AudioFlinger.cpp revision b9980659501d0428d65d8292f3c32da69d37fbd2
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include <math.h>
23#include <signal.h>
24#include <sys/time.h>
25#include <sys/resource.h>
26
27#include <binder/IPCThreadState.h>
28#include <binder/IServiceManager.h>
29#include <utils/Log.h>
30#include <binder/Parcel.h>
31#include <binder/IPCThreadState.h>
32#include <utils/String16.h>
33#include <utils/threads.h>
34#include <utils/Atomic.h>
35
36#include <cutils/bitops.h>
37#include <cutils/properties.h>
38#include <cutils/compiler.h>
39
40#include <media/AudioTrack.h>
41#include <media/AudioRecord.h>
42#include <media/IMediaPlayerService.h>
43#include <media/IMediaDeathNotifier.h>
44
45#include <private/media/AudioTrackShared.h>
46#include <private/media/AudioEffectShared.h>
47
48#include <system/audio.h>
49#include <hardware/audio.h>
50
51#include "AudioMixer.h"
52#include "AudioFlinger.h"
53
54#include <media/EffectsFactoryApi.h>
55#include <audio_effects/effect_visualizer.h>
56#include <audio_effects/effect_ns.h>
57#include <audio_effects/effect_aec.h>
58
59#include <audio_utils/primitives.h>
60
61#include <cpustats/ThreadCpuUsage.h>
62#include <powermanager/PowerManager.h>
63// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
64
65// ----------------------------------------------------------------------------
66
67
68namespace android {
69
70static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
71static const char kHardwareLockedString[] = "Hardware lock is taken\n";
72
73//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
74static const float MAX_GAIN = 4096.0f;
75static const float MAX_GAIN_INT = 0x1000;
76
77// retry counts for buffer fill timeout
78// 50 * ~20msecs = 1 second
79static const int8_t kMaxTrackRetries = 50;
80static const int8_t kMaxTrackStartupRetries = 50;
81// allow less retry attempts on direct output thread.
82// direct outputs can be a scarce resource in audio hardware and should
83// be released as quickly as possible.
84static const int8_t kMaxTrackRetriesDirect = 2;
85
86static const int kDumpLockRetries = 50;
87static const int kDumpLockSleepUs = 20000;
88
89// don't warn about blocked writes or record buffer overflows more often than this
90static const nsecs_t kWarningThrottleNs = seconds(5);
91
92// RecordThread loop sleep time upon application overrun or audio HAL read error
93static const int kRecordThreadSleepUs = 5000;
94
95// maximum time to wait for setParameters to complete
96static const nsecs_t kSetParametersTimeoutNs = seconds(2);
97
98// minimum sleep time for the mixer thread loop when tracks are active but in underrun
99static const uint32_t kMinThreadSleepTimeUs = 5000;
100// maximum divider applied to the active sleep time in the mixer thread loop
101static const uint32_t kMaxThreadSleepTimeShift = 2;
102
103
104// ----------------------------------------------------------------------------
105
106static bool recordingAllowed() {
107    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
108    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
109    if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
110    return ok;
111}
112
113static bool settingsAllowed() {
114    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
115    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
116    if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
117    return ok;
118}
119
120// To collect the amplifier usage
121static void addBatteryData(uint32_t params) {
122    sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
123    if (service == NULL) {
124        // it already logged
125        return;
126    }
127
128    service->addBatteryData(params);
129}
130
131static int load_audio_interface(const char *if_name, const hw_module_t **mod,
132                                audio_hw_device_t **dev)
133{
134    int rc;
135
136    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
137    if (rc)
138        goto out;
139
140    rc = audio_hw_device_open(*mod, dev);
141    ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
142            AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
143    if (rc)
144        goto out;
145
146    return 0;
147
148out:
149    *mod = NULL;
150    *dev = NULL;
151    return rc;
152}
153
154static const char * const audio_interfaces[] = {
155    "primary",
156    "a2dp",
157    "usb",
158};
159#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
160
161// ----------------------------------------------------------------------------
162
163AudioFlinger::AudioFlinger()
164    : BnAudioFlinger(),
165        mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
166        mBtNrecIsOff(false)
167{
168}
169
170void AudioFlinger::onFirstRef()
171{
172    int rc = 0;
173
174    Mutex::Autolock _l(mLock);
175
176    /* TODO: move all this work into an Init() function */
177    mHardwareStatus = AUDIO_HW_IDLE;
178
179    for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
180        const hw_module_t *mod;
181        audio_hw_device_t *dev;
182
183        rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
184        if (rc)
185            continue;
186
187        ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
188             mod->name, mod->id);
189        mAudioHwDevs.push(dev);
190
191        if (!mPrimaryHardwareDev) {
192            mPrimaryHardwareDev = dev;
193            ALOGI("Using '%s' (%s.%s) as the primary audio interface",
194                 mod->name, mod->id, audio_interfaces[i]);
195        }
196    }
197
198    mHardwareStatus = AUDIO_HW_INIT;
199
200    if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
201        ALOGE("Primary audio interface not found");
202        return;
203    }
204
205    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
206        audio_hw_device_t *dev = mAudioHwDevs[i];
207
208        mHardwareStatus = AUDIO_HW_INIT;
209        rc = dev->init_check(dev);
210        if (rc == 0) {
211            AutoMutex lock(mHardwareLock);
212
213            mMode = AUDIO_MODE_NORMAL;
214            mHardwareStatus = AUDIO_HW_SET_MODE;
215            dev->set_mode(dev, mMode);
216            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
217            dev->set_master_volume(dev, 1.0f);
218            mHardwareStatus = AUDIO_HW_IDLE;
219        }
220    }
221}
222
223status_t AudioFlinger::initCheck() const
224{
225    Mutex::Autolock _l(mLock);
226    if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
227        return NO_INIT;
228    return NO_ERROR;
229}
230
231AudioFlinger::~AudioFlinger()
232{
233    int num_devs = mAudioHwDevs.size();
234
235    while (!mRecordThreads.isEmpty()) {
236        // closeInput() will remove first entry from mRecordThreads
237        closeInput(mRecordThreads.keyAt(0));
238    }
239    while (!mPlaybackThreads.isEmpty()) {
240        // closeOutput() will remove first entry from mPlaybackThreads
241        closeOutput(mPlaybackThreads.keyAt(0));
242    }
243
244    for (int i = 0; i < num_devs; i++) {
245        audio_hw_device_t *dev = mAudioHwDevs[i];
246        audio_hw_device_close(dev);
247    }
248    mAudioHwDevs.clear();
249}
250
251audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
252{
253    /* first matching HW device is returned */
254    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
255        audio_hw_device_t *dev = mAudioHwDevs[i];
256        if ((dev->get_supported_devices(dev) & devices) == devices)
257            return dev;
258    }
259    return NULL;
260}
261
262status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
263{
264    const size_t SIZE = 256;
265    char buffer[SIZE];
266    String8 result;
267
268    result.append("Clients:\n");
269    for (size_t i = 0; i < mClients.size(); ++i) {
270        wp<Client> wClient = mClients.valueAt(i);
271        if (wClient != 0) {
272            sp<Client> client = wClient.promote();
273            if (client != 0) {
274                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
275                result.append(buffer);
276            }
277        }
278    }
279
280    result.append("Global session refs:\n");
281    result.append(" session pid cnt\n");
282    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
283        AudioSessionRef *r = mAudioSessionRefs[i];
284        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
285        result.append(buffer);
286    }
287    write(fd, result.string(), result.size());
288    return NO_ERROR;
289}
290
291
292status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
293{
294    const size_t SIZE = 256;
295    char buffer[SIZE];
296    String8 result;
297    hardware_call_state hardwareStatus = mHardwareStatus;
298
299    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
300    result.append(buffer);
301    write(fd, result.string(), result.size());
302    return NO_ERROR;
303}
304
305status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
306{
307    const size_t SIZE = 256;
308    char buffer[SIZE];
309    String8 result;
310    snprintf(buffer, SIZE, "Permission Denial: "
311            "can't dump AudioFlinger from pid=%d, uid=%d\n",
312            IPCThreadState::self()->getCallingPid(),
313            IPCThreadState::self()->getCallingUid());
314    result.append(buffer);
315    write(fd, result.string(), result.size());
316    return NO_ERROR;
317}
318
319static bool tryLock(Mutex& mutex)
320{
321    bool locked = false;
322    for (int i = 0; i < kDumpLockRetries; ++i) {
323        if (mutex.tryLock() == NO_ERROR) {
324            locked = true;
325            break;
326        }
327        usleep(kDumpLockSleepUs);
328    }
329    return locked;
330}
331
332status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
333{
334    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
335        dumpPermissionDenial(fd, args);
336    } else {
337        // get state of hardware lock
338        bool hardwareLocked = tryLock(mHardwareLock);
339        if (!hardwareLocked) {
340            String8 result(kHardwareLockedString);
341            write(fd, result.string(), result.size());
342        } else {
343            mHardwareLock.unlock();
344        }
345
346        bool locked = tryLock(mLock);
347
348        // failed to lock - AudioFlinger is probably deadlocked
349        if (!locked) {
350            String8 result(kDeadlockedString);
351            write(fd, result.string(), result.size());
352        }
353
354        dumpClients(fd, args);
355        dumpInternals(fd, args);
356
357        // dump playback threads
358        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
359            mPlaybackThreads.valueAt(i)->dump(fd, args);
360        }
361
362        // dump record threads
363        for (size_t i = 0; i < mRecordThreads.size(); i++) {
364            mRecordThreads.valueAt(i)->dump(fd, args);
365        }
366
367        // dump all hardware devs
368        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
369            audio_hw_device_t *dev = mAudioHwDevs[i];
370            dev->dump(dev, fd);
371        }
372        if (locked) mLock.unlock();
373    }
374    return NO_ERROR;
375}
376
377
378// IAudioFlinger interface
379
380
381sp<IAudioTrack> AudioFlinger::createTrack(
382        pid_t pid,
383        int streamType,
384        uint32_t sampleRate,
385        uint32_t format,
386        uint32_t channelMask,
387        int frameCount,
388        uint32_t flags,
389        const sp<IMemory>& sharedBuffer,
390        int output,
391        int *sessionId,
392        status_t *status)
393{
394    sp<PlaybackThread::Track> track;
395    sp<TrackHandle> trackHandle;
396    sp<Client> client;
397    wp<Client> wclient;
398    status_t lStatus;
399    int lSessionId;
400
401    if (streamType >= AUDIO_STREAM_CNT) {
402        ALOGE("createTrack() invalid stream type %d", streamType);
403        lStatus = BAD_VALUE;
404        goto Exit;
405    }
406
407    {
408        Mutex::Autolock _l(mLock);
409        PlaybackThread *thread = checkPlaybackThread_l(output);
410        PlaybackThread *effectThread = NULL;
411        if (thread == NULL) {
412            ALOGE("unknown output thread");
413            lStatus = BAD_VALUE;
414            goto Exit;
415        }
416
417        wclient = mClients.valueFor(pid);
418
419        if (wclient != NULL) {
420            client = wclient.promote();
421        } else {
422            client = new Client(this, pid);
423            mClients.add(pid, client);
424        }
425
426        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
427        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
428            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
429                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
430                if (mPlaybackThreads.keyAt(i) != output) {
431                    // prevent same audio session on different output threads
432                    uint32_t sessions = t->hasAudioSession(*sessionId);
433                    if (sessions & PlaybackThread::TRACK_SESSION) {
434                        ALOGE("createTrack() session ID %d already in use", *sessionId);
435                        lStatus = BAD_VALUE;
436                        goto Exit;
437                    }
438                    // check if an effect with same session ID is waiting for a track to be created
439                    if (sessions & PlaybackThread::EFFECT_SESSION) {
440                        effectThread = t.get();
441                    }
442                }
443            }
444            lSessionId = *sessionId;
445        } else {
446            // if no audio session id is provided, create one here
447            lSessionId = nextUniqueId();
448            if (sessionId != NULL) {
449                *sessionId = lSessionId;
450            }
451        }
452        ALOGV("createTrack() lSessionId: %d", lSessionId);
453
454        track = thread->createTrack_l(client, streamType, sampleRate, format,
455                channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
456
457        // move effect chain to this output thread if an effect on same session was waiting
458        // for a track to be created
459        if (lStatus == NO_ERROR && effectThread != NULL) {
460            Mutex::Autolock _dl(thread->mLock);
461            Mutex::Autolock _sl(effectThread->mLock);
462            moveEffectChain_l(lSessionId, effectThread, thread, true);
463        }
464    }
465    if (lStatus == NO_ERROR) {
466        trackHandle = new TrackHandle(track);
467    } else {
468        // remove local strong reference to Client before deleting the Track so that the Client
469        // destructor is called by the TrackBase destructor with mLock held
470        client.clear();
471        track.clear();
472    }
473
474Exit:
475    if(status) {
476        *status = lStatus;
477    }
478    return trackHandle;
479}
480
481uint32_t AudioFlinger::sampleRate(int output) const
482{
483    Mutex::Autolock _l(mLock);
484    PlaybackThread *thread = checkPlaybackThread_l(output);
485    if (thread == NULL) {
486        ALOGW("sampleRate() unknown thread %d", output);
487        return 0;
488    }
489    return thread->sampleRate();
490}
491
492int AudioFlinger::channelCount(int output) const
493{
494    Mutex::Autolock _l(mLock);
495    PlaybackThread *thread = checkPlaybackThread_l(output);
496    if (thread == NULL) {
497        ALOGW("channelCount() unknown thread %d", output);
498        return 0;
499    }
500    return thread->channelCount();
501}
502
503uint32_t AudioFlinger::format(int output) const
504{
505    Mutex::Autolock _l(mLock);
506    PlaybackThread *thread = checkPlaybackThread_l(output);
507    if (thread == NULL) {
508        ALOGW("format() unknown thread %d", output);
509        return 0;
510    }
511    return thread->format();
512}
513
514size_t AudioFlinger::frameCount(int output) const
515{
516    Mutex::Autolock _l(mLock);
517    PlaybackThread *thread = checkPlaybackThread_l(output);
518    if (thread == NULL) {
519        ALOGW("frameCount() unknown thread %d", output);
520        return 0;
521    }
522    return thread->frameCount();
523}
524
525uint32_t AudioFlinger::latency(int output) const
526{
527    Mutex::Autolock _l(mLock);
528    PlaybackThread *thread = checkPlaybackThread_l(output);
529    if (thread == NULL) {
530        ALOGW("latency() unknown thread %d", output);
531        return 0;
532    }
533    return thread->latency();
534}
535
536status_t AudioFlinger::setMasterVolume(float value)
537{
538    status_t ret = initCheck();
539    if (ret != NO_ERROR) {
540        return ret;
541    }
542
543    // check calling permissions
544    if (!settingsAllowed()) {
545        return PERMISSION_DENIED;
546    }
547
548    // when hw supports master volume, don't scale in sw mixer
549    { // scope for the lock
550        AutoMutex lock(mHardwareLock);
551        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
552        if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
553            value = 1.0f;
554        }
555        mHardwareStatus = AUDIO_HW_IDLE;
556    }
557
558    Mutex::Autolock _l(mLock);
559    mMasterVolume = value;
560    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
561       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
562
563    return NO_ERROR;
564}
565
566status_t AudioFlinger::setMode(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(int stream, float value, int output)
658{
659    // check calling permissions
660    if (!settingsAllowed()) {
661        return PERMISSION_DENIED;
662    }
663
664    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
665        ALOGE("setStreamVolume() invalid stream %d", stream);
666        return BAD_VALUE;
667    }
668
669    AutoMutex lock(mLock);
670    PlaybackThread *thread = NULL;
671    if (output) {
672        thread = checkPlaybackThread_l(output);
673        if (thread == NULL) {
674            return BAD_VALUE;
675        }
676    }
677
678    mStreamTypes[stream].volume = value;
679
680    if (thread == NULL) {
681        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
682           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
683        }
684    } else {
685        thread->setStreamVolume(stream, value);
686    }
687
688    return NO_ERROR;
689}
690
691status_t AudioFlinger::setStreamMute(int stream, bool muted)
692{
693    // check calling permissions
694    if (!settingsAllowed()) {
695        return PERMISSION_DENIED;
696    }
697
698    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
699        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
700        ALOGE("setStreamMute() invalid stream %d", stream);
701        return BAD_VALUE;
702    }
703
704    AutoMutex lock(mLock);
705    mStreamTypes[stream].mute = muted;
706    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
707       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
708
709    return NO_ERROR;
710}
711
712float AudioFlinger::streamVolume(int stream, int output) const
713{
714    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
715        return 0.0f;
716    }
717
718    AutoMutex lock(mLock);
719    float volume;
720    if (output) {
721        PlaybackThread *thread = checkPlaybackThread_l(output);
722        if (thread == NULL) {
723            return 0.0f;
724        }
725        volume = thread->streamVolume(stream);
726    } else {
727        volume = mStreamTypes[stream].volume;
728    }
729
730    return volume;
731}
732
733bool AudioFlinger::streamMute(int stream) const
734{
735    if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
736        return true;
737    }
738
739    return mStreamTypes[stream].mute;
740}
741
742status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
743{
744    status_t result;
745
746    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
747            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
748    // check calling permissions
749    if (!settingsAllowed()) {
750        return PERMISSION_DENIED;
751    }
752
753    // ioHandle == 0 means the parameters are global to the audio hardware interface
754    if (ioHandle == 0) {
755        AutoMutex lock(mHardwareLock);
756        mHardwareStatus = AUDIO_SET_PARAMETER;
757        status_t final_result = NO_ERROR;
758        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
759            audio_hw_device_t *dev = mAudioHwDevs[i];
760            result = dev->set_parameters(dev, keyValuePairs.string());
761            final_result = result ?: final_result;
762        }
763        mHardwareStatus = AUDIO_HW_IDLE;
764        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
765        AudioParameter param = AudioParameter(keyValuePairs);
766        String8 value;
767        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
768            Mutex::Autolock _l(mLock);
769            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
770            if (mBtNrecIsOff != btNrecIsOff) {
771                for (size_t i = 0; i < mRecordThreads.size(); i++) {
772                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
773                    RecordThread::RecordTrack *track = thread->track();
774                    if (track != NULL) {
775                        audio_devices_t device = (audio_devices_t)(
776                                thread->device() & AUDIO_DEVICE_IN_ALL);
777                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
778                        thread->setEffectSuspended(FX_IID_AEC,
779                                                   suspend,
780                                                   track->sessionId());
781                        thread->setEffectSuspended(FX_IID_NS,
782                                                   suspend,
783                                                   track->sessionId());
784                    }
785                }
786                mBtNrecIsOff = btNrecIsOff;
787            }
788        }
789        return final_result;
790    }
791
792    // hold a strong ref on thread in case closeOutput() or closeInput() is called
793    // and the thread is exited once the lock is released
794    sp<ThreadBase> thread;
795    {
796        Mutex::Autolock _l(mLock);
797        thread = checkPlaybackThread_l(ioHandle);
798        if (thread == NULL) {
799            thread = checkRecordThread_l(ioHandle);
800        } else if (thread.get() == primaryPlaybackThread_l()) {
801            // indicate output device change to all input threads for pre processing
802            AudioParameter param = AudioParameter(keyValuePairs);
803            int value;
804            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
805                for (size_t i = 0; i < mRecordThreads.size(); i++) {
806                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
807                }
808            }
809        }
810    }
811    if (thread != NULL) {
812        result = thread->setParameters(keyValuePairs);
813        return result;
814    }
815    return BAD_VALUE;
816}
817
818String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
819{
820//    ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
821//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
822
823    if (ioHandle == 0) {
824        String8 out_s8;
825
826        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
827            audio_hw_device_t *dev = mAudioHwDevs[i];
828            char *s = dev->get_parameters(dev, keys.string());
829            out_s8 += String8(s);
830            free(s);
831        }
832        return out_s8;
833    }
834
835    Mutex::Autolock _l(mLock);
836
837    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
838    if (playbackThread != NULL) {
839        return playbackThread->getParameters(keys);
840    }
841    RecordThread *recordThread = checkRecordThread_l(ioHandle);
842    if (recordThread != NULL) {
843        return recordThread->getParameters(keys);
844    }
845    return String8("");
846}
847
848size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
849{
850    status_t ret = initCheck();
851    if (ret != NO_ERROR) {
852        return 0;
853    }
854
855    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
856}
857
858unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
859{
860    if (ioHandle == 0) {
861        return 0;
862    }
863
864    Mutex::Autolock _l(mLock);
865
866    RecordThread *recordThread = checkRecordThread_l(ioHandle);
867    if (recordThread != NULL) {
868        return recordThread->getInputFramesLost();
869    }
870    return 0;
871}
872
873status_t AudioFlinger::setVoiceVolume(float value)
874{
875    status_t ret = initCheck();
876    if (ret != NO_ERROR) {
877        return ret;
878    }
879
880    // check calling permissions
881    if (!settingsAllowed()) {
882        return PERMISSION_DENIED;
883    }
884
885    AutoMutex lock(mHardwareLock);
886    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
887    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
888    mHardwareStatus = AUDIO_HW_IDLE;
889
890    return ret;
891}
892
893status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
894{
895    status_t status;
896
897    Mutex::Autolock _l(mLock);
898
899    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
900    if (playbackThread != NULL) {
901        return playbackThread->getRenderPosition(halFrames, dspFrames);
902    }
903
904    return BAD_VALUE;
905}
906
907void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
908{
909
910    Mutex::Autolock _l(mLock);
911
912    int pid = IPCThreadState::self()->getCallingPid();
913    if (mNotificationClients.indexOfKey(pid) < 0) {
914        sp<NotificationClient> notificationClient = new NotificationClient(this,
915                                                                            client,
916                                                                            pid);
917        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
918
919        mNotificationClients.add(pid, notificationClient);
920
921        sp<IBinder> binder = client->asBinder();
922        binder->linkToDeath(notificationClient);
923
924        // the config change is always sent from playback or record threads to avoid deadlock
925        // with AudioSystem::gLock
926        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
927            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
928        }
929
930        for (size_t i = 0; i < mRecordThreads.size(); i++) {
931            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
932        }
933    }
934}
935
936void AudioFlinger::removeNotificationClient(pid_t pid)
937{
938    Mutex::Autolock _l(mLock);
939
940    int index = mNotificationClients.indexOfKey(pid);
941    if (index >= 0) {
942        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
943        ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
944        mNotificationClients.removeItem(pid);
945    }
946
947    ALOGV("%d died, releasing its sessions", pid);
948    int num = mAudioSessionRefs.size();
949    bool removed = false;
950    for (int i = 0; i< num; i++) {
951        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
952        ALOGV(" pid %d @ %d", ref->pid, i);
953        if (ref->pid == pid) {
954            ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
955            mAudioSessionRefs.removeAt(i);
956            delete ref;
957            removed = true;
958            i--;
959            num--;
960        }
961    }
962    if (removed) {
963        purgeStaleEffects_l();
964    }
965}
966
967// audioConfigChanged_l() must be called with AudioFlinger::mLock held
968void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
969{
970    size_t size = mNotificationClients.size();
971    for (size_t i = 0; i < size; i++) {
972        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
973    }
974}
975
976// removeClient_l() must be called with AudioFlinger::mLock held
977void AudioFlinger::removeClient_l(pid_t pid)
978{
979    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
980    mClients.removeItem(pid);
981}
982
983
984// ----------------------------------------------------------------------------
985
986AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
987    :   Thread(false),
988        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
989        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
990        mDevice(device)
991{
992    mDeathRecipient = new PMDeathRecipient(this);
993}
994
995AudioFlinger::ThreadBase::~ThreadBase()
996{
997    mParamCond.broadcast();
998    // do not lock the mutex in destructor
999    releaseWakeLock_l();
1000    if (mPowerManager != 0) {
1001        sp<IBinder> binder = mPowerManager->asBinder();
1002        binder->unlinkToDeath(mDeathRecipient);
1003    }
1004}
1005
1006void AudioFlinger::ThreadBase::exit()
1007{
1008    // keep a strong ref on ourself so that we won't get
1009    // destroyed in the middle of requestExitAndWait()
1010    sp <ThreadBase> strongMe = this;
1011
1012    ALOGV("ThreadBase::exit");
1013    {
1014        AutoMutex lock(mLock);
1015        mExiting = true;
1016        requestExit();
1017        mWaitWorkCV.signal();
1018    }
1019    requestExitAndWait();
1020}
1021
1022uint32_t AudioFlinger::ThreadBase::sampleRate() const
1023{
1024    return mSampleRate;
1025}
1026
1027int AudioFlinger::ThreadBase::channelCount() const
1028{
1029    return (int)mChannelCount;
1030}
1031
1032uint32_t AudioFlinger::ThreadBase::format() const
1033{
1034    return mFormat;
1035}
1036
1037size_t AudioFlinger::ThreadBase::frameCount() const
1038{
1039    return mFrameCount;
1040}
1041
1042status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1043{
1044    status_t status;
1045
1046    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1047    Mutex::Autolock _l(mLock);
1048
1049    mNewParameters.add(keyValuePairs);
1050    mWaitWorkCV.signal();
1051    // wait condition with timeout in case the thread loop has exited
1052    // before the request could be processed
1053    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
1054        status = mParamStatus;
1055        mWaitWorkCV.signal();
1056    } else {
1057        status = TIMED_OUT;
1058    }
1059    return status;
1060}
1061
1062void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1063{
1064    Mutex::Autolock _l(mLock);
1065    sendConfigEvent_l(event, param);
1066}
1067
1068// sendConfigEvent_l() must be called with ThreadBase::mLock held
1069void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1070{
1071    ConfigEvent configEvent;
1072    configEvent.mEvent = event;
1073    configEvent.mParam = param;
1074    mConfigEvents.add(configEvent);
1075    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1076    mWaitWorkCV.signal();
1077}
1078
1079void AudioFlinger::ThreadBase::processConfigEvents()
1080{
1081    mLock.lock();
1082    while(!mConfigEvents.isEmpty()) {
1083        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1084        ConfigEvent configEvent = mConfigEvents[0];
1085        mConfigEvents.removeAt(0);
1086        // release mLock before locking AudioFlinger mLock: lock order is always
1087        // AudioFlinger then ThreadBase to avoid cross deadlock
1088        mLock.unlock();
1089        mAudioFlinger->mLock.lock();
1090        audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
1091        mAudioFlinger->mLock.unlock();
1092        mLock.lock();
1093    }
1094    mLock.unlock();
1095}
1096
1097status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1098{
1099    const size_t SIZE = 256;
1100    char buffer[SIZE];
1101    String8 result;
1102
1103    bool locked = tryLock(mLock);
1104    if (!locked) {
1105        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1106        write(fd, buffer, strlen(buffer));
1107    }
1108
1109    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1110    result.append(buffer);
1111    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1112    result.append(buffer);
1113    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1114    result.append(buffer);
1115    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1116    result.append(buffer);
1117    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1118    result.append(buffer);
1119    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1120    result.append(buffer);
1121    snprintf(buffer, SIZE, "Frame size: %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    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
1386        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1387        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1388        mStreamTypes[stream].valid = true;
1389    }
1390}
1391
1392AudioFlinger::PlaybackThread::~PlaybackThread()
1393{
1394    delete [] mMixBuffer;
1395}
1396
1397status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1398{
1399    dumpInternals(fd, args);
1400    dumpTracks(fd, args);
1401    dumpEffectChains(fd, args);
1402    return NO_ERROR;
1403}
1404
1405status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1406{
1407    const size_t SIZE = 256;
1408    char buffer[SIZE];
1409    String8 result;
1410
1411    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1412    result.append(buffer);
1413    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1414    for (size_t i = 0; i < mTracks.size(); ++i) {
1415        sp<Track> track = mTracks[i];
1416        if (track != 0) {
1417            track->dump(buffer, SIZE);
1418            result.append(buffer);
1419        }
1420    }
1421
1422    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1423    result.append(buffer);
1424    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1425    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1426        wp<Track> wTrack = mActiveTracks[i];
1427        if (wTrack != 0) {
1428            sp<Track> track = wTrack.promote();
1429            if (track != 0) {
1430                track->dump(buffer, SIZE);
1431                result.append(buffer);
1432            }
1433        }
1434    }
1435    write(fd, result.string(), result.size());
1436    return NO_ERROR;
1437}
1438
1439status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1440{
1441    const size_t SIZE = 256;
1442    char buffer[SIZE];
1443    String8 result;
1444
1445    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1446    result.append(buffer);
1447    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1448    result.append(buffer);
1449    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1450    result.append(buffer);
1451    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1452    result.append(buffer);
1453    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1454    result.append(buffer);
1455    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1456    result.append(buffer);
1457    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1458    result.append(buffer);
1459    write(fd, result.string(), result.size());
1460
1461    dumpBase(fd, args);
1462
1463    return NO_ERROR;
1464}
1465
1466// Thread virtuals
1467status_t AudioFlinger::PlaybackThread::readyToRun()
1468{
1469    status_t status = initCheck();
1470    if (status == NO_ERROR) {
1471        ALOGI("AudioFlinger's thread %p ready to run", this);
1472    } else {
1473        ALOGE("No working audio driver found.");
1474    }
1475    return status;
1476}
1477
1478void AudioFlinger::PlaybackThread::onFirstRef()
1479{
1480    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1481}
1482
1483// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1484sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1485        const sp<AudioFlinger::Client>& client,
1486        int streamType,
1487        uint32_t sampleRate,
1488        uint32_t format,
1489        uint32_t channelMask,
1490        int frameCount,
1491        const sp<IMemory>& sharedBuffer,
1492        int sessionId,
1493        status_t *status)
1494{
1495    sp<Track> track;
1496    status_t lStatus;
1497
1498    if (mType == DIRECT) {
1499        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1500            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1501                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1502                        "for output %p with format %d",
1503                        sampleRate, format, channelMask, mOutput, mFormat);
1504                lStatus = BAD_VALUE;
1505                goto Exit;
1506            }
1507        }
1508    } else {
1509        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1510        if (sampleRate > mSampleRate*2) {
1511            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1512            lStatus = BAD_VALUE;
1513            goto Exit;
1514        }
1515    }
1516
1517    lStatus = initCheck();
1518    if (lStatus != NO_ERROR) {
1519        ALOGE("Audio driver not initialized.");
1520        goto Exit;
1521    }
1522
1523    { // scope for mLock
1524        Mutex::Autolock _l(mLock);
1525
1526        // all tracks in same audio session must share the same routing strategy otherwise
1527        // conflicts will happen when tracks are moved from one output to another by audio policy
1528        // manager
1529        uint32_t strategy =
1530                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1531        for (size_t i = 0; i < mTracks.size(); ++i) {
1532            sp<Track> t = mTracks[i];
1533            if (t != 0) {
1534                uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
1535                if (sessionId == t->sessionId() && strategy != actual) {
1536                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1537                            strategy, actual);
1538                    lStatus = BAD_VALUE;
1539                    goto Exit;
1540                }
1541            }
1542        }
1543
1544        track = new Track(this, client, streamType, sampleRate, format,
1545                channelMask, frameCount, sharedBuffer, sessionId);
1546        if (track->getCblk() == NULL || track->name() < 0) {
1547            lStatus = NO_MEMORY;
1548            goto Exit;
1549        }
1550        mTracks.add(track);
1551
1552        sp<EffectChain> chain = getEffectChain_l(sessionId);
1553        if (chain != 0) {
1554            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1555            track->setMainBuffer(chain->inBuffer());
1556            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1557            chain->incTrackCnt();
1558        }
1559
1560        // invalidate track immediately if the stream type was moved to another thread since
1561        // createTrack() was called by the client process.
1562        if (!mStreamTypes[streamType].valid) {
1563            ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1564                 this, streamType);
1565            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1566        }
1567    }
1568    lStatus = NO_ERROR;
1569
1570Exit:
1571    if(status) {
1572        *status = lStatus;
1573    }
1574    return track;
1575}
1576
1577uint32_t AudioFlinger::PlaybackThread::latency() const
1578{
1579    Mutex::Autolock _l(mLock);
1580    if (initCheck() == NO_ERROR) {
1581        return mOutput->stream->get_latency(mOutput->stream);
1582    } else {
1583        return 0;
1584    }
1585}
1586
1587status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1588{
1589    mMasterVolume = value;
1590    return NO_ERROR;
1591}
1592
1593status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1594{
1595    mMasterMute = muted;
1596    return NO_ERROR;
1597}
1598
1599float AudioFlinger::PlaybackThread::masterVolume() const
1600{
1601    return mMasterVolume;
1602}
1603
1604bool AudioFlinger::PlaybackThread::masterMute() const
1605{
1606    return mMasterMute;
1607}
1608
1609status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
1610{
1611    mStreamTypes[stream].volume = value;
1612    return NO_ERROR;
1613}
1614
1615status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
1616{
1617    mStreamTypes[stream].mute = muted;
1618    return NO_ERROR;
1619}
1620
1621float AudioFlinger::PlaybackThread::streamVolume(int stream) const
1622{
1623    return mStreamTypes[stream].volume;
1624}
1625
1626bool AudioFlinger::PlaybackThread::streamMute(int stream) const
1627{
1628    return mStreamTypes[stream].mute;
1629}
1630
1631// addTrack_l() must be called with ThreadBase::mLock held
1632status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1633{
1634    status_t status = ALREADY_EXISTS;
1635
1636    // set retry count for buffer fill
1637    track->mRetryCount = kMaxTrackStartupRetries;
1638    if (mActiveTracks.indexOf(track) < 0) {
1639        // the track is newly added, make sure it fills up all its
1640        // buffers before playing. This is to ensure the client will
1641        // effectively get the latency it requested.
1642        track->mFillingUpStatus = Track::FS_FILLING;
1643        track->mResetDone = false;
1644        mActiveTracks.add(track);
1645        if (track->mainBuffer() != mMixBuffer) {
1646            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1647            if (chain != 0) {
1648                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1649                chain->incActiveTrackCnt();
1650            }
1651        }
1652
1653        status = NO_ERROR;
1654    }
1655
1656    ALOGV("mWaitWorkCV.broadcast");
1657    mWaitWorkCV.broadcast();
1658
1659    return status;
1660}
1661
1662// destroyTrack_l() must be called with ThreadBase::mLock held
1663void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1664{
1665    track->mState = TrackBase::TERMINATED;
1666    if (mActiveTracks.indexOf(track) < 0) {
1667        removeTrack_l(track);
1668    }
1669}
1670
1671void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1672{
1673    mTracks.remove(track);
1674    deleteTrackName_l(track->name());
1675    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1676    if (chain != 0) {
1677        chain->decTrackCnt();
1678    }
1679}
1680
1681String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1682{
1683    String8 out_s8 = String8("");
1684    char *s;
1685
1686    Mutex::Autolock _l(mLock);
1687    if (initCheck() != NO_ERROR) {
1688        return out_s8;
1689    }
1690
1691    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1692    out_s8 = String8(s);
1693    free(s);
1694    return out_s8;
1695}
1696
1697// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1698void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1699    AudioSystem::OutputDescriptor desc;
1700    void *param2 = 0;
1701
1702    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1703
1704    switch (event) {
1705    case AudioSystem::OUTPUT_OPENED:
1706    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1707        desc.channels = mChannelMask;
1708        desc.samplingRate = mSampleRate;
1709        desc.format = mFormat;
1710        desc.frameCount = mFrameCount;
1711        desc.latency = latency();
1712        param2 = &desc;
1713        break;
1714
1715    case AudioSystem::STREAM_CONFIG_CHANGED:
1716        param2 = &param;
1717    case AudioSystem::OUTPUT_CLOSED:
1718    default:
1719        break;
1720    }
1721    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1722}
1723
1724void AudioFlinger::PlaybackThread::readOutputParameters()
1725{
1726    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1727    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1728    mChannelCount = (uint16_t)popcount(mChannelMask);
1729    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1730    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
1731    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1732
1733    // FIXME - Current mixer implementation only supports stereo output: Always
1734    // Allocate a stereo buffer even if HW output is mono.
1735    if (mMixBuffer != NULL) delete[] mMixBuffer;
1736    mMixBuffer = new int16_t[mFrameCount * 2];
1737    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1738
1739    // force reconfiguration of effect chains and engines to take new buffer size and audio
1740    // parameters into account
1741    // Note that mLock is not held when readOutputParameters() is called from the constructor
1742    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1743    // matter.
1744    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1745    Vector< sp<EffectChain> > effectChains = mEffectChains;
1746    for (size_t i = 0; i < effectChains.size(); i ++) {
1747        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1748    }
1749}
1750
1751status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1752{
1753    if (halFrames == 0 || dspFrames == 0) {
1754        return BAD_VALUE;
1755    }
1756    Mutex::Autolock _l(mLock);
1757    if (initCheck() != NO_ERROR) {
1758        return INVALID_OPERATION;
1759    }
1760    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1761
1762    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1763}
1764
1765uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1766{
1767    Mutex::Autolock _l(mLock);
1768    uint32_t result = 0;
1769    if (getEffectChain_l(sessionId) != 0) {
1770        result = EFFECT_SESSION;
1771    }
1772
1773    for (size_t i = 0; i < mTracks.size(); ++i) {
1774        sp<Track> track = mTracks[i];
1775        if (sessionId == track->sessionId() &&
1776                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1777            result |= TRACK_SESSION;
1778            break;
1779        }
1780    }
1781
1782    return result;
1783}
1784
1785uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1786{
1787    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1788    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1789    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1790        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1791    }
1792    for (size_t i = 0; i < mTracks.size(); i++) {
1793        sp<Track> track = mTracks[i];
1794        if (sessionId == track->sessionId() &&
1795                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1796            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1797        }
1798    }
1799    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1800}
1801
1802
1803AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1804{
1805    Mutex::Autolock _l(mLock);
1806    return mOutput;
1807}
1808
1809AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1810{
1811    Mutex::Autolock _l(mLock);
1812    AudioStreamOut *output = mOutput;
1813    mOutput = NULL;
1814    return output;
1815}
1816
1817// this method must always be called either with ThreadBase mLock held or inside the thread loop
1818audio_stream_t* AudioFlinger::PlaybackThread::stream()
1819{
1820    if (mOutput == NULL) {
1821        return NULL;
1822    }
1823    return &mOutput->stream->common;
1824}
1825
1826uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
1827{
1828    // A2DP output latency is not due only to buffering capacity. It also reflects encoding,
1829    // decoding and transfer time. So sleeping for half of the latency would likely cause
1830    // underruns
1831    if (audio_is_a2dp_device((audio_devices_t)mDevice)) {
1832        return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000);
1833    } else {
1834        return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
1835    }
1836}
1837
1838// ----------------------------------------------------------------------------
1839
1840AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1841    :   PlaybackThread(audioFlinger, output, id, device),
1842        mAudioMixer(NULL)
1843{
1844    mType = ThreadBase::MIXER;
1845    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1846
1847    // FIXME - Current mixer implementation only supports stereo output
1848    if (mChannelCount == 1) {
1849        ALOGE("Invalid audio hardware channel count");
1850    }
1851}
1852
1853AudioFlinger::MixerThread::~MixerThread()
1854{
1855    delete mAudioMixer;
1856}
1857
1858bool AudioFlinger::MixerThread::threadLoop()
1859{
1860    Vector< sp<Track> > tracksToRemove;
1861    uint32_t mixerStatus = MIXER_IDLE;
1862    nsecs_t standbyTime = systemTime();
1863    size_t mixBufferSize = mFrameCount * mFrameSize;
1864    // FIXME: Relaxed timing because of a certain device that can't meet latency
1865    // Should be reduced to 2x after the vendor fixes the driver issue
1866    // increase threshold again due to low power audio mode. The way this warning threshold is
1867    // calculated and its usefulness should be reconsidered anyway.
1868    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1869    nsecs_t lastWarning = 0;
1870    bool longStandbyExit = false;
1871    uint32_t activeSleepTime = activeSleepTimeUs();
1872    uint32_t idleSleepTime = idleSleepTimeUs();
1873    uint32_t sleepTime = idleSleepTime;
1874    uint32_t sleepTimeShift = 0;
1875    Vector< sp<EffectChain> > effectChains;
1876#ifdef DEBUG_CPU_USAGE
1877    ThreadCpuUsage cpu;
1878    const CentralTendencyStatistics& stats = cpu.statistics();
1879#endif
1880
1881    acquireWakeLock();
1882
1883    while (!exitPending())
1884    {
1885#ifdef DEBUG_CPU_USAGE
1886        cpu.sampleAndEnable();
1887        unsigned n = stats.n();
1888        // cpu.elapsed() is expensive, so don't call it every loop
1889        if ((n & 127) == 1) {
1890            long long elapsed = cpu.elapsed();
1891            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1892                double perLoop = elapsed / (double) n;
1893                double perLoop100 = perLoop * 0.01;
1894                double mean = stats.mean();
1895                double stddev = stats.stddev();
1896                double minimum = stats.minimum();
1897                double maximum = stats.maximum();
1898                cpu.resetStatistics();
1899                ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
1900                        elapsed * .000000001, n, perLoop * .000001,
1901                        mean * .001,
1902                        stddev * .001,
1903                        minimum * .001,
1904                        maximum * .001,
1905                        mean / perLoop100,
1906                        stddev / perLoop100,
1907                        minimum / perLoop100,
1908                        maximum / perLoop100);
1909            }
1910        }
1911#endif
1912        processConfigEvents();
1913
1914        mixerStatus = MIXER_IDLE;
1915        { // scope for mLock
1916
1917            Mutex::Autolock _l(mLock);
1918
1919            if (checkForNewParameters_l()) {
1920                mixBufferSize = mFrameCount * mFrameSize;
1921                // FIXME: Relaxed timing because of a certain device that can't meet latency
1922                // Should be reduced to 2x after the vendor fixes the driver issue
1923                // increase threshold again due to low power audio mode. The way this warning
1924                // threshold is calculated and its usefulness should be reconsidered anyway.
1925                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1926                activeSleepTime = activeSleepTimeUs();
1927                idleSleepTime = idleSleepTimeUs();
1928            }
1929
1930            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1931
1932            // put audio hardware into standby after short delay
1933            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1934                        mSuspended)) {
1935                if (!mStandby) {
1936                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1937                    mOutput->stream->common.standby(&mOutput->stream->common);
1938                    mStandby = true;
1939                    mBytesWritten = 0;
1940                }
1941
1942                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1943                    // we're about to wait, flush the binder command buffer
1944                    IPCThreadState::self()->flushCommands();
1945
1946                    if (exitPending()) break;
1947
1948                    releaseWakeLock_l();
1949                    // wait until we have something to do...
1950                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1951                    mWaitWorkCV.wait(mLock);
1952                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1953                    acquireWakeLock_l();
1954
1955                    if (mMasterMute == false) {
1956                        char value[PROPERTY_VALUE_MAX];
1957                        property_get("ro.audio.silent", value, "0");
1958                        if (atoi(value)) {
1959                            ALOGD("Silence is golden");
1960                            setMasterMute(true);
1961                        }
1962                    }
1963
1964                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1965                    sleepTime = idleSleepTime;
1966                    sleepTimeShift = 0;
1967                    continue;
1968                }
1969            }
1970
1971            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1972
1973            // prevent any changes in effect chain list and in each effect chain
1974            // during mixing and effect process as the audio buffers could be deleted
1975            // or modified if an effect is created or deleted
1976            lockEffectChains_l(effectChains);
1977        }
1978
1979        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1980            // mix buffers...
1981            mAudioMixer->process();
1982            sleepTime = 0;
1983            // increase sleep time progressively when application underrun condition clears
1984            if (sleepTimeShift > 0) {
1985                sleepTimeShift--;
1986            }
1987            standbyTime = systemTime() + kStandbyTimeInNsecs;
1988            //TODO: delay standby when effects have a tail
1989        } else {
1990            // If no tracks are ready, sleep once for the duration of an output
1991            // buffer size, then write 0s to the output
1992            if (sleepTime == 0) {
1993                if (mixerStatus == MIXER_TRACKS_ENABLED) {
1994                    sleepTime = activeSleepTime >> sleepTimeShift;
1995                    if (sleepTime < kMinThreadSleepTimeUs) {
1996                        sleepTime = kMinThreadSleepTimeUs;
1997                    }
1998                    // reduce sleep time in case of consecutive application underruns to avoid
1999                    // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
2000                    // duration we would end up writing less data than needed by the audio HAL if
2001                    // the condition persists.
2002                    if (sleepTimeShift < kMaxThreadSleepTimeShift) {
2003                        sleepTimeShift++;
2004                    }
2005                } else {
2006                    sleepTime = idleSleepTime;
2007                }
2008            } else if (mBytesWritten != 0 ||
2009                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
2010                memset (mMixBuffer, 0, mixBufferSize);
2011                sleepTime = 0;
2012                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
2013            }
2014            // TODO add standby time extension fct of effect tail
2015        }
2016
2017        if (mSuspended) {
2018            sleepTime = suspendSleepTimeUs();
2019        }
2020        // sleepTime == 0 means we must write to audio hardware
2021        if (sleepTime == 0) {
2022            for (size_t i = 0; i < effectChains.size(); i ++) {
2023                effectChains[i]->process_l();
2024            }
2025            // enable changes in effect chain
2026            unlockEffectChains(effectChains);
2027            mLastWriteTime = systemTime();
2028            mInWrite = true;
2029            mBytesWritten += mixBufferSize;
2030
2031            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2032            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2033            mNumWrites++;
2034            mInWrite = false;
2035            nsecs_t now = systemTime();
2036            nsecs_t delta = now - mLastWriteTime;
2037            if (!mStandby && delta > maxPeriod) {
2038                mNumDelayedWrites++;
2039                if ((now - lastWarning) > kWarningThrottleNs) {
2040                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2041                            ns2ms(delta), mNumDelayedWrites, this);
2042                    lastWarning = now;
2043                }
2044                if (mStandby) {
2045                    longStandbyExit = true;
2046                }
2047            }
2048            mStandby = false;
2049        } else {
2050            // enable changes in effect chain
2051            unlockEffectChains(effectChains);
2052            usleep(sleepTime);
2053        }
2054
2055        // finally let go of all our tracks, without the lock held
2056        // since we can't guarantee the destructors won't acquire that
2057        // same lock.
2058        tracksToRemove.clear();
2059
2060        // Effect chains will be actually deleted here if they were removed from
2061        // mEffectChains list during mixing or effects processing
2062        effectChains.clear();
2063    }
2064
2065    if (!mStandby) {
2066        mOutput->stream->common.standby(&mOutput->stream->common);
2067    }
2068
2069    releaseWakeLock();
2070
2071    ALOGV("MixerThread %p exiting", this);
2072    return false;
2073}
2074
2075// prepareTracks_l() must be called with ThreadBase::mLock held
2076uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2077{
2078
2079    uint32_t mixerStatus = MIXER_IDLE;
2080    // find out which tracks need to be processed
2081    size_t count = activeTracks.size();
2082    size_t mixedTracks = 0;
2083    size_t tracksWithEffect = 0;
2084
2085    float masterVolume = mMasterVolume;
2086    bool  masterMute = mMasterMute;
2087
2088    if (masterMute) {
2089        masterVolume = 0;
2090    }
2091    // Delegate master volume control to effect in output mix effect chain if needed
2092    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2093    if (chain != 0) {
2094        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2095        chain->setVolume_l(&v, &v);
2096        masterVolume = (float)((v + (1 << 23)) >> 24);
2097        chain.clear();
2098    }
2099
2100    for (size_t i=0 ; i<count ; i++) {
2101        sp<Track> t = activeTracks[i].promote();
2102        if (t == 0) continue;
2103
2104        // this const just means the local variable doesn't change
2105        Track* const track = t.get();
2106        audio_track_cblk_t* cblk = track->cblk();
2107
2108        // The first time a track is added we wait
2109        // for all its buffers to be filled before processing it
2110        int name = track->name();
2111        // make sure that we have enough frames to mix one full buffer.
2112        // enforce this condition only once to enable draining the buffer in case the client
2113        // app does not call stop() and relies on underrun to stop:
2114        // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed
2115        // during last round
2116        uint32_t minFrames = 1;
2117        if (!track->isStopped() && !track->isPausing() &&
2118                (track->mRetryCount >= kMaxTrackRetries)) {
2119            if (t->sampleRate() == (int)mSampleRate) {
2120                minFrames = mFrameCount;
2121            } else {
2122                // +1 for rounding and +1 for additional sample needed for interpolation
2123                minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
2124                // add frames already consumed but not yet released by the resampler
2125                // because cblk->framesReady() will  include these frames
2126                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
2127                // the minimum track buffer size is normally twice the number of frames necessary
2128                // to fill one buffer and the resampler should not leave more than one buffer worth
2129                // of unreleased frames after each pass, but just in case...
2130                ALOG_ASSERT(minFrames <= cblk->frameCount);
2131            }
2132        }
2133        if ((cblk->framesReady() >= minFrames) && track->isReady() &&
2134                !track->isPaused() && !track->isTerminated())
2135        {
2136            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
2137
2138            mixedTracks++;
2139
2140            // track->mainBuffer() != mMixBuffer means there is an effect chain
2141            // connected to the track
2142            chain.clear();
2143            if (track->mainBuffer() != mMixBuffer) {
2144                chain = getEffectChain_l(track->sessionId());
2145                // Delegate volume control to effect in track effect chain if needed
2146                if (chain != 0) {
2147                    tracksWithEffect++;
2148                } else {
2149                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
2150                            name, track->sessionId());
2151                }
2152            }
2153
2154
2155            int param = AudioMixer::VOLUME;
2156            if (track->mFillingUpStatus == Track::FS_FILLED) {
2157                // no ramp for the first volume setting
2158                track->mFillingUpStatus = Track::FS_ACTIVE;
2159                if (track->mState == TrackBase::RESUMING) {
2160                    track->mState = TrackBase::ACTIVE;
2161                    param = AudioMixer::RAMP_VOLUME;
2162                }
2163                mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2164            } else if (cblk->server != 0) {
2165                // If the track is stopped before the first frame was mixed,
2166                // do not apply ramp
2167                param = AudioMixer::RAMP_VOLUME;
2168            }
2169
2170            // compute volume for this track
2171            uint32_t vl, vr, va;
2172            if (track->isMuted() || track->isPausing() ||
2173                mStreamTypes[track->type()].mute) {
2174                vl = vr = va = 0;
2175                if (track->isPausing()) {
2176                    track->setPaused();
2177                }
2178            } else {
2179
2180                // read original volumes with volume control
2181                float typeVolume = mStreamTypes[track->type()].volume;
2182                float v = masterVolume * typeVolume;
2183                vl = (uint32_t)(v * cblk->volume[0]) << 12;
2184                vr = (uint32_t)(v * cblk->volume[1]) << 12;
2185
2186                va = (uint32_t)(v * cblk->sendLevel);
2187            }
2188            // Delegate volume control to effect in track effect chain if needed
2189            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2190                // Do not ramp volume if volume is controlled by effect
2191                param = AudioMixer::VOLUME;
2192                track->mHasVolumeController = true;
2193            } else {
2194                // force no volume ramp when volume controller was just disabled or removed
2195                // from effect chain to avoid volume spike
2196                if (track->mHasVolumeController) {
2197                    param = AudioMixer::VOLUME;
2198                }
2199                track->mHasVolumeController = false;
2200            }
2201
2202            // Convert volumes from 8.24 to 4.12 format
2203            int16_t left, right, aux;
2204            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2205            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2206            left = int16_t(v_clamped);
2207            v_clamped = (vr + (1 << 11)) >> 12;
2208            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2209            right = int16_t(v_clamped);
2210
2211            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2212            aux = int16_t(va);
2213
2214            // XXX: these things DON'T need to be done each time
2215            mAudioMixer->setBufferProvider(name, track);
2216            mAudioMixer->enable(name);
2217
2218            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left);
2219            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right);
2220            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux);
2221            mAudioMixer->setParameter(
2222                name,
2223                AudioMixer::TRACK,
2224                AudioMixer::FORMAT, (void *)track->format());
2225            mAudioMixer->setParameter(
2226                name,
2227                AudioMixer::TRACK,
2228                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2229            mAudioMixer->setParameter(
2230                name,
2231                AudioMixer::RESAMPLE,
2232                AudioMixer::SAMPLE_RATE,
2233                (void *)(cblk->sampleRate));
2234            mAudioMixer->setParameter(
2235                name,
2236                AudioMixer::TRACK,
2237                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2238            mAudioMixer->setParameter(
2239                name,
2240                AudioMixer::TRACK,
2241                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2242
2243            // reset retry count
2244            track->mRetryCount = kMaxTrackRetries;
2245            mixerStatus = MIXER_TRACKS_READY;
2246        } else {
2247            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
2248            if (track->isStopped()) {
2249                track->reset();
2250            }
2251            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2252                // We have consumed all the buffers of this track.
2253                // Remove it from the list of active tracks.
2254                tracksToRemove->add(track);
2255            } else {
2256                // No buffers for this track. Give it a few chances to
2257                // fill a buffer, then remove it from active list.
2258                if (--(track->mRetryCount) <= 0) {
2259                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
2260                    tracksToRemove->add(track);
2261                    // indicate to client process that the track was disabled because of underrun
2262                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2263                } else if (mixerStatus != MIXER_TRACKS_READY) {
2264                    mixerStatus = MIXER_TRACKS_ENABLED;
2265                }
2266            }
2267            mAudioMixer->disable(name);
2268        }
2269    }
2270
2271    // remove all the tracks that need to be...
2272    count = tracksToRemove->size();
2273    if (CC_UNLIKELY(count)) {
2274        for (size_t i=0 ; i<count ; i++) {
2275            const sp<Track>& track = tracksToRemove->itemAt(i);
2276            mActiveTracks.remove(track);
2277            if (track->mainBuffer() != mMixBuffer) {
2278                chain = getEffectChain_l(track->sessionId());
2279                if (chain != 0) {
2280                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2281                    chain->decActiveTrackCnt();
2282                }
2283            }
2284            if (track->isTerminated()) {
2285                removeTrack_l(track);
2286            }
2287        }
2288    }
2289
2290    // mix buffer must be cleared if all tracks are connected to an
2291    // effect chain as in this case the mixer will not write to
2292    // mix buffer and track effects will accumulate into it
2293    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2294        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2295    }
2296
2297    return mixerStatus;
2298}
2299
2300void AudioFlinger::MixerThread::invalidateTracks(int streamType)
2301{
2302    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2303            this,  streamType, mTracks.size());
2304    Mutex::Autolock _l(mLock);
2305
2306    size_t size = mTracks.size();
2307    for (size_t i = 0; i < size; i++) {
2308        sp<Track> t = mTracks[i];
2309        if (t->type() == streamType) {
2310            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2311            t->mCblk->cv.signal();
2312        }
2313    }
2314}
2315
2316void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid)
2317{
2318    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2319            this,  streamType, valid);
2320    Mutex::Autolock _l(mLock);
2321
2322    mStreamTypes[streamType].valid = valid;
2323}
2324
2325// getTrackName_l() must be called with ThreadBase::mLock held
2326int AudioFlinger::MixerThread::getTrackName_l()
2327{
2328    return mAudioMixer->getTrackName();
2329}
2330
2331// deleteTrackName_l() must be called with ThreadBase::mLock held
2332void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2333{
2334    ALOGV("remove track (%d) and delete from mixer", name);
2335    mAudioMixer->deleteTrackName(name);
2336}
2337
2338// checkForNewParameters_l() must be called with ThreadBase::mLock held
2339bool AudioFlinger::MixerThread::checkForNewParameters_l()
2340{
2341    bool reconfig = false;
2342
2343    while (!mNewParameters.isEmpty()) {
2344        status_t status = NO_ERROR;
2345        String8 keyValuePair = mNewParameters[0];
2346        AudioParameter param = AudioParameter(keyValuePair);
2347        int value;
2348
2349        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2350            reconfig = true;
2351        }
2352        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2353            if (value != AUDIO_FORMAT_PCM_16_BIT) {
2354                status = BAD_VALUE;
2355            } else {
2356                reconfig = true;
2357            }
2358        }
2359        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2360            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2361                status = BAD_VALUE;
2362            } else {
2363                reconfig = true;
2364            }
2365        }
2366        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2367            // do not accept frame count changes if tracks are open as the track buffer
2368            // size depends on frame count and correct behavior would not be guaranteed
2369            // if frame count is changed after track creation
2370            if (!mTracks.isEmpty()) {
2371                status = INVALID_OPERATION;
2372            } else {
2373                reconfig = true;
2374            }
2375        }
2376        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2377            // when changing the audio output device, call addBatteryData to notify
2378            // the change
2379            if ((int)mDevice != value) {
2380                uint32_t params = 0;
2381                // check whether speaker is on
2382                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2383                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2384                }
2385
2386                int deviceWithoutSpeaker
2387                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2388                // check if any other device (except speaker) is on
2389                if (value & deviceWithoutSpeaker ) {
2390                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2391                }
2392
2393                if (params != 0) {
2394                    addBatteryData(params);
2395                }
2396            }
2397
2398            // forward device change to effects that have requested to be
2399            // aware of attached audio device.
2400            mDevice = (uint32_t)value;
2401            for (size_t i = 0; i < mEffectChains.size(); i++) {
2402                mEffectChains[i]->setDevice_l(mDevice);
2403            }
2404        }
2405
2406        if (status == NO_ERROR) {
2407            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2408                                                    keyValuePair.string());
2409            if (!mStandby && status == INVALID_OPERATION) {
2410               mOutput->stream->common.standby(&mOutput->stream->common);
2411               mStandby = true;
2412               mBytesWritten = 0;
2413               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2414                                                       keyValuePair.string());
2415            }
2416            if (status == NO_ERROR && reconfig) {
2417                delete mAudioMixer;
2418                readOutputParameters();
2419                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2420                for (size_t i = 0; i < mTracks.size() ; i++) {
2421                    int name = getTrackName_l();
2422                    if (name < 0) break;
2423                    mTracks[i]->mName = name;
2424                    // limit track sample rate to 2 x new output sample rate
2425                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2426                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2427                    }
2428                }
2429                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2430            }
2431        }
2432
2433        mNewParameters.removeAt(0);
2434
2435        mParamStatus = status;
2436        mParamCond.signal();
2437        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2438        // already timed out waiting for the status and will never signal the condition.
2439        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2440    }
2441    return reconfig;
2442}
2443
2444status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2445{
2446    const size_t SIZE = 256;
2447    char buffer[SIZE];
2448    String8 result;
2449
2450    PlaybackThread::dumpInternals(fd, args);
2451
2452    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2453    result.append(buffer);
2454    write(fd, result.string(), result.size());
2455    return NO_ERROR;
2456}
2457
2458uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2459{
2460    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2461}
2462
2463uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2464{
2465    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2466}
2467
2468// ----------------------------------------------------------------------------
2469AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2470    :   PlaybackThread(audioFlinger, output, id, device)
2471{
2472    mType = ThreadBase::DIRECT;
2473}
2474
2475AudioFlinger::DirectOutputThread::~DirectOutputThread()
2476{
2477}
2478
2479static inline
2480int32_t mul(int16_t in, int16_t v)
2481{
2482#if defined(__arm__) && !defined(__thumb__)
2483    int32_t out;
2484    asm( "smulbb %[out], %[in], %[v] \n"
2485         : [out]"=r"(out)
2486         : [in]"%r"(in), [v]"r"(v)
2487         : );
2488    return out;
2489#else
2490    return in * int32_t(v);
2491#endif
2492}
2493
2494void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2495{
2496    // Do not apply volume on compressed audio
2497    if (!audio_is_linear_pcm(mFormat)) {
2498        return;
2499    }
2500
2501    // convert to signed 16 bit before volume calculation
2502    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2503        size_t count = mFrameCount * mChannelCount;
2504        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2505        int16_t *dst = mMixBuffer + count-1;
2506        while(count--) {
2507            *dst-- = (int16_t)(*src--^0x80) << 8;
2508        }
2509    }
2510
2511    size_t frameCount = mFrameCount;
2512    int16_t *out = mMixBuffer;
2513    if (ramp) {
2514        if (mChannelCount == 1) {
2515            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2516            int32_t vlInc = d / (int32_t)frameCount;
2517            int32_t vl = ((int32_t)mLeftVolShort << 16);
2518            do {
2519                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2520                out++;
2521                vl += vlInc;
2522            } while (--frameCount);
2523
2524        } else {
2525            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2526            int32_t vlInc = d / (int32_t)frameCount;
2527            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2528            int32_t vrInc = d / (int32_t)frameCount;
2529            int32_t vl = ((int32_t)mLeftVolShort << 16);
2530            int32_t vr = ((int32_t)mRightVolShort << 16);
2531            do {
2532                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2533                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2534                out += 2;
2535                vl += vlInc;
2536                vr += vrInc;
2537            } while (--frameCount);
2538        }
2539    } else {
2540        if (mChannelCount == 1) {
2541            do {
2542                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2543                out++;
2544            } while (--frameCount);
2545        } else {
2546            do {
2547                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2548                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2549                out += 2;
2550            } while (--frameCount);
2551        }
2552    }
2553
2554    // convert back to unsigned 8 bit after volume calculation
2555    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2556        size_t count = mFrameCount * mChannelCount;
2557        int16_t *src = mMixBuffer;
2558        uint8_t *dst = (uint8_t *)mMixBuffer;
2559        while(count--) {
2560            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2561        }
2562    }
2563
2564    mLeftVolShort = leftVol;
2565    mRightVolShort = rightVol;
2566}
2567
2568bool AudioFlinger::DirectOutputThread::threadLoop()
2569{
2570    uint32_t mixerStatus = MIXER_IDLE;
2571    sp<Track> trackToRemove;
2572    sp<Track> activeTrack;
2573    nsecs_t standbyTime = systemTime();
2574    int8_t *curBuf;
2575    size_t mixBufferSize = mFrameCount*mFrameSize;
2576    uint32_t activeSleepTime = activeSleepTimeUs();
2577    uint32_t idleSleepTime = idleSleepTimeUs();
2578    uint32_t sleepTime = idleSleepTime;
2579    // use shorter standby delay as on normal output to release
2580    // hardware resources as soon as possible
2581    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2582
2583    acquireWakeLock();
2584
2585    while (!exitPending())
2586    {
2587        bool rampVolume;
2588        uint16_t leftVol;
2589        uint16_t rightVol;
2590        Vector< sp<EffectChain> > effectChains;
2591
2592        processConfigEvents();
2593
2594        mixerStatus = MIXER_IDLE;
2595
2596        { // scope for the mLock
2597
2598            Mutex::Autolock _l(mLock);
2599
2600            if (checkForNewParameters_l()) {
2601                mixBufferSize = mFrameCount*mFrameSize;
2602                activeSleepTime = activeSleepTimeUs();
2603                idleSleepTime = idleSleepTimeUs();
2604                standbyDelay = microseconds(activeSleepTime*2);
2605            }
2606
2607            // put audio hardware into standby after short delay
2608            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2609                        mSuspended)) {
2610                // wait until we have something to do...
2611                if (!mStandby) {
2612                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2613                    mOutput->stream->common.standby(&mOutput->stream->common);
2614                    mStandby = true;
2615                    mBytesWritten = 0;
2616                }
2617
2618                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2619                    // we're about to wait, flush the binder command buffer
2620                    IPCThreadState::self()->flushCommands();
2621
2622                    if (exitPending()) break;
2623
2624                    releaseWakeLock_l();
2625                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2626                    mWaitWorkCV.wait(mLock);
2627                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2628                    acquireWakeLock_l();
2629
2630                    if (mMasterMute == false) {
2631                        char value[PROPERTY_VALUE_MAX];
2632                        property_get("ro.audio.silent", value, "0");
2633                        if (atoi(value)) {
2634                            ALOGD("Silence is golden");
2635                            setMasterMute(true);
2636                        }
2637                    }
2638
2639                    standbyTime = systemTime() + standbyDelay;
2640                    sleepTime = idleSleepTime;
2641                    continue;
2642                }
2643            }
2644
2645            effectChains = mEffectChains;
2646
2647            // find out which tracks need to be processed
2648            if (mActiveTracks.size() != 0) {
2649                sp<Track> t = mActiveTracks[0].promote();
2650                if (t == 0) continue;
2651
2652                Track* const track = t.get();
2653                audio_track_cblk_t* cblk = track->cblk();
2654
2655                // The first time a track is added we wait
2656                // for all its buffers to be filled before processing it
2657                if (cblk->framesReady() && track->isReady() &&
2658                        !track->isPaused() && !track->isTerminated())
2659                {
2660                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2661
2662                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2663                        track->mFillingUpStatus = Track::FS_ACTIVE;
2664                        mLeftVolFloat = mRightVolFloat = 0;
2665                        mLeftVolShort = mRightVolShort = 0;
2666                        if (track->mState == TrackBase::RESUMING) {
2667                            track->mState = TrackBase::ACTIVE;
2668                            rampVolume = true;
2669                        }
2670                    } else if (cblk->server != 0) {
2671                        // If the track is stopped before the first frame was mixed,
2672                        // do not apply ramp
2673                        rampVolume = true;
2674                    }
2675                    // compute volume for this track
2676                    float left, right;
2677                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2678                        mStreamTypes[track->type()].mute) {
2679                        left = right = 0;
2680                        if (track->isPausing()) {
2681                            track->setPaused();
2682                        }
2683                    } else {
2684                        float typeVolume = mStreamTypes[track->type()].volume;
2685                        float v = mMasterVolume * typeVolume;
2686                        float v_clamped = v * cblk->volume[0];
2687                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2688                        left = v_clamped/MAX_GAIN;
2689                        v_clamped = v * cblk->volume[1];
2690                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2691                        right = v_clamped/MAX_GAIN;
2692                    }
2693
2694                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2695                        mLeftVolFloat = left;
2696                        mRightVolFloat = right;
2697
2698                        // If audio HAL implements volume control,
2699                        // force software volume to nominal value
2700                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2701                            left = 1.0f;
2702                            right = 1.0f;
2703                        }
2704
2705                        // Convert volumes from float to 8.24
2706                        uint32_t vl = (uint32_t)(left * (1 << 24));
2707                        uint32_t vr = (uint32_t)(right * (1 << 24));
2708
2709                        // Delegate volume control to effect in track effect chain if needed
2710                        // only one effect chain can be present on DirectOutputThread, so if
2711                        // there is one, the track is connected to it
2712                        if (!effectChains.isEmpty()) {
2713                            // Do not ramp volume if volume is controlled by effect
2714                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2715                                rampVolume = false;
2716                            }
2717                        }
2718
2719                        // Convert volumes from 8.24 to 4.12 format
2720                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2721                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2722                        leftVol = (uint16_t)v_clamped;
2723                        v_clamped = (vr + (1 << 11)) >> 12;
2724                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2725                        rightVol = (uint16_t)v_clamped;
2726                    } else {
2727                        leftVol = mLeftVolShort;
2728                        rightVol = mRightVolShort;
2729                        rampVolume = false;
2730                    }
2731
2732                    // reset retry count
2733                    track->mRetryCount = kMaxTrackRetriesDirect;
2734                    activeTrack = t;
2735                    mixerStatus = MIXER_TRACKS_READY;
2736                } else {
2737                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2738                    if (track->isStopped()) {
2739                        track->reset();
2740                    }
2741                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2742                        // We have consumed all the buffers of this track.
2743                        // Remove it from the list of active tracks.
2744                        trackToRemove = track;
2745                    } else {
2746                        // No buffers for this track. Give it a few chances to
2747                        // fill a buffer, then remove it from active list.
2748                        if (--(track->mRetryCount) <= 0) {
2749                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2750                            trackToRemove = track;
2751                        } else {
2752                            mixerStatus = MIXER_TRACKS_ENABLED;
2753                        }
2754                    }
2755                }
2756            }
2757
2758            // remove all the tracks that need to be...
2759            if (CC_UNLIKELY(trackToRemove != 0)) {
2760                mActiveTracks.remove(trackToRemove);
2761                if (!effectChains.isEmpty()) {
2762                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2763                            trackToRemove->sessionId());
2764                    effectChains[0]->decActiveTrackCnt();
2765                }
2766                if (trackToRemove->isTerminated()) {
2767                    removeTrack_l(trackToRemove);
2768                }
2769            }
2770
2771            lockEffectChains_l(effectChains);
2772       }
2773
2774        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2775            AudioBufferProvider::Buffer buffer;
2776            size_t frameCount = mFrameCount;
2777            curBuf = (int8_t *)mMixBuffer;
2778            // output audio to hardware
2779            while (frameCount) {
2780                buffer.frameCount = frameCount;
2781                activeTrack->getNextBuffer(&buffer);
2782                if (CC_UNLIKELY(buffer.raw == NULL)) {
2783                    memset(curBuf, 0, frameCount * mFrameSize);
2784                    break;
2785                }
2786                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2787                frameCount -= buffer.frameCount;
2788                curBuf += buffer.frameCount * mFrameSize;
2789                activeTrack->releaseBuffer(&buffer);
2790            }
2791            sleepTime = 0;
2792            standbyTime = systemTime() + standbyDelay;
2793        } else {
2794            if (sleepTime == 0) {
2795                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2796                    sleepTime = activeSleepTime;
2797                } else {
2798                    sleepTime = idleSleepTime;
2799                }
2800            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2801                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2802                sleepTime = 0;
2803            }
2804        }
2805
2806        if (mSuspended) {
2807            sleepTime = suspendSleepTimeUs();
2808        }
2809        // sleepTime == 0 means we must write to audio hardware
2810        if (sleepTime == 0) {
2811            if (mixerStatus == MIXER_TRACKS_READY) {
2812                applyVolume(leftVol, rightVol, rampVolume);
2813            }
2814            for (size_t i = 0; i < effectChains.size(); i ++) {
2815                effectChains[i]->process_l();
2816            }
2817            unlockEffectChains(effectChains);
2818
2819            mLastWriteTime = systemTime();
2820            mInWrite = true;
2821            mBytesWritten += mixBufferSize;
2822            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2823            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2824            mNumWrites++;
2825            mInWrite = false;
2826            mStandby = false;
2827        } else {
2828            unlockEffectChains(effectChains);
2829            usleep(sleepTime);
2830        }
2831
2832        // finally let go of removed track, without the lock held
2833        // since we can't guarantee the destructors won't acquire that
2834        // same lock.
2835        trackToRemove.clear();
2836        activeTrack.clear();
2837
2838        // Effect chains will be actually deleted here if they were removed from
2839        // mEffectChains list during mixing or effects processing
2840        effectChains.clear();
2841    }
2842
2843    if (!mStandby) {
2844        mOutput->stream->common.standby(&mOutput->stream->common);
2845    }
2846
2847    releaseWakeLock();
2848
2849    ALOGV("DirectOutputThread %p exiting", this);
2850    return false;
2851}
2852
2853// getTrackName_l() must be called with ThreadBase::mLock held
2854int AudioFlinger::DirectOutputThread::getTrackName_l()
2855{
2856    return 0;
2857}
2858
2859// deleteTrackName_l() must be called with ThreadBase::mLock held
2860void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2861{
2862}
2863
2864// checkForNewParameters_l() must be called with ThreadBase::mLock held
2865bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2866{
2867    bool reconfig = false;
2868
2869    while (!mNewParameters.isEmpty()) {
2870        status_t status = NO_ERROR;
2871        String8 keyValuePair = mNewParameters[0];
2872        AudioParameter param = AudioParameter(keyValuePair);
2873        int value;
2874
2875        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2876            // do not accept frame count changes if tracks are open as the track buffer
2877            // size depends on frame count and correct behavior would not be garantied
2878            // if frame count is changed after track creation
2879            if (!mTracks.isEmpty()) {
2880                status = INVALID_OPERATION;
2881            } else {
2882                reconfig = true;
2883            }
2884        }
2885        if (status == NO_ERROR) {
2886            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2887                                                    keyValuePair.string());
2888            if (!mStandby && status == INVALID_OPERATION) {
2889               mOutput->stream->common.standby(&mOutput->stream->common);
2890               mStandby = true;
2891               mBytesWritten = 0;
2892               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2893                                                       keyValuePair.string());
2894            }
2895            if (status == NO_ERROR && reconfig) {
2896                readOutputParameters();
2897                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2898            }
2899        }
2900
2901        mNewParameters.removeAt(0);
2902
2903        mParamStatus = status;
2904        mParamCond.signal();
2905        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2906        // already timed out waiting for the status and will never signal the condition.
2907        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2908    }
2909    return reconfig;
2910}
2911
2912uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2913{
2914    uint32_t time;
2915    if (audio_is_linear_pcm(mFormat)) {
2916        time = PlaybackThread::activeSleepTimeUs();
2917    } else {
2918        time = 10000;
2919    }
2920    return time;
2921}
2922
2923uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2924{
2925    uint32_t time;
2926    if (audio_is_linear_pcm(mFormat)) {
2927        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2928    } else {
2929        time = 10000;
2930    }
2931    return time;
2932}
2933
2934uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2935{
2936    uint32_t time;
2937    if (audio_is_linear_pcm(mFormat)) {
2938        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2939    } else {
2940        time = 10000;
2941    }
2942    return time;
2943}
2944
2945
2946// ----------------------------------------------------------------------------
2947
2948AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2949    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2950{
2951    mType = ThreadBase::DUPLICATING;
2952    addOutputTrack(mainThread);
2953}
2954
2955AudioFlinger::DuplicatingThread::~DuplicatingThread()
2956{
2957    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2958        mOutputTracks[i]->destroy();
2959    }
2960    mOutputTracks.clear();
2961}
2962
2963bool AudioFlinger::DuplicatingThread::threadLoop()
2964{
2965    Vector< sp<Track> > tracksToRemove;
2966    uint32_t mixerStatus = MIXER_IDLE;
2967    nsecs_t standbyTime = systemTime();
2968    size_t mixBufferSize = mFrameCount*mFrameSize;
2969    SortedVector< sp<OutputTrack> > outputTracks;
2970    uint32_t writeFrames = 0;
2971    uint32_t activeSleepTime = activeSleepTimeUs();
2972    uint32_t idleSleepTime = idleSleepTimeUs();
2973    uint32_t sleepTime = idleSleepTime;
2974    Vector< sp<EffectChain> > effectChains;
2975
2976    acquireWakeLock();
2977
2978    while (!exitPending())
2979    {
2980        processConfigEvents();
2981
2982        mixerStatus = MIXER_IDLE;
2983        { // scope for the mLock
2984
2985            Mutex::Autolock _l(mLock);
2986
2987            if (checkForNewParameters_l()) {
2988                mixBufferSize = mFrameCount*mFrameSize;
2989                updateWaitTime();
2990                activeSleepTime = activeSleepTimeUs();
2991                idleSleepTime = idleSleepTimeUs();
2992            }
2993
2994            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
2995
2996            for (size_t i = 0; i < mOutputTracks.size(); i++) {
2997                outputTracks.add(mOutputTracks[i]);
2998            }
2999
3000            // put audio hardware into standby after short delay
3001            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
3002                         mSuspended)) {
3003                if (!mStandby) {
3004                    for (size_t i = 0; i < outputTracks.size(); i++) {
3005                        outputTracks[i]->stop();
3006                    }
3007                    mStandby = true;
3008                    mBytesWritten = 0;
3009                }
3010
3011                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
3012                    // we're about to wait, flush the binder command buffer
3013                    IPCThreadState::self()->flushCommands();
3014                    outputTracks.clear();
3015
3016                    if (exitPending()) break;
3017
3018                    releaseWakeLock_l();
3019                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3020                    mWaitWorkCV.wait(mLock);
3021                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3022                    acquireWakeLock_l();
3023
3024                    if (mMasterMute == false) {
3025                        char value[PROPERTY_VALUE_MAX];
3026                        property_get("ro.audio.silent", value, "0");
3027                        if (atoi(value)) {
3028                            ALOGD("Silence is golden");
3029                            setMasterMute(true);
3030                        }
3031                    }
3032
3033                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3034                    sleepTime = idleSleepTime;
3035                    continue;
3036                }
3037            }
3038
3039            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3040
3041            // prevent any changes in effect chain list and in each effect chain
3042            // during mixing and effect process as the audio buffers could be deleted
3043            // or modified if an effect is created or deleted
3044            lockEffectChains_l(effectChains);
3045        }
3046
3047        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3048            // mix buffers...
3049            if (outputsReady(outputTracks)) {
3050                mAudioMixer->process();
3051            } else {
3052                memset(mMixBuffer, 0, mixBufferSize);
3053            }
3054            sleepTime = 0;
3055            writeFrames = mFrameCount;
3056        } else {
3057            if (sleepTime == 0) {
3058                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3059                    sleepTime = activeSleepTime;
3060                } else {
3061                    sleepTime = idleSleepTime;
3062                }
3063            } else if (mBytesWritten != 0) {
3064                // flush remaining overflow buffers in output tracks
3065                for (size_t i = 0; i < outputTracks.size(); i++) {
3066                    if (outputTracks[i]->isActive()) {
3067                        sleepTime = 0;
3068                        writeFrames = 0;
3069                        memset(mMixBuffer, 0, mixBufferSize);
3070                        break;
3071                    }
3072                }
3073            }
3074        }
3075
3076        if (mSuspended) {
3077            sleepTime = suspendSleepTimeUs();
3078        }
3079        // sleepTime == 0 means we must write to audio hardware
3080        if (sleepTime == 0) {
3081            for (size_t i = 0; i < effectChains.size(); i ++) {
3082                effectChains[i]->process_l();
3083            }
3084            // enable changes in effect chain
3085            unlockEffectChains(effectChains);
3086
3087            standbyTime = systemTime() + kStandbyTimeInNsecs;
3088            for (size_t i = 0; i < outputTracks.size(); i++) {
3089                outputTracks[i]->write(mMixBuffer, writeFrames);
3090            }
3091            mStandby = false;
3092            mBytesWritten += mixBufferSize;
3093        } else {
3094            // enable changes in effect chain
3095            unlockEffectChains(effectChains);
3096            usleep(sleepTime);
3097        }
3098
3099        // finally let go of all our tracks, without the lock held
3100        // since we can't guarantee the destructors won't acquire that
3101        // same lock.
3102        tracksToRemove.clear();
3103        outputTracks.clear();
3104
3105        // Effect chains will be actually deleted here if they were removed from
3106        // mEffectChains list during mixing or effects processing
3107        effectChains.clear();
3108    }
3109
3110    releaseWakeLock();
3111
3112    return false;
3113}
3114
3115void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3116{
3117    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3118    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3119                                            this,
3120                                            mSampleRate,
3121                                            mFormat,
3122                                            mChannelMask,
3123                                            frameCount);
3124    if (outputTrack->cblk() != NULL) {
3125        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3126        mOutputTracks.add(outputTrack);
3127        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3128        updateWaitTime();
3129    }
3130}
3131
3132void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3133{
3134    Mutex::Autolock _l(mLock);
3135    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3136        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3137            mOutputTracks[i]->destroy();
3138            mOutputTracks.removeAt(i);
3139            updateWaitTime();
3140            return;
3141        }
3142    }
3143    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3144}
3145
3146void AudioFlinger::DuplicatingThread::updateWaitTime()
3147{
3148    mWaitTimeMs = UINT_MAX;
3149    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3150        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3151        if (strong != NULL) {
3152            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3153            if (waitTimeMs < mWaitTimeMs) {
3154                mWaitTimeMs = waitTimeMs;
3155            }
3156        }
3157    }
3158}
3159
3160
3161bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3162{
3163    for (size_t i = 0; i < outputTracks.size(); i++) {
3164        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3165        if (thread == 0) {
3166            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3167            return false;
3168        }
3169        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3170        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3171            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3172            return false;
3173        }
3174    }
3175    return true;
3176}
3177
3178uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3179{
3180    return (mWaitTimeMs * 1000) / 2;
3181}
3182
3183// ----------------------------------------------------------------------------
3184
3185// TrackBase constructor must be called with AudioFlinger::mLock held
3186AudioFlinger::ThreadBase::TrackBase::TrackBase(
3187            const wp<ThreadBase>& thread,
3188            const sp<Client>& client,
3189            uint32_t sampleRate,
3190            uint32_t format,
3191            uint32_t channelMask,
3192            int frameCount,
3193            uint32_t flags,
3194            const sp<IMemory>& sharedBuffer,
3195            int sessionId)
3196    :   RefBase(),
3197        mThread(thread),
3198        mClient(client),
3199        mCblk(0),
3200        mFrameCount(0),
3201        mState(IDLE),
3202        mClientTid(-1),
3203        mFormat(format),
3204        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3205        mSessionId(sessionId)
3206{
3207    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3208
3209    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3210   size_t size = sizeof(audio_track_cblk_t);
3211   uint8_t channelCount = popcount(channelMask);
3212   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3213   if (sharedBuffer == 0) {
3214       size += bufferSize;
3215   }
3216
3217   if (client != NULL) {
3218        mCblkMemory = client->heap()->allocate(size);
3219        if (mCblkMemory != 0) {
3220            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3221            if (mCblk) { // construct the shared structure in-place.
3222                new(mCblk) audio_track_cblk_t();
3223                // clear all buffers
3224                mCblk->frameCount = frameCount;
3225                mCblk->sampleRate = sampleRate;
3226                mChannelCount = channelCount;
3227                mChannelMask = channelMask;
3228                if (sharedBuffer == 0) {
3229                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3230                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3231                    // Force underrun condition to avoid false underrun callback until first data is
3232                    // written to buffer (other flags are cleared)
3233                    mCblk->flags = CBLK_UNDERRUN_ON;
3234                } else {
3235                    mBuffer = sharedBuffer->pointer();
3236                }
3237                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3238            }
3239        } else {
3240            ALOGE("not enough memory for AudioTrack size=%u", size);
3241            client->heap()->dump("AudioTrack");
3242            return;
3243        }
3244   } else {
3245       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3246           // construct the shared structure in-place.
3247           new(mCblk) audio_track_cblk_t();
3248           // clear all buffers
3249           mCblk->frameCount = frameCount;
3250           mCblk->sampleRate = sampleRate;
3251           mChannelCount = channelCount;
3252           mChannelMask = channelMask;
3253           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3254           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3255           // Force underrun condition to avoid false underrun callback until first data is
3256           // written to buffer (other flags are cleared)
3257           mCblk->flags = CBLK_UNDERRUN_ON;
3258           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3259   }
3260}
3261
3262AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3263{
3264    if (mCblk) {
3265        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3266        if (mClient == NULL) {
3267            delete mCblk;
3268        }
3269    }
3270    mCblkMemory.clear();            // and free the shared memory
3271    if (mClient != NULL) {
3272        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3273        mClient.clear();
3274    }
3275}
3276
3277void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3278{
3279    buffer->raw = NULL;
3280    mFrameCount = buffer->frameCount;
3281    step();
3282    buffer->frameCount = 0;
3283}
3284
3285bool AudioFlinger::ThreadBase::TrackBase::step() {
3286    bool result;
3287    audio_track_cblk_t* cblk = this->cblk();
3288
3289    result = cblk->stepServer(mFrameCount);
3290    if (!result) {
3291        ALOGV("stepServer failed acquiring cblk mutex");
3292        mFlags |= STEPSERVER_FAILED;
3293    }
3294    return result;
3295}
3296
3297void AudioFlinger::ThreadBase::TrackBase::reset() {
3298    audio_track_cblk_t* cblk = this->cblk();
3299
3300    cblk->user = 0;
3301    cblk->server = 0;
3302    cblk->userBase = 0;
3303    cblk->serverBase = 0;
3304    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3305    ALOGV("TrackBase::reset");
3306}
3307
3308sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3309{
3310    return mCblkMemory;
3311}
3312
3313int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3314    return (int)mCblk->sampleRate;
3315}
3316
3317int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3318    return (const int)mChannelCount;
3319}
3320
3321uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3322    return mChannelMask;
3323}
3324
3325void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3326    audio_track_cblk_t* cblk = this->cblk();
3327    size_t frameSize = cblk->frameSize;
3328    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
3329    int8_t *bufferEnd = bufferStart + frames * frameSize;
3330
3331    // Check validity of returned pointer in case the track control block would have been corrupted.
3332    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3333        ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) {
3334        ALOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3335                server %d, serverBase %d, user %d, userBase %d",
3336                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3337                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3338        return 0;
3339    }
3340
3341    return bufferStart;
3342}
3343
3344// ----------------------------------------------------------------------------
3345
3346// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3347AudioFlinger::PlaybackThread::Track::Track(
3348            const wp<ThreadBase>& thread,
3349            const sp<Client>& client,
3350            int streamType,
3351            uint32_t sampleRate,
3352            uint32_t format,
3353            uint32_t channelMask,
3354            int frameCount,
3355            const sp<IMemory>& sharedBuffer,
3356            int sessionId)
3357    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3358    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3359    mAuxEffectId(0), mHasVolumeController(false)
3360{
3361    if (mCblk != NULL) {
3362        sp<ThreadBase> baseThread = thread.promote();
3363        if (baseThread != 0) {
3364            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3365            mName = playbackThread->getTrackName_l();
3366            mMainBuffer = playbackThread->mixBuffer();
3367        }
3368        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3369        if (mName < 0) {
3370            ALOGE("no more track names available");
3371        }
3372        mVolume[0] = 1.0f;
3373        mVolume[1] = 1.0f;
3374        mStreamType = streamType;
3375        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3376        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3377        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3378    }
3379}
3380
3381AudioFlinger::PlaybackThread::Track::~Track()
3382{
3383    ALOGV("PlaybackThread::Track destructor");
3384    sp<ThreadBase> thread = mThread.promote();
3385    if (thread != 0) {
3386        Mutex::Autolock _l(thread->mLock);
3387        mState = TERMINATED;
3388    }
3389}
3390
3391void AudioFlinger::PlaybackThread::Track::destroy()
3392{
3393    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3394    // by removing it from mTracks vector, so there is a risk that this Tracks's
3395    // desctructor is called. As the destructor needs to lock mLock,
3396    // we must acquire a strong reference on this Track before locking mLock
3397    // here so that the destructor is called only when exiting this function.
3398    // On the other hand, as long as Track::destroy() is only called by
3399    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3400    // this Track with its member mTrack.
3401    sp<Track> keep(this);
3402    { // scope for mLock
3403        sp<ThreadBase> thread = mThread.promote();
3404        if (thread != 0) {
3405            if (!isOutputTrack()) {
3406                if (mState == ACTIVE || mState == RESUMING) {
3407                    AudioSystem::stopOutput(thread->id(),
3408                                            (audio_stream_type_t)mStreamType,
3409                                            mSessionId);
3410
3411                    // to track the speaker usage
3412                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3413                }
3414                AudioSystem::releaseOutput(thread->id());
3415            }
3416            Mutex::Autolock _l(thread->mLock);
3417            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3418            playbackThread->destroyTrack_l(this);
3419        }
3420    }
3421}
3422
3423void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3424{
3425    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",
3426            mName - AudioMixer::TRACK0,
3427            (mClient == NULL) ? getpid() : mClient->pid(),
3428            mStreamType,
3429            mFormat,
3430            mChannelMask,
3431            mSessionId,
3432            mFrameCount,
3433            mState,
3434            mMute,
3435            mFillingUpStatus,
3436            mCblk->sampleRate,
3437            mCblk->volume[0],
3438            mCblk->volume[1],
3439            mCblk->server,
3440            mCblk->user,
3441            (int)mMainBuffer,
3442            (int)mAuxBuffer);
3443}
3444
3445status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3446{
3447     audio_track_cblk_t* cblk = this->cblk();
3448     uint32_t framesReady;
3449     uint32_t framesReq = buffer->frameCount;
3450
3451     // Check if last stepServer failed, try to step now
3452     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3453         if (!step())  goto getNextBuffer_exit;
3454         ALOGV("stepServer recovered");
3455         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3456     }
3457
3458     framesReady = cblk->framesReady();
3459
3460     if (CC_LIKELY(framesReady)) {
3461        uint32_t s = cblk->server;
3462        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3463
3464        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3465        if (framesReq > framesReady) {
3466            framesReq = framesReady;
3467        }
3468        if (s + framesReq > bufferEnd) {
3469            framesReq = bufferEnd - s;
3470        }
3471
3472         buffer->raw = getBuffer(s, framesReq);
3473         if (buffer->raw == NULL) goto getNextBuffer_exit;
3474
3475         buffer->frameCount = framesReq;
3476        return NO_ERROR;
3477     }
3478
3479getNextBuffer_exit:
3480     buffer->raw = NULL;
3481     buffer->frameCount = 0;
3482     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3483     return NOT_ENOUGH_DATA;
3484}
3485
3486bool AudioFlinger::PlaybackThread::Track::isReady() const {
3487    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3488
3489    if (mCblk->framesReady() >= mCblk->frameCount ||
3490            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3491        mFillingUpStatus = FS_FILLED;
3492        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3493        return true;
3494    }
3495    return false;
3496}
3497
3498status_t AudioFlinger::PlaybackThread::Track::start()
3499{
3500    status_t status = NO_ERROR;
3501    ALOGV("start(%d), calling thread %d session %d",
3502            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3503    sp<ThreadBase> thread = mThread.promote();
3504    if (thread != 0) {
3505        Mutex::Autolock _l(thread->mLock);
3506        int state = mState;
3507        // here the track could be either new, or restarted
3508        // in both cases "unstop" the track
3509        if (mState == PAUSED) {
3510            mState = TrackBase::RESUMING;
3511            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3512        } else {
3513            mState = TrackBase::ACTIVE;
3514            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3515        }
3516
3517        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3518            thread->mLock.unlock();
3519            status = AudioSystem::startOutput(thread->id(),
3520                                              (audio_stream_type_t)mStreamType,
3521                                              mSessionId);
3522            thread->mLock.lock();
3523
3524            // to track the speaker usage
3525            if (status == NO_ERROR) {
3526                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3527            }
3528        }
3529        if (status == NO_ERROR) {
3530            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3531            playbackThread->addTrack_l(this);
3532        } else {
3533            mState = state;
3534        }
3535    } else {
3536        status = BAD_VALUE;
3537    }
3538    return status;
3539}
3540
3541void AudioFlinger::PlaybackThread::Track::stop()
3542{
3543    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3544    sp<ThreadBase> thread = mThread.promote();
3545    if (thread != 0) {
3546        Mutex::Autolock _l(thread->mLock);
3547        int state = mState;
3548        if (mState > STOPPED) {
3549            mState = STOPPED;
3550            // If the track is not active (PAUSED and buffers full), flush buffers
3551            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3552            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3553                reset();
3554            }
3555            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3556        }
3557        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3558            thread->mLock.unlock();
3559            AudioSystem::stopOutput(thread->id(),
3560                                    (audio_stream_type_t)mStreamType,
3561                                    mSessionId);
3562            thread->mLock.lock();
3563
3564            // to track the speaker usage
3565            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3566        }
3567    }
3568}
3569
3570void AudioFlinger::PlaybackThread::Track::pause()
3571{
3572    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3573    sp<ThreadBase> thread = mThread.promote();
3574    if (thread != 0) {
3575        Mutex::Autolock _l(thread->mLock);
3576        if (mState == ACTIVE || mState == RESUMING) {
3577            mState = PAUSING;
3578            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3579            if (!isOutputTrack()) {
3580                thread->mLock.unlock();
3581                AudioSystem::stopOutput(thread->id(),
3582                                        (audio_stream_type_t)mStreamType,
3583                                        mSessionId);
3584                thread->mLock.lock();
3585
3586                // to track the speaker usage
3587                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3588            }
3589        }
3590    }
3591}
3592
3593void AudioFlinger::PlaybackThread::Track::flush()
3594{
3595    ALOGV("flush(%d)", mName);
3596    sp<ThreadBase> thread = mThread.promote();
3597    if (thread != 0) {
3598        Mutex::Autolock _l(thread->mLock);
3599        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3600            return;
3601        }
3602        // No point remaining in PAUSED state after a flush => go to
3603        // STOPPED state
3604        mState = STOPPED;
3605
3606        // do not reset the track if it is still in the process of being stopped or paused.
3607        // this will be done by prepareTracks_l() when the track is stopped.
3608        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3609        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3610            reset();
3611        }
3612    }
3613}
3614
3615void AudioFlinger::PlaybackThread::Track::reset()
3616{
3617    // Do not reset twice to avoid discarding data written just after a flush and before
3618    // the audioflinger thread detects the track is stopped.
3619    if (!mResetDone) {
3620        TrackBase::reset();
3621        // Force underrun condition to avoid false underrun callback until first data is
3622        // written to buffer
3623        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3624        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3625        mFillingUpStatus = FS_FILLING;
3626        mResetDone = true;
3627    }
3628}
3629
3630void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3631{
3632    mMute = muted;
3633}
3634
3635void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3636{
3637    mVolume[0] = left;
3638    mVolume[1] = right;
3639}
3640
3641status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3642{
3643    status_t status = DEAD_OBJECT;
3644    sp<ThreadBase> thread = mThread.promote();
3645    if (thread != 0) {
3646       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3647       status = playbackThread->attachAuxEffect(this, EffectId);
3648    }
3649    return status;
3650}
3651
3652void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3653{
3654    mAuxEffectId = EffectId;
3655    mAuxBuffer = buffer;
3656}
3657
3658// ----------------------------------------------------------------------------
3659
3660// RecordTrack constructor must be called with AudioFlinger::mLock held
3661AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3662            const wp<ThreadBase>& thread,
3663            const sp<Client>& client,
3664            uint32_t sampleRate,
3665            uint32_t format,
3666            uint32_t channelMask,
3667            int frameCount,
3668            uint32_t flags,
3669            int sessionId)
3670    :   TrackBase(thread, client, sampleRate, format,
3671                  channelMask, frameCount, flags, 0, sessionId),
3672        mOverflow(false)
3673{
3674    if (mCblk != NULL) {
3675       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3676       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3677           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3678       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3679           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3680       } else {
3681           mCblk->frameSize = sizeof(int8_t);
3682       }
3683    }
3684}
3685
3686AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3687{
3688    sp<ThreadBase> thread = mThread.promote();
3689    if (thread != 0) {
3690        AudioSystem::releaseInput(thread->id());
3691    }
3692}
3693
3694status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3695{
3696    audio_track_cblk_t* cblk = this->cblk();
3697    uint32_t framesAvail;
3698    uint32_t framesReq = buffer->frameCount;
3699
3700     // Check if last stepServer failed, try to step now
3701    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3702        if (!step()) goto getNextBuffer_exit;
3703        ALOGV("stepServer recovered");
3704        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3705    }
3706
3707    framesAvail = cblk->framesAvailable_l();
3708
3709    if (CC_LIKELY(framesAvail)) {
3710        uint32_t s = cblk->server;
3711        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3712
3713        if (framesReq > framesAvail) {
3714            framesReq = framesAvail;
3715        }
3716        if (s + framesReq > bufferEnd) {
3717            framesReq = bufferEnd - s;
3718        }
3719
3720        buffer->raw = getBuffer(s, framesReq);
3721        if (buffer->raw == NULL) goto getNextBuffer_exit;
3722
3723        buffer->frameCount = framesReq;
3724        return NO_ERROR;
3725    }
3726
3727getNextBuffer_exit:
3728    buffer->raw = NULL;
3729    buffer->frameCount = 0;
3730    return NOT_ENOUGH_DATA;
3731}
3732
3733status_t AudioFlinger::RecordThread::RecordTrack::start()
3734{
3735    sp<ThreadBase> thread = mThread.promote();
3736    if (thread != 0) {
3737        RecordThread *recordThread = (RecordThread *)thread.get();
3738        return recordThread->start(this);
3739    } else {
3740        return BAD_VALUE;
3741    }
3742}
3743
3744void AudioFlinger::RecordThread::RecordTrack::stop()
3745{
3746    sp<ThreadBase> thread = mThread.promote();
3747    if (thread != 0) {
3748        RecordThread *recordThread = (RecordThread *)thread.get();
3749        recordThread->stop(this);
3750        TrackBase::reset();
3751        // Force overerrun condition to avoid false overrun callback until first data is
3752        // read from buffer
3753        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3754    }
3755}
3756
3757void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3758{
3759    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3760            (mClient == NULL) ? getpid() : mClient->pid(),
3761            mFormat,
3762            mChannelMask,
3763            mSessionId,
3764            mFrameCount,
3765            mState,
3766            mCblk->sampleRate,
3767            mCblk->server,
3768            mCblk->user);
3769}
3770
3771
3772// ----------------------------------------------------------------------------
3773
3774AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3775            const wp<ThreadBase>& thread,
3776            DuplicatingThread *sourceThread,
3777            uint32_t sampleRate,
3778            uint32_t format,
3779            uint32_t channelMask,
3780            int frameCount)
3781    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3782    mActive(false), mSourceThread(sourceThread)
3783{
3784
3785    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3786    if (mCblk != NULL) {
3787        mCblk->flags |= CBLK_DIRECTION_OUT;
3788        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3789        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
3790        mOutBuffer.frameCount = 0;
3791        playbackThread->mTracks.add(this);
3792        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3793                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3794                mCblk, mBuffer, mCblk->buffers,
3795                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3796    } else {
3797        ALOGW("Error creating output track on thread %p", playbackThread);
3798    }
3799}
3800
3801AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3802{
3803    clearBufferQueue();
3804}
3805
3806status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3807{
3808    status_t status = Track::start();
3809    if (status != NO_ERROR) {
3810        return status;
3811    }
3812
3813    mActive = true;
3814    mRetryCount = 127;
3815    return status;
3816}
3817
3818void AudioFlinger::PlaybackThread::OutputTrack::stop()
3819{
3820    Track::stop();
3821    clearBufferQueue();
3822    mOutBuffer.frameCount = 0;
3823    mActive = false;
3824}
3825
3826bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3827{
3828    Buffer *pInBuffer;
3829    Buffer inBuffer;
3830    uint32_t channelCount = mChannelCount;
3831    bool outputBufferFull = false;
3832    inBuffer.frameCount = frames;
3833    inBuffer.i16 = data;
3834
3835    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3836
3837    if (!mActive && frames != 0) {
3838        start();
3839        sp<ThreadBase> thread = mThread.promote();
3840        if (thread != 0) {
3841            MixerThread *mixerThread = (MixerThread *)thread.get();
3842            if (mCblk->frameCount > frames){
3843                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3844                    uint32_t startFrames = (mCblk->frameCount - frames);
3845                    pInBuffer = new Buffer;
3846                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3847                    pInBuffer->frameCount = startFrames;
3848                    pInBuffer->i16 = pInBuffer->mBuffer;
3849                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3850                    mBufferQueue.add(pInBuffer);
3851                } else {
3852                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
3853                }
3854            }
3855        }
3856    }
3857
3858    while (waitTimeLeftMs) {
3859        // First write pending buffers, then new data
3860        if (mBufferQueue.size()) {
3861            pInBuffer = mBufferQueue.itemAt(0);
3862        } else {
3863            pInBuffer = &inBuffer;
3864        }
3865
3866        if (pInBuffer->frameCount == 0) {
3867            break;
3868        }
3869
3870        if (mOutBuffer.frameCount == 0) {
3871            mOutBuffer.frameCount = pInBuffer->frameCount;
3872            nsecs_t startTime = systemTime();
3873            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
3874                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3875                outputBufferFull = true;
3876                break;
3877            }
3878            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3879            if (waitTimeLeftMs >= waitTimeMs) {
3880                waitTimeLeftMs -= waitTimeMs;
3881            } else {
3882                waitTimeLeftMs = 0;
3883            }
3884        }
3885
3886        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3887        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3888        mCblk->stepUser(outFrames);
3889        pInBuffer->frameCount -= outFrames;
3890        pInBuffer->i16 += outFrames * channelCount;
3891        mOutBuffer.frameCount -= outFrames;
3892        mOutBuffer.i16 += outFrames * channelCount;
3893
3894        if (pInBuffer->frameCount == 0) {
3895            if (mBufferQueue.size()) {
3896                mBufferQueue.removeAt(0);
3897                delete [] pInBuffer->mBuffer;
3898                delete pInBuffer;
3899                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3900            } else {
3901                break;
3902            }
3903        }
3904    }
3905
3906    // If we could not write all frames, allocate a buffer and queue it for next time.
3907    if (inBuffer.frameCount) {
3908        sp<ThreadBase> thread = mThread.promote();
3909        if (thread != 0 && !thread->standby()) {
3910            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3911                pInBuffer = new Buffer;
3912                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3913                pInBuffer->frameCount = inBuffer.frameCount;
3914                pInBuffer->i16 = pInBuffer->mBuffer;
3915                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3916                mBufferQueue.add(pInBuffer);
3917                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3918            } else {
3919                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3920            }
3921        }
3922    }
3923
3924    // Calling write() with a 0 length buffer, means that no more data will be written:
3925    // If no more buffers are pending, fill output track buffer to make sure it is started
3926    // by output mixer.
3927    if (frames == 0 && mBufferQueue.size() == 0) {
3928        if (mCblk->user < mCblk->frameCount) {
3929            frames = mCblk->frameCount - mCblk->user;
3930            pInBuffer = new Buffer;
3931            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3932            pInBuffer->frameCount = frames;
3933            pInBuffer->i16 = pInBuffer->mBuffer;
3934            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3935            mBufferQueue.add(pInBuffer);
3936        } else if (mActive) {
3937            stop();
3938        }
3939    }
3940
3941    return outputBufferFull;
3942}
3943
3944status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3945{
3946    int active;
3947    status_t result;
3948    audio_track_cblk_t* cblk = mCblk;
3949    uint32_t framesReq = buffer->frameCount;
3950
3951//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3952    buffer->frameCount  = 0;
3953
3954    uint32_t framesAvail = cblk->framesAvailable();
3955
3956
3957    if (framesAvail == 0) {
3958        Mutex::Autolock _l(cblk->lock);
3959        goto start_loop_here;
3960        while (framesAvail == 0) {
3961            active = mActive;
3962            if (CC_UNLIKELY(!active)) {
3963                ALOGV("Not active and NO_MORE_BUFFERS");
3964                return AudioTrack::NO_MORE_BUFFERS;
3965            }
3966            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
3967            if (result != NO_ERROR) {
3968                return AudioTrack::NO_MORE_BUFFERS;
3969            }
3970            // read the server count again
3971        start_loop_here:
3972            framesAvail = cblk->framesAvailable_l();
3973        }
3974    }
3975
3976//    if (framesAvail < framesReq) {
3977//        return AudioTrack::NO_MORE_BUFFERS;
3978//    }
3979
3980    if (framesReq > framesAvail) {
3981        framesReq = framesAvail;
3982    }
3983
3984    uint32_t u = cblk->user;
3985    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
3986
3987    if (u + framesReq > bufferEnd) {
3988        framesReq = bufferEnd - u;
3989    }
3990
3991    buffer->frameCount  = framesReq;
3992    buffer->raw         = (void *)cblk->buffer(u);
3993    return NO_ERROR;
3994}
3995
3996
3997void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
3998{
3999    size_t size = mBufferQueue.size();
4000    Buffer *pBuffer;
4001
4002    for (size_t i = 0; i < size; i++) {
4003        pBuffer = mBufferQueue.itemAt(i);
4004        delete [] pBuffer->mBuffer;
4005        delete pBuffer;
4006    }
4007    mBufferQueue.clear();
4008}
4009
4010// ----------------------------------------------------------------------------
4011
4012AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
4013    :   RefBase(),
4014        mAudioFlinger(audioFlinger),
4015        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
4016        mPid(pid)
4017{
4018    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
4019}
4020
4021// Client destructor must be called with AudioFlinger::mLock held
4022AudioFlinger::Client::~Client()
4023{
4024    mAudioFlinger->removeClient_l(mPid);
4025}
4026
4027const sp<MemoryDealer>& AudioFlinger::Client::heap() const
4028{
4029    return mMemoryDealer;
4030}
4031
4032// ----------------------------------------------------------------------------
4033
4034AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4035                                                     const sp<IAudioFlingerClient>& client,
4036                                                     pid_t pid)
4037    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
4038{
4039}
4040
4041AudioFlinger::NotificationClient::~NotificationClient()
4042{
4043    mClient.clear();
4044}
4045
4046void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4047{
4048    sp<NotificationClient> keep(this);
4049    {
4050        mAudioFlinger->removeNotificationClient(mPid);
4051    }
4052}
4053
4054// ----------------------------------------------------------------------------
4055
4056AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4057    : BnAudioTrack(),
4058      mTrack(track)
4059{
4060}
4061
4062AudioFlinger::TrackHandle::~TrackHandle() {
4063    // just stop the track on deletion, associated resources
4064    // will be freed from the main thread once all pending buffers have
4065    // been played. Unless it's not in the active track list, in which
4066    // case we free everything now...
4067    mTrack->destroy();
4068}
4069
4070status_t AudioFlinger::TrackHandle::start() {
4071    return mTrack->start();
4072}
4073
4074void AudioFlinger::TrackHandle::stop() {
4075    mTrack->stop();
4076}
4077
4078void AudioFlinger::TrackHandle::flush() {
4079    mTrack->flush();
4080}
4081
4082void AudioFlinger::TrackHandle::mute(bool e) {
4083    mTrack->mute(e);
4084}
4085
4086void AudioFlinger::TrackHandle::pause() {
4087    mTrack->pause();
4088}
4089
4090void AudioFlinger::TrackHandle::setVolume(float left, float right) {
4091    mTrack->setVolume(left, right);
4092}
4093
4094sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4095    return mTrack->getCblk();
4096}
4097
4098status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4099{
4100    return mTrack->attachAuxEffect(EffectId);
4101}
4102
4103status_t AudioFlinger::TrackHandle::onTransact(
4104    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4105{
4106    return BnAudioTrack::onTransact(code, data, reply, flags);
4107}
4108
4109// ----------------------------------------------------------------------------
4110
4111sp<IAudioRecord> AudioFlinger::openRecord(
4112        pid_t pid,
4113        int input,
4114        uint32_t sampleRate,
4115        uint32_t format,
4116        uint32_t channelMask,
4117        int frameCount,
4118        uint32_t flags,
4119        int *sessionId,
4120        status_t *status)
4121{
4122    sp<RecordThread::RecordTrack> recordTrack;
4123    sp<RecordHandle> recordHandle;
4124    sp<Client> client;
4125    wp<Client> wclient;
4126    status_t lStatus;
4127    RecordThread *thread;
4128    size_t inFrameCount;
4129    int lSessionId;
4130
4131    // check calling permissions
4132    if (!recordingAllowed()) {
4133        lStatus = PERMISSION_DENIED;
4134        goto Exit;
4135    }
4136
4137    // add client to list
4138    { // scope for mLock
4139        Mutex::Autolock _l(mLock);
4140        thread = checkRecordThread_l(input);
4141        if (thread == NULL) {
4142            lStatus = BAD_VALUE;
4143            goto Exit;
4144        }
4145
4146        wclient = mClients.valueFor(pid);
4147        if (wclient != NULL) {
4148            client = wclient.promote();
4149        } else {
4150            client = new Client(this, pid);
4151            mClients.add(pid, client);
4152        }
4153
4154        // If no audio session id is provided, create one here
4155        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4156            lSessionId = *sessionId;
4157        } else {
4158            lSessionId = nextUniqueId();
4159            if (sessionId != NULL) {
4160                *sessionId = lSessionId;
4161            }
4162        }
4163        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4164        recordTrack = thread->createRecordTrack_l(client,
4165                                                sampleRate,
4166                                                format,
4167                                                channelMask,
4168                                                frameCount,
4169                                                flags,
4170                                                lSessionId,
4171                                                &lStatus);
4172    }
4173    if (lStatus != NO_ERROR) {
4174        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4175        // destructor is called by the TrackBase destructor with mLock held
4176        client.clear();
4177        recordTrack.clear();
4178        goto Exit;
4179    }
4180
4181    // return to handle to client
4182    recordHandle = new RecordHandle(recordTrack);
4183    lStatus = NO_ERROR;
4184
4185Exit:
4186    if (status) {
4187        *status = lStatus;
4188    }
4189    return recordHandle;
4190}
4191
4192// ----------------------------------------------------------------------------
4193
4194AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4195    : BnAudioRecord(),
4196    mRecordTrack(recordTrack)
4197{
4198}
4199
4200AudioFlinger::RecordHandle::~RecordHandle() {
4201    stop();
4202}
4203
4204status_t AudioFlinger::RecordHandle::start() {
4205    ALOGV("RecordHandle::start()");
4206    return mRecordTrack->start();
4207}
4208
4209void AudioFlinger::RecordHandle::stop() {
4210    ALOGV("RecordHandle::stop()");
4211    mRecordTrack->stop();
4212}
4213
4214sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4215    return mRecordTrack->getCblk();
4216}
4217
4218status_t AudioFlinger::RecordHandle::onTransact(
4219    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4220{
4221    return BnAudioRecord::onTransact(code, data, reply, flags);
4222}
4223
4224// ----------------------------------------------------------------------------
4225
4226AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4227                                         AudioStreamIn *input,
4228                                         uint32_t sampleRate,
4229                                         uint32_t channels,
4230                                         int id,
4231                                         uint32_t device) :
4232    ThreadBase(audioFlinger, id, device),
4233    mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
4234{
4235    mType = ThreadBase::RECORD;
4236
4237    snprintf(mName, kNameLength, "AudioIn_%d", id);
4238
4239    mReqChannelCount = popcount(channels);
4240    mReqSampleRate = sampleRate;
4241    readInputParameters();
4242}
4243
4244
4245AudioFlinger::RecordThread::~RecordThread()
4246{
4247    delete[] mRsmpInBuffer;
4248    if (mResampler != NULL) {
4249        delete mResampler;
4250        delete[] mRsmpOutBuffer;
4251    }
4252}
4253
4254void AudioFlinger::RecordThread::onFirstRef()
4255{
4256    run(mName, PRIORITY_URGENT_AUDIO);
4257}
4258
4259status_t AudioFlinger::RecordThread::readyToRun()
4260{
4261    status_t status = initCheck();
4262    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4263    return status;
4264}
4265
4266bool AudioFlinger::RecordThread::threadLoop()
4267{
4268    AudioBufferProvider::Buffer buffer;
4269    sp<RecordTrack> activeTrack;
4270    Vector< sp<EffectChain> > effectChains;
4271
4272    nsecs_t lastWarning = 0;
4273
4274    acquireWakeLock();
4275
4276    // start recording
4277    while (!exitPending()) {
4278
4279        processConfigEvents();
4280
4281        { // scope for mLock
4282            Mutex::Autolock _l(mLock);
4283            checkForNewParameters_l();
4284            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4285                if (!mStandby) {
4286                    mInput->stream->common.standby(&mInput->stream->common);
4287                    mStandby = true;
4288                }
4289
4290                if (exitPending()) break;
4291
4292                releaseWakeLock_l();
4293                ALOGV("RecordThread: loop stopping");
4294                // go to sleep
4295                mWaitWorkCV.wait(mLock);
4296                ALOGV("RecordThread: loop starting");
4297                acquireWakeLock_l();
4298                continue;
4299            }
4300            if (mActiveTrack != 0) {
4301                if (mActiveTrack->mState == TrackBase::PAUSING) {
4302                    if (!mStandby) {
4303                        mInput->stream->common.standby(&mInput->stream->common);
4304                        mStandby = true;
4305                    }
4306                    mActiveTrack.clear();
4307                    mStartStopCond.broadcast();
4308                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4309                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4310                        mActiveTrack.clear();
4311                        mStartStopCond.broadcast();
4312                    } else if (mBytesRead != 0) {
4313                        // record start succeeds only if first read from audio input
4314                        // succeeds
4315                        if (mBytesRead > 0) {
4316                            mActiveTrack->mState = TrackBase::ACTIVE;
4317                        } else {
4318                            mActiveTrack.clear();
4319                        }
4320                        mStartStopCond.broadcast();
4321                    }
4322                    mStandby = false;
4323                }
4324            }
4325            lockEffectChains_l(effectChains);
4326        }
4327
4328        if (mActiveTrack != 0) {
4329            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4330                mActiveTrack->mState != TrackBase::RESUMING) {
4331                unlockEffectChains(effectChains);
4332                usleep(kRecordThreadSleepUs);
4333                continue;
4334            }
4335            for (size_t i = 0; i < effectChains.size(); i ++) {
4336                effectChains[i]->process_l();
4337            }
4338
4339            buffer.frameCount = mFrameCount;
4340            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4341                size_t framesOut = buffer.frameCount;
4342                if (mResampler == NULL) {
4343                    // no resampling
4344                    while (framesOut) {
4345                        size_t framesIn = mFrameCount - mRsmpInIndex;
4346                        if (framesIn) {
4347                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4348                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4349                            if (framesIn > framesOut)
4350                                framesIn = framesOut;
4351                            mRsmpInIndex += framesIn;
4352                            framesOut -= framesIn;
4353                            if ((int)mChannelCount == mReqChannelCount ||
4354                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4355                                memcpy(dst, src, framesIn * mFrameSize);
4356                            } else {
4357                                int16_t *src16 = (int16_t *)src;
4358                                int16_t *dst16 = (int16_t *)dst;
4359                                if (mChannelCount == 1) {
4360                                    while (framesIn--) {
4361                                        *dst16++ = *src16;
4362                                        *dst16++ = *src16++;
4363                                    }
4364                                } else {
4365                                    while (framesIn--) {
4366                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4367                                        src16 += 2;
4368                                    }
4369                                }
4370                            }
4371                        }
4372                        if (framesOut && mFrameCount == mRsmpInIndex) {
4373                            if (framesOut == mFrameCount &&
4374                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4375                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4376                                framesOut = 0;
4377                            } else {
4378                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4379                                mRsmpInIndex = 0;
4380                            }
4381                            if (mBytesRead < 0) {
4382                                ALOGE("Error reading audio input");
4383                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4384                                    // Force input into standby so that it tries to
4385                                    // recover at next read attempt
4386                                    mInput->stream->common.standby(&mInput->stream->common);
4387                                    usleep(kRecordThreadSleepUs);
4388                                }
4389                                mRsmpInIndex = mFrameCount;
4390                                framesOut = 0;
4391                                buffer.frameCount = 0;
4392                            }
4393                        }
4394                    }
4395                } else {
4396                    // resampling
4397
4398                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4399                    // alter output frame count as if we were expecting stereo samples
4400                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4401                        framesOut >>= 1;
4402                    }
4403                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4404                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4405                    // are 32 bit aligned which should be always true.
4406                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4407                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4408                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4409                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4410                        int16_t *dst = buffer.i16;
4411                        while (framesOut--) {
4412                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4413                            src += 2;
4414                        }
4415                    } else {
4416                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4417                    }
4418
4419                }
4420                mActiveTrack->releaseBuffer(&buffer);
4421                mActiveTrack->overflow();
4422            }
4423            // client isn't retrieving buffers fast enough
4424            else {
4425                if (!mActiveTrack->setOverflow()) {
4426                    nsecs_t now = systemTime();
4427                    if ((now - lastWarning) > kWarningThrottleNs) {
4428                        ALOGW("RecordThread: buffer overflow");
4429                        lastWarning = now;
4430                    }
4431                }
4432                // Release the processor for a while before asking for a new buffer.
4433                // This will give the application more chance to read from the buffer and
4434                // clear the overflow.
4435                usleep(kRecordThreadSleepUs);
4436            }
4437        }
4438        // enable changes in effect chain
4439        unlockEffectChains(effectChains);
4440        effectChains.clear();
4441    }
4442
4443    if (!mStandby) {
4444        mInput->stream->common.standby(&mInput->stream->common);
4445    }
4446    mActiveTrack.clear();
4447
4448    mStartStopCond.broadcast();
4449
4450    releaseWakeLock();
4451
4452    ALOGV("RecordThread %p exiting", this);
4453    return false;
4454}
4455
4456
4457sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4458        const sp<AudioFlinger::Client>& client,
4459        uint32_t sampleRate,
4460        int format,
4461        int channelMask,
4462        int frameCount,
4463        uint32_t flags,
4464        int sessionId,
4465        status_t *status)
4466{
4467    sp<RecordTrack> track;
4468    status_t lStatus;
4469
4470    lStatus = initCheck();
4471    if (lStatus != NO_ERROR) {
4472        ALOGE("Audio driver not initialized.");
4473        goto Exit;
4474    }
4475
4476    { // scope for mLock
4477        Mutex::Autolock _l(mLock);
4478
4479        track = new RecordTrack(this, client, sampleRate,
4480                      format, channelMask, frameCount, flags, sessionId);
4481
4482        if (track->getCblk() == NULL) {
4483            lStatus = NO_MEMORY;
4484            goto Exit;
4485        }
4486
4487        mTrack = track.get();
4488        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4489        bool suspend = audio_is_bluetooth_sco_device(
4490                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4491        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4492        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4493    }
4494    lStatus = NO_ERROR;
4495
4496Exit:
4497    if (status) {
4498        *status = lStatus;
4499    }
4500    return track;
4501}
4502
4503status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4504{
4505    ALOGV("RecordThread::start");
4506    sp <ThreadBase> strongMe = this;
4507    status_t status = NO_ERROR;
4508    {
4509        AutoMutex lock(mLock);
4510        if (mActiveTrack != 0) {
4511            if (recordTrack != mActiveTrack.get()) {
4512                status = -EBUSY;
4513            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4514                mActiveTrack->mState = TrackBase::ACTIVE;
4515            }
4516            return status;
4517        }
4518
4519        recordTrack->mState = TrackBase::IDLE;
4520        mActiveTrack = recordTrack;
4521        mLock.unlock();
4522        status_t status = AudioSystem::startInput(mId);
4523        mLock.lock();
4524        if (status != NO_ERROR) {
4525            mActiveTrack.clear();
4526            return status;
4527        }
4528        mRsmpInIndex = mFrameCount;
4529        mBytesRead = 0;
4530        if (mResampler != NULL) {
4531            mResampler->reset();
4532        }
4533        mActiveTrack->mState = TrackBase::RESUMING;
4534        // signal thread to start
4535        ALOGV("Signal record thread");
4536        mWaitWorkCV.signal();
4537        // do not wait for mStartStopCond if exiting
4538        if (mExiting) {
4539            mActiveTrack.clear();
4540            status = INVALID_OPERATION;
4541            goto startError;
4542        }
4543        mStartStopCond.wait(mLock);
4544        if (mActiveTrack == 0) {
4545            ALOGV("Record failed to start");
4546            status = BAD_VALUE;
4547            goto startError;
4548        }
4549        ALOGV("Record started OK");
4550        return status;
4551    }
4552startError:
4553    AudioSystem::stopInput(mId);
4554    return status;
4555}
4556
4557void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4558    ALOGV("RecordThread::stop");
4559    sp <ThreadBase> strongMe = this;
4560    {
4561        AutoMutex lock(mLock);
4562        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4563            mActiveTrack->mState = TrackBase::PAUSING;
4564            // do not wait for mStartStopCond if exiting
4565            if (mExiting) {
4566                return;
4567            }
4568            mStartStopCond.wait(mLock);
4569            // if we have been restarted, recordTrack == mActiveTrack.get() here
4570            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4571                mLock.unlock();
4572                AudioSystem::stopInput(mId);
4573                mLock.lock();
4574                ALOGV("Record stopped OK");
4575            }
4576        }
4577    }
4578}
4579
4580status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4581{
4582    const size_t SIZE = 256;
4583    char buffer[SIZE];
4584    String8 result;
4585    pid_t pid = 0;
4586
4587    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4588    result.append(buffer);
4589
4590    if (mActiveTrack != 0) {
4591        result.append("Active Track:\n");
4592        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4593        mActiveTrack->dump(buffer, SIZE);
4594        result.append(buffer);
4595
4596        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4597        result.append(buffer);
4598        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4599        result.append(buffer);
4600        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4601        result.append(buffer);
4602        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4603        result.append(buffer);
4604        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4605        result.append(buffer);
4606
4607
4608    } else {
4609        result.append("No record client\n");
4610    }
4611    write(fd, result.string(), result.size());
4612
4613    dumpBase(fd, args);
4614    dumpEffectChains(fd, args);
4615
4616    return NO_ERROR;
4617}
4618
4619status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4620{
4621    size_t framesReq = buffer->frameCount;
4622    size_t framesReady = mFrameCount - mRsmpInIndex;
4623    int channelCount;
4624
4625    if (framesReady == 0) {
4626        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4627        if (mBytesRead < 0) {
4628            ALOGE("RecordThread::getNextBuffer() Error reading audio input");
4629            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4630                // Force input into standby so that it tries to
4631                // recover at next read attempt
4632                mInput->stream->common.standby(&mInput->stream->common);
4633                usleep(kRecordThreadSleepUs);
4634            }
4635            buffer->raw = NULL;
4636            buffer->frameCount = 0;
4637            return NOT_ENOUGH_DATA;
4638        }
4639        mRsmpInIndex = 0;
4640        framesReady = mFrameCount;
4641    }
4642
4643    if (framesReq > framesReady) {
4644        framesReq = framesReady;
4645    }
4646
4647    if (mChannelCount == 1 && mReqChannelCount == 2) {
4648        channelCount = 1;
4649    } else {
4650        channelCount = 2;
4651    }
4652    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4653    buffer->frameCount = framesReq;
4654    return NO_ERROR;
4655}
4656
4657void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4658{
4659    mRsmpInIndex += buffer->frameCount;
4660    buffer->frameCount = 0;
4661}
4662
4663bool AudioFlinger::RecordThread::checkForNewParameters_l()
4664{
4665    bool reconfig = false;
4666
4667    while (!mNewParameters.isEmpty()) {
4668        status_t status = NO_ERROR;
4669        String8 keyValuePair = mNewParameters[0];
4670        AudioParameter param = AudioParameter(keyValuePair);
4671        int value;
4672        int reqFormat = mFormat;
4673        int reqSamplingRate = mReqSampleRate;
4674        int reqChannelCount = mReqChannelCount;
4675
4676        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4677            reqSamplingRate = value;
4678            reconfig = true;
4679        }
4680        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4681            reqFormat = value;
4682            reconfig = true;
4683        }
4684        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4685            reqChannelCount = popcount(value);
4686            reconfig = true;
4687        }
4688        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4689            // do not accept frame count changes if tracks are open as the track buffer
4690            // size depends on frame count and correct behavior would not be garantied
4691            // if frame count is changed after track creation
4692            if (mActiveTrack != 0) {
4693                status = INVALID_OPERATION;
4694            } else {
4695                reconfig = true;
4696            }
4697        }
4698        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4699            // forward device change to effects that have requested to be
4700            // aware of attached audio device.
4701            for (size_t i = 0; i < mEffectChains.size(); i++) {
4702                mEffectChains[i]->setDevice_l(value);
4703            }
4704            // store input device and output device but do not forward output device to audio HAL.
4705            // Note that status is ignored by the caller for output device
4706            // (see AudioFlinger::setParameters()
4707            if (value & AUDIO_DEVICE_OUT_ALL) {
4708                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4709                status = BAD_VALUE;
4710            } else {
4711                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4712                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4713                if (mTrack != NULL) {
4714                    bool suspend = audio_is_bluetooth_sco_device(
4715                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4716                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4717                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4718                }
4719            }
4720            mDevice |= (uint32_t)value;
4721        }
4722        if (status == NO_ERROR) {
4723            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4724            if (status == INVALID_OPERATION) {
4725               mInput->stream->common.standby(&mInput->stream->common);
4726               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4727            }
4728            if (reconfig) {
4729                if (status == BAD_VALUE &&
4730                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4731                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4732                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4733                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4734                    (reqChannelCount < 3)) {
4735                    status = NO_ERROR;
4736                }
4737                if (status == NO_ERROR) {
4738                    readInputParameters();
4739                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4740                }
4741            }
4742        }
4743
4744        mNewParameters.removeAt(0);
4745
4746        mParamStatus = status;
4747        mParamCond.signal();
4748        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4749        // already timed out waiting for the status and will never signal the condition.
4750        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4751    }
4752    return reconfig;
4753}
4754
4755String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4756{
4757    char *s;
4758    String8 out_s8 = String8();
4759
4760    Mutex::Autolock _l(mLock);
4761    if (initCheck() != NO_ERROR) {
4762        return out_s8;
4763    }
4764
4765    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4766    out_s8 = String8(s);
4767    free(s);
4768    return out_s8;
4769}
4770
4771void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4772    AudioSystem::OutputDescriptor desc;
4773    void *param2 = 0;
4774
4775    switch (event) {
4776    case AudioSystem::INPUT_OPENED:
4777    case AudioSystem::INPUT_CONFIG_CHANGED:
4778        desc.channels = mChannelMask;
4779        desc.samplingRate = mSampleRate;
4780        desc.format = mFormat;
4781        desc.frameCount = mFrameCount;
4782        desc.latency = 0;
4783        param2 = &desc;
4784        break;
4785
4786    case AudioSystem::INPUT_CLOSED:
4787    default:
4788        break;
4789    }
4790    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4791}
4792
4793void AudioFlinger::RecordThread::readInputParameters()
4794{
4795    if (mRsmpInBuffer) delete mRsmpInBuffer;
4796    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4797    if (mResampler) delete mResampler;
4798    mResampler = NULL;
4799
4800    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4801    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4802    mChannelCount = (uint16_t)popcount(mChannelMask);
4803    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4804    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
4805    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4806    mFrameCount = mInputBytes / mFrameSize;
4807    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4808
4809    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4810    {
4811        int channelCount;
4812         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4813         // stereo to mono post process as the resampler always outputs stereo.
4814        if (mChannelCount == 1 && mReqChannelCount == 2) {
4815            channelCount = 1;
4816        } else {
4817            channelCount = 2;
4818        }
4819        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4820        mResampler->setSampleRate(mSampleRate);
4821        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4822        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4823
4824        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4825        if (mChannelCount == 1 && mReqChannelCount == 1) {
4826            mFrameCount >>= 1;
4827        }
4828
4829    }
4830    mRsmpInIndex = mFrameCount;
4831}
4832
4833unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4834{
4835    Mutex::Autolock _l(mLock);
4836    if (initCheck() != NO_ERROR) {
4837        return 0;
4838    }
4839
4840    return mInput->stream->get_input_frames_lost(mInput->stream);
4841}
4842
4843uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4844{
4845    Mutex::Autolock _l(mLock);
4846    uint32_t result = 0;
4847    if (getEffectChain_l(sessionId) != 0) {
4848        result = EFFECT_SESSION;
4849    }
4850
4851    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4852        result |= TRACK_SESSION;
4853    }
4854
4855    return result;
4856}
4857
4858AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4859{
4860    Mutex::Autolock _l(mLock);
4861    return mTrack;
4862}
4863
4864AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
4865{
4866    Mutex::Autolock _l(mLock);
4867    return mInput;
4868}
4869
4870AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4871{
4872    Mutex::Autolock _l(mLock);
4873    AudioStreamIn *input = mInput;
4874    mInput = NULL;
4875    return input;
4876}
4877
4878// this method must always be called either with ThreadBase mLock held or inside the thread loop
4879audio_stream_t* AudioFlinger::RecordThread::stream()
4880{
4881    if (mInput == NULL) {
4882        return NULL;
4883    }
4884    return &mInput->stream->common;
4885}
4886
4887
4888// ----------------------------------------------------------------------------
4889
4890int AudioFlinger::openOutput(uint32_t *pDevices,
4891                                uint32_t *pSamplingRate,
4892                                uint32_t *pFormat,
4893                                uint32_t *pChannels,
4894                                uint32_t *pLatencyMs,
4895                                uint32_t flags)
4896{
4897    status_t status;
4898    PlaybackThread *thread = NULL;
4899    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4900    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4901    uint32_t format = pFormat ? *pFormat : 0;
4902    uint32_t channels = pChannels ? *pChannels : 0;
4903    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4904    audio_stream_out_t *outStream;
4905    audio_hw_device_t *outHwDev;
4906
4907    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4908            pDevices ? *pDevices : 0,
4909            samplingRate,
4910            format,
4911            channels,
4912            flags);
4913
4914    if (pDevices == NULL || *pDevices == 0) {
4915        return 0;
4916    }
4917
4918    Mutex::Autolock _l(mLock);
4919
4920    outHwDev = findSuitableHwDev_l(*pDevices);
4921    if (outHwDev == NULL)
4922        return 0;
4923
4924    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
4925                                          &channels, &samplingRate, &outStream);
4926    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4927            outStream,
4928            samplingRate,
4929            format,
4930            channels,
4931            status);
4932
4933    mHardwareStatus = AUDIO_HW_IDLE;
4934    if (outStream != NULL) {
4935        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4936        int id = nextUniqueId();
4937
4938        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4939            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4940            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4941            thread = new DirectOutputThread(this, output, id, *pDevices);
4942            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4943        } else {
4944            thread = new MixerThread(this, output, id, *pDevices);
4945            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4946        }
4947        mPlaybackThreads.add(id, thread);
4948
4949        if (pSamplingRate) *pSamplingRate = samplingRate;
4950        if (pFormat) *pFormat = format;
4951        if (pChannels) *pChannels = channels;
4952        if (pLatencyMs) *pLatencyMs = thread->latency();
4953
4954        // notify client processes of the new output creation
4955        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4956        return id;
4957    }
4958
4959    return 0;
4960}
4961
4962int AudioFlinger::openDuplicateOutput(int output1, int output2)
4963{
4964    Mutex::Autolock _l(mLock);
4965    MixerThread *thread1 = checkMixerThread_l(output1);
4966    MixerThread *thread2 = checkMixerThread_l(output2);
4967
4968    if (thread1 == NULL || thread2 == NULL) {
4969        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
4970        return 0;
4971    }
4972
4973    int id = nextUniqueId();
4974    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
4975    thread->addOutputTrack(thread2);
4976    mPlaybackThreads.add(id, thread);
4977    // notify client processes of the new output creation
4978    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4979    return id;
4980}
4981
4982status_t AudioFlinger::closeOutput(int output)
4983{
4984    // keep strong reference on the playback thread so that
4985    // it is not destroyed while exit() is executed
4986    sp <PlaybackThread> thread;
4987    {
4988        Mutex::Autolock _l(mLock);
4989        thread = checkPlaybackThread_l(output);
4990        if (thread == NULL) {
4991            return BAD_VALUE;
4992        }
4993
4994        ALOGV("closeOutput() %d", output);
4995
4996        if (thread->type() == ThreadBase::MIXER) {
4997            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4998                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
4999                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
5000                    dupThread->removeOutputTrack((MixerThread *)thread.get());
5001                }
5002            }
5003        }
5004        void *param2 = 0;
5005        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
5006        mPlaybackThreads.removeItem(output);
5007    }
5008    thread->exit();
5009
5010    if (thread->type() != ThreadBase::DUPLICATING) {
5011        AudioStreamOut *out = thread->clearOutput();
5012        // from now on thread->mOutput is NULL
5013        out->hwDev->close_output_stream(out->hwDev, out->stream);
5014        delete out;
5015    }
5016    return NO_ERROR;
5017}
5018
5019status_t AudioFlinger::suspendOutput(int output)
5020{
5021    Mutex::Autolock _l(mLock);
5022    PlaybackThread *thread = checkPlaybackThread_l(output);
5023
5024    if (thread == NULL) {
5025        return BAD_VALUE;
5026    }
5027
5028    ALOGV("suspendOutput() %d", output);
5029    thread->suspend();
5030
5031    return NO_ERROR;
5032}
5033
5034status_t AudioFlinger::restoreOutput(int output)
5035{
5036    Mutex::Autolock _l(mLock);
5037    PlaybackThread *thread = checkPlaybackThread_l(output);
5038
5039    if (thread == NULL) {
5040        return BAD_VALUE;
5041    }
5042
5043    ALOGV("restoreOutput() %d", output);
5044
5045    thread->restore();
5046
5047    return NO_ERROR;
5048}
5049
5050int AudioFlinger::openInput(uint32_t *pDevices,
5051                                uint32_t *pSamplingRate,
5052                                uint32_t *pFormat,
5053                                uint32_t *pChannels,
5054                                uint32_t acoustics)
5055{
5056    status_t status;
5057    RecordThread *thread = NULL;
5058    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5059    uint32_t format = pFormat ? *pFormat : 0;
5060    uint32_t channels = pChannels ? *pChannels : 0;
5061    uint32_t reqSamplingRate = samplingRate;
5062    uint32_t reqFormat = format;
5063    uint32_t reqChannels = channels;
5064    audio_stream_in_t *inStream;
5065    audio_hw_device_t *inHwDev;
5066
5067    if (pDevices == NULL || *pDevices == 0) {
5068        return 0;
5069    }
5070
5071    Mutex::Autolock _l(mLock);
5072
5073    inHwDev = findSuitableHwDev_l(*pDevices);
5074    if (inHwDev == NULL)
5075        return 0;
5076
5077    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5078                                        &channels, &samplingRate,
5079                                        (audio_in_acoustics_t)acoustics,
5080                                        &inStream);
5081    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5082            inStream,
5083            samplingRate,
5084            format,
5085            channels,
5086            acoustics,
5087            status);
5088
5089    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5090    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5091    // or stereo to mono conversions on 16 bit PCM inputs.
5092    if (inStream == NULL && status == BAD_VALUE &&
5093        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5094        (samplingRate <= 2 * reqSamplingRate) &&
5095        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5096        ALOGV("openInput() reopening with proposed sampling rate and channels");
5097        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5098                                            &channels, &samplingRate,
5099                                            (audio_in_acoustics_t)acoustics,
5100                                            &inStream);
5101    }
5102
5103    if (inStream != NULL) {
5104        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5105
5106        int id = nextUniqueId();
5107        // Start record thread
5108        // RecorThread require both input and output device indication to forward to audio
5109        // pre processing modules
5110        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5111        thread = new RecordThread(this,
5112                                  input,
5113                                  reqSamplingRate,
5114                                  reqChannels,
5115                                  id,
5116                                  device);
5117        mRecordThreads.add(id, thread);
5118        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5119        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5120        if (pFormat) *pFormat = format;
5121        if (pChannels) *pChannels = reqChannels;
5122
5123        input->stream->common.standby(&input->stream->common);
5124
5125        // notify client processes of the new input creation
5126        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5127        return id;
5128    }
5129
5130    return 0;
5131}
5132
5133status_t AudioFlinger::closeInput(int input)
5134{
5135    // keep strong reference on the record thread so that
5136    // it is not destroyed while exit() is executed
5137    sp <RecordThread> thread;
5138    {
5139        Mutex::Autolock _l(mLock);
5140        thread = checkRecordThread_l(input);
5141        if (thread == NULL) {
5142            return BAD_VALUE;
5143        }
5144
5145        ALOGV("closeInput() %d", input);
5146        void *param2 = 0;
5147        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5148        mRecordThreads.removeItem(input);
5149    }
5150    thread->exit();
5151
5152    AudioStreamIn *in = thread->clearInput();
5153    // from now on thread->mInput is NULL
5154    in->hwDev->close_input_stream(in->hwDev, in->stream);
5155    delete in;
5156
5157    return NO_ERROR;
5158}
5159
5160status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
5161{
5162    Mutex::Autolock _l(mLock);
5163    MixerThread *dstThread = checkMixerThread_l(output);
5164    if (dstThread == NULL) {
5165        ALOGW("setStreamOutput() bad output id %d", output);
5166        return BAD_VALUE;
5167    }
5168
5169    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5170    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5171
5172    dstThread->setStreamValid(stream, true);
5173
5174    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5175        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5176        if (thread != dstThread &&
5177            thread->type() != ThreadBase::DIRECT) {
5178            MixerThread *srcThread = (MixerThread *)thread;
5179            srcThread->setStreamValid(stream, false);
5180            srcThread->invalidateTracks(stream);
5181        }
5182    }
5183
5184    return NO_ERROR;
5185}
5186
5187
5188int AudioFlinger::newAudioSessionId()
5189{
5190    return nextUniqueId();
5191}
5192
5193void AudioFlinger::acquireAudioSessionId(int audioSession)
5194{
5195    Mutex::Autolock _l(mLock);
5196    int caller = IPCThreadState::self()->getCallingPid();
5197    ALOGV("acquiring %d from %d", audioSession, caller);
5198    int num = mAudioSessionRefs.size();
5199    for (int i = 0; i< num; i++) {
5200        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5201        if (ref->sessionid == audioSession && ref->pid == caller) {
5202            ref->cnt++;
5203            ALOGV(" incremented refcount to %d", ref->cnt);
5204            return;
5205        }
5206    }
5207    AudioSessionRef *ref = new AudioSessionRef();
5208    ref->sessionid = audioSession;
5209    ref->pid = caller;
5210    ref->cnt = 1;
5211    mAudioSessionRefs.push(ref);
5212    ALOGV(" added new entry for %d", ref->sessionid);
5213}
5214
5215void AudioFlinger::releaseAudioSessionId(int audioSession)
5216{
5217    Mutex::Autolock _l(mLock);
5218    int caller = IPCThreadState::self()->getCallingPid();
5219    ALOGV("releasing %d from %d", audioSession, caller);
5220    int num = mAudioSessionRefs.size();
5221    for (int i = 0; i< num; i++) {
5222        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5223        if (ref->sessionid == audioSession && ref->pid == caller) {
5224            ref->cnt--;
5225            ALOGV(" decremented refcount to %d", ref->cnt);
5226            if (ref->cnt == 0) {
5227                mAudioSessionRefs.removeAt(i);
5228                delete ref;
5229                purgeStaleEffects_l();
5230            }
5231            return;
5232        }
5233    }
5234    ALOGW("session id %d not found for pid %d", audioSession, caller);
5235}
5236
5237void AudioFlinger::purgeStaleEffects_l() {
5238
5239    ALOGV("purging stale effects");
5240
5241    Vector< sp<EffectChain> > chains;
5242
5243    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5244        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5245        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5246            sp<EffectChain> ec = t->mEffectChains[j];
5247            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5248                chains.push(ec);
5249            }
5250        }
5251    }
5252    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5253        sp<RecordThread> t = mRecordThreads.valueAt(i);
5254        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5255            sp<EffectChain> ec = t->mEffectChains[j];
5256            chains.push(ec);
5257        }
5258    }
5259
5260    for (size_t i = 0; i < chains.size(); i++) {
5261        sp<EffectChain> ec = chains[i];
5262        int sessionid = ec->sessionId();
5263        sp<ThreadBase> t = ec->mThread.promote();
5264        if (t == 0) {
5265            continue;
5266        }
5267        size_t numsessionrefs = mAudioSessionRefs.size();
5268        bool found = false;
5269        for (size_t k = 0; k < numsessionrefs; k++) {
5270            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5271            if (ref->sessionid == sessionid) {
5272                ALOGV(" session %d still exists for %d with %d refs",
5273                     sessionid, ref->pid, ref->cnt);
5274                found = true;
5275                break;
5276            }
5277        }
5278        if (!found) {
5279            // remove all effects from the chain
5280            while (ec->mEffects.size()) {
5281                sp<EffectModule> effect = ec->mEffects[0];
5282                effect->unPin();
5283                Mutex::Autolock _l (t->mLock);
5284                t->removeEffect_l(effect);
5285                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5286                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5287                    if (handle != 0) {
5288                        handle->mEffect.clear();
5289                        if (handle->mHasControl && handle->mEnabled) {
5290                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5291                        }
5292                    }
5293                }
5294                AudioSystem::unregisterEffect(effect->id());
5295            }
5296        }
5297    }
5298    return;
5299}
5300
5301// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5302AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5303{
5304    PlaybackThread *thread = NULL;
5305    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5306        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5307    }
5308    return thread;
5309}
5310
5311// checkMixerThread_l() must be called with AudioFlinger::mLock held
5312AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5313{
5314    PlaybackThread *thread = checkPlaybackThread_l(output);
5315    if (thread != NULL) {
5316        if (thread->type() == ThreadBase::DIRECT) {
5317            thread = NULL;
5318        }
5319    }
5320    return (MixerThread *)thread;
5321}
5322
5323// checkRecordThread_l() must be called with AudioFlinger::mLock held
5324AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5325{
5326    RecordThread *thread = NULL;
5327    if (mRecordThreads.indexOfKey(input) >= 0) {
5328        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5329    }
5330    return thread;
5331}
5332
5333uint32_t AudioFlinger::nextUniqueId()
5334{
5335    return android_atomic_inc(&mNextUniqueId);
5336}
5337
5338AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5339{
5340    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5341        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5342        AudioStreamOut *output = thread->getOutput();
5343        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5344            return thread;
5345        }
5346    }
5347    return NULL;
5348}
5349
5350uint32_t AudioFlinger::primaryOutputDevice_l()
5351{
5352    PlaybackThread *thread = primaryPlaybackThread_l();
5353
5354    if (thread == NULL) {
5355        return 0;
5356    }
5357
5358    return thread->device();
5359}
5360
5361
5362// ----------------------------------------------------------------------------
5363//  Effect management
5364// ----------------------------------------------------------------------------
5365
5366
5367status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5368{
5369    Mutex::Autolock _l(mLock);
5370    return EffectQueryNumberEffects(numEffects);
5371}
5372
5373status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5374{
5375    Mutex::Autolock _l(mLock);
5376    return EffectQueryEffect(index, descriptor);
5377}
5378
5379status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5380{
5381    Mutex::Autolock _l(mLock);
5382    return EffectGetDescriptor(pUuid, descriptor);
5383}
5384
5385
5386sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5387        effect_descriptor_t *pDesc,
5388        const sp<IEffectClient>& effectClient,
5389        int32_t priority,
5390        int io,
5391        int sessionId,
5392        status_t *status,
5393        int *id,
5394        int *enabled)
5395{
5396    status_t lStatus = NO_ERROR;
5397    sp<EffectHandle> handle;
5398    effect_descriptor_t desc;
5399    sp<Client> client;
5400    wp<Client> wclient;
5401
5402    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5403            pid, effectClient.get(), priority, sessionId, io);
5404
5405    if (pDesc == NULL) {
5406        lStatus = BAD_VALUE;
5407        goto Exit;
5408    }
5409
5410    // check audio settings permission for global effects
5411    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5412        lStatus = PERMISSION_DENIED;
5413        goto Exit;
5414    }
5415
5416    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5417    // that can only be created by audio policy manager (running in same process)
5418    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5419        lStatus = PERMISSION_DENIED;
5420        goto Exit;
5421    }
5422
5423    if (io == 0) {
5424        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5425            // output must be specified by AudioPolicyManager when using session
5426            // AUDIO_SESSION_OUTPUT_STAGE
5427            lStatus = BAD_VALUE;
5428            goto Exit;
5429        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5430            // if the output returned by getOutputForEffect() is removed before we lock the
5431            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5432            // and we will exit safely
5433            io = AudioSystem::getOutputForEffect(&desc);
5434        }
5435    }
5436
5437    {
5438        Mutex::Autolock _l(mLock);
5439
5440
5441        if (!EffectIsNullUuid(&pDesc->uuid)) {
5442            // if uuid is specified, request effect descriptor
5443            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5444            if (lStatus < 0) {
5445                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5446                goto Exit;
5447            }
5448        } else {
5449            // if uuid is not specified, look for an available implementation
5450            // of the required type in effect factory
5451            if (EffectIsNullUuid(&pDesc->type)) {
5452                ALOGW("createEffect() no effect type");
5453                lStatus = BAD_VALUE;
5454                goto Exit;
5455            }
5456            uint32_t numEffects = 0;
5457            effect_descriptor_t d;
5458            d.flags = 0; // prevent compiler warning
5459            bool found = false;
5460
5461            lStatus = EffectQueryNumberEffects(&numEffects);
5462            if (lStatus < 0) {
5463                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5464                goto Exit;
5465            }
5466            for (uint32_t i = 0; i < numEffects; i++) {
5467                lStatus = EffectQueryEffect(i, &desc);
5468                if (lStatus < 0) {
5469                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5470                    continue;
5471                }
5472                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5473                    // If matching type found save effect descriptor. If the session is
5474                    // 0 and the effect is not auxiliary, continue enumeration in case
5475                    // an auxiliary version of this effect type is available
5476                    found = true;
5477                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5478                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5479                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5480                        break;
5481                    }
5482                }
5483            }
5484            if (!found) {
5485                lStatus = BAD_VALUE;
5486                ALOGW("createEffect() effect not found");
5487                goto Exit;
5488            }
5489            // For same effect type, chose auxiliary version over insert version if
5490            // connect to output mix (Compliance to OpenSL ES)
5491            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5492                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5493                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5494            }
5495        }
5496
5497        // Do not allow auxiliary effects on a session different from 0 (output mix)
5498        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5499             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5500            lStatus = INVALID_OPERATION;
5501            goto Exit;
5502        }
5503
5504        // check recording permission for visualizer
5505        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5506            !recordingAllowed()) {
5507            lStatus = PERMISSION_DENIED;
5508            goto Exit;
5509        }
5510
5511        // return effect descriptor
5512        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5513
5514        // If output is not specified try to find a matching audio session ID in one of the
5515        // output threads.
5516        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5517        // because of code checking output when entering the function.
5518        // Note: io is never 0 when creating an effect on an input
5519        if (io == 0) {
5520             // look for the thread where the specified audio session is present
5521            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5522                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5523                    io = mPlaybackThreads.keyAt(i);
5524                    break;
5525                }
5526            }
5527            if (io == 0) {
5528               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5529                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5530                       io = mRecordThreads.keyAt(i);
5531                       break;
5532                   }
5533               }
5534            }
5535            // If no output thread contains the requested session ID, default to
5536            // first output. The effect chain will be moved to the correct output
5537            // thread when a track with the same session ID is created
5538            if (io == 0 && mPlaybackThreads.size()) {
5539                io = mPlaybackThreads.keyAt(0);
5540            }
5541            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5542        }
5543        ThreadBase *thread = checkRecordThread_l(io);
5544        if (thread == NULL) {
5545            thread = checkPlaybackThread_l(io);
5546            if (thread == NULL) {
5547                ALOGE("createEffect() unknown output thread");
5548                lStatus = BAD_VALUE;
5549                goto Exit;
5550            }
5551        }
5552
5553        wclient = mClients.valueFor(pid);
5554
5555        if (wclient != NULL) {
5556            client = wclient.promote();
5557        } else {
5558            client = new Client(this, pid);
5559            mClients.add(pid, client);
5560        }
5561
5562        // create effect on selected output thread
5563        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5564                &desc, enabled, &lStatus);
5565        if (handle != 0 && id != NULL) {
5566            *id = handle->id();
5567        }
5568    }
5569
5570Exit:
5571    if(status) {
5572        *status = lStatus;
5573    }
5574    return handle;
5575}
5576
5577status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5578{
5579    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5580            sessionId, srcOutput, dstOutput);
5581    Mutex::Autolock _l(mLock);
5582    if (srcOutput == dstOutput) {
5583        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
5584        return NO_ERROR;
5585    }
5586    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5587    if (srcThread == NULL) {
5588        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
5589        return BAD_VALUE;
5590    }
5591    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5592    if (dstThread == NULL) {
5593        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
5594        return BAD_VALUE;
5595    }
5596
5597    Mutex::Autolock _dl(dstThread->mLock);
5598    Mutex::Autolock _sl(srcThread->mLock);
5599    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5600
5601    return NO_ERROR;
5602}
5603
5604// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5605status_t AudioFlinger::moveEffectChain_l(int sessionId,
5606                                   AudioFlinger::PlaybackThread *srcThread,
5607                                   AudioFlinger::PlaybackThread *dstThread,
5608                                   bool reRegister)
5609{
5610    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5611            sessionId, srcThread, dstThread);
5612
5613    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5614    if (chain == 0) {
5615        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5616                sessionId, srcThread);
5617        return INVALID_OPERATION;
5618    }
5619
5620    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5621    // so that a new chain is created with correct parameters when first effect is added. This is
5622    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5623    // removed.
5624    srcThread->removeEffectChain_l(chain);
5625
5626    // transfer all effects one by one so that new effect chain is created on new thread with
5627    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5628    int dstOutput = dstThread->id();
5629    sp<EffectChain> dstChain;
5630    uint32_t strategy = 0; // prevent compiler warning
5631    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5632    while (effect != 0) {
5633        srcThread->removeEffect_l(effect);
5634        dstThread->addEffect_l(effect);
5635        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5636        if (effect->state() == EffectModule::ACTIVE ||
5637                effect->state() == EffectModule::STOPPING) {
5638            effect->start();
5639        }
5640        // if the move request is not received from audio policy manager, the effect must be
5641        // re-registered with the new strategy and output
5642        if (dstChain == 0) {
5643            dstChain = effect->chain().promote();
5644            if (dstChain == 0) {
5645                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5646                srcThread->addEffect_l(effect);
5647                return NO_INIT;
5648            }
5649            strategy = dstChain->strategy();
5650        }
5651        if (reRegister) {
5652            AudioSystem::unregisterEffect(effect->id());
5653            AudioSystem::registerEffect(&effect->desc(),
5654                                        dstOutput,
5655                                        strategy,
5656                                        sessionId,
5657                                        effect->id());
5658        }
5659        effect = chain->getEffectFromId_l(0);
5660    }
5661
5662    return NO_ERROR;
5663}
5664
5665
5666// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5667sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5668        const sp<AudioFlinger::Client>& client,
5669        const sp<IEffectClient>& effectClient,
5670        int32_t priority,
5671        int sessionId,
5672        effect_descriptor_t *desc,
5673        int *enabled,
5674        status_t *status
5675        )
5676{
5677    sp<EffectModule> effect;
5678    sp<EffectHandle> handle;
5679    status_t lStatus;
5680    sp<EffectChain> chain;
5681    bool chainCreated = false;
5682    bool effectCreated = false;
5683    bool effectRegistered = false;
5684
5685    lStatus = initCheck();
5686    if (lStatus != NO_ERROR) {
5687        ALOGW("createEffect_l() Audio driver not initialized.");
5688        goto Exit;
5689    }
5690
5691    // Do not allow effects with session ID 0 on direct output or duplicating threads
5692    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5693    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5694        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5695                desc->name, sessionId);
5696        lStatus = BAD_VALUE;
5697        goto Exit;
5698    }
5699    // Only Pre processor effects are allowed on input threads and only on input threads
5700    if ((mType == RECORD &&
5701            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5702            (mType != RECORD &&
5703                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5704        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5705                desc->name, desc->flags, mType);
5706        lStatus = BAD_VALUE;
5707        goto Exit;
5708    }
5709
5710    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5711
5712    { // scope for mLock
5713        Mutex::Autolock _l(mLock);
5714
5715        // check for existing effect chain with the requested audio session
5716        chain = getEffectChain_l(sessionId);
5717        if (chain == 0) {
5718            // create a new chain for this session
5719            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5720            chain = new EffectChain(this, sessionId);
5721            addEffectChain_l(chain);
5722            chain->setStrategy(getStrategyForSession_l(sessionId));
5723            chainCreated = true;
5724        } else {
5725            effect = chain->getEffectFromDesc_l(desc);
5726        }
5727
5728        ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
5729
5730        if (effect == 0) {
5731            int id = mAudioFlinger->nextUniqueId();
5732            // Check CPU and memory usage
5733            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5734            if (lStatus != NO_ERROR) {
5735                goto Exit;
5736            }
5737            effectRegistered = true;
5738            // create a new effect module if none present in the chain
5739            effect = new EffectModule(this, chain, desc, id, sessionId);
5740            lStatus = effect->status();
5741            if (lStatus != NO_ERROR) {
5742                goto Exit;
5743            }
5744            lStatus = chain->addEffect_l(effect);
5745            if (lStatus != NO_ERROR) {
5746                goto Exit;
5747            }
5748            effectCreated = true;
5749
5750            effect->setDevice(mDevice);
5751            effect->setMode(mAudioFlinger->getMode());
5752        }
5753        // create effect handle and connect it to effect module
5754        handle = new EffectHandle(effect, client, effectClient, priority);
5755        lStatus = effect->addHandle(handle);
5756        if (enabled) {
5757            *enabled = (int)effect->isEnabled();
5758        }
5759    }
5760
5761Exit:
5762    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5763        Mutex::Autolock _l(mLock);
5764        if (effectCreated) {
5765            chain->removeEffect_l(effect);
5766        }
5767        if (effectRegistered) {
5768            AudioSystem::unregisterEffect(effect->id());
5769        }
5770        if (chainCreated) {
5771            removeEffectChain_l(chain);
5772        }
5773        handle.clear();
5774    }
5775
5776    if(status) {
5777        *status = lStatus;
5778    }
5779    return handle;
5780}
5781
5782sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5783{
5784    sp<EffectModule> effect;
5785
5786    sp<EffectChain> chain = getEffectChain_l(sessionId);
5787    if (chain != 0) {
5788        effect = chain->getEffectFromId_l(effectId);
5789    }
5790    return effect;
5791}
5792
5793// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5794// PlaybackThread::mLock held
5795status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5796{
5797    // check for existing effect chain with the requested audio session
5798    int sessionId = effect->sessionId();
5799    sp<EffectChain> chain = getEffectChain_l(sessionId);
5800    bool chainCreated = false;
5801
5802    if (chain == 0) {
5803        // create a new chain for this session
5804        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5805        chain = new EffectChain(this, sessionId);
5806        addEffectChain_l(chain);
5807        chain->setStrategy(getStrategyForSession_l(sessionId));
5808        chainCreated = true;
5809    }
5810    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5811
5812    if (chain->getEffectFromId_l(effect->id()) != 0) {
5813        ALOGW("addEffect_l() %p effect %s already present in chain %p",
5814                this, effect->desc().name, chain.get());
5815        return BAD_VALUE;
5816    }
5817
5818    status_t status = chain->addEffect_l(effect);
5819    if (status != NO_ERROR) {
5820        if (chainCreated) {
5821            removeEffectChain_l(chain);
5822        }
5823        return status;
5824    }
5825
5826    effect->setDevice(mDevice);
5827    effect->setMode(mAudioFlinger->getMode());
5828    return NO_ERROR;
5829}
5830
5831void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5832
5833    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5834    effect_descriptor_t desc = effect->desc();
5835    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5836        detachAuxEffect_l(effect->id());
5837    }
5838
5839    sp<EffectChain> chain = effect->chain().promote();
5840    if (chain != 0) {
5841        // remove effect chain if removing last effect
5842        if (chain->removeEffect_l(effect) == 0) {
5843            removeEffectChain_l(chain);
5844        }
5845    } else {
5846        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5847    }
5848}
5849
5850void AudioFlinger::ThreadBase::lockEffectChains_l(
5851        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5852{
5853    effectChains = mEffectChains;
5854    for (size_t i = 0; i < mEffectChains.size(); i++) {
5855        mEffectChains[i]->lock();
5856    }
5857}
5858
5859void AudioFlinger::ThreadBase::unlockEffectChains(
5860        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5861{
5862    for (size_t i = 0; i < effectChains.size(); i++) {
5863        effectChains[i]->unlock();
5864    }
5865}
5866
5867sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5868{
5869    Mutex::Autolock _l(mLock);
5870    return getEffectChain_l(sessionId);
5871}
5872
5873sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5874{
5875    sp<EffectChain> chain;
5876
5877    size_t size = mEffectChains.size();
5878    for (size_t i = 0; i < size; i++) {
5879        if (mEffectChains[i]->sessionId() == sessionId) {
5880            chain = mEffectChains[i];
5881            break;
5882        }
5883    }
5884    return chain;
5885}
5886
5887void AudioFlinger::ThreadBase::setMode(uint32_t mode)
5888{
5889    Mutex::Autolock _l(mLock);
5890    size_t size = mEffectChains.size();
5891    for (size_t i = 0; i < size; i++) {
5892        mEffectChains[i]->setMode_l(mode);
5893    }
5894}
5895
5896void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5897                                                    const wp<EffectHandle>& handle,
5898                                                    bool unpiniflast) {
5899
5900    Mutex::Autolock _l(mLock);
5901    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5902    // delete the effect module if removing last handle on it
5903    if (effect->removeHandle(handle) == 0) {
5904        if (!effect->isPinned() || unpiniflast) {
5905            removeEffect_l(effect);
5906            AudioSystem::unregisterEffect(effect->id());
5907        }
5908    }
5909}
5910
5911status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5912{
5913    int session = chain->sessionId();
5914    int16_t *buffer = mMixBuffer;
5915    bool ownsBuffer = false;
5916
5917    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5918    if (session > 0) {
5919        // Only one effect chain can be present in direct output thread and it uses
5920        // the mix buffer as input
5921        if (mType != DIRECT) {
5922            size_t numSamples = mFrameCount * mChannelCount;
5923            buffer = new int16_t[numSamples];
5924            memset(buffer, 0, numSamples * sizeof(int16_t));
5925            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5926            ownsBuffer = true;
5927        }
5928
5929        // Attach all tracks with same session ID to this chain.
5930        for (size_t i = 0; i < mTracks.size(); ++i) {
5931            sp<Track> track = mTracks[i];
5932            if (session == track->sessionId()) {
5933                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5934                track->setMainBuffer(buffer);
5935                chain->incTrackCnt();
5936            }
5937        }
5938
5939        // indicate all active tracks in the chain
5940        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5941            sp<Track> track = mActiveTracks[i].promote();
5942            if (track == 0) continue;
5943            if (session == track->sessionId()) {
5944                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5945                chain->incActiveTrackCnt();
5946            }
5947        }
5948    }
5949
5950    chain->setInBuffer(buffer, ownsBuffer);
5951    chain->setOutBuffer(mMixBuffer);
5952    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5953    // chains list in order to be processed last as it contains output stage effects
5954    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5955    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5956    // after track specific effects and before output stage
5957    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5958    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5959    // Effect chain for other sessions are inserted at beginning of effect
5960    // chains list to be processed before output mix effects. Relative order between other
5961    // sessions is not important
5962    size_t size = mEffectChains.size();
5963    size_t i = 0;
5964    for (i = 0; i < size; i++) {
5965        if (mEffectChains[i]->sessionId() < session) break;
5966    }
5967    mEffectChains.insertAt(chain, i);
5968    checkSuspendOnAddEffectChain_l(chain);
5969
5970    return NO_ERROR;
5971}
5972
5973size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
5974{
5975    int session = chain->sessionId();
5976
5977    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
5978
5979    for (size_t i = 0; i < mEffectChains.size(); i++) {
5980        if (chain == mEffectChains[i]) {
5981            mEffectChains.removeAt(i);
5982            // detach all active tracks from the chain
5983            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5984                sp<Track> track = mActiveTracks[i].promote();
5985                if (track == 0) continue;
5986                if (session == track->sessionId()) {
5987                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
5988                            chain.get(), session);
5989                    chain->decActiveTrackCnt();
5990                }
5991            }
5992
5993            // detach all tracks with same session ID from this chain
5994            for (size_t i = 0; i < mTracks.size(); ++i) {
5995                sp<Track> track = mTracks[i];
5996                if (session == track->sessionId()) {
5997                    track->setMainBuffer(mMixBuffer);
5998                    chain->decTrackCnt();
5999                }
6000            }
6001            break;
6002        }
6003    }
6004    return mEffectChains.size();
6005}
6006
6007status_t AudioFlinger::PlaybackThread::attachAuxEffect(
6008        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6009{
6010    Mutex::Autolock _l(mLock);
6011    return attachAuxEffect_l(track, EffectId);
6012}
6013
6014status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6015        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6016{
6017    status_t status = NO_ERROR;
6018
6019    if (EffectId == 0) {
6020        track->setAuxBuffer(0, NULL);
6021    } else {
6022        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6023        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6024        if (effect != 0) {
6025            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6026                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6027            } else {
6028                status = INVALID_OPERATION;
6029            }
6030        } else {
6031            status = BAD_VALUE;
6032        }
6033    }
6034    return status;
6035}
6036
6037void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6038{
6039     for (size_t i = 0; i < mTracks.size(); ++i) {
6040        sp<Track> track = mTracks[i];
6041        if (track->auxEffectId() == effectId) {
6042            attachAuxEffect_l(track, 0);
6043        }
6044    }
6045}
6046
6047status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6048{
6049    // only one chain per input thread
6050    if (mEffectChains.size() != 0) {
6051        return INVALID_OPERATION;
6052    }
6053    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6054
6055    chain->setInBuffer(NULL);
6056    chain->setOutBuffer(NULL);
6057
6058    checkSuspendOnAddEffectChain_l(chain);
6059
6060    mEffectChains.add(chain);
6061
6062    return NO_ERROR;
6063}
6064
6065size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6066{
6067    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6068    ALOGW_IF(mEffectChains.size() != 1,
6069            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6070            chain.get(), mEffectChains.size(), this);
6071    if (mEffectChains.size() == 1) {
6072        mEffectChains.removeAt(0);
6073    }
6074    return 0;
6075}
6076
6077// ----------------------------------------------------------------------------
6078//  EffectModule implementation
6079// ----------------------------------------------------------------------------
6080
6081#undef LOG_TAG
6082#define LOG_TAG "AudioFlinger::EffectModule"
6083
6084AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6085                                        const wp<AudioFlinger::EffectChain>& chain,
6086                                        effect_descriptor_t *desc,
6087                                        int id,
6088                                        int sessionId)
6089    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6090      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6091{
6092    ALOGV("Constructor %p", this);
6093    int lStatus;
6094    sp<ThreadBase> thread = mThread.promote();
6095    if (thread == 0) {
6096        return;
6097    }
6098
6099    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6100
6101    // create effect engine from effect factory
6102    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6103
6104    if (mStatus != NO_ERROR) {
6105        return;
6106    }
6107    lStatus = init();
6108    if (lStatus < 0) {
6109        mStatus = lStatus;
6110        goto Error;
6111    }
6112
6113    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6114        mPinned = true;
6115    }
6116    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6117    return;
6118Error:
6119    EffectRelease(mEffectInterface);
6120    mEffectInterface = NULL;
6121    ALOGV("Constructor Error %d", mStatus);
6122}
6123
6124AudioFlinger::EffectModule::~EffectModule()
6125{
6126    ALOGV("Destructor %p", this);
6127    if (mEffectInterface != NULL) {
6128        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6129                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6130            sp<ThreadBase> thread = mThread.promote();
6131            if (thread != 0) {
6132                audio_stream_t *stream = thread->stream();
6133                if (stream != NULL) {
6134                    stream->remove_audio_effect(stream, mEffectInterface);
6135                }
6136            }
6137        }
6138        // release effect engine
6139        EffectRelease(mEffectInterface);
6140    }
6141}
6142
6143status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6144{
6145    status_t status;
6146
6147    Mutex::Autolock _l(mLock);
6148    // First handle in mHandles has highest priority and controls the effect module
6149    int priority = handle->priority();
6150    size_t size = mHandles.size();
6151    sp<EffectHandle> h;
6152    size_t i;
6153    for (i = 0; i < size; i++) {
6154        h = mHandles[i].promote();
6155        if (h == 0) continue;
6156        if (h->priority() <= priority) break;
6157    }
6158    // if inserted in first place, move effect control from previous owner to this handle
6159    if (i == 0) {
6160        bool enabled = false;
6161        if (h != 0) {
6162            enabled = h->enabled();
6163            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6164        }
6165        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6166        status = NO_ERROR;
6167    } else {
6168        status = ALREADY_EXISTS;
6169    }
6170    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6171    mHandles.insertAt(handle, i);
6172    return status;
6173}
6174
6175size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6176{
6177    Mutex::Autolock _l(mLock);
6178    size_t size = mHandles.size();
6179    size_t i;
6180    for (i = 0; i < size; i++) {
6181        if (mHandles[i] == handle) break;
6182    }
6183    if (i == size) {
6184        return size;
6185    }
6186    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6187
6188    bool enabled = false;
6189    EffectHandle *hdl = handle.unsafe_get();
6190    if (hdl) {
6191        ALOGV("removeHandle() unsafe_get OK");
6192        enabled = hdl->enabled();
6193    }
6194    mHandles.removeAt(i);
6195    size = mHandles.size();
6196    // if removed from first place, move effect control from this handle to next in line
6197    if (i == 0 && size != 0) {
6198        sp<EffectHandle> h = mHandles[0].promote();
6199        if (h != 0) {
6200            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6201        }
6202    }
6203
6204    // Prevent calls to process() and other functions on effect interface from now on.
6205    // The effect engine will be released by the destructor when the last strong reference on
6206    // this object is released which can happen after next process is called.
6207    if (size == 0 && !mPinned) {
6208        mState = DESTROYED;
6209    }
6210
6211    return size;
6212}
6213
6214sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6215{
6216    Mutex::Autolock _l(mLock);
6217    sp<EffectHandle> handle;
6218    if (mHandles.size() != 0) {
6219        handle = mHandles[0].promote();
6220    }
6221    return handle;
6222}
6223
6224void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6225{
6226    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6227    // keep a strong reference on this EffectModule to avoid calling the
6228    // destructor before we exit
6229    sp<EffectModule> keep(this);
6230    {
6231        sp<ThreadBase> thread = mThread.promote();
6232        if (thread != 0) {
6233            thread->disconnectEffect(keep, handle, unpiniflast);
6234        }
6235    }
6236}
6237
6238void AudioFlinger::EffectModule::updateState() {
6239    Mutex::Autolock _l(mLock);
6240
6241    switch (mState) {
6242    case RESTART:
6243        reset_l();
6244        // FALL THROUGH
6245
6246    case STARTING:
6247        // clear auxiliary effect input buffer for next accumulation
6248        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6249            memset(mConfig.inputCfg.buffer.raw,
6250                   0,
6251                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6252        }
6253        start_l();
6254        mState = ACTIVE;
6255        break;
6256    case STOPPING:
6257        stop_l();
6258        mDisableWaitCnt = mMaxDisableWaitCnt;
6259        mState = STOPPED;
6260        break;
6261    case STOPPED:
6262        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6263        // turn off sequence.
6264        if (--mDisableWaitCnt == 0) {
6265            reset_l();
6266            mState = IDLE;
6267        }
6268        break;
6269    default: //IDLE , ACTIVE, DESTROYED
6270        break;
6271    }
6272}
6273
6274void AudioFlinger::EffectModule::process()
6275{
6276    Mutex::Autolock _l(mLock);
6277
6278    if (mState == DESTROYED || mEffectInterface == NULL ||
6279            mConfig.inputCfg.buffer.raw == NULL ||
6280            mConfig.outputCfg.buffer.raw == NULL) {
6281        return;
6282    }
6283
6284    if (isProcessEnabled()) {
6285        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6286        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6287            ditherAndClamp(mConfig.inputCfg.buffer.s32,
6288                                        mConfig.inputCfg.buffer.s32,
6289                                        mConfig.inputCfg.buffer.frameCount/2);
6290        }
6291
6292        // do the actual processing in the effect engine
6293        int ret = (*mEffectInterface)->process(mEffectInterface,
6294                                               &mConfig.inputCfg.buffer,
6295                                               &mConfig.outputCfg.buffer);
6296
6297        // force transition to IDLE state when engine is ready
6298        if (mState == STOPPED && ret == -ENODATA) {
6299            mDisableWaitCnt = 1;
6300        }
6301
6302        // clear auxiliary effect input buffer for next accumulation
6303        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6304            memset(mConfig.inputCfg.buffer.raw, 0,
6305                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6306        }
6307    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6308                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6309        // If an insert effect is idle and input buffer is different from output buffer,
6310        // accumulate input onto output
6311        sp<EffectChain> chain = mChain.promote();
6312        if (chain != 0 && chain->activeTrackCnt() != 0) {
6313            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6314            int16_t *in = mConfig.inputCfg.buffer.s16;
6315            int16_t *out = mConfig.outputCfg.buffer.s16;
6316            for (size_t i = 0; i < frameCnt; i++) {
6317                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6318            }
6319        }
6320    }
6321}
6322
6323void AudioFlinger::EffectModule::reset_l()
6324{
6325    if (mEffectInterface == NULL) {
6326        return;
6327    }
6328    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6329}
6330
6331status_t AudioFlinger::EffectModule::configure()
6332{
6333    uint32_t channels;
6334    if (mEffectInterface == NULL) {
6335        return NO_INIT;
6336    }
6337
6338    sp<ThreadBase> thread = mThread.promote();
6339    if (thread == 0) {
6340        return DEAD_OBJECT;
6341    }
6342
6343    // TODO: handle configuration of effects replacing track process
6344    if (thread->channelCount() == 1) {
6345        channels = AUDIO_CHANNEL_OUT_MONO;
6346    } else {
6347        channels = AUDIO_CHANNEL_OUT_STEREO;
6348    }
6349
6350    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6351        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6352    } else {
6353        mConfig.inputCfg.channels = channels;
6354    }
6355    mConfig.outputCfg.channels = channels;
6356    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6357    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6358    mConfig.inputCfg.samplingRate = thread->sampleRate();
6359    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6360    mConfig.inputCfg.bufferProvider.cookie = NULL;
6361    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6362    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6363    mConfig.outputCfg.bufferProvider.cookie = NULL;
6364    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6365    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6366    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6367    // Insert effect:
6368    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6369    // always overwrites output buffer: input buffer == output buffer
6370    // - in other sessions:
6371    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6372    //      other effect: overwrites output buffer: input buffer == output buffer
6373    // Auxiliary effect:
6374    //      accumulates in output buffer: input buffer != output buffer
6375    // Therefore: accumulate <=> input buffer != output buffer
6376    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6377        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6378    } else {
6379        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6380    }
6381    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6382    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6383    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6384    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6385
6386    ALOGV("configure() %p thread %p buffer %p framecount %d",
6387            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6388
6389    status_t cmdStatus;
6390    uint32_t size = sizeof(int);
6391    status_t status = (*mEffectInterface)->command(mEffectInterface,
6392                                                   EFFECT_CMD_SET_CONFIG,
6393                                                   sizeof(effect_config_t),
6394                                                   &mConfig,
6395                                                   &size,
6396                                                   &cmdStatus);
6397    if (status == 0) {
6398        status = cmdStatus;
6399    }
6400
6401    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6402            (1000 * mConfig.outputCfg.buffer.frameCount);
6403
6404    return status;
6405}
6406
6407status_t AudioFlinger::EffectModule::init()
6408{
6409    Mutex::Autolock _l(mLock);
6410    if (mEffectInterface == NULL) {
6411        return NO_INIT;
6412    }
6413    status_t cmdStatus;
6414    uint32_t size = sizeof(status_t);
6415    status_t status = (*mEffectInterface)->command(mEffectInterface,
6416                                                   EFFECT_CMD_INIT,
6417                                                   0,
6418                                                   NULL,
6419                                                   &size,
6420                                                   &cmdStatus);
6421    if (status == 0) {
6422        status = cmdStatus;
6423    }
6424    return status;
6425}
6426
6427status_t AudioFlinger::EffectModule::start()
6428{
6429    Mutex::Autolock _l(mLock);
6430    return start_l();
6431}
6432
6433status_t AudioFlinger::EffectModule::start_l()
6434{
6435    if (mEffectInterface == NULL) {
6436        return NO_INIT;
6437    }
6438    status_t cmdStatus;
6439    uint32_t size = sizeof(status_t);
6440    status_t status = (*mEffectInterface)->command(mEffectInterface,
6441                                                   EFFECT_CMD_ENABLE,
6442                                                   0,
6443                                                   NULL,
6444                                                   &size,
6445                                                   &cmdStatus);
6446    if (status == 0) {
6447        status = cmdStatus;
6448    }
6449    if (status == 0 &&
6450            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6451             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6452        sp<ThreadBase> thread = mThread.promote();
6453        if (thread != 0) {
6454            audio_stream_t *stream = thread->stream();
6455            if (stream != NULL) {
6456                stream->add_audio_effect(stream, mEffectInterface);
6457            }
6458        }
6459    }
6460    return status;
6461}
6462
6463status_t AudioFlinger::EffectModule::stop()
6464{
6465    Mutex::Autolock _l(mLock);
6466    return stop_l();
6467}
6468
6469status_t AudioFlinger::EffectModule::stop_l()
6470{
6471    if (mEffectInterface == NULL) {
6472        return NO_INIT;
6473    }
6474    status_t cmdStatus;
6475    uint32_t size = sizeof(status_t);
6476    status_t status = (*mEffectInterface)->command(mEffectInterface,
6477                                                   EFFECT_CMD_DISABLE,
6478                                                   0,
6479                                                   NULL,
6480                                                   &size,
6481                                                   &cmdStatus);
6482    if (status == 0) {
6483        status = cmdStatus;
6484    }
6485    if (status == 0 &&
6486            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6487             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6488        sp<ThreadBase> thread = mThread.promote();
6489        if (thread != 0) {
6490            audio_stream_t *stream = thread->stream();
6491            if (stream != NULL) {
6492                stream->remove_audio_effect(stream, mEffectInterface);
6493            }
6494        }
6495    }
6496    return status;
6497}
6498
6499status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6500                                             uint32_t cmdSize,
6501                                             void *pCmdData,
6502                                             uint32_t *replySize,
6503                                             void *pReplyData)
6504{
6505    Mutex::Autolock _l(mLock);
6506//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6507
6508    if (mState == DESTROYED || mEffectInterface == NULL) {
6509        return NO_INIT;
6510    }
6511    status_t status = (*mEffectInterface)->command(mEffectInterface,
6512                                                   cmdCode,
6513                                                   cmdSize,
6514                                                   pCmdData,
6515                                                   replySize,
6516                                                   pReplyData);
6517    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6518        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6519        for (size_t i = 1; i < mHandles.size(); i++) {
6520            sp<EffectHandle> h = mHandles[i].promote();
6521            if (h != 0) {
6522                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6523            }
6524        }
6525    }
6526    return status;
6527}
6528
6529status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6530{
6531
6532    Mutex::Autolock _l(mLock);
6533    ALOGV("setEnabled %p enabled %d", this, enabled);
6534
6535    if (enabled != isEnabled()) {
6536        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6537        if (enabled && status != NO_ERROR) {
6538            return status;
6539        }
6540
6541        switch (mState) {
6542        // going from disabled to enabled
6543        case IDLE:
6544            mState = STARTING;
6545            break;
6546        case STOPPED:
6547            mState = RESTART;
6548            break;
6549        case STOPPING:
6550            mState = ACTIVE;
6551            break;
6552
6553        // going from enabled to disabled
6554        case RESTART:
6555            mState = STOPPED;
6556            break;
6557        case STARTING:
6558            mState = IDLE;
6559            break;
6560        case ACTIVE:
6561            mState = STOPPING;
6562            break;
6563        case DESTROYED:
6564            return NO_ERROR; // simply ignore as we are being destroyed
6565        }
6566        for (size_t i = 1; i < mHandles.size(); i++) {
6567            sp<EffectHandle> h = mHandles[i].promote();
6568            if (h != 0) {
6569                h->setEnabled(enabled);
6570            }
6571        }
6572    }
6573    return NO_ERROR;
6574}
6575
6576bool AudioFlinger::EffectModule::isEnabled()
6577{
6578    switch (mState) {
6579    case RESTART:
6580    case STARTING:
6581    case ACTIVE:
6582        return true;
6583    case IDLE:
6584    case STOPPING:
6585    case STOPPED:
6586    case DESTROYED:
6587    default:
6588        return false;
6589    }
6590}
6591
6592bool AudioFlinger::EffectModule::isProcessEnabled()
6593{
6594    switch (mState) {
6595    case RESTART:
6596    case ACTIVE:
6597    case STOPPING:
6598    case STOPPED:
6599        return true;
6600    case IDLE:
6601    case STARTING:
6602    case DESTROYED:
6603    default:
6604        return false;
6605    }
6606}
6607
6608status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6609{
6610    Mutex::Autolock _l(mLock);
6611    status_t status = NO_ERROR;
6612
6613    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6614    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6615    if (isProcessEnabled() &&
6616            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6617            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6618        status_t cmdStatus;
6619        uint32_t volume[2];
6620        uint32_t *pVolume = NULL;
6621        uint32_t size = sizeof(volume);
6622        volume[0] = *left;
6623        volume[1] = *right;
6624        if (controller) {
6625            pVolume = volume;
6626        }
6627        status = (*mEffectInterface)->command(mEffectInterface,
6628                                              EFFECT_CMD_SET_VOLUME,
6629                                              size,
6630                                              volume,
6631                                              &size,
6632                                              pVolume);
6633        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6634            *left = volume[0];
6635            *right = volume[1];
6636        }
6637    }
6638    return status;
6639}
6640
6641status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6642{
6643    Mutex::Autolock _l(mLock);
6644    status_t status = NO_ERROR;
6645    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6646        // audio pre processing modules on RecordThread can receive both output and
6647        // input device indication in the same call
6648        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6649        if (dev) {
6650            status_t cmdStatus;
6651            uint32_t size = sizeof(status_t);
6652
6653            status = (*mEffectInterface)->command(mEffectInterface,
6654                                                  EFFECT_CMD_SET_DEVICE,
6655                                                  sizeof(uint32_t),
6656                                                  &dev,
6657                                                  &size,
6658                                                  &cmdStatus);
6659            if (status == NO_ERROR) {
6660                status = cmdStatus;
6661            }
6662        }
6663        dev = device & AUDIO_DEVICE_IN_ALL;
6664        if (dev) {
6665            status_t cmdStatus;
6666            uint32_t size = sizeof(status_t);
6667
6668            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6669                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6670                                                  sizeof(uint32_t),
6671                                                  &dev,
6672                                                  &size,
6673                                                  &cmdStatus);
6674            if (status2 == NO_ERROR) {
6675                status2 = cmdStatus;
6676            }
6677            if (status == NO_ERROR) {
6678                status = status2;
6679            }
6680        }
6681    }
6682    return status;
6683}
6684
6685status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
6686{
6687    Mutex::Autolock _l(mLock);
6688    status_t status = NO_ERROR;
6689    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6690        status_t cmdStatus;
6691        uint32_t size = sizeof(status_t);
6692        status = (*mEffectInterface)->command(mEffectInterface,
6693                                              EFFECT_CMD_SET_AUDIO_MODE,
6694                                              sizeof(int),
6695                                              &mode,
6696                                              &size,
6697                                              &cmdStatus);
6698        if (status == NO_ERROR) {
6699            status = cmdStatus;
6700        }
6701    }
6702    return status;
6703}
6704
6705void AudioFlinger::EffectModule::setSuspended(bool suspended)
6706{
6707    Mutex::Autolock _l(mLock);
6708    mSuspended = suspended;
6709}
6710
6711bool AudioFlinger::EffectModule::suspended() const
6712{
6713    Mutex::Autolock _l(mLock);
6714    return mSuspended;
6715}
6716
6717status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6718{
6719    const size_t SIZE = 256;
6720    char buffer[SIZE];
6721    String8 result;
6722
6723    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6724    result.append(buffer);
6725
6726    bool locked = tryLock(mLock);
6727    // failed to lock - AudioFlinger is probably deadlocked
6728    if (!locked) {
6729        result.append("\t\tCould not lock Fx mutex:\n");
6730    }
6731
6732    result.append("\t\tSession Status State Engine:\n");
6733    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6734            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6735    result.append(buffer);
6736
6737    result.append("\t\tDescriptor:\n");
6738    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6739            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6740            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6741            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6742    result.append(buffer);
6743    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6744                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6745                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6746                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6747    result.append(buffer);
6748    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6749            mDescriptor.apiVersion,
6750            mDescriptor.flags);
6751    result.append(buffer);
6752    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6753            mDescriptor.name);
6754    result.append(buffer);
6755    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6756            mDescriptor.implementor);
6757    result.append(buffer);
6758
6759    result.append("\t\t- Input configuration:\n");
6760    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6761    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6762            (uint32_t)mConfig.inputCfg.buffer.raw,
6763            mConfig.inputCfg.buffer.frameCount,
6764            mConfig.inputCfg.samplingRate,
6765            mConfig.inputCfg.channels,
6766            mConfig.inputCfg.format);
6767    result.append(buffer);
6768
6769    result.append("\t\t- Output configuration:\n");
6770    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6771    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6772            (uint32_t)mConfig.outputCfg.buffer.raw,
6773            mConfig.outputCfg.buffer.frameCount,
6774            mConfig.outputCfg.samplingRate,
6775            mConfig.outputCfg.channels,
6776            mConfig.outputCfg.format);
6777    result.append(buffer);
6778
6779    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6780    result.append(buffer);
6781    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6782    for (size_t i = 0; i < mHandles.size(); ++i) {
6783        sp<EffectHandle> handle = mHandles[i].promote();
6784        if (handle != 0) {
6785            handle->dump(buffer, SIZE);
6786            result.append(buffer);
6787        }
6788    }
6789
6790    result.append("\n");
6791
6792    write(fd, result.string(), result.length());
6793
6794    if (locked) {
6795        mLock.unlock();
6796    }
6797
6798    return NO_ERROR;
6799}
6800
6801// ----------------------------------------------------------------------------
6802//  EffectHandle implementation
6803// ----------------------------------------------------------------------------
6804
6805#undef LOG_TAG
6806#define LOG_TAG "AudioFlinger::EffectHandle"
6807
6808AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6809                                        const sp<AudioFlinger::Client>& client,
6810                                        const sp<IEffectClient>& effectClient,
6811                                        int32_t priority)
6812    : BnEffect(),
6813    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6814    mPriority(priority), mHasControl(false), mEnabled(false)
6815{
6816    ALOGV("constructor %p", this);
6817
6818    if (client == 0) {
6819        return;
6820    }
6821    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6822    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6823    if (mCblkMemory != 0) {
6824        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6825
6826        if (mCblk) {
6827            new(mCblk) effect_param_cblk_t();
6828            mBuffer = (uint8_t *)mCblk + bufOffset;
6829         }
6830    } else {
6831        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6832        return;
6833    }
6834}
6835
6836AudioFlinger::EffectHandle::~EffectHandle()
6837{
6838    ALOGV("Destructor %p", this);
6839    disconnect(false);
6840    ALOGV("Destructor DONE %p", this);
6841}
6842
6843status_t AudioFlinger::EffectHandle::enable()
6844{
6845    ALOGV("enable %p", this);
6846    if (!mHasControl) return INVALID_OPERATION;
6847    if (mEffect == 0) return DEAD_OBJECT;
6848
6849    if (mEnabled) {
6850        return NO_ERROR;
6851    }
6852
6853    mEnabled = true;
6854
6855    sp<ThreadBase> thread = mEffect->thread().promote();
6856    if (thread != 0) {
6857        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6858    }
6859
6860    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6861    if (mEffect->suspended()) {
6862        return NO_ERROR;
6863    }
6864
6865    status_t status = mEffect->setEnabled(true);
6866    if (status != NO_ERROR) {
6867        if (thread != 0) {
6868            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6869        }
6870        mEnabled = false;
6871    }
6872    return status;
6873}
6874
6875status_t AudioFlinger::EffectHandle::disable()
6876{
6877    ALOGV("disable %p", this);
6878    if (!mHasControl) return INVALID_OPERATION;
6879    if (mEffect == 0) return DEAD_OBJECT;
6880
6881    if (!mEnabled) {
6882        return NO_ERROR;
6883    }
6884    mEnabled = false;
6885
6886    if (mEffect->suspended()) {
6887        return NO_ERROR;
6888    }
6889
6890    status_t status = mEffect->setEnabled(false);
6891
6892    sp<ThreadBase> thread = mEffect->thread().promote();
6893    if (thread != 0) {
6894        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6895    }
6896
6897    return status;
6898}
6899
6900void AudioFlinger::EffectHandle::disconnect()
6901{
6902    disconnect(true);
6903}
6904
6905void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6906{
6907    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6908    if (mEffect == 0) {
6909        return;
6910    }
6911    mEffect->disconnect(this, unpiniflast);
6912
6913    if (mHasControl && mEnabled) {
6914        sp<ThreadBase> thread = mEffect->thread().promote();
6915        if (thread != 0) {
6916            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6917        }
6918    }
6919
6920    // release sp on module => module destructor can be called now
6921    mEffect.clear();
6922    if (mClient != 0) {
6923        if (mCblk) {
6924            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6925        }
6926        mCblkMemory.clear();            // and free the shared memory
6927        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6928        mClient.clear();
6929    }
6930}
6931
6932status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6933                                             uint32_t cmdSize,
6934                                             void *pCmdData,
6935                                             uint32_t *replySize,
6936                                             void *pReplyData)
6937{
6938//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6939//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6940
6941    // only get parameter command is permitted for applications not controlling the effect
6942    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6943        return INVALID_OPERATION;
6944    }
6945    if (mEffect == 0) return DEAD_OBJECT;
6946    if (mClient == 0) return INVALID_OPERATION;
6947
6948    // handle commands that are not forwarded transparently to effect engine
6949    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6950        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6951        // no risk to block the whole media server process or mixer threads is we are stuck here
6952        Mutex::Autolock _l(mCblk->lock);
6953        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6954            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6955            mCblk->serverIndex = 0;
6956            mCblk->clientIndex = 0;
6957            return BAD_VALUE;
6958        }
6959        status_t status = NO_ERROR;
6960        while (mCblk->serverIndex < mCblk->clientIndex) {
6961            int reply;
6962            uint32_t rsize = sizeof(int);
6963            int *p = (int *)(mBuffer + mCblk->serverIndex);
6964            int size = *p++;
6965            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
6966                ALOGW("command(): invalid parameter block size");
6967                break;
6968            }
6969            effect_param_t *param = (effect_param_t *)p;
6970            if (param->psize == 0 || param->vsize == 0) {
6971                ALOGW("command(): null parameter or value size");
6972                mCblk->serverIndex += size;
6973                continue;
6974            }
6975            uint32_t psize = sizeof(effect_param_t) +
6976                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
6977                             param->vsize;
6978            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
6979                                            psize,
6980                                            p,
6981                                            &rsize,
6982                                            &reply);
6983            // stop at first error encountered
6984            if (ret != NO_ERROR) {
6985                status = ret;
6986                *(int *)pReplyData = reply;
6987                break;
6988            } else if (reply != NO_ERROR) {
6989                *(int *)pReplyData = reply;
6990                break;
6991            }
6992            mCblk->serverIndex += size;
6993        }
6994        mCblk->serverIndex = 0;
6995        mCblk->clientIndex = 0;
6996        return status;
6997    } else if (cmdCode == EFFECT_CMD_ENABLE) {
6998        *(int *)pReplyData = NO_ERROR;
6999        return enable();
7000    } else if (cmdCode == EFFECT_CMD_DISABLE) {
7001        *(int *)pReplyData = NO_ERROR;
7002        return disable();
7003    }
7004
7005    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7006}
7007
7008sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
7009    return mCblkMemory;
7010}
7011
7012void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
7013{
7014    ALOGV("setControl %p control %d", this, hasControl);
7015
7016    mHasControl = hasControl;
7017    mEnabled = enabled;
7018
7019    if (signal && mEffectClient != 0) {
7020        mEffectClient->controlStatusChanged(hasControl);
7021    }
7022}
7023
7024void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7025                                                 uint32_t cmdSize,
7026                                                 void *pCmdData,
7027                                                 uint32_t replySize,
7028                                                 void *pReplyData)
7029{
7030    if (mEffectClient != 0) {
7031        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7032    }
7033}
7034
7035
7036
7037void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7038{
7039    if (mEffectClient != 0) {
7040        mEffectClient->enableStatusChanged(enabled);
7041    }
7042}
7043
7044status_t AudioFlinger::EffectHandle::onTransact(
7045    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7046{
7047    return BnEffect::onTransact(code, data, reply, flags);
7048}
7049
7050
7051void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7052{
7053    bool locked = mCblk ? tryLock(mCblk->lock) : false;
7054
7055    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7056            (mClient == NULL) ? getpid() : mClient->pid(),
7057            mPriority,
7058            mHasControl,
7059            !locked,
7060            mCblk ? mCblk->clientIndex : 0,
7061            mCblk ? mCblk->serverIndex : 0
7062            );
7063
7064    if (locked) {
7065        mCblk->lock.unlock();
7066    }
7067}
7068
7069#undef LOG_TAG
7070#define LOG_TAG "AudioFlinger::EffectChain"
7071
7072AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7073                                        int sessionId)
7074    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7075      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7076      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7077{
7078    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7079    sp<ThreadBase> thread = mThread.promote();
7080    if (thread == 0) {
7081        return;
7082    }
7083    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7084                                    thread->frameCount();
7085}
7086
7087AudioFlinger::EffectChain::~EffectChain()
7088{
7089    if (mOwnInBuffer) {
7090        delete mInBuffer;
7091    }
7092
7093}
7094
7095// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7096sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7097{
7098    sp<EffectModule> effect;
7099    size_t size = mEffects.size();
7100
7101    for (size_t i = 0; i < size; i++) {
7102        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7103            effect = mEffects[i];
7104            break;
7105        }
7106    }
7107    return effect;
7108}
7109
7110// getEffectFromId_l() must be called with ThreadBase::mLock held
7111sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7112{
7113    sp<EffectModule> effect;
7114    size_t size = mEffects.size();
7115
7116    for (size_t i = 0; i < size; i++) {
7117        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7118        if (id == 0 || mEffects[i]->id() == id) {
7119            effect = mEffects[i];
7120            break;
7121        }
7122    }
7123    return effect;
7124}
7125
7126// getEffectFromType_l() must be called with ThreadBase::mLock held
7127sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7128        const effect_uuid_t *type)
7129{
7130    sp<EffectModule> effect;
7131    size_t size = mEffects.size();
7132
7133    for (size_t i = 0; i < size; i++) {
7134        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7135            effect = mEffects[i];
7136            break;
7137        }
7138    }
7139    return effect;
7140}
7141
7142// Must be called with EffectChain::mLock locked
7143void AudioFlinger::EffectChain::process_l()
7144{
7145    sp<ThreadBase> thread = mThread.promote();
7146    if (thread == 0) {
7147        ALOGW("process_l(): cannot promote mixer thread");
7148        return;
7149    }
7150    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7151            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7152    // always process effects unless no more tracks are on the session and the effect tail
7153    // has been rendered
7154    bool doProcess = true;
7155    if (!isGlobalSession) {
7156        bool tracksOnSession = (trackCnt() != 0);
7157
7158        if (!tracksOnSession && mTailBufferCount == 0) {
7159            doProcess = false;
7160        }
7161
7162        if (activeTrackCnt() == 0) {
7163            // if no track is active and the effect tail has not been rendered,
7164            // the input buffer must be cleared here as the mixer process will not do it
7165            if (tracksOnSession || mTailBufferCount > 0) {
7166                size_t numSamples = thread->frameCount() * thread->channelCount();
7167                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7168                if (mTailBufferCount > 0) {
7169                    mTailBufferCount--;
7170                }
7171            }
7172        }
7173    }
7174
7175    size_t size = mEffects.size();
7176    if (doProcess) {
7177        for (size_t i = 0; i < size; i++) {
7178            mEffects[i]->process();
7179        }
7180    }
7181    for (size_t i = 0; i < size; i++) {
7182        mEffects[i]->updateState();
7183    }
7184}
7185
7186// addEffect_l() must be called with PlaybackThread::mLock held
7187status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7188{
7189    effect_descriptor_t desc = effect->desc();
7190    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7191
7192    Mutex::Autolock _l(mLock);
7193    effect->setChain(this);
7194    sp<ThreadBase> thread = mThread.promote();
7195    if (thread == 0) {
7196        return NO_INIT;
7197    }
7198    effect->setThread(thread);
7199
7200    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7201        // Auxiliary effects are inserted at the beginning of mEffects vector as
7202        // they are processed first and accumulated in chain input buffer
7203        mEffects.insertAt(effect, 0);
7204
7205        // the input buffer for auxiliary effect contains mono samples in
7206        // 32 bit format. This is to avoid saturation in AudoMixer
7207        // accumulation stage. Saturation is done in EffectModule::process() before
7208        // calling the process in effect engine
7209        size_t numSamples = thread->frameCount();
7210        int32_t *buffer = new int32_t[numSamples];
7211        memset(buffer, 0, numSamples * sizeof(int32_t));
7212        effect->setInBuffer((int16_t *)buffer);
7213        // auxiliary effects output samples to chain input buffer for further processing
7214        // by insert effects
7215        effect->setOutBuffer(mInBuffer);
7216    } else {
7217        // Insert effects are inserted at the end of mEffects vector as they are processed
7218        //  after track and auxiliary effects.
7219        // Insert effect order as a function of indicated preference:
7220        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7221        //  another effect is present
7222        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7223        //  last effect claiming first position
7224        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7225        //  first effect claiming last position
7226        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7227        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7228        // already present
7229
7230        int size = (int)mEffects.size();
7231        int idx_insert = size;
7232        int idx_insert_first = -1;
7233        int idx_insert_last = -1;
7234
7235        for (int i = 0; i < size; i++) {
7236            effect_descriptor_t d = mEffects[i]->desc();
7237            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7238            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7239            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7240                // check invalid effect chaining combinations
7241                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7242                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7243                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7244                    return INVALID_OPERATION;
7245                }
7246                // remember position of first insert effect and by default
7247                // select this as insert position for new effect
7248                if (idx_insert == size) {
7249                    idx_insert = i;
7250                }
7251                // remember position of last insert effect claiming
7252                // first position
7253                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7254                    idx_insert_first = i;
7255                }
7256                // remember position of first insert effect claiming
7257                // last position
7258                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7259                    idx_insert_last == -1) {
7260                    idx_insert_last = i;
7261                }
7262            }
7263        }
7264
7265        // modify idx_insert from first position if needed
7266        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7267            if (idx_insert_last != -1) {
7268                idx_insert = idx_insert_last;
7269            } else {
7270                idx_insert = size;
7271            }
7272        } else {
7273            if (idx_insert_first != -1) {
7274                idx_insert = idx_insert_first + 1;
7275            }
7276        }
7277
7278        // always read samples from chain input buffer
7279        effect->setInBuffer(mInBuffer);
7280
7281        // if last effect in the chain, output samples to chain
7282        // output buffer, otherwise to chain input buffer
7283        if (idx_insert == size) {
7284            if (idx_insert != 0) {
7285                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7286                mEffects[idx_insert-1]->configure();
7287            }
7288            effect->setOutBuffer(mOutBuffer);
7289        } else {
7290            effect->setOutBuffer(mInBuffer);
7291        }
7292        mEffects.insertAt(effect, idx_insert);
7293
7294        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7295    }
7296    effect->configure();
7297    return NO_ERROR;
7298}
7299
7300// removeEffect_l() must be called with PlaybackThread::mLock held
7301size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7302{
7303    Mutex::Autolock _l(mLock);
7304    int size = (int)mEffects.size();
7305    int i;
7306    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7307
7308    for (i = 0; i < size; i++) {
7309        if (effect == mEffects[i]) {
7310            // calling stop here will remove pre-processing effect from the audio HAL.
7311            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7312            // the middle of a read from audio HAL
7313            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7314                    mEffects[i]->state() == EffectModule::STOPPING) {
7315                mEffects[i]->stop();
7316            }
7317            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7318                delete[] effect->inBuffer();
7319            } else {
7320                if (i == size - 1 && i != 0) {
7321                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7322                    mEffects[i - 1]->configure();
7323                }
7324            }
7325            mEffects.removeAt(i);
7326            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7327            break;
7328        }
7329    }
7330
7331    return mEffects.size();
7332}
7333
7334// setDevice_l() must be called with PlaybackThread::mLock held
7335void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7336{
7337    size_t size = mEffects.size();
7338    for (size_t i = 0; i < size; i++) {
7339        mEffects[i]->setDevice(device);
7340    }
7341}
7342
7343// setMode_l() must be called with PlaybackThread::mLock held
7344void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
7345{
7346    size_t size = mEffects.size();
7347    for (size_t i = 0; i < size; i++) {
7348        mEffects[i]->setMode(mode);
7349    }
7350}
7351
7352// setVolume_l() must be called with PlaybackThread::mLock held
7353bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7354{
7355    uint32_t newLeft = *left;
7356    uint32_t newRight = *right;
7357    bool hasControl = false;
7358    int ctrlIdx = -1;
7359    size_t size = mEffects.size();
7360
7361    // first update volume controller
7362    for (size_t i = size; i > 0; i--) {
7363        if (mEffects[i - 1]->isProcessEnabled() &&
7364            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7365            ctrlIdx = i - 1;
7366            hasControl = true;
7367            break;
7368        }
7369    }
7370
7371    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7372        if (hasControl) {
7373            *left = mNewLeftVolume;
7374            *right = mNewRightVolume;
7375        }
7376        return hasControl;
7377    }
7378
7379    mVolumeCtrlIdx = ctrlIdx;
7380    mLeftVolume = newLeft;
7381    mRightVolume = newRight;
7382
7383    // second get volume update from volume controller
7384    if (ctrlIdx >= 0) {
7385        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7386        mNewLeftVolume = newLeft;
7387        mNewRightVolume = newRight;
7388    }
7389    // then indicate volume to all other effects in chain.
7390    // Pass altered volume to effects before volume controller
7391    // and requested volume to effects after controller
7392    uint32_t lVol = newLeft;
7393    uint32_t rVol = newRight;
7394
7395    for (size_t i = 0; i < size; i++) {
7396        if ((int)i == ctrlIdx) continue;
7397        // this also works for ctrlIdx == -1 when there is no volume controller
7398        if ((int)i > ctrlIdx) {
7399            lVol = *left;
7400            rVol = *right;
7401        }
7402        mEffects[i]->setVolume(&lVol, &rVol, false);
7403    }
7404    *left = newLeft;
7405    *right = newRight;
7406
7407    return hasControl;
7408}
7409
7410status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7411{
7412    const size_t SIZE = 256;
7413    char buffer[SIZE];
7414    String8 result;
7415
7416    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7417    result.append(buffer);
7418
7419    bool locked = tryLock(mLock);
7420    // failed to lock - AudioFlinger is probably deadlocked
7421    if (!locked) {
7422        result.append("\tCould not lock mutex:\n");
7423    }
7424
7425    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7426    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7427            mEffects.size(),
7428            (uint32_t)mInBuffer,
7429            (uint32_t)mOutBuffer,
7430            mActiveTrackCnt);
7431    result.append(buffer);
7432    write(fd, result.string(), result.size());
7433
7434    for (size_t i = 0; i < mEffects.size(); ++i) {
7435        sp<EffectModule> effect = mEffects[i];
7436        if (effect != 0) {
7437            effect->dump(fd, args);
7438        }
7439    }
7440
7441    if (locked) {
7442        mLock.unlock();
7443    }
7444
7445    return NO_ERROR;
7446}
7447
7448// must be called with ThreadBase::mLock held
7449void AudioFlinger::EffectChain::setEffectSuspended_l(
7450        const effect_uuid_t *type, bool suspend)
7451{
7452    sp<SuspendedEffectDesc> desc;
7453    // use effect type UUID timelow as key as there is no real risk of identical
7454    // timeLow fields among effect type UUIDs.
7455    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7456    if (suspend) {
7457        if (index >= 0) {
7458            desc = mSuspendedEffects.valueAt(index);
7459        } else {
7460            desc = new SuspendedEffectDesc();
7461            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7462            mSuspendedEffects.add(type->timeLow, desc);
7463            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7464        }
7465        if (desc->mRefCount++ == 0) {
7466            sp<EffectModule> effect = getEffectIfEnabled(type);
7467            if (effect != 0) {
7468                desc->mEffect = effect;
7469                effect->setSuspended(true);
7470                effect->setEnabled(false);
7471            }
7472        }
7473    } else {
7474        if (index < 0) {
7475            return;
7476        }
7477        desc = mSuspendedEffects.valueAt(index);
7478        if (desc->mRefCount <= 0) {
7479            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7480            desc->mRefCount = 1;
7481        }
7482        if (--desc->mRefCount == 0) {
7483            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7484            if (desc->mEffect != 0) {
7485                sp<EffectModule> effect = desc->mEffect.promote();
7486                if (effect != 0) {
7487                    effect->setSuspended(false);
7488                    sp<EffectHandle> handle = effect->controlHandle();
7489                    if (handle != 0) {
7490                        effect->setEnabled(handle->enabled());
7491                    }
7492                }
7493                desc->mEffect.clear();
7494            }
7495            mSuspendedEffects.removeItemsAt(index);
7496        }
7497    }
7498}
7499
7500// must be called with ThreadBase::mLock held
7501void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7502{
7503    sp<SuspendedEffectDesc> desc;
7504
7505    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7506    if (suspend) {
7507        if (index >= 0) {
7508            desc = mSuspendedEffects.valueAt(index);
7509        } else {
7510            desc = new SuspendedEffectDesc();
7511            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7512            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7513        }
7514        if (desc->mRefCount++ == 0) {
7515            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7516            for (size_t i = 0; i < effects.size(); i++) {
7517                setEffectSuspended_l(&effects[i]->desc().type, true);
7518            }
7519        }
7520    } else {
7521        if (index < 0) {
7522            return;
7523        }
7524        desc = mSuspendedEffects.valueAt(index);
7525        if (desc->mRefCount <= 0) {
7526            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7527            desc->mRefCount = 1;
7528        }
7529        if (--desc->mRefCount == 0) {
7530            Vector<const effect_uuid_t *> types;
7531            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7532                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7533                    continue;
7534                }
7535                types.add(&mSuspendedEffects.valueAt(i)->mType);
7536            }
7537            for (size_t i = 0; i < types.size(); i++) {
7538                setEffectSuspended_l(types[i], false);
7539            }
7540            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7541            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7542        }
7543    }
7544}
7545
7546
7547// The volume effect is used for automated tests only
7548#ifndef OPENSL_ES_H_
7549static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7550                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7551const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7552#endif //OPENSL_ES_H_
7553
7554bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7555{
7556    // auxiliary effects and visualizer are never suspended on output mix
7557    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7558        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7559         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7560         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7561        return false;
7562    }
7563    return true;
7564}
7565
7566Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7567{
7568    Vector< sp<EffectModule> > effects;
7569    for (size_t i = 0; i < mEffects.size(); i++) {
7570        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7571            continue;
7572        }
7573        effects.add(mEffects[i]);
7574    }
7575    return effects;
7576}
7577
7578sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7579                                                            const effect_uuid_t *type)
7580{
7581    sp<EffectModule> effect;
7582    effect = getEffectFromType_l(type);
7583    if (effect != 0 && !effect->isEnabled()) {
7584        effect.clear();
7585    }
7586    return effect;
7587}
7588
7589void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7590                                                            bool enabled)
7591{
7592    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7593    if (enabled) {
7594        if (index < 0) {
7595            // if the effect is not suspend check if all effects are suspended
7596            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7597            if (index < 0) {
7598                return;
7599            }
7600            if (!isEffectEligibleForSuspend(effect->desc())) {
7601                return;
7602            }
7603            setEffectSuspended_l(&effect->desc().type, enabled);
7604            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7605            if (index < 0) {
7606                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7607                return;
7608            }
7609        }
7610        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7611             effect->desc().type.timeLow);
7612        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7613        // if effect is requested to suspended but was not yet enabled, supend it now.
7614        if (desc->mEffect == 0) {
7615            desc->mEffect = effect;
7616            effect->setEnabled(false);
7617            effect->setSuspended(true);
7618        }
7619    } else {
7620        if (index < 0) {
7621            return;
7622        }
7623        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7624             effect->desc().type.timeLow);
7625        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7626        desc->mEffect.clear();
7627        effect->setSuspended(false);
7628    }
7629}
7630
7631#undef LOG_TAG
7632#define LOG_TAG "AudioFlinger"
7633
7634// ----------------------------------------------------------------------------
7635
7636status_t AudioFlinger::onTransact(
7637        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7638{
7639    return BnAudioFlinger::onTransact(code, data, reply, flags);
7640}
7641
7642}; // namespace android
7643