate volume to all other effects in chain.
7398    // Pass altered volume to effects before volume controller
7399    // and requested volume to effects after controller
7400    uint32_t lVol = newLeft;
7401    uint32_t rVol = newRight;
7402
7403    for (size_t i = 0; i < size; i++) {
7404        if ((int)i == ctrlIdx) continue;
7405        // this also works for ctrlIdx == -1 when there is no volume controller
7406        if ((int)i > ctrlIdx) {
7407            lVol = *left;
7408            rVol = *right;
7409        }
7410        mEffects[i]->setVolume(&lVol, &rVol, false);
7411    }
7412    *left = newLeft;
7413    *right = newRight;
7414
7415    return hasControl;
7416}
7417
7418status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7419{
7420    const size_t SIZE = 256;
7421    char buffer[SIZE];
7422    String8 result;
7423
7424    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7425    result.append(buffer);
7426
7427    bool locked = tryLock(mLock);
7428    // failed to lock - AudioFlinger is probably deadlocked
7429    if (!locked) {
7430        result.append("\tCould not lock mutex:\n");
7431    }
7432
7433    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7434    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7435            mEffects.size(),
7436            (uint32_t)mInBuffer,
7437            (uint32_t)mOutBuffer,
7438            mActiveTrackCnt);
7439    result.append(buffer);
7440    write(fd, result.string(), result.size());
7441
7442    for (size_t i = 0; i < mEffects.size(); ++i) {
7443        sp<EffectModule> effect = mEffects[i];
7444        if (effect != 0) {
7445            effect->dump(fd, args);
7446        }
7447    }
7448
7449    if (locked) {
7450        mLock.unlock();
7451    }
7452
7453    return NO_ERROR;
7454}
7455
7456// must be called with ThreadBase::mLock held
7457void AudioFlinger::EffectChain::setEffectSuspended_l(
7458        const effect_uuid_t *type, bool suspend)
7459{
7460    sp<SuspendedEffectDesc> desc;
7461    // use effect type UUID timelow as key as there is no real risk of identical
7462    // timeLow fields among effect type UUIDs.
7463    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7464    if (suspend) {
7465        if (index >= 0) {
7466            desc = mSuspendedEffects.valueAt(index);
7467        } else {
7468            desc = new SuspendedEffectDesc();
7469            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7470            mSuspendedEffects.add(type->timeLow, desc);
7471            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7472        }
7473        if (desc->mRefCount++ == 0) {
7474            sp<EffectModule> effect = getEffectIfEnabled(type);
7475            if (effect != 0) {
7476                desc->mEffect = effect;
7477                effect->setSuspended(true);
7478                effect->setEnabled(false);
7479            }
7480        }
7481    } else {
7482        if (index < 0) {
7483            return;
7484        }
7485        desc = mSuspendedEffects.valueAt(index);
7486        if (desc->mRefCount <= 0) {
7487            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7488            desc->mRefCount = 1;
7489        }
7490        if (--desc->mRefCount == 0) {
7491            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7492            if (desc->mEffect != 0) {
7493                sp<EffectModule> effect = desc->mEffect.promote();
7494                if (effect != 0) {
7495                    effect->setSuspended(false);
7496                    sp<EffectHandle> handle = effect->controlHandle();
7497                    if (handle != 0) {
7498                        effect->setEnabled(handle->enabled());
7499                    }
7500                }
7501                desc->mEffect.clear();
7502            }
7503            mSuspendedEffects.removeItemsAt(index);
7504        }
7505    }
7506}
7507
7508// must be called with ThreadBase::mLock held
7509void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7510{
7511    sp<SuspendedEffectDesc> desc;
7512
7513    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7514    if (suspend) {
7515        if (index >= 0) {
7516            desc = mSuspendedEffects.valueAt(index);
7517        } else {
7518            desc = new SuspendedEffectDesc();
7519            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7520            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7521        }
7522        if (desc->mRefCount++ == 0) {
7523            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7524            for (size_t i = 0; i < effects.size(); i++) {
7525                setEffectSuspended_l(&effects[i]->desc().type, true);
7526            }
7527        }
7528    } else {
7529        if (index < 0) {
7530            return;
7531        }
7532        desc = mSuspendedEffects.valueAt(index);
7533        if (desc->mRefCount <= 0) {
7534            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7535            desc->mRefCount = 1;
7536        }
7537        if (--desc->mRefCount == 0) {
7538            Vector<const effect_uuid_t *> types;
7539            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7540                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7541                    continue;
7542                }
7543                types.add(&mSuspendedEffects.valueAt(i)->mType);
7544            }
7545            for (size_t i = 0; i < types.size(); i++) {
7546                setEffectSuspended_l(types[i], false);
7547            }
7548            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7549            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7550        }
7551    }
7552}
7553
7554
7555// The volume effect is used for automated tests only
7556#ifndef OPENSL_ES_H_
7557static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7558                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7559const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7560#endif //OPENSL_ES_H_
7561
7562bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7563{
7564    // auxiliary effects and visualizer are never suspended on output mix
7565    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7566        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7567         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7568         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7569        return false;
7570    }
7571    return true;
7572}
7573
7574Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7575{
7576    Vector< sp<EffectModule> > effects;
7577    for (size_t i = 0; i < mEffects.size(); i++) {
7578        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7579            continue;
7580        }
7581        effects.add(mEffects[i]);
7582    }
7583    return effects;
7584}
7585
7586sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7587                                                            const effect_uuid_t *type)
7588{
7589    sp<EffectModule> effect;
7590    effect = getEffectFromType_l(type);
7591    if (effect != 0 && !effect->isEnabled()) {
7592        effect.clear();
7593    }
7594    return effect;
7595}
7596
7597void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7598                                                            bool enabled)
7599{
7600    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7601    if (enabled) {
7602        if (index < 0) {
7603            // if the effect is not suspend check if all effects are suspended
7604            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7605            if (index < 0) {
7606                return;
7607            }
7608            if (!isEffectEligibleForSuspend(effect->desc())) {
7609                return;
7610            }
7611            setEffectSuspended_l(&effect->desc().type, enabled);
7612            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7613            if (index < 0) {
7614                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7615                return;
7616            }
7617        }
7618        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7619             effect->desc().type.timeLow);
7620        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7621        // if effect is requested to suspended but was not yet enabled, supend it now.
7622        if (desc->mEffect == 0) {
7623            desc->mEffect = effect;
7624            effect->setEnabled(false);
7625            effect->setSuspended(true);
7626        }
7627    } else {
7628        if (index < 0) {
7629            return;
7630        }
7631        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7632             effect->desc().type.timeLow);
7633        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7634        desc->mEffect.clear();
7635        effect->setSuspended(false);
7636    }
7637}
7638
7639#undef LOG_TAG
7640#define LOG_TAG "AudioFlinger"
7641
7642// ----------------------------------------------------------------------------
7643
7644status_t AudioFlinger::onTransact(
7645        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7646{
7647    return BnAudioFlinger::onTransact(code, data, reply, flags);
7648}
7649
7650}; // namespace android
7651