AudioFlinger.cpp revision 926798f8c21ab002d9797ef8973852a2612c1f75 
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include <math.h>
23#include <signal.h>
24#include <sys/time.h>
25#include <sys/resource.h>
26
27#include <binder/IPCThreadState.h>
28#include <binder/IServiceManager.h>
29#include <utils/Log.h>
30#include <binder/Parcel.h>
31#include <binder/IPCThreadState.h>
32#include <utils/String16.h>
33#include <utils/threads.h>
34#include <utils/Atomic.h>
35
36#include <cutils/bitops.h>
37#include <cutils/properties.h>
38
39#include <media/AudioTrack.h>
40#include <media/AudioRecord.h>
41#include <media/IMediaPlayerService.h>
42
43#include <private/media/AudioTrackShared.h>
44#include <private/media/AudioEffectShared.h>
45
46#include <system/audio.h>
47#include <hardware/audio.h>
48
49#include "AudioMixer.h"
50#include "AudioFlinger.h"
51
52#include <media/EffectsFactoryApi.h>
53#include <audio_effects/effect_visualizer.h>
54#include <audio_effects/effect_ns.h>
55#include <audio_effects/effect_aec.h>
56
57#include <cpustats/ThreadCpuUsage.h>
58#include <powermanager/PowerManager.h>
59// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
60
61// ----------------------------------------------------------------------------
62
63
64namespace android {
65
66static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
67static const char* kHardwareLockedString = "Hardware lock is taken\n";
68
69//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
70static const float MAX_GAIN = 4096.0f;
71static const float MAX_GAIN_INT = 0x1000;
72
73// retry counts for buffer fill timeout
74// 50 * ~20msecs = 1 second
75static const int8_t kMaxTrackRetries = 50;
76static const int8_t kMaxTrackStartupRetries = 50;
77// allow less retry attempts on direct output thread.
78// direct outputs can be a scarce resource in audio hardware and should
79// be released as quickly as possible.
80static const int8_t kMaxTrackRetriesDirect = 2;
81
82static const int kDumpLockRetries = 50;
83static const int kDumpLockSleep = 20000;
84
85static const nsecs_t kWarningThrottle = seconds(5);
86
87// RecordThread loop sleep time upon application overrun or audio HAL read error
88static const int kRecordThreadSleepUs = 5000;
89
90static const nsecs_t kSetParametersTimeout = seconds(2);
91
92// minimum sleep time for the mixer thread loop when tracks are active but in underrun
93static const uint32_t kMinThreadSleepTimeUs = 5000;
94// maximum divider applied to the active sleep time in the mixer thread loop
95static const uint32_t kMaxThreadSleepTimeShift = 2;
96
97
98// ----------------------------------------------------------------------------
99
100static bool recordingAllowed() {
101    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
102    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
103    if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
104    return ok;
105}
106
107static bool settingsAllowed() {
108    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
109    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
110    if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
111    return ok;
112}
113
114// To collect the amplifier usage
115static void addBatteryData(uint32_t params) {
116    sp<IBinder> binder =
117        defaultServiceManager()->getService(String16("media.player"));
118    sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder);
119    if (service.get() == NULL) {
120        LOGW("Cannot connect to the MediaPlayerService for battery tracking");
121        return;
122    }
123
124    service->addBatteryData(params);
125}
126
127static int load_audio_interface(const char *if_name, const hw_module_t **mod,
128                                audio_hw_device_t **dev)
129{
130    int rc;
131
132    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
133    if (rc)
134        goto out;
135
136    rc = audio_hw_device_open(*mod, dev);
137    LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
138            AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
139    if (rc)
140        goto out;
141
142    return 0;
143
144out:
145    *mod = NULL;
146    *dev = NULL;
147    return rc;
148}
149
150static const char *audio_interfaces[] = {
151    "primary",
152    "a2dp",
153    "usb",
154};
155#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
156
157// ----------------------------------------------------------------------------
158
159AudioFlinger::AudioFlinger()
160    : BnAudioFlinger(),
161        mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
162        mBtNrecIsOff(false)
163{
164}
165
166void AudioFlinger::onFirstRef()
167{
168    int rc = 0;
169
170    Mutex::Autolock _l(mLock);
171
172    /* TODO: move all this work into an Init() function */
173    mHardwareStatus = AUDIO_HW_IDLE;
174
175    for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
176        const hw_module_t *mod;
177        audio_hw_device_t *dev;
178
179        rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
180        if (rc)
181            continue;
182
183        LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
184             mod->name, mod->id);
185        mAudioHwDevs.push(dev);
186
187        if (!mPrimaryHardwareDev) {
188            mPrimaryHardwareDev = dev;
189            LOGI("Using '%s' (%s.%s) as the primary audio interface",
190                 mod->name, mod->id, audio_interfaces[i]);
191        }
192    }
193
194    mHardwareStatus = AUDIO_HW_INIT;
195
196    if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
197        LOGE("Primary audio interface not found");
198        return;
199    }
200
201    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
202        audio_hw_device_t *dev = mAudioHwDevs[i];
203
204        mHardwareStatus = AUDIO_HW_INIT;
205        rc = dev->init_check(dev);
206        if (rc == 0) {
207            AutoMutex lock(mHardwareLock);
208
209            mMode = AUDIO_MODE_NORMAL;
210            mHardwareStatus = AUDIO_HW_SET_MODE;
211            dev->set_mode(dev, mMode);
212            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
213            dev->set_master_volume(dev, 1.0f);
214            mHardwareStatus = AUDIO_HW_IDLE;
215        }
216    }
217}
218
219status_t AudioFlinger::initCheck() const
220{
221    Mutex::Autolock _l(mLock);
222    if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
223        return NO_INIT;
224    return NO_ERROR;
225}
226
227AudioFlinger::~AudioFlinger()
228{
229    int num_devs = mAudioHwDevs.size();
230
231    while (!mRecordThreads.isEmpty()) {
232        // closeInput() will remove first entry from mRecordThreads
233        closeInput(mRecordThreads.keyAt(0));
234    }
235    while (!mPlaybackThreads.isEmpty()) {
236        // closeOutput() will remove first entry from mPlaybackThreads
237        closeOutput(mPlaybackThreads.keyAt(0));
238    }
239
240    for (int i = 0; i < num_devs; i++) {
241        audio_hw_device_t *dev = mAudioHwDevs[i];
242        audio_hw_device_close(dev);
243    }
244    mAudioHwDevs.clear();
245}
246
247audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
248{
249    /* first matching HW device is returned */
250    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
251        audio_hw_device_t *dev = mAudioHwDevs[i];
252        if ((dev->get_supported_devices(dev) & devices) == devices)
253            return dev;
254    }
255    return NULL;
256}
257
258status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
259{
260    const size_t SIZE = 256;
261    char buffer[SIZE];
262    String8 result;
263
264    result.append("Clients:\n");
265    for (size_t i = 0; i < mClients.size(); ++i) {
266        wp<Client> wClient = mClients.valueAt(i);
267        if (wClient != 0) {
268            sp<Client> client = wClient.promote();
269            if (client != 0) {
270                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
271                result.append(buffer);
272            }
273        }
274    }
275
276    result.append("Global session refs:\n");
277    result.append(" session pid cnt\n");
278    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
279        AudioSessionRef *r = mAudioSessionRefs[i];
280        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
281        result.append(buffer);
282    }
283    write(fd, result.string(), result.size());
284    return NO_ERROR;
285}
286
287
288status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
289{
290    const size_t SIZE = 256;
291    char buffer[SIZE];
292    String8 result;
293    int hardwareStatus = mHardwareStatus;
294
295    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
296    result.append(buffer);
297    write(fd, result.string(), result.size());
298    return NO_ERROR;
299}
300
301status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
302{
303    const size_t SIZE = 256;
304    char buffer[SIZE];
305    String8 result;
306    snprintf(buffer, SIZE, "Permission Denial: "
307            "can't dump AudioFlinger from pid=%d, uid=%d\n",
308            IPCThreadState::self()->getCallingPid(),
309            IPCThreadState::self()->getCallingUid());
310    result.append(buffer);
311    write(fd, result.string(), result.size());
312    return NO_ERROR;
313}
314
315static bool tryLock(Mutex& mutex)
316{
317    bool locked = false;
318    for (int i = 0; i < kDumpLockRetries; ++i) {
319        if (mutex.tryLock() == NO_ERROR) {
320            locked = true;
321            break;
322        }
323        usleep(kDumpLockSleep);
324    }
325    return locked;
326}
327
328status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
329{
330    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
331        dumpPermissionDenial(fd, args);
332    } else {
333        // get state of hardware lock
334        bool hardwareLocked = tryLock(mHardwareLock);
335        if (!hardwareLocked) {
336            String8 result(kHardwareLockedString);
337            write(fd, result.string(), result.size());
338        } else {
339            mHardwareLock.unlock();
340        }
341
342        bool locked = tryLock(mLock);
343
344        // failed to lock - AudioFlinger is probably deadlocked
345        if (!locked) {
346            String8 result(kDeadlockedString);
347            write(fd, result.string(), result.size());
348        }
349
350        dumpClients(fd, args);
351        dumpInternals(fd, args);
352
353        // dump playback threads
354        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
355            mPlaybackThreads.valueAt(i)->dump(fd, args);
356        }
357
358        // dump record threads
359        for (size_t i = 0; i < mRecordThreads.size(); i++) {
360            mRecordThreads.valueAt(i)->dump(fd, args);
361        }
362
363        // dump all hardware devs
364        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
365            audio_hw_device_t *dev = mAudioHwDevs[i];
366            dev->dump(dev, fd);
367        }
368        if (locked) mLock.unlock();
369    }
370    return NO_ERROR;
371}
372
373
374// IAudioFlinger interface
375
376
377sp<IAudioTrack> AudioFlinger::createTrack(
378        pid_t pid,
379        int streamType,
380        uint32_t sampleRate,
381        uint32_t format,
382        uint32_t channelMask,
383        int frameCount,
384        uint32_t flags,
385        const sp<IMemory>& sharedBuffer,
386        int output,
387        int *sessionId,
388        status_t *status)
389{
390    sp<PlaybackThread::Track> track;
391    sp<TrackHandle> trackHandle;
392    sp<Client> client;
393    wp<Client> wclient;
394    status_t lStatus;
395    int lSessionId;
396
397    if (streamType >= AUDIO_STREAM_CNT) {
398        LOGE("invalid stream type");
399        lStatus = BAD_VALUE;
400        goto Exit;
401    }
402
403    {
404        Mutex::Autolock _l(mLock);
405        PlaybackThread *thread = checkPlaybackThread_l(output);
406        PlaybackThread *effectThread = NULL;
407        if (thread == NULL) {
408            LOGE("unknown output thread");
409            lStatus = BAD_VALUE;
410            goto Exit;
411        }
412
413        wclient = mClients.valueFor(pid);
414
415        if (wclient != NULL) {
416            client = wclient.promote();
417        } else {
418            client = new Client(this, pid);
419            mClients.add(pid, client);
420        }
421
422        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
423        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
424            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
425                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
426                if (mPlaybackThreads.keyAt(i) != output) {
427                    // prevent same audio session on different output threads
428                    uint32_t sessions = t->hasAudioSession(*sessionId);
429                    if (sessions & PlaybackThread::TRACK_SESSION) {
430                        lStatus = BAD_VALUE;
431                        goto Exit;
432                    }
433                    // check if an effect with same session ID is waiting for a track to be created
434                    if (sessions & PlaybackThread::EFFECT_SESSION) {
435                        effectThread = t.get();
436                    }
437                }
438            }
439            lSessionId = *sessionId;
440        } else {
441            // if no audio session id is provided, create one here
442            lSessionId = nextUniqueId();
443            if (sessionId != NULL) {
444                *sessionId = lSessionId;
445            }
446        }
447        ALOGV("createTrack() lSessionId: %d", lSessionId);
448
449        track = thread->createTrack_l(client, streamType, sampleRate, format,
450                channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
451
452        // move effect chain to this output thread if an effect on same session was waiting
453        // for a track to be created
454        if (lStatus == NO_ERROR && effectThread != NULL) {
455            Mutex::Autolock _dl(thread->mLock);
456            Mutex::Autolock _sl(effectThread->mLock);
457            moveEffectChain_l(lSessionId, effectThread, thread, true);
458        }
459    }
460    if (lStatus == NO_ERROR) {
461        trackHandle = new TrackHandle(track);
462    } else {
463        // remove local strong reference to Client before deleting the Track so that the Client
464        // destructor is called by the TrackBase destructor with mLock held
465        client.clear();
466        track.clear();
467    }
468
469Exit:
470    if(status) {
471        *status = lStatus;
472    }
473    return trackHandle;
474}
475
476uint32_t AudioFlinger::sampleRate(int output) const
477{
478    Mutex::Autolock _l(mLock);
479    PlaybackThread *thread = checkPlaybackThread_l(output);
480    if (thread == NULL) {
481        LOGW("sampleRate() unknown thread %d", output);
482        return 0;
483    }
484    return thread->sampleRate();
485}
486
487int AudioFlinger::channelCount(int output) const
488{
489    Mutex::Autolock _l(mLock);
490    PlaybackThread *thread = checkPlaybackThread_l(output);
491    if (thread == NULL) {
492        LOGW("channelCount() unknown thread %d", output);
493        return 0;
494    }
495    return thread->channelCount();
496}
497
498uint32_t AudioFlinger::format(int output) const
499{
500    Mutex::Autolock _l(mLock);
501    PlaybackThread *thread = checkPlaybackThread_l(output);
502    if (thread == NULL) {
503        LOGW("format() unknown thread %d", output);
504        return 0;
505    }
506    return thread->format();
507}
508
509size_t AudioFlinger::frameCount(int output) const
510{
511    Mutex::Autolock _l(mLock);
512    PlaybackThread *thread = checkPlaybackThread_l(output);
513    if (thread == NULL) {
514        LOGW("frameCount() unknown thread %d", output);
515        return 0;
516    }
517    return thread->frameCount();
518}
519
520uint32_t AudioFlinger::latency(int output) const
521{
522    Mutex::Autolock _l(mLock);
523    PlaybackThread *thread = checkPlaybackThread_l(output);
524    if (thread == NULL) {
525        LOGW("latency() unknown thread %d", output);
526        return 0;
527    }
528    return thread->latency();
529}
530
531status_t AudioFlinger::setMasterVolume(float value)
532{
533    status_t ret = initCheck();
534    if (ret != NO_ERROR) {
535        return ret;
536    }
537
538    // check calling permissions
539    if (!settingsAllowed()) {
540        return PERMISSION_DENIED;
541    }
542
543    // when hw supports master volume, don't scale in sw mixer
544    { // scope for the lock
545        AutoMutex lock(mHardwareLock);
546        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
547        if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
548            value = 1.0f;
549        }
550        mHardwareStatus = AUDIO_HW_IDLE;
551    }
552
553    Mutex::Autolock _l(mLock);
554    mMasterVolume = value;
555    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
556       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
557
558    return NO_ERROR;
559}
560
561status_t AudioFlinger::setMode(int mode)
562{
563    status_t ret = initCheck();
564    if (ret != NO_ERROR) {
565        return ret;
566    }
567
568    // check calling permissions
569    if (!settingsAllowed()) {
570        return PERMISSION_DENIED;
571    }
572    if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) {
573        LOGW("Illegal value: setMode(%d)", mode);
574        return BAD_VALUE;
575    }
576
577    { // scope for the lock
578        AutoMutex lock(mHardwareLock);
579        mHardwareStatus = AUDIO_HW_SET_MODE;
580        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
581        mHardwareStatus = AUDIO_HW_IDLE;
582    }
583
584    if (NO_ERROR == ret) {
585        Mutex::Autolock _l(mLock);
586        mMode = mode;
587        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
588           mPlaybackThreads.valueAt(i)->setMode(mode);
589    }
590
591    return ret;
592}
593
594status_t AudioFlinger::setMicMute(bool state)
595{
596    status_t ret = initCheck();
597    if (ret != NO_ERROR) {
598        return ret;
599    }
600
601    // check calling permissions
602    if (!settingsAllowed()) {
603        return PERMISSION_DENIED;
604    }
605
606    AutoMutex lock(mHardwareLock);
607    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
608    ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
609    mHardwareStatus = AUDIO_HW_IDLE;
610    return ret;
611}
612
613bool AudioFlinger::getMicMute() const
614{
615    status_t ret = initCheck();
616    if (ret != NO_ERROR) {
617        return false;
618    }
619
620    bool state = AUDIO_MODE_INVALID;
621    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
622    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
623    mHardwareStatus = AUDIO_HW_IDLE;
624    return state;
625}
626
627status_t AudioFlinger::setMasterMute(bool muted)
628{
629    // check calling permissions
630    if (!settingsAllowed()) {
631        return PERMISSION_DENIED;
632    }
633
634    Mutex::Autolock _l(mLock);
635    mMasterMute = muted;
636    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
637       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
638
639    return NO_ERROR;
640}
641
642float AudioFlinger::masterVolume() const
643{
644    return mMasterVolume;
645}
646
647bool AudioFlinger::masterMute() const
648{
649    return mMasterMute;
650}
651
652status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
653{
654    // check calling permissions
655    if (!settingsAllowed()) {
656        return PERMISSION_DENIED;
657    }
658
659    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
660        return BAD_VALUE;
661    }
662
663    AutoMutex lock(mLock);
664    PlaybackThread *thread = NULL;
665    if (output) {
666        thread = checkPlaybackThread_l(output);
667        if (thread == NULL) {
668            return BAD_VALUE;
669        }
670    }
671
672    mStreamTypes[stream].volume = value;
673
674    if (thread == NULL) {
675        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
676           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
677        }
678    } else {
679        thread->setStreamVolume(stream, value);
680    }
681
682    return NO_ERROR;
683}
684
685status_t AudioFlinger::setStreamMute(int stream, bool muted)
686{
687    // check calling permissions
688    if (!settingsAllowed()) {
689        return PERMISSION_DENIED;
690    }
691
692    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
693        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
694        return BAD_VALUE;
695    }
696
697    AutoMutex lock(mLock);
698    mStreamTypes[stream].mute = muted;
699    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
700       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
701
702    return NO_ERROR;
703}
704
705float AudioFlinger::streamVolume(int stream, int output) const
706{
707    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
708        return 0.0f;
709    }
710
711    AutoMutex lock(mLock);
712    float volume;
713    if (output) {
714        PlaybackThread *thread = checkPlaybackThread_l(output);
715        if (thread == NULL) {
716            return 0.0f;
717        }
718        volume = thread->streamVolume(stream);
719    } else {
720        volume = mStreamTypes[stream].volume;
721    }
722
723    return volume;
724}
725
726bool AudioFlinger::streamMute(int stream) const
727{
728    if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
729        return true;
730    }
731
732    return mStreamTypes[stream].mute;
733}
734
735status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
736{
737    status_t result;
738
739    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
740            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
741    // check calling permissions
742    if (!settingsAllowed()) {
743        return PERMISSION_DENIED;
744    }
745
746    // ioHandle == 0 means the parameters are global to the audio hardware interface
747    if (ioHandle == 0) {
748        AutoMutex lock(mHardwareLock);
749        mHardwareStatus = AUDIO_SET_PARAMETER;
750        status_t final_result = NO_ERROR;
751        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
752            audio_hw_device_t *dev = mAudioHwDevs[i];
753            result = dev->set_parameters(dev, keyValuePairs.string());
754            final_result = result ?: final_result;
755        }
756        mHardwareStatus = AUDIO_HW_IDLE;
757        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
758        AudioParameter param = AudioParameter(keyValuePairs);
759        String8 value;
760        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
761            Mutex::Autolock _l(mLock);
762            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
763            if (mBtNrecIsOff != btNrecIsOff) {
764                for (size_t i = 0; i < mRecordThreads.size(); i++) {
765                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
766                    RecordThread::RecordTrack *track = thread->track();
767                    if (track != NULL) {
768                        audio_devices_t device = (audio_devices_t)(
769                                thread->device() & AUDIO_DEVICE_IN_ALL);
770                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
771                        thread->setEffectSuspended(FX_IID_AEC,
772                                                   suspend,
773                                                   track->sessionId());
774                        thread->setEffectSuspended(FX_IID_NS,
775                                                   suspend,
776                                                   track->sessionId());
777                    }
778                }
779                mBtNrecIsOff = btNrecIsOff;
780            }
781        }
782        return final_result;
783    }
784
785    // hold a strong ref on thread in case closeOutput() or closeInput() is called
786    // and the thread is exited once the lock is released
787    sp<ThreadBase> thread;
788    {
789        Mutex::Autolock _l(mLock);
790        thread = checkPlaybackThread_l(ioHandle);
791        if (thread == NULL) {
792            thread = checkRecordThread_l(ioHandle);
793        } else if (thread.get() == primaryPlaybackThread_l()) {
794            // indicate output device change to all input threads for pre processing
795            AudioParameter param = AudioParameter(keyValuePairs);
796            int value;
797            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
798                for (size_t i = 0; i < mRecordThreads.size(); i++) {
799                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
800                }
801            }
802        }
803    }
804    if (thread != NULL) {
805        result = thread->setParameters(keyValuePairs);
806        return result;
807    }
808    return BAD_VALUE;
809}
810
811String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
812{
813//    ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
814//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
815
816    if (ioHandle == 0) {
817        String8 out_s8;
818
819        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
820            audio_hw_device_t *dev = mAudioHwDevs[i];
821            char *s = dev->get_parameters(dev, keys.string());
822            out_s8 += String8(s);
823            free(s);
824        }
825        return out_s8;
826    }
827
828    Mutex::Autolock _l(mLock);
829
830    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
831    if (playbackThread != NULL) {
832        return playbackThread->getParameters(keys);
833    }
834    RecordThread *recordThread = checkRecordThread_l(ioHandle);
835    if (recordThread != NULL) {
836        return recordThread->getParameters(keys);
837    }
838    return String8("");
839}
840
841size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
842{
843    status_t ret = initCheck();
844    if (ret != NO_ERROR) {
845        return 0;
846    }
847
848    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
849}
850
851unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
852{
853    if (ioHandle == 0) {
854        return 0;
855    }
856
857    Mutex::Autolock _l(mLock);
858
859    RecordThread *recordThread = checkRecordThread_l(ioHandle);
860    if (recordThread != NULL) {
861        return recordThread->getInputFramesLost();
862    }
863    return 0;
864}
865
866status_t AudioFlinger::setVoiceVolume(float value)
867{
868    status_t ret = initCheck();
869    if (ret != NO_ERROR) {
870        return ret;
871    }
872
873    // check calling permissions
874    if (!settingsAllowed()) {
875        return PERMISSION_DENIED;
876    }
877
878    AutoMutex lock(mHardwareLock);
879    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
880    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
881    mHardwareStatus = AUDIO_HW_IDLE;
882
883    return ret;
884}
885
886status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
887{
888    status_t status;
889
890    Mutex::Autolock _l(mLock);
891
892    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
893    if (playbackThread != NULL) {
894        return playbackThread->getRenderPosition(halFrames, dspFrames);
895    }
896
897    return BAD_VALUE;
898}
899
900void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
901{
902
903    Mutex::Autolock _l(mLock);
904
905    int pid = IPCThreadState::self()->getCallingPid();
906    if (mNotificationClients.indexOfKey(pid) < 0) {
907        sp<NotificationClient> notificationClient = new NotificationClient(this,
908                                                                            client,
909                                                                            pid);
910        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
911
912        mNotificationClients.add(pid, notificationClient);
913
914        sp<IBinder> binder = client->asBinder();
915        binder->linkToDeath(notificationClient);
916
917        // the config change is always sent from playback or record threads to avoid deadlock
918        // with AudioSystem::gLock
919        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
920            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
921        }
922
923        for (size_t i = 0; i < mRecordThreads.size(); i++) {
924            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
925        }
926    }
927}
928
929void AudioFlinger::removeNotificationClient(pid_t pid)
930{
931    Mutex::Autolock _l(mLock);
932
933    int index = mNotificationClients.indexOfKey(pid);
934    if (index >= 0) {
935        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
936        ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
937        mNotificationClients.removeItem(pid);
938    }
939
940    ALOGV("%d died, releasing its sessions", pid);
941    int num = mAudioSessionRefs.size();
942    bool removed = false;
943    for (int i = 0; i< num; i++) {
944        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
945        ALOGV(" pid %d @ %d", ref->pid, i);
946        if (ref->pid == pid) {
947            ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
948            mAudioSessionRefs.removeAt(i);
949            delete ref;
950            removed = true;
951            i--;
952            num--;
953        }
954    }
955    if (removed) {
956        purgeStaleEffects_l();
957    }
958}
959
960// audioConfigChanged_l() must be called with AudioFlinger::mLock held
961void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
962{
963    size_t size = mNotificationClients.size();
964    for (size_t i = 0; i < size; i++) {
965        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
966    }
967}
968
969// removeClient_l() must be called with AudioFlinger::mLock held
970void AudioFlinger::removeClient_l(pid_t pid)
971{
972    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
973    mClients.removeItem(pid);
974}
975
976
977// ----------------------------------------------------------------------------
978
979AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
980    :   Thread(false),
981        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
982        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
983        mDevice(device)
984{
985    mDeathRecipient = new PMDeathRecipient(this);
986}
987
988AudioFlinger::ThreadBase::~ThreadBase()
989{
990    mParamCond.broadcast();
991    mNewParameters.clear();
992    // do not lock the mutex in destructor
993    releaseWakeLock_l();
994    if (mPowerManager != 0) {
995        sp<IBinder> binder = mPowerManager->asBinder();
996        binder->unlinkToDeath(mDeathRecipient);
997    }
998}
999
1000void AudioFlinger::ThreadBase::exit()
1001{
1002    // keep a strong ref on ourself so that we wont get
1003    // destroyed in the middle of requestExitAndWait()
1004    sp <ThreadBase> strongMe = this;
1005
1006    ALOGV("ThreadBase::exit");
1007    {
1008        AutoMutex lock(&mLock);
1009        mExiting = true;
1010        requestExit();
1011        mWaitWorkCV.signal();
1012    }
1013    requestExitAndWait();
1014}
1015
1016uint32_t AudioFlinger::ThreadBase::sampleRate() const
1017{
1018    return mSampleRate;
1019}
1020
1021int AudioFlinger::ThreadBase::channelCount() const
1022{
1023    return (int)mChannelCount;
1024}
1025
1026uint32_t AudioFlinger::ThreadBase::format() const
1027{
1028    return mFormat;
1029}
1030
1031size_t AudioFlinger::ThreadBase::frameCount() const
1032{
1033    return mFrameCount;
1034}
1035
1036status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1037{
1038    status_t status;
1039
1040    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1041    Mutex::Autolock _l(mLock);
1042
1043    mNewParameters.add(keyValuePairs);
1044    mWaitWorkCV.signal();
1045    // wait condition with timeout in case the thread loop has exited
1046    // before the request could be processed
1047    if (mParamCond.waitRelative(mLock, kSetParametersTimeout) == NO_ERROR) {
1048        status = mParamStatus;
1049        mWaitWorkCV.signal();
1050    } else {
1051        status = TIMED_OUT;
1052    }
1053    return status;
1054}
1055
1056void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1057{
1058    Mutex::Autolock _l(mLock);
1059    sendConfigEvent_l(event, param);
1060}
1061
1062// sendConfigEvent_l() must be called with ThreadBase::mLock held
1063void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1064{
1065    ConfigEvent *configEvent = new ConfigEvent();
1066    configEvent->mEvent = event;
1067    configEvent->mParam = param;
1068    mConfigEvents.add(configEvent);
1069    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1070    mWaitWorkCV.signal();
1071}
1072
1073void AudioFlinger::ThreadBase::processConfigEvents()
1074{
1075    mLock.lock();
1076    while(!mConfigEvents.isEmpty()) {
1077        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1078        ConfigEvent *configEvent = mConfigEvents[0];
1079        mConfigEvents.removeAt(0);
1080        // release mLock before locking AudioFlinger mLock: lock order is always
1081        // AudioFlinger then ThreadBase to avoid cross deadlock
1082        mLock.unlock();
1083        mAudioFlinger->mLock.lock();
1084        audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
1085        mAudioFlinger->mLock.unlock();
1086        delete configEvent;
1087        mLock.lock();
1088    }
1089    mLock.unlock();
1090}
1091
1092status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1093{
1094    const size_t SIZE = 256;
1095    char buffer[SIZE];
1096    String8 result;
1097
1098    bool locked = tryLock(mLock);
1099    if (!locked) {
1100        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1101        write(fd, buffer, strlen(buffer));
1102    }
1103
1104    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1105    result.append(buffer);
1106    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1107    result.append(buffer);
1108    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1109    result.append(buffer);
1110    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1111    result.append(buffer);
1112    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1113    result.append(buffer);
1114    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1115    result.append(buffer);
1116    snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
1117    result.append(buffer);
1118
1119    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1120    result.append(buffer);
1121    result.append(" Index Command");
1122    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1123        snprintf(buffer, SIZE, "\n %02d    ", i);
1124        result.append(buffer);
1125        result.append(mNewParameters[i]);
1126    }
1127
1128    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1129    result.append(buffer);
1130    snprintf(buffer, SIZE, " Index event param\n");
1131    result.append(buffer);
1132    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1133        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
1134        result.append(buffer);
1135    }
1136    result.append("\n");
1137
1138    write(fd, result.string(), result.size());
1139
1140    if (locked) {
1141        mLock.unlock();
1142    }
1143    return NO_ERROR;
1144}
1145
1146status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1147{
1148    const size_t SIZE = 256;
1149    char buffer[SIZE];
1150    String8 result;
1151
1152    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1153    write(fd, buffer, strlen(buffer));
1154
1155    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1156        sp<EffectChain> chain = mEffectChains[i];
1157        if (chain != 0) {
1158            chain->dump(fd, args);
1159        }
1160    }
1161    return NO_ERROR;
1162}
1163
1164void AudioFlinger::ThreadBase::acquireWakeLock()
1165{
1166    Mutex::Autolock _l(mLock);
1167    acquireWakeLock_l();
1168}
1169
1170void AudioFlinger::ThreadBase::acquireWakeLock_l()
1171{
1172    if (mPowerManager == 0) {
1173        // use checkService() to avoid blocking if power service is not up yet
1174        sp<IBinder> binder =
1175            defaultServiceManager()->checkService(String16("power"));
1176        if (binder == 0) {
1177            LOGW("Thread %s cannot connect to the power manager service", mName);
1178        } else {
1179            mPowerManager = interface_cast<IPowerManager>(binder);
1180            binder->linkToDeath(mDeathRecipient);
1181        }
1182    }
1183    if (mPowerManager != 0) {
1184        sp<IBinder> binder = new BBinder();
1185        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
1186                                                         binder,
1187                                                         String16(mName));
1188        if (status == NO_ERROR) {
1189            mWakeLockToken = binder;
1190        }
1191        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
1192    }
1193}
1194
1195void AudioFlinger::ThreadBase::releaseWakeLock()
1196{
1197    Mutex::Autolock _l(mLock);
1198    releaseWakeLock_l();
1199}
1200
1201void AudioFlinger::ThreadBase::releaseWakeLock_l()
1202{
1203    if (mWakeLockToken != 0) {
1204        ALOGV("releaseWakeLock_l() %s", mName);
1205        if (mPowerManager != 0) {
1206            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
1207        }
1208        mWakeLockToken.clear();
1209    }
1210}
1211
1212void AudioFlinger::ThreadBase::clearPowerManager()
1213{
1214    Mutex::Autolock _l(mLock);
1215    releaseWakeLock_l();
1216    mPowerManager.clear();
1217}
1218
1219void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
1220{
1221    sp<ThreadBase> thread = mThread.promote();
1222    if (thread != 0) {
1223        thread->clearPowerManager();
1224    }
1225    LOGW("power manager service died !!!");
1226}
1227
1228void AudioFlinger::ThreadBase::setEffectSuspended(
1229        const effect_uuid_t *type, bool suspend, int sessionId)
1230{
1231    Mutex::Autolock _l(mLock);
1232    setEffectSuspended_l(type, suspend, sessionId);
1233}
1234
1235void AudioFlinger::ThreadBase::setEffectSuspended_l(
1236        const effect_uuid_t *type, bool suspend, int sessionId)
1237{
1238    sp<EffectChain> chain;
1239    chain = getEffectChain_l(sessionId);
1240    if (chain != 0) {
1241        if (type != NULL) {
1242            chain->setEffectSuspended_l(type, suspend);
1243        } else {
1244            chain->setEffectSuspendedAll_l(suspend);
1245        }
1246    }
1247
1248    updateSuspendedSessions_l(type, suspend, sessionId);
1249}
1250
1251void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1252{
1253    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1254    if (index < 0) {
1255        return;
1256    }
1257
1258    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1259            mSuspendedSessions.editValueAt(index);
1260
1261    for (size_t i = 0; i < sessionEffects.size(); i++) {
1262        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1263        for (int j = 0; j < desc->mRefCount; j++) {
1264            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1265                chain->setEffectSuspendedAll_l(true);
1266            } else {
1267                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1268                     desc->mType.timeLow);
1269                chain->setEffectSuspended_l(&desc->mType, true);
1270            }
1271        }
1272    }
1273}
1274
1275void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1276                                                         bool suspend,
1277                                                         int sessionId)
1278{
1279    int index = mSuspendedSessions.indexOfKey(sessionId);
1280
1281    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1282
1283    if (suspend) {
1284        if (index >= 0) {
1285            sessionEffects = mSuspendedSessions.editValueAt(index);
1286        } else {
1287            mSuspendedSessions.add(sessionId, sessionEffects);
1288        }
1289    } else {
1290        if (index < 0) {
1291            return;
1292        }
1293        sessionEffects = mSuspendedSessions.editValueAt(index);
1294    }
1295
1296
1297    int key = EffectChain::kKeyForSuspendAll;
1298    if (type != NULL) {
1299        key = type->timeLow;
1300    }
1301    index = sessionEffects.indexOfKey(key);
1302
1303    sp <SuspendedSessionDesc> desc;
1304    if (suspend) {
1305        if (index >= 0) {
1306            desc = sessionEffects.valueAt(index);
1307        } else {
1308            desc = new SuspendedSessionDesc();
1309            if (type != NULL) {
1310                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1311            }
1312            sessionEffects.add(key, desc);
1313            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1314        }
1315        desc->mRefCount++;
1316    } else {
1317        if (index < 0) {
1318            return;
1319        }
1320        desc = sessionEffects.valueAt(index);
1321        if (--desc->mRefCount == 0) {
1322            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1323            sessionEffects.removeItemsAt(index);
1324            if (sessionEffects.isEmpty()) {
1325                ALOGV("updateSuspendedSessions_l() restore removing session %d",
1326                                 sessionId);
1327                mSuspendedSessions.removeItem(sessionId);
1328            }
1329        }
1330    }
1331    if (!sessionEffects.isEmpty()) {
1332        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1333    }
1334}
1335
1336void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1337                                                            bool enabled,
1338                                                            int sessionId)
1339{
1340    Mutex::Autolock _l(mLock);
1341    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1342}
1343
1344void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1345                                                            bool enabled,
1346                                                            int sessionId)
1347{
1348    if (mType != RECORD) {
1349        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1350        // another session. This gives the priority to well behaved effect control panels
1351        // and applications not using global effects.
1352        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1353            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1354        }
1355    }
1356
1357    sp<EffectChain> chain = getEffectChain_l(sessionId);
1358    if (chain != 0) {
1359        chain->checkSuspendOnEffectEnabled(effect, enabled);
1360    }
1361}
1362
1363// ----------------------------------------------------------------------------
1364
1365AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1366                                             AudioStreamOut* output,
1367                                             int id,
1368                                             uint32_t device)
1369    :   ThreadBase(audioFlinger, id, device),
1370        mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
1371        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1372{
1373    snprintf(mName, kNameLength, "AudioOut_%d", id);
1374
1375    readOutputParameters();
1376
1377    mMasterVolume = mAudioFlinger->masterVolume();
1378    mMasterMute = mAudioFlinger->masterMute();
1379
1380    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
1381        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1382        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1383        mStreamTypes[stream].valid = true;
1384    }
1385}
1386
1387AudioFlinger::PlaybackThread::~PlaybackThread()
1388{
1389    delete [] mMixBuffer;
1390}
1391
1392status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1393{
1394    dumpInternals(fd, args);
1395    dumpTracks(fd, args);
1396    dumpEffectChains(fd, args);
1397    return NO_ERROR;
1398}
1399
1400status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1401{
1402    const size_t SIZE = 256;
1403    char buffer[SIZE];
1404    String8 result;
1405
1406    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1407    result.append(buffer);
1408    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1409    for (size_t i = 0; i < mTracks.size(); ++i) {
1410        sp<Track> track = mTracks[i];
1411        if (track != 0) {
1412            track->dump(buffer, SIZE);
1413            result.append(buffer);
1414        }
1415    }
1416
1417    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1418    result.append(buffer);
1419    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1420    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1421        wp<Track> wTrack = mActiveTracks[i];
1422        if (wTrack != 0) {
1423            sp<Track> track = wTrack.promote();
1424            if (track != 0) {
1425                track->dump(buffer, SIZE);
1426                result.append(buffer);
1427            }
1428        }
1429    }
1430    write(fd, result.string(), result.size());
1431    return NO_ERROR;
1432}
1433
1434status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1435{
1436    const size_t SIZE = 256;
1437    char buffer[SIZE];
1438    String8 result;
1439
1440    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1441    result.append(buffer);
1442    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1443    result.append(buffer);
1444    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1445    result.append(buffer);
1446    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1447    result.append(buffer);
1448    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1449    result.append(buffer);
1450    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1451    result.append(buffer);
1452    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1453    result.append(buffer);
1454    write(fd, result.string(), result.size());
1455
1456    dumpBase(fd, args);
1457
1458    return NO_ERROR;
1459}
1460
1461// Thread virtuals
1462status_t AudioFlinger::PlaybackThread::readyToRun()
1463{
1464    status_t status = initCheck();
1465    if (status == NO_ERROR) {
1466        LOGI("AudioFlinger's thread %p ready to run", this);
1467    } else {
1468        LOGE("No working audio driver found.");
1469    }
1470    return status;
1471}
1472
1473void AudioFlinger::PlaybackThread::onFirstRef()
1474{
1475    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1476}
1477
1478// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1479sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1480        const sp<AudioFlinger::Client>& client,
1481        int streamType,
1482        uint32_t sampleRate,
1483        uint32_t format,
1484        uint32_t channelMask,
1485        int frameCount,
1486        const sp<IMemory>& sharedBuffer,
1487        int sessionId,
1488        status_t *status)
1489{
1490    sp<Track> track;
1491    status_t lStatus;
1492
1493    if (mType == DIRECT) {
1494        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1495            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1496                LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1497                        "for output %p with format %d",
1498                        sampleRate, format, channelMask, mOutput, mFormat);
1499                lStatus = BAD_VALUE;
1500                goto Exit;
1501            }
1502        }
1503    } else {
1504        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1505        if (sampleRate > mSampleRate*2) {
1506            LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1507            lStatus = BAD_VALUE;
1508            goto Exit;
1509        }
1510    }
1511
1512    lStatus = initCheck();
1513    if (lStatus != NO_ERROR) {
1514        LOGE("Audio driver not initialized.");
1515        goto Exit;
1516    }
1517
1518    { // scope for mLock
1519        Mutex::Autolock _l(mLock);
1520
1521        // all tracks in same audio session must share the same routing strategy otherwise
1522        // conflicts will happen when tracks are moved from one output to another by audio policy
1523        // manager
1524        uint32_t strategy =
1525                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1526        for (size_t i = 0; i < mTracks.size(); ++i) {
1527            sp<Track> t = mTracks[i];
1528            if (t != 0) {
1529                if (sessionId == t->sessionId() &&
1530                        strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
1531                    lStatus = BAD_VALUE;
1532                    goto Exit;
1533                }
1534            }
1535        }
1536
1537        track = new Track(this, client, streamType, sampleRate, format,
1538                channelMask, frameCount, sharedBuffer, sessionId);
1539        if (track->getCblk() == NULL || track->name() < 0) {
1540            lStatus = NO_MEMORY;
1541            goto Exit;
1542        }
1543        mTracks.add(track);
1544
1545        sp<EffectChain> chain = getEffectChain_l(sessionId);
1546        if (chain != 0) {
1547            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1548            track->setMainBuffer(chain->inBuffer());
1549            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1550            chain->incTrackCnt();
1551        }
1552
1553        // invalidate track immediately if the stream type was moved to another thread since
1554        // createTrack() was called by the client process.
1555        if (!mStreamTypes[streamType].valid) {
1556            LOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1557                 this, streamType);
1558            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1559        }
1560    }
1561    lStatus = NO_ERROR;
1562
1563Exit:
1564    if(status) {
1565        *status = lStatus;
1566    }
1567    return track;
1568}
1569
1570uint32_t AudioFlinger::PlaybackThread::latency() const
1571{
1572    Mutex::Autolock _l(mLock);
1573    if (initCheck() == NO_ERROR) {
1574        return mOutput->stream->get_latency(mOutput->stream);
1575    } else {
1576        return 0;
1577    }
1578}
1579
1580status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1581{
1582    mMasterVolume = value;
1583    return NO_ERROR;
1584}
1585
1586status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1587{
1588    mMasterMute = muted;
1589    return NO_ERROR;
1590}
1591
1592float AudioFlinger::PlaybackThread::masterVolume() const
1593{
1594    return mMasterVolume;
1595}
1596
1597bool AudioFlinger::PlaybackThread::masterMute() const
1598{
1599    return mMasterMute;
1600}
1601
1602status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
1603{
1604    mStreamTypes[stream].volume = value;
1605    return NO_ERROR;
1606}
1607
1608status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
1609{
1610    mStreamTypes[stream].mute = muted;
1611    return NO_ERROR;
1612}
1613
1614float AudioFlinger::PlaybackThread::streamVolume(int stream) const
1615{
1616    return mStreamTypes[stream].volume;
1617}
1618
1619bool AudioFlinger::PlaybackThread::streamMute(int stream) const
1620{
1621    return mStreamTypes[stream].mute;
1622}
1623
1624// addTrack_l() must be called with ThreadBase::mLock held
1625status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1626{
1627    status_t status = ALREADY_EXISTS;
1628
1629    // set retry count for buffer fill
1630    track->mRetryCount = kMaxTrackStartupRetries;
1631    if (mActiveTracks.indexOf(track) < 0) {
1632        // the track is newly added, make sure it fills up all its
1633        // buffers before playing. This is to ensure the client will
1634        // effectively get the latency it requested.
1635        track->mFillingUpStatus = Track::FS_FILLING;
1636        track->mResetDone = false;
1637        mActiveTracks.add(track);
1638        if (track->mainBuffer() != mMixBuffer) {
1639            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1640            if (chain != 0) {
1641                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1642                chain->incActiveTrackCnt();
1643            }
1644        }
1645
1646        status = NO_ERROR;
1647    }
1648
1649    ALOGV("mWaitWorkCV.broadcast");
1650    mWaitWorkCV.broadcast();
1651
1652    return status;
1653}
1654
1655// destroyTrack_l() must be called with ThreadBase::mLock held
1656void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1657{
1658    track->mState = TrackBase::TERMINATED;
1659    if (mActiveTracks.indexOf(track) < 0) {
1660        removeTrack_l(track);
1661    }
1662}
1663
1664void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1665{
1666    mTracks.remove(track);
1667    deleteTrackName_l(track->name());
1668    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1669    if (chain != 0) {
1670        chain->decTrackCnt();
1671    }
1672}
1673
1674String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1675{
1676    String8 out_s8 = String8("");
1677    char *s;
1678
1679    Mutex::Autolock _l(mLock);
1680    if (initCheck() != NO_ERROR) {
1681        return out_s8;
1682    }
1683
1684    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1685    out_s8 = String8(s);
1686    free(s);
1687    return out_s8;
1688}
1689
1690// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1691void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1692    AudioSystem::OutputDescriptor desc;
1693    void *param2 = 0;
1694
1695    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1696
1697    switch (event) {
1698    case AudioSystem::OUTPUT_OPENED:
1699    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1700        desc.channels = mChannelMask;
1701        desc.samplingRate = mSampleRate;
1702        desc.format = mFormat;
1703        desc.frameCount = mFrameCount;
1704        desc.latency = latency();
1705        param2 = &desc;
1706        break;
1707
1708    case AudioSystem::STREAM_CONFIG_CHANGED:
1709        param2 = &param;
1710    case AudioSystem::OUTPUT_CLOSED:
1711    default:
1712        break;
1713    }
1714    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1715}
1716
1717void AudioFlinger::PlaybackThread::readOutputParameters()
1718{
1719    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1720    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1721    mChannelCount = (uint16_t)popcount(mChannelMask);
1722    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1723    mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
1724    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1725
1726    // FIXME - Current mixer implementation only supports stereo output: Always
1727    // Allocate a stereo buffer even if HW output is mono.
1728    if (mMixBuffer != NULL) delete[] mMixBuffer;
1729    mMixBuffer = new int16_t[mFrameCount * 2];
1730    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1731
1732    // force reconfiguration of effect chains and engines to take new buffer size and audio
1733    // parameters into account
1734    // Note that mLock is not held when readOutputParameters() is called from the constructor
1735    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1736    // matter.
1737    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1738    Vector< sp<EffectChain> > effectChains = mEffectChains;
1739    for (size_t i = 0; i < effectChains.size(); i ++) {
1740        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1741    }
1742}
1743
1744status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1745{
1746    if (halFrames == 0 || dspFrames == 0) {
1747        return BAD_VALUE;
1748    }
1749    Mutex::Autolock _l(mLock);
1750    if (initCheck() != NO_ERROR) {
1751        return INVALID_OPERATION;
1752    }
1753    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1754
1755    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1756}
1757
1758uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1759{
1760    Mutex::Autolock _l(mLock);
1761    uint32_t result = 0;
1762    if (getEffectChain_l(sessionId) != 0) {
1763        result = EFFECT_SESSION;
1764    }
1765
1766    for (size_t i = 0; i < mTracks.size(); ++i) {
1767        sp<Track> track = mTracks[i];
1768        if (sessionId == track->sessionId() &&
1769                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1770            result |= TRACK_SESSION;
1771            break;
1772        }
1773    }
1774
1775    return result;
1776}
1777
1778uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1779{
1780    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1781    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1782    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1783        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1784    }
1785    for (size_t i = 0; i < mTracks.size(); i++) {
1786        sp<Track> track = mTracks[i];
1787        if (sessionId == track->sessionId() &&
1788                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1789            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1790        }
1791    }
1792    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1793}
1794
1795
1796AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1797{
1798    Mutex::Autolock _l(mLock);
1799    return mOutput;
1800}
1801
1802AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1803{
1804    Mutex::Autolock _l(mLock);
1805    AudioStreamOut *output = mOutput;
1806    mOutput = NULL;
1807    return output;
1808}
1809
1810// this method must always be called either with ThreadBase mLock held or inside the thread loop
1811audio_stream_t* AudioFlinger::PlaybackThread::stream()
1812{
1813    if (mOutput == NULL) {
1814        return NULL;
1815    }
1816    return &mOutput->stream->common;
1817}
1818
1819// ----------------------------------------------------------------------------
1820
1821AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1822    :   PlaybackThread(audioFlinger, output, id, device),
1823        mAudioMixer(0)
1824{
1825    mType = ThreadBase::MIXER;
1826    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1827
1828    // FIXME - Current mixer implementation only supports stereo output
1829    if (mChannelCount == 1) {
1830        LOGE("Invalid audio hardware channel count");
1831    }
1832}
1833
1834AudioFlinger::MixerThread::~MixerThread()
1835{
1836    delete mAudioMixer;
1837}
1838
1839bool AudioFlinger::MixerThread::threadLoop()
1840{
1841    Vector< sp<Track> > tracksToRemove;
1842    uint32_t mixerStatus = MIXER_IDLE;
1843    nsecs_t standbyTime = systemTime();
1844    size_t mixBufferSize = mFrameCount * mFrameSize;
1845    // FIXME: Relaxed timing because of a certain device that can't meet latency
1846    // Should be reduced to 2x after the vendor fixes the driver issue
1847    // increase threshold again due to low power audio mode. The way this warning threshold is
1848    // calculated and its usefulness should be reconsidered anyway.
1849    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1850    nsecs_t lastWarning = 0;
1851    bool longStandbyExit = false;
1852    uint32_t activeSleepTime = activeSleepTimeUs();
1853    uint32_t idleSleepTime = idleSleepTimeUs();
1854    uint32_t sleepTime = idleSleepTime;
1855    uint32_t sleepTimeShift = 0;
1856    Vector< sp<EffectChain> > effectChains;
1857#ifdef DEBUG_CPU_USAGE
1858    ThreadCpuUsage cpu;
1859    const CentralTendencyStatistics& stats = cpu.statistics();
1860#endif
1861
1862    acquireWakeLock();
1863
1864    while (!exitPending())
1865    {
1866#ifdef DEBUG_CPU_USAGE
1867        cpu.sampleAndEnable();
1868        unsigned n = stats.n();
1869        // cpu.elapsed() is expensive, so don't call it every loop
1870        if ((n & 127) == 1) {
1871            long long elapsed = cpu.elapsed();
1872            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1873                double perLoop = elapsed / (double) n;
1874                double perLoop100 = perLoop * 0.01;
1875                double mean = stats.mean();
1876                double stddev = stats.stddev();
1877                double minimum = stats.minimum();
1878                double maximum = stats.maximum();
1879                cpu.resetStatistics();
1880                LOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
1881                        elapsed * .000000001, n, perLoop * .000001,
1882                        mean * .001,
1883                        stddev * .001,
1884                        minimum * .001,
1885                        maximum * .001,
1886                        mean / perLoop100,
1887                        stddev / perLoop100,
1888                        minimum / perLoop100,
1889                        maximum / perLoop100);
1890            }
1891        }
1892#endif
1893        processConfigEvents();
1894
1895        mixerStatus = MIXER_IDLE;
1896        { // scope for mLock
1897
1898            Mutex::Autolock _l(mLock);
1899
1900            if (checkForNewParameters_l()) {
1901                mixBufferSize = mFrameCount * mFrameSize;
1902                // FIXME: Relaxed timing because of a certain device that can't meet latency
1903                // Should be reduced to 2x after the vendor fixes the driver issue
1904                // increase threshold again due to low power audio mode. The way this warning
1905                // threshold is calculated and its usefulness should be reconsidered anyway.
1906                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1907                activeSleepTime = activeSleepTimeUs();
1908                idleSleepTime = idleSleepTimeUs();
1909            }
1910
1911            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1912
1913            // put audio hardware into standby after short delay
1914            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1915                        mSuspended) {
1916                if (!mStandby) {
1917                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1918                    mOutput->stream->common.standby(&mOutput->stream->common);
1919                    mStandby = true;
1920                    mBytesWritten = 0;
1921                }
1922
1923                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1924                    // we're about to wait, flush the binder command buffer
1925                    IPCThreadState::self()->flushCommands();
1926
1927                    if (exitPending()) break;
1928
1929                    releaseWakeLock_l();
1930                    // wait until we have something to do...
1931                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1932                    mWaitWorkCV.wait(mLock);
1933                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1934                    acquireWakeLock_l();
1935
1936                    if (mMasterMute == false) {
1937                        char value[PROPERTY_VALUE_MAX];
1938                        property_get("ro.audio.silent", value, "0");
1939                        if (atoi(value)) {
1940                            LOGD("Silence is golden");
1941                            setMasterMute(true);
1942                        }
1943                    }
1944
1945                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1946                    sleepTime = idleSleepTime;
1947                    sleepTimeShift = 0;
1948                    continue;
1949                }
1950            }
1951
1952            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1953
1954            // prevent any changes in effect chain list and in each effect chain
1955            // during mixing and effect process as the audio buffers could be deleted
1956            // or modified if an effect is created or deleted
1957            lockEffectChains_l(effectChains);
1958       }
1959
1960        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1961            // mix buffers...
1962            mAudioMixer->process();
1963            sleepTime = 0;
1964            // increase sleep time progressively when application underrun condition clears
1965            if (sleepTimeShift > 0) {
1966                sleepTimeShift--;
1967            }
1968            standbyTime = systemTime() + kStandbyTimeInNsecs;
1969            //TODO: delay standby when effects have a tail
1970        } else {
1971            // If no tracks are ready, sleep once for the duration of an output
1972            // buffer size, then write 0s to the output
1973            if (sleepTime == 0) {
1974                if (mixerStatus == MIXER_TRACKS_ENABLED) {
1975                    sleepTime = activeSleepTime >> sleepTimeShift;
1976                    if (sleepTime < kMinThreadSleepTimeUs) {
1977                        sleepTime = kMinThreadSleepTimeUs;
1978                    }
1979                    // reduce sleep time in case of consecutive application underruns to avoid
1980                    // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
1981                    // duration we would end up writing less data than needed by the audio HAL if
1982                    // the condition persists.
1983                    if (sleepTimeShift < kMaxThreadSleepTimeShift) {
1984                        sleepTimeShift++;
1985                    }
1986                } else {
1987                    sleepTime = idleSleepTime;
1988                }
1989            } else if (mBytesWritten != 0 ||
1990                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
1991                memset (mMixBuffer, 0, mixBufferSize);
1992                sleepTime = 0;
1993                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
1994            }
1995            // TODO add standby time extension fct of effect tail
1996        }
1997
1998        if (mSuspended) {
1999            sleepTime = suspendSleepTimeUs();
2000        }
2001        // sleepTime == 0 means we must write to audio hardware
2002        if (sleepTime == 0) {
2003             for (size_t i = 0; i < effectChains.size(); i ++) {
2004                 effectChains[i]->process_l();
2005             }
2006             // enable changes in effect chain
2007             unlockEffectChains(effectChains);
2008            mLastWriteTime = systemTime();
2009            mInWrite = true;
2010            mBytesWritten += mixBufferSize;
2011
2012            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2013            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2014            mNumWrites++;
2015            mInWrite = false;
2016            nsecs_t now = systemTime();
2017            nsecs_t delta = now - mLastWriteTime;
2018            if (!mStandby && delta > maxPeriod) {
2019                mNumDelayedWrites++;
2020                if ((now - lastWarning) > kWarningThrottle) {
2021                    LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2022                            ns2ms(delta), mNumDelayedWrites, this);
2023                    lastWarning = now;
2024                }
2025                if (mStandby) {
2026                    longStandbyExit = true;
2027                }
2028            }
2029            mStandby = false;
2030        } else {
2031            // enable changes in effect chain
2032            unlockEffectChains(effectChains);
2033            usleep(sleepTime);
2034        }
2035
2036        // finally let go of all our tracks, without the lock held
2037        // since we can't guarantee the destructors won't acquire that
2038        // same lock.
2039        tracksToRemove.clear();
2040
2041        // Effect chains will be actually deleted here if they were removed from
2042        // mEffectChains list during mixing or effects processing
2043        effectChains.clear();
2044    }
2045
2046    if (!mStandby) {
2047        mOutput->stream->common.standby(&mOutput->stream->common);
2048    }
2049
2050    releaseWakeLock();
2051
2052    ALOGV("MixerThread %p exiting", this);
2053    return false;
2054}
2055
2056// prepareTracks_l() must be called with ThreadBase::mLock held
2057uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2058{
2059
2060    uint32_t mixerStatus = MIXER_IDLE;
2061    // find out which tracks need to be processed
2062    size_t count = activeTracks.size();
2063    size_t mixedTracks = 0;
2064    size_t tracksWithEffect = 0;
2065
2066    float masterVolume = mMasterVolume;
2067    bool  masterMute = mMasterMute;
2068
2069    if (masterMute) {
2070        masterVolume = 0;
2071    }
2072    // Delegate master volume control to effect in output mix effect chain if needed
2073    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2074    if (chain != 0) {
2075        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2076        chain->setVolume_l(&v, &v);
2077        masterVolume = (float)((v + (1 << 23)) >> 24);
2078        chain.clear();
2079    }
2080
2081    for (size_t i=0 ; i<count ; i++) {
2082        sp<Track> t = activeTracks[i].promote();
2083        if (t == 0) continue;
2084
2085        Track* const track = t.get();
2086        audio_track_cblk_t* cblk = track->cblk();
2087
2088        // The first time a track is added we wait
2089        // for all its buffers to be filled before processing it
2090        mAudioMixer->setActiveTrack(track->name());
2091        // make sure that we have enough frames to mix one full buffer.
2092        // enforce this condition only once to enable draining the buffer in case the client
2093        // app does not call stop() and relies on underrun to stop:
2094        // hence the test on (track->mRetryCount >= kMaxTrackRetries) meaning the track was mixed
2095        // during last round
2096        uint32_t minFrames = 1;
2097        if (!track->isStopped() && !track->isPausing() &&
2098                (track->mRetryCount >= kMaxTrackRetries)) {
2099            if (t->sampleRate() == (int)mSampleRate) {
2100                minFrames = mFrameCount;
2101            } else {
2102                minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1;
2103            }
2104        }
2105        if ((cblk->framesReady() >= minFrames) && track->isReady() &&
2106                !track->isPaused() && !track->isTerminated())
2107        {
2108            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
2109
2110            mixedTracks++;
2111
2112            // track->mainBuffer() != mMixBuffer means there is an effect chain
2113            // connected to the track
2114            chain.clear();
2115            if (track->mainBuffer() != mMixBuffer) {
2116                chain = getEffectChain_l(track->sessionId());
2117                // Delegate volume control to effect in track effect chain if needed
2118                if (chain != 0) {
2119                    tracksWithEffect++;
2120                } else {
2121                    LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
2122                            track->name(), track->sessionId());
2123                }
2124            }
2125
2126
2127            int param = AudioMixer::VOLUME;
2128            if (track->mFillingUpStatus == Track::FS_FILLED) {
2129                // no ramp for the first volume setting
2130                track->mFillingUpStatus = Track::FS_ACTIVE;
2131                if (track->mState == TrackBase::RESUMING) {
2132                    track->mState = TrackBase::ACTIVE;
2133                    param = AudioMixer::RAMP_VOLUME;
2134                }
2135                mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2136            } else if (cblk->server != 0) {
2137                // If the track is stopped before the first frame was mixed,
2138                // do not apply ramp
2139                param = AudioMixer::RAMP_VOLUME;
2140            }
2141
2142            // compute volume for this track
2143            uint32_t vl, vr, va;
2144            if (track->isMuted() || track->isPausing() ||
2145                mStreamTypes[track->type()].mute) {
2146                vl = vr = va = 0;
2147                if (track->isPausing()) {
2148                    track->setPaused();
2149                }
2150            } else {
2151
2152                // read original volumes with volume control
2153                float typeVolume = mStreamTypes[track->type()].volume;
2154                float v = masterVolume * typeVolume;
2155                vl = (uint32_t)(v * cblk->volume[0]) << 12;
2156                vr = (uint32_t)(v * cblk->volume[1]) << 12;
2157
2158                va = (uint32_t)(v * cblk->sendLevel);
2159            }
2160            // Delegate volume control to effect in track effect chain if needed
2161            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2162                // Do not ramp volume if volume is controlled by effect
2163                param = AudioMixer::VOLUME;
2164                track->mHasVolumeController = true;
2165            } else {
2166                // force no volume ramp when volume controller was just disabled or removed
2167                // from effect chain to avoid volume spike
2168                if (track->mHasVolumeController) {
2169                    param = AudioMixer::VOLUME;
2170                }
2171                track->mHasVolumeController = false;
2172            }
2173
2174            // Convert volumes from 8.24 to 4.12 format
2175            int16_t left, right, aux;
2176            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2177            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2178            left = int16_t(v_clamped);
2179            v_clamped = (vr + (1 << 11)) >> 12;
2180            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2181            right = int16_t(v_clamped);
2182
2183            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2184            aux = int16_t(va);
2185
2186            // XXX: these things DON'T need to be done each time
2187            mAudioMixer->setBufferProvider(track);
2188            mAudioMixer->enable(AudioMixer::MIXING);
2189
2190            mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
2191            mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
2192            mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
2193            mAudioMixer->setParameter(
2194                AudioMixer::TRACK,
2195                AudioMixer::FORMAT, (void *)track->format());
2196            mAudioMixer->setParameter(
2197                AudioMixer::TRACK,
2198                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2199            mAudioMixer->setParameter(
2200                AudioMixer::RESAMPLE,
2201                AudioMixer::SAMPLE_RATE,
2202                (void *)(cblk->sampleRate));
2203            mAudioMixer->setParameter(
2204                AudioMixer::TRACK,
2205                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2206            mAudioMixer->setParameter(
2207                AudioMixer::TRACK,
2208                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2209
2210            // reset retry count
2211            track->mRetryCount = kMaxTrackRetries;
2212            mixerStatus = MIXER_TRACKS_READY;
2213        } else {
2214            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
2215            if (track->isStopped()) {
2216                track->reset();
2217            }
2218            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2219                // We have consumed all the buffers of this track.
2220                // Remove it from the list of active tracks.
2221                tracksToRemove->add(track);
2222            } else {
2223                // No buffers for this track. Give it a few chances to
2224                // fill a buffer, then remove it from active list.
2225                if (--(track->mRetryCount) <= 0) {
2226                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
2227                    tracksToRemove->add(track);
2228                    // indicate to client process that the track was disabled because of underrun
2229                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2230                } else if (mixerStatus != MIXER_TRACKS_READY) {
2231                    mixerStatus = MIXER_TRACKS_ENABLED;
2232                }
2233            }
2234            mAudioMixer->disable(AudioMixer::MIXING);
2235        }
2236    }
2237
2238    // remove all the tracks that need to be...
2239    count = tracksToRemove->size();
2240    if (UNLIKELY(count)) {
2241        for (size_t i=0 ; i<count ; i++) {
2242            const sp<Track>& track = tracksToRemove->itemAt(i);
2243            mActiveTracks.remove(track);
2244            if (track->mainBuffer() != mMixBuffer) {
2245                chain = getEffectChain_l(track->sessionId());
2246                if (chain != 0) {
2247                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2248                    chain->decActiveTrackCnt();
2249                }
2250            }
2251            if (track->isTerminated()) {
2252                removeTrack_l(track);
2253            }
2254        }
2255    }
2256
2257    // mix buffer must be cleared if all tracks are connected to an
2258    // effect chain as in this case the mixer will not write to
2259    // mix buffer and track effects will accumulate into it
2260    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2261        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2262    }
2263
2264    return mixerStatus;
2265}
2266
2267void AudioFlinger::MixerThread::invalidateTracks(int streamType)
2268{
2269    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2270            this,  streamType, mTracks.size());
2271    Mutex::Autolock _l(mLock);
2272
2273    size_t size = mTracks.size();
2274    for (size_t i = 0; i < size; i++) {
2275        sp<Track> t = mTracks[i];
2276        if (t->type() == streamType) {
2277            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2278            t->mCblk->cv.signal();
2279        }
2280    }
2281}
2282
2283void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid)
2284{
2285    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2286            this,  streamType, valid);
2287    Mutex::Autolock _l(mLock);
2288
2289    mStreamTypes[streamType].valid = valid;
2290}
2291
2292// getTrackName_l() must be called with ThreadBase::mLock held
2293int AudioFlinger::MixerThread::getTrackName_l()
2294{
2295    return mAudioMixer->getTrackName();
2296}
2297
2298// deleteTrackName_l() must be called with ThreadBase::mLock held
2299void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2300{
2301    ALOGV("remove track (%d) and delete from mixer", name);
2302    mAudioMixer->deleteTrackName(name);
2303}
2304
2305// checkForNewParameters_l() must be called with ThreadBase::mLock held
2306bool AudioFlinger::MixerThread::checkForNewParameters_l()
2307{
2308    bool reconfig = false;
2309
2310    while (!mNewParameters.isEmpty()) {
2311        status_t status = NO_ERROR;
2312        String8 keyValuePair = mNewParameters[0];
2313        AudioParameter param = AudioParameter(keyValuePair);
2314        int value;
2315
2316        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2317            reconfig = true;
2318        }
2319        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2320            if (value != AUDIO_FORMAT_PCM_16_BIT) {
2321                status = BAD_VALUE;
2322            } else {
2323                reconfig = true;
2324            }
2325        }
2326        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2327            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2328                status = BAD_VALUE;
2329            } else {
2330                reconfig = true;
2331            }
2332        }
2333        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2334            // do not accept frame count changes if tracks are open as the track buffer
2335            // size depends on frame count and correct behavior would not be garantied
2336            // if frame count is changed after track creation
2337            if (!mTracks.isEmpty()) {
2338                status = INVALID_OPERATION;
2339            } else {
2340                reconfig = true;
2341            }
2342        }
2343        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2344            // when changing the audio output device, call addBatteryData to notify
2345            // the change
2346            if ((int)mDevice != value) {
2347                uint32_t params = 0;
2348                // check whether speaker is on
2349                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2350                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2351                }
2352
2353                int deviceWithoutSpeaker
2354                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2355                // check if any other device (except speaker) is on
2356                if (value & deviceWithoutSpeaker ) {
2357                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2358                }
2359
2360                if (params != 0) {
2361                    addBatteryData(params);
2362                }
2363            }
2364
2365            // forward device change to effects that have requested to be
2366            // aware of attached audio device.
2367            mDevice = (uint32_t)value;
2368            for (size_t i = 0; i < mEffectChains.size(); i++) {
2369                mEffectChains[i]->setDevice_l(mDevice);
2370            }
2371        }
2372
2373        if (status == NO_ERROR) {
2374            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2375                                                    keyValuePair.string());
2376            if (!mStandby && status == INVALID_OPERATION) {
2377               mOutput->stream->common.standby(&mOutput->stream->common);
2378               mStandby = true;
2379               mBytesWritten = 0;
2380               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2381                                                       keyValuePair.string());
2382            }
2383            if (status == NO_ERROR && reconfig) {
2384                delete mAudioMixer;
2385                readOutputParameters();
2386                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2387                for (size_t i = 0; i < mTracks.size() ; i++) {
2388                    int name = getTrackName_l();
2389                    if (name < 0) break;
2390                    mTracks[i]->mName = name;
2391                    // limit track sample rate to 2 x new output sample rate
2392                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2393                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2394                    }
2395                }
2396                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2397            }
2398        }
2399
2400        mNewParameters.removeAt(0);
2401
2402        mParamStatus = status;
2403        mParamCond.signal();
2404        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2405        // already timed out waiting for the status and will never signal the condition.
2406        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
2407    }
2408    return reconfig;
2409}
2410
2411status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2412{
2413    const size_t SIZE = 256;
2414    char buffer[SIZE];
2415    String8 result;
2416
2417    PlaybackThread::dumpInternals(fd, args);
2418
2419    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2420    result.append(buffer);
2421    write(fd, result.string(), result.size());
2422    return NO_ERROR;
2423}
2424
2425uint32_t AudioFlinger::MixerThread::activeSleepTimeUs()
2426{
2427    return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2428}
2429
2430uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2431{
2432    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2433}
2434
2435uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2436{
2437    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2438}
2439
2440// ----------------------------------------------------------------------------
2441AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2442    :   PlaybackThread(audioFlinger, output, id, device)
2443{
2444    mType = ThreadBase::DIRECT;
2445}
2446
2447AudioFlinger::DirectOutputThread::~DirectOutputThread()
2448{
2449}
2450
2451
2452static inline int16_t clamp16(int32_t sample)
2453{
2454    if ((sample>>15) ^ (sample>>31))
2455        sample = 0x7FFF ^ (sample>>31);
2456    return sample;
2457}
2458
2459static inline
2460int32_t mul(int16_t in, int16_t v)
2461{
2462#if defined(__arm__) && !defined(__thumb__)
2463    int32_t out;
2464    asm( "smulbb %[out], %[in], %[v] \n"
2465         : [out]"=r"(out)
2466         : [in]"%r"(in), [v]"r"(v)
2467         : );
2468    return out;
2469#else
2470    return in * int32_t(v);
2471#endif
2472}
2473
2474void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2475{
2476    // Do not apply volume on compressed audio
2477    if (!audio_is_linear_pcm(mFormat)) {
2478        return;
2479    }
2480
2481    // convert to signed 16 bit before volume calculation
2482    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2483        size_t count = mFrameCount * mChannelCount;
2484        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2485        int16_t *dst = mMixBuffer + count-1;
2486        while(count--) {
2487            *dst-- = (int16_t)(*src--^0x80) << 8;
2488        }
2489    }
2490
2491    size_t frameCount = mFrameCount;
2492    int16_t *out = mMixBuffer;
2493    if (ramp) {
2494        if (mChannelCount == 1) {
2495            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2496            int32_t vlInc = d / (int32_t)frameCount;
2497            int32_t vl = ((int32_t)mLeftVolShort << 16);
2498            do {
2499                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2500                out++;
2501                vl += vlInc;
2502            } while (--frameCount);
2503
2504        } else {
2505            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2506            int32_t vlInc = d / (int32_t)frameCount;
2507            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2508            int32_t vrInc = d / (int32_t)frameCount;
2509            int32_t vl = ((int32_t)mLeftVolShort << 16);
2510            int32_t vr = ((int32_t)mRightVolShort << 16);
2511            do {
2512                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2513                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2514                out += 2;
2515                vl += vlInc;
2516                vr += vrInc;
2517            } while (--frameCount);
2518        }
2519    } else {
2520        if (mChannelCount == 1) {
2521            do {
2522                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2523                out++;
2524            } while (--frameCount);
2525        } else {
2526            do {
2527                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2528                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2529                out += 2;
2530            } while (--frameCount);
2531        }
2532    }
2533
2534    // convert back to unsigned 8 bit after volume calculation
2535    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2536        size_t count = mFrameCount * mChannelCount;
2537        int16_t *src = mMixBuffer;
2538        uint8_t *dst = (uint8_t *)mMixBuffer;
2539        while(count--) {
2540            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2541        }
2542    }
2543
2544    mLeftVolShort = leftVol;
2545    mRightVolShort = rightVol;
2546}
2547
2548bool AudioFlinger::DirectOutputThread::threadLoop()
2549{
2550    uint32_t mixerStatus = MIXER_IDLE;
2551    sp<Track> trackToRemove;
2552    sp<Track> activeTrack;
2553    nsecs_t standbyTime = systemTime();
2554    int8_t *curBuf;
2555    size_t mixBufferSize = mFrameCount*mFrameSize;
2556    uint32_t activeSleepTime = activeSleepTimeUs();
2557    uint32_t idleSleepTime = idleSleepTimeUs();
2558    uint32_t sleepTime = idleSleepTime;
2559    // use shorter standby delay as on normal output to release
2560    // hardware resources as soon as possible
2561    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2562
2563    acquireWakeLock();
2564
2565    while (!exitPending())
2566    {
2567        bool rampVolume;
2568        uint16_t leftVol;
2569        uint16_t rightVol;
2570        Vector< sp<EffectChain> > effectChains;
2571
2572        processConfigEvents();
2573
2574        mixerStatus = MIXER_IDLE;
2575
2576        { // scope for the mLock
2577
2578            Mutex::Autolock _l(mLock);
2579
2580            if (checkForNewParameters_l()) {
2581                mixBufferSize = mFrameCount*mFrameSize;
2582                activeSleepTime = activeSleepTimeUs();
2583                idleSleepTime = idleSleepTimeUs();
2584                standbyDelay = microseconds(activeSleepTime*2);
2585            }
2586
2587            // put audio hardware into standby after short delay
2588            if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2589                        mSuspended) {
2590                // wait until we have something to do...
2591                if (!mStandby) {
2592                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2593                    mOutput->stream->common.standby(&mOutput->stream->common);
2594                    mStandby = true;
2595                    mBytesWritten = 0;
2596                }
2597
2598                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2599                    // we're about to wait, flush the binder command buffer
2600                    IPCThreadState::self()->flushCommands();
2601
2602                    if (exitPending()) break;
2603
2604                    releaseWakeLock_l();
2605                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2606                    mWaitWorkCV.wait(mLock);
2607                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2608                    acquireWakeLock_l();
2609
2610                    if (mMasterMute == false) {
2611                        char value[PROPERTY_VALUE_MAX];
2612                        property_get("ro.audio.silent", value, "0");
2613                        if (atoi(value)) {
2614                            LOGD("Silence is golden");
2615                            setMasterMute(true);
2616                        }
2617                    }
2618
2619                    standbyTime = systemTime() + standbyDelay;
2620                    sleepTime = idleSleepTime;
2621                    continue;
2622                }
2623            }
2624
2625            effectChains = mEffectChains;
2626
2627            // find out which tracks need to be processed
2628            if (mActiveTracks.size() != 0) {
2629                sp<Track> t = mActiveTracks[0].promote();
2630                if (t == 0) continue;
2631
2632                Track* const track = t.get();
2633                audio_track_cblk_t* cblk = track->cblk();
2634
2635                // The first time a track is added we wait
2636                // for all its buffers to be filled before processing it
2637                if (cblk->framesReady() && track->isReady() &&
2638                        !track->isPaused() && !track->isTerminated())
2639                {
2640                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2641
2642                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2643                        track->mFillingUpStatus = Track::FS_ACTIVE;
2644                        mLeftVolFloat = mRightVolFloat = 0;
2645                        mLeftVolShort = mRightVolShort = 0;
2646                        if (track->mState == TrackBase::RESUMING) {
2647                            track->mState = TrackBase::ACTIVE;
2648                            rampVolume = true;
2649                        }
2650                    } else if (cblk->server != 0) {
2651                        // If the track is stopped before the first frame was mixed,
2652                        // do not apply ramp
2653                        rampVolume = true;
2654                    }
2655                    // compute volume for this track
2656                    float left, right;
2657                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2658                        mStreamTypes[track->type()].mute) {
2659                        left = right = 0;
2660                        if (track->isPausing()) {
2661                            track->setPaused();
2662                        }
2663                    } else {
2664                        float typeVolume = mStreamTypes[track->type()].volume;
2665                        float v = mMasterVolume * typeVolume;
2666                        float v_clamped = v * cblk->volume[0];
2667                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2668                        left = v_clamped/MAX_GAIN;
2669                        v_clamped = v * cblk->volume[1];
2670                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2671                        right = v_clamped/MAX_GAIN;
2672                    }
2673
2674                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2675                        mLeftVolFloat = left;
2676                        mRightVolFloat = right;
2677
2678                        // If audio HAL implements volume control,
2679                        // force software volume to nominal value
2680                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2681                            left = 1.0f;
2682                            right = 1.0f;
2683                        }
2684
2685                        // Convert volumes from float to 8.24
2686                        uint32_t vl = (uint32_t)(left * (1 << 24));
2687                        uint32_t vr = (uint32_t)(right * (1 << 24));
2688
2689                        // Delegate volume control to effect in track effect chain if needed
2690                        // only one effect chain can be present on DirectOutputThread, so if
2691                        // there is one, the track is connected to it
2692                        if (!effectChains.isEmpty()) {
2693                            // Do not ramp volume if volume is controlled by effect
2694                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2695                                rampVolume = false;
2696                            }
2697                        }
2698
2699                        // Convert volumes from 8.24 to 4.12 format
2700                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2701                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2702                        leftVol = (uint16_t)v_clamped;
2703                        v_clamped = (vr + (1 << 11)) >> 12;
2704                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2705                        rightVol = (uint16_t)v_clamped;
2706                    } else {
2707                        leftVol = mLeftVolShort;
2708                        rightVol = mRightVolShort;
2709                        rampVolume = false;
2710                    }
2711
2712                    // reset retry count
2713                    track->mRetryCount = kMaxTrackRetriesDirect;
2714                    activeTrack = t;
2715                    mixerStatus = MIXER_TRACKS_READY;
2716                } else {
2717                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2718                    if (track->isStopped()) {
2719                        track->reset();
2720                    }
2721                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2722                        // We have consumed all the buffers of this track.
2723                        // Remove it from the list of active tracks.
2724                        trackToRemove = track;
2725                    } else {
2726                        // No buffers for this track. Give it a few chances to
2727                        // fill a buffer, then remove it from active list.
2728                        if (--(track->mRetryCount) <= 0) {
2729                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2730                            trackToRemove = track;
2731                        } else {
2732                            mixerStatus = MIXER_TRACKS_ENABLED;
2733                        }
2734                    }
2735                }
2736            }
2737
2738            // remove all the tracks that need to be...
2739            if (UNLIKELY(trackToRemove != 0)) {
2740                mActiveTracks.remove(trackToRemove);
2741                if (!effectChains.isEmpty()) {
2742                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2743                            trackToRemove->sessionId());
2744                    effectChains[0]->decActiveTrackCnt();
2745                }
2746                if (trackToRemove->isTerminated()) {
2747                    removeTrack_l(trackToRemove);
2748                }
2749            }
2750
2751            lockEffectChains_l(effectChains);
2752       }
2753
2754        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2755            AudioBufferProvider::Buffer buffer;
2756            size_t frameCount = mFrameCount;
2757            curBuf = (int8_t *)mMixBuffer;
2758            // output audio to hardware
2759            while (frameCount) {
2760                buffer.frameCount = frameCount;
2761                activeTrack->getNextBuffer(&buffer);
2762                if (UNLIKELY(buffer.raw == 0)) {
2763                    memset(curBuf, 0, frameCount * mFrameSize);
2764                    break;
2765                }
2766                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2767                frameCount -= buffer.frameCount;
2768                curBuf += buffer.frameCount * mFrameSize;
2769                activeTrack->releaseBuffer(&buffer);
2770            }
2771            sleepTime = 0;
2772            standbyTime = systemTime() + standbyDelay;
2773        } else {
2774            if (sleepTime == 0) {
2775                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2776                    sleepTime = activeSleepTime;
2777                } else {
2778                    sleepTime = idleSleepTime;
2779                }
2780            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2781                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2782                sleepTime = 0;
2783            }
2784        }
2785
2786        if (mSuspended) {
2787            sleepTime = suspendSleepTimeUs();
2788        }
2789        // sleepTime == 0 means we must write to audio hardware
2790        if (sleepTime == 0) {
2791            if (mixerStatus == MIXER_TRACKS_READY) {
2792                applyVolume(leftVol, rightVol, rampVolume);
2793            }
2794            for (size_t i = 0; i < effectChains.size(); i ++) {
2795                effectChains[i]->process_l();
2796            }
2797            unlockEffectChains(effectChains);
2798
2799            mLastWriteTime = systemTime();
2800            mInWrite = true;
2801            mBytesWritten += mixBufferSize;
2802            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2803            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2804            mNumWrites++;
2805            mInWrite = false;
2806            mStandby = false;
2807        } else {
2808            unlockEffectChains(effectChains);
2809            usleep(sleepTime);
2810        }
2811
2812        // finally let go of removed track, without the lock held
2813        // since we can't guarantee the destructors won't acquire that
2814        // same lock.
2815        trackToRemove.clear();
2816        activeTrack.clear();
2817
2818        // Effect chains will be actually deleted here if they were removed from
2819        // mEffectChains list during mixing or effects processing
2820        effectChains.clear();
2821    }
2822
2823    if (!mStandby) {
2824        mOutput->stream->common.standby(&mOutput->stream->common);
2825    }
2826
2827    releaseWakeLock();
2828
2829    ALOGV("DirectOutputThread %p exiting", this);
2830    return false;
2831}
2832
2833// getTrackName_l() must be called with ThreadBase::mLock held
2834int AudioFlinger::DirectOutputThread::getTrackName_l()
2835{
2836    return 0;
2837}
2838
2839// deleteTrackName_l() must be called with ThreadBase::mLock held
2840void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2841{
2842}
2843
2844// checkForNewParameters_l() must be called with ThreadBase::mLock held
2845bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2846{
2847    bool reconfig = false;
2848
2849    while (!mNewParameters.isEmpty()) {
2850        status_t status = NO_ERROR;
2851        String8 keyValuePair = mNewParameters[0];
2852        AudioParameter param = AudioParameter(keyValuePair);
2853        int value;
2854
2855        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2856            // do not accept frame count changes if tracks are open as the track buffer
2857            // size depends on frame count and correct behavior would not be garantied
2858            // if frame count is changed after track creation
2859            if (!mTracks.isEmpty()) {
2860                status = INVALID_OPERATION;
2861            } else {
2862                reconfig = true;
2863            }
2864        }
2865        if (status == NO_ERROR) {
2866            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2867                                                    keyValuePair.string());
2868            if (!mStandby && status == INVALID_OPERATION) {
2869               mOutput->stream->common.standby(&mOutput->stream->common);
2870               mStandby = true;
2871               mBytesWritten = 0;
2872               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2873                                                       keyValuePair.string());
2874            }
2875            if (status == NO_ERROR && reconfig) {
2876                readOutputParameters();
2877                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2878            }
2879        }
2880
2881        mNewParameters.removeAt(0);
2882
2883        mParamStatus = status;
2884        mParamCond.signal();
2885        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2886        // already timed out waiting for the status and will never signal the condition.
2887        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
2888    }
2889    return reconfig;
2890}
2891
2892uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2893{
2894    uint32_t time;
2895    if (audio_is_linear_pcm(mFormat)) {
2896        time = (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2897    } else {
2898        time = 10000;
2899    }
2900    return time;
2901}
2902
2903uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2904{
2905    uint32_t time;
2906    if (audio_is_linear_pcm(mFormat)) {
2907        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2908    } else {
2909        time = 10000;
2910    }
2911    return time;
2912}
2913
2914uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2915{
2916    uint32_t time;
2917    if (audio_is_linear_pcm(mFormat)) {
2918        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2919    } else {
2920        time = 10000;
2921    }
2922    return time;
2923}
2924
2925
2926// ----------------------------------------------------------------------------
2927
2928AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2929    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2930{
2931    mType = ThreadBase::DUPLICATING;
2932    addOutputTrack(mainThread);
2933}
2934
2935AudioFlinger::DuplicatingThread::~DuplicatingThread()
2936{
2937    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2938        mOutputTracks[i]->destroy();
2939    }
2940    mOutputTracks.clear();
2941}
2942
2943bool AudioFlinger::DuplicatingThread::threadLoop()
2944{
2945    Vector< sp<Track> > tracksToRemove;
2946    uint32_t mixerStatus = MIXER_IDLE;
2947    nsecs_t standbyTime = systemTime();
2948    size_t mixBufferSize = mFrameCount*mFrameSize;
2949    SortedVector< sp<OutputTrack> > outputTracks;
2950    uint32_t writeFrames = 0;
2951    uint32_t activeSleepTime = activeSleepTimeUs();
2952    uint32_t idleSleepTime = idleSleepTimeUs();
2953    uint32_t sleepTime = idleSleepTime;
2954    Vector< sp<EffectChain> > effectChains;
2955
2956    acquireWakeLock();
2957
2958    while (!exitPending())
2959    {
2960        processConfigEvents();
2961
2962        mixerStatus = MIXER_IDLE;
2963        { // scope for the mLock
2964
2965            Mutex::Autolock _l(mLock);
2966
2967            if (checkForNewParameters_l()) {
2968                mixBufferSize = mFrameCount*mFrameSize;
2969                updateWaitTime();
2970                activeSleepTime = activeSleepTimeUs();
2971                idleSleepTime = idleSleepTimeUs();
2972            }
2973
2974            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
2975
2976            for (size_t i = 0; i < mOutputTracks.size(); i++) {
2977                outputTracks.add(mOutputTracks[i]);
2978            }
2979
2980            // put audio hardware into standby after short delay
2981            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
2982                         mSuspended) {
2983                if (!mStandby) {
2984                    for (size_t i = 0; i < outputTracks.size(); i++) {
2985                        outputTracks[i]->stop();
2986                    }
2987                    mStandby = true;
2988                    mBytesWritten = 0;
2989                }
2990
2991                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
2992                    // we're about to wait, flush the binder command buffer
2993                    IPCThreadState::self()->flushCommands();
2994                    outputTracks.clear();
2995
2996                    if (exitPending()) break;
2997
2998                    releaseWakeLock_l();
2999                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3000                    mWaitWorkCV.wait(mLock);
3001                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3002                    acquireWakeLock_l();
3003
3004                    if (mMasterMute == false) {
3005                        char value[PROPERTY_VALUE_MAX];
3006                        property_get("ro.audio.silent", value, "0");
3007                        if (atoi(value)) {
3008                            LOGD("Silence is golden");
3009                            setMasterMute(true);
3010                        }
3011                    }
3012
3013                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3014                    sleepTime = idleSleepTime;
3015                    continue;
3016                }
3017            }
3018
3019            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3020
3021            // prevent any changes in effect chain list and in each effect chain
3022            // during mixing and effect process as the audio buffers could be deleted
3023            // or modified if an effect is created or deleted
3024            lockEffectChains_l(effectChains);
3025        }
3026
3027        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3028            // mix buffers...
3029            if (outputsReady(outputTracks)) {
3030                mAudioMixer->process();
3031            } else {
3032                memset(mMixBuffer, 0, mixBufferSize);
3033            }
3034            sleepTime = 0;
3035            writeFrames = mFrameCount;
3036        } else {
3037            if (sleepTime == 0) {
3038                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3039                    sleepTime = activeSleepTime;
3040                } else {
3041                    sleepTime = idleSleepTime;
3042                }
3043            } else if (mBytesWritten != 0) {
3044                // flush remaining overflow buffers in output tracks
3045                for (size_t i = 0; i < outputTracks.size(); i++) {
3046                    if (outputTracks[i]->isActive()) {
3047                        sleepTime = 0;
3048                        writeFrames = 0;
3049                        memset(mMixBuffer, 0, mixBufferSize);
3050                        break;
3051                    }
3052                }
3053            }
3054        }
3055
3056        if (mSuspended) {
3057            sleepTime = suspendSleepTimeUs();
3058        }
3059        // sleepTime == 0 means we must write to audio hardware
3060        if (sleepTime == 0) {
3061            for (size_t i = 0; i < effectChains.size(); i ++) {
3062                effectChains[i]->process_l();
3063            }
3064            // enable changes in effect chain
3065            unlockEffectChains(effectChains);
3066
3067            standbyTime = systemTime() + kStandbyTimeInNsecs;
3068            for (size_t i = 0; i < outputTracks.size(); i++) {
3069                outputTracks[i]->write(mMixBuffer, writeFrames);
3070            }
3071            mStandby = false;
3072            mBytesWritten += mixBufferSize;
3073        } else {
3074            // enable changes in effect chain
3075            unlockEffectChains(effectChains);
3076            usleep(sleepTime);
3077        }
3078
3079        // finally let go of all our tracks, without the lock held
3080        // since we can't guarantee the destructors won't acquire that
3081        // same lock.
3082        tracksToRemove.clear();
3083        outputTracks.clear();
3084
3085        // Effect chains will be actually deleted here if they were removed from
3086        // mEffectChains list during mixing or effects processing
3087        effectChains.clear();
3088    }
3089
3090    releaseWakeLock();
3091
3092    return false;
3093}
3094
3095void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3096{
3097    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3098    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3099                                            this,
3100                                            mSampleRate,
3101                                            mFormat,
3102                                            mChannelMask,
3103                                            frameCount);
3104    if (outputTrack->cblk() != NULL) {
3105        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3106        mOutputTracks.add(outputTrack);
3107        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3108        updateWaitTime();
3109    }
3110}
3111
3112void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3113{
3114    Mutex::Autolock _l(mLock);
3115    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3116        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3117            mOutputTracks[i]->destroy();
3118            mOutputTracks.removeAt(i);
3119            updateWaitTime();
3120            return;
3121        }
3122    }
3123    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3124}
3125
3126void AudioFlinger::DuplicatingThread::updateWaitTime()
3127{
3128    mWaitTimeMs = UINT_MAX;
3129    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3130        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3131        if (strong != NULL) {
3132            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3133            if (waitTimeMs < mWaitTimeMs) {
3134                mWaitTimeMs = waitTimeMs;
3135            }
3136        }
3137    }
3138}
3139
3140
3141bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3142{
3143    for (size_t i = 0; i < outputTracks.size(); i++) {
3144        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3145        if (thread == 0) {
3146            LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3147            return false;
3148        }
3149        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3150        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3151            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3152            return false;
3153        }
3154    }
3155    return true;
3156}
3157
3158uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3159{
3160    return (mWaitTimeMs * 1000) / 2;
3161}
3162
3163// ----------------------------------------------------------------------------
3164
3165// TrackBase constructor must be called with AudioFlinger::mLock held
3166AudioFlinger::ThreadBase::TrackBase::TrackBase(
3167            const wp<ThreadBase>& thread,
3168            const sp<Client>& client,
3169            uint32_t sampleRate,
3170            uint32_t format,
3171            uint32_t channelMask,
3172            int frameCount,
3173            uint32_t flags,
3174            const sp<IMemory>& sharedBuffer,
3175            int sessionId)
3176    :   RefBase(),
3177        mThread(thread),
3178        mClient(client),
3179        mCblk(0),
3180        mFrameCount(0),
3181        mState(IDLE),
3182        mClientTid(-1),
3183        mFormat(format),
3184        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3185        mSessionId(sessionId)
3186{
3187    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3188
3189    // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3190   size_t size = sizeof(audio_track_cblk_t);
3191   uint8_t channelCount = popcount(channelMask);
3192   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3193   if (sharedBuffer == 0) {
3194       size += bufferSize;
3195   }
3196
3197   if (client != NULL) {
3198        mCblkMemory = client->heap()->allocate(size);
3199        if (mCblkMemory != 0) {
3200            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3201            if (mCblk) { // construct the shared structure in-place.
3202                new(mCblk) audio_track_cblk_t();
3203                // clear all buffers
3204                mCblk->frameCount = frameCount;
3205                mCblk->sampleRate = sampleRate;
3206                mChannelCount = channelCount;
3207                mChannelMask = channelMask;
3208                if (sharedBuffer == 0) {
3209                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3210                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3211                    // Force underrun condition to avoid false underrun callback until first data is
3212                    // written to buffer (other flags are cleared)
3213                    mCblk->flags = CBLK_UNDERRUN_ON;
3214                } else {
3215                    mBuffer = sharedBuffer->pointer();
3216                }
3217                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3218            }
3219        } else {
3220            LOGE("not enough memory for AudioTrack size=%u", size);
3221            client->heap()->dump("AudioTrack");
3222            return;
3223        }
3224   } else {
3225       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3226       if (mCblk) { // construct the shared structure in-place.
3227           new(mCblk) audio_track_cblk_t();
3228           // clear all buffers
3229           mCblk->frameCount = frameCount;
3230           mCblk->sampleRate = sampleRate;
3231           mChannelCount = channelCount;
3232           mChannelMask = channelMask;
3233           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3234           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3235           // Force underrun condition to avoid false underrun callback until first data is
3236           // written to buffer (other flags are cleared)
3237           mCblk->flags = CBLK_UNDERRUN_ON;
3238           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3239       }
3240   }
3241}
3242
3243AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3244{
3245    if (mCblk) {
3246        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3247        if (mClient == NULL) {
3248            delete mCblk;
3249        }
3250    }
3251    mCblkMemory.clear();            // and free the shared memory
3252    if (mClient != NULL) {
3253        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3254        mClient.clear();
3255    }
3256}
3257
3258void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3259{
3260    buffer->raw = 0;
3261    mFrameCount = buffer->frameCount;
3262    step();
3263    buffer->frameCount = 0;
3264}
3265
3266bool AudioFlinger::ThreadBase::TrackBase::step() {
3267    bool result;
3268    audio_track_cblk_t* cblk = this->cblk();
3269
3270    result = cblk->stepServer(mFrameCount);
3271    if (!result) {
3272        ALOGV("stepServer failed acquiring cblk mutex");
3273        mFlags |= STEPSERVER_FAILED;
3274    }
3275    return result;
3276}
3277
3278void AudioFlinger::ThreadBase::TrackBase::reset() {
3279    audio_track_cblk_t* cblk = this->cblk();
3280
3281    cblk->user = 0;
3282    cblk->server = 0;
3283    cblk->userBase = 0;
3284    cblk->serverBase = 0;
3285    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3286    ALOGV("TrackBase::reset");
3287}
3288
3289sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3290{
3291    return mCblkMemory;
3292}
3293
3294int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3295    return (int)mCblk->sampleRate;
3296}
3297
3298int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3299    return (const int)mChannelCount;
3300}
3301
3302uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3303    return mChannelMask;
3304}
3305
3306void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3307    audio_track_cblk_t* cblk = this->cblk();
3308    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
3309    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
3310
3311    // Check validity of returned pointer in case the track control block would have been corrupted.
3312    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3313        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
3314        LOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3315                server %d, serverBase %d, user %d, userBase %d",
3316                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3317                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3318        return 0;
3319    }
3320
3321    return bufferStart;
3322}
3323
3324// ----------------------------------------------------------------------------
3325
3326// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3327AudioFlinger::PlaybackThread::Track::Track(
3328            const wp<ThreadBase>& thread,
3329            const sp<Client>& client,
3330            int streamType,
3331            uint32_t sampleRate,
3332            uint32_t format,
3333            uint32_t channelMask,
3334            int frameCount,
3335            const sp<IMemory>& sharedBuffer,
3336            int sessionId)
3337    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3338    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3339    mAuxEffectId(0), mHasVolumeController(false)
3340{
3341    if (mCblk != NULL) {
3342        sp<ThreadBase> baseThread = thread.promote();
3343        if (baseThread != 0) {
3344            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3345            mName = playbackThread->getTrackName_l();
3346            mMainBuffer = playbackThread->mixBuffer();
3347        }
3348        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3349        if (mName < 0) {
3350            LOGE("no more track names available");
3351        }
3352        mVolume[0] = 1.0f;
3353        mVolume[1] = 1.0f;
3354        mStreamType = streamType;
3355        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3356        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3357        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3358    }
3359}
3360
3361AudioFlinger::PlaybackThread::Track::~Track()
3362{
3363    ALOGV("PlaybackThread::Track destructor");
3364    sp<ThreadBase> thread = mThread.promote();
3365    if (thread != 0) {
3366        Mutex::Autolock _l(thread->mLock);
3367        mState = TERMINATED;
3368    }
3369}
3370
3371void AudioFlinger::PlaybackThread::Track::destroy()
3372{
3373    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3374    // by removing it from mTracks vector, so there is a risk that this Tracks's
3375    // desctructor is called. As the destructor needs to lock mLock,
3376    // we must acquire a strong reference on this Track before locking mLock
3377    // here so that the destructor is called only when exiting this function.
3378    // On the other hand, as long as Track::destroy() is only called by
3379    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3380    // this Track with its member mTrack.
3381    sp<Track> keep(this);
3382    { // scope for mLock
3383        sp<ThreadBase> thread = mThread.promote();
3384        if (thread != 0) {
3385            if (!isOutputTrack()) {
3386                if (mState == ACTIVE || mState == RESUMING) {
3387                    AudioSystem::stopOutput(thread->id(),
3388                                            (audio_stream_type_t)mStreamType,
3389                                            mSessionId);
3390
3391                    // to track the speaker usage
3392                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3393                }
3394                AudioSystem::releaseOutput(thread->id());
3395            }
3396            Mutex::Autolock _l(thread->mLock);
3397            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3398            playbackThread->destroyTrack_l(this);
3399        }
3400    }
3401}
3402
3403void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3404{
3405    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",
3406            mName - AudioMixer::TRACK0,
3407            (mClient == NULL) ? getpid() : mClient->pid(),
3408            mStreamType,
3409            mFormat,
3410            mChannelMask,
3411            mSessionId,
3412            mFrameCount,
3413            mState,
3414            mMute,
3415            mFillingUpStatus,
3416            mCblk->sampleRate,
3417            mCblk->volume[0],
3418            mCblk->volume[1],
3419            mCblk->server,
3420            mCblk->user,
3421            (int)mMainBuffer,
3422            (int)mAuxBuffer);
3423}
3424
3425status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3426{
3427     audio_track_cblk_t* cblk = this->cblk();
3428     uint32_t framesReady;
3429     uint32_t framesReq = buffer->frameCount;
3430
3431     // Check if last stepServer failed, try to step now
3432     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3433         if (!step())  goto getNextBuffer_exit;
3434         ALOGV("stepServer recovered");
3435         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3436     }
3437
3438     framesReady = cblk->framesReady();
3439
3440     if (LIKELY(framesReady)) {
3441        uint32_t s = cblk->server;
3442        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3443
3444        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3445        if (framesReq > framesReady) {
3446            framesReq = framesReady;
3447        }
3448        if (s + framesReq > bufferEnd) {
3449            framesReq = bufferEnd - s;
3450        }
3451
3452         buffer->raw = getBuffer(s, framesReq);
3453         if (buffer->raw == 0) goto getNextBuffer_exit;
3454
3455         buffer->frameCount = framesReq;
3456        return NO_ERROR;
3457     }
3458
3459getNextBuffer_exit:
3460     buffer->raw = 0;
3461     buffer->frameCount = 0;
3462     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3463     return NOT_ENOUGH_DATA;
3464}
3465
3466bool AudioFlinger::PlaybackThread::Track::isReady() const {
3467    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3468
3469    if (mCblk->framesReady() >= mCblk->frameCount ||
3470            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3471        mFillingUpStatus = FS_FILLED;
3472        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3473        return true;
3474    }
3475    return false;
3476}
3477
3478status_t AudioFlinger::PlaybackThread::Track::start()
3479{
3480    status_t status = NO_ERROR;
3481    ALOGV("start(%d), calling thread %d session %d",
3482            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3483    sp<ThreadBase> thread = mThread.promote();
3484    if (thread != 0) {
3485        Mutex::Autolock _l(thread->mLock);
3486        int state = mState;
3487        // here the track could be either new, or restarted
3488        // in both cases "unstop" the track
3489        if (mState == PAUSED) {
3490            mState = TrackBase::RESUMING;
3491            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3492        } else {
3493            mState = TrackBase::ACTIVE;
3494            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3495        }
3496
3497        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3498            thread->mLock.unlock();
3499            status = AudioSystem::startOutput(thread->id(),
3500                                              (audio_stream_type_t)mStreamType,
3501                                              mSessionId);
3502            thread->mLock.lock();
3503
3504            // to track the speaker usage
3505            if (status == NO_ERROR) {
3506                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3507            }
3508        }
3509        if (status == NO_ERROR) {
3510            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3511            playbackThread->addTrack_l(this);
3512        } else {
3513            mState = state;
3514        }
3515    } else {
3516        status = BAD_VALUE;
3517    }
3518    return status;
3519}
3520
3521void AudioFlinger::PlaybackThread::Track::stop()
3522{
3523    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3524    sp<ThreadBase> thread = mThread.promote();
3525    if (thread != 0) {
3526        Mutex::Autolock _l(thread->mLock);
3527        int state = mState;
3528        if (mState > STOPPED) {
3529            mState = STOPPED;
3530            // If the track is not active (PAUSED and buffers full), flush buffers
3531            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3532            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3533                reset();
3534            }
3535            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3536        }
3537        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3538            thread->mLock.unlock();
3539            AudioSystem::stopOutput(thread->id(),
3540                                    (audio_stream_type_t)mStreamType,
3541                                    mSessionId);
3542            thread->mLock.lock();
3543
3544            // to track the speaker usage
3545            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3546        }
3547    }
3548}
3549
3550void AudioFlinger::PlaybackThread::Track::pause()
3551{
3552    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3553    sp<ThreadBase> thread = mThread.promote();
3554    if (thread != 0) {
3555        Mutex::Autolock _l(thread->mLock);
3556        if (mState == ACTIVE || mState == RESUMING) {
3557            mState = PAUSING;
3558            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3559            if (!isOutputTrack()) {
3560                thread->mLock.unlock();
3561                AudioSystem::stopOutput(thread->id(),
3562                                        (audio_stream_type_t)mStreamType,
3563                                        mSessionId);
3564                thread->mLock.lock();
3565
3566                // to track the speaker usage
3567                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3568            }
3569        }
3570    }
3571}
3572
3573void AudioFlinger::PlaybackThread::Track::flush()
3574{
3575    ALOGV("flush(%d)", mName);
3576    sp<ThreadBase> thread = mThread.promote();
3577    if (thread != 0) {
3578        Mutex::Autolock _l(thread->mLock);
3579        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3580            return;
3581        }
3582        // No point remaining in PAUSED state after a flush => go to
3583        // STOPPED state
3584        mState = STOPPED;
3585
3586        // do not reset the track if it is still in the process of being stopped or paused.
3587        // this will be done by prepareTracks_l() when the track is stopped.
3588        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3589        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3590            reset();
3591        }
3592    }
3593}
3594
3595void AudioFlinger::PlaybackThread::Track::reset()
3596{
3597    // Do not reset twice to avoid discarding data written just after a flush and before
3598    // the audioflinger thread detects the track is stopped.
3599    if (!mResetDone) {
3600        TrackBase::reset();
3601        // Force underrun condition to avoid false underrun callback until first data is
3602        // written to buffer
3603        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3604        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3605        mFillingUpStatus = FS_FILLING;
3606        mResetDone = true;
3607    }
3608}
3609
3610void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3611{
3612    mMute = muted;
3613}
3614
3615void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3616{
3617    mVolume[0] = left;
3618    mVolume[1] = right;
3619}
3620
3621status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3622{
3623    status_t status = DEAD_OBJECT;
3624    sp<ThreadBase> thread = mThread.promote();
3625    if (thread != 0) {
3626       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3627       status = playbackThread->attachAuxEffect(this, EffectId);
3628    }
3629    return status;
3630}
3631
3632void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3633{
3634    mAuxEffectId = EffectId;
3635    mAuxBuffer = buffer;
3636}
3637
3638// ----------------------------------------------------------------------------
3639
3640// RecordTrack constructor must be called with AudioFlinger::mLock held
3641AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3642            const wp<ThreadBase>& thread,
3643            const sp<Client>& client,
3644            uint32_t sampleRate,
3645            uint32_t format,
3646            uint32_t channelMask,
3647            int frameCount,
3648            uint32_t flags,
3649            int sessionId)
3650    :   TrackBase(thread, client, sampleRate, format,
3651                  channelMask, frameCount, flags, 0, sessionId),
3652        mOverflow(false)
3653{
3654    if (mCblk != NULL) {
3655       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3656       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3657           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3658       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3659           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3660       } else {
3661           mCblk->frameSize = sizeof(int8_t);
3662       }
3663    }
3664}
3665
3666AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3667{
3668    sp<ThreadBase> thread = mThread.promote();
3669    if (thread != 0) {
3670        AudioSystem::releaseInput(thread->id());
3671    }
3672}
3673
3674status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3675{
3676    audio_track_cblk_t* cblk = this->cblk();
3677    uint32_t framesAvail;
3678    uint32_t framesReq = buffer->frameCount;
3679
3680     // Check if last stepServer failed, try to step now
3681    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3682        if (!step()) goto getNextBuffer_exit;
3683        ALOGV("stepServer recovered");
3684        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3685    }
3686
3687    framesAvail = cblk->framesAvailable_l();
3688
3689    if (LIKELY(framesAvail)) {
3690        uint32_t s = cblk->server;
3691        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3692
3693        if (framesReq > framesAvail) {
3694            framesReq = framesAvail;
3695        }
3696        if (s + framesReq > bufferEnd) {
3697            framesReq = bufferEnd - s;
3698        }
3699
3700        buffer->raw = getBuffer(s, framesReq);
3701        if (buffer->raw == 0) goto getNextBuffer_exit;
3702
3703        buffer->frameCount = framesReq;
3704        return NO_ERROR;
3705    }
3706
3707getNextBuffer_exit:
3708    buffer->raw = 0;
3709    buffer->frameCount = 0;
3710    return NOT_ENOUGH_DATA;
3711}
3712
3713status_t AudioFlinger::RecordThread::RecordTrack::start()
3714{
3715    sp<ThreadBase> thread = mThread.promote();
3716    if (thread != 0) {
3717        RecordThread *recordThread = (RecordThread *)thread.get();
3718        return recordThread->start(this);
3719    } else {
3720        return BAD_VALUE;
3721    }
3722}
3723
3724void AudioFlinger::RecordThread::RecordTrack::stop()
3725{
3726    sp<ThreadBase> thread = mThread.promote();
3727    if (thread != 0) {
3728        RecordThread *recordThread = (RecordThread *)thread.get();
3729        recordThread->stop(this);
3730        TrackBase::reset();
3731        // Force overerrun condition to avoid false overrun callback until first data is
3732        // read from buffer
3733        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3734    }
3735}
3736
3737void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3738{
3739    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3740            (mClient == NULL) ? getpid() : mClient->pid(),
3741            mFormat,
3742            mChannelMask,
3743            mSessionId,
3744            mFrameCount,
3745            mState,
3746            mCblk->sampleRate,
3747            mCblk->server,
3748            mCblk->user);
3749}
3750
3751
3752// ----------------------------------------------------------------------------
3753
3754AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3755            const wp<ThreadBase>& thread,
3756            DuplicatingThread *sourceThread,
3757            uint32_t sampleRate,
3758            uint32_t format,
3759            uint32_t channelMask,
3760            int frameCount)
3761    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3762    mActive(false), mSourceThread(sourceThread)
3763{
3764
3765    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3766    if (mCblk != NULL) {
3767        mCblk->flags |= CBLK_DIRECTION_OUT;
3768        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3769        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
3770        mOutBuffer.frameCount = 0;
3771        playbackThread->mTracks.add(this);
3772        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3773                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3774                mCblk, mBuffer, mCblk->buffers,
3775                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3776    } else {
3777        LOGW("Error creating output track on thread %p", playbackThread);
3778    }
3779}
3780
3781AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3782{
3783    clearBufferQueue();
3784}
3785
3786status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3787{
3788    status_t status = Track::start();
3789    if (status != NO_ERROR) {
3790        return status;
3791    }
3792
3793    mActive = true;
3794    mRetryCount = 127;
3795    return status;
3796}
3797
3798void AudioFlinger::PlaybackThread::OutputTrack::stop()
3799{
3800    Track::stop();
3801    clearBufferQueue();
3802    mOutBuffer.frameCount = 0;
3803    mActive = false;
3804}
3805
3806bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3807{
3808    Buffer *pInBuffer;
3809    Buffer inBuffer;
3810    uint32_t channelCount = mChannelCount;
3811    bool outputBufferFull = false;
3812    inBuffer.frameCount = frames;
3813    inBuffer.i16 = data;
3814
3815    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3816
3817    if (!mActive && frames != 0) {
3818        start();
3819        sp<ThreadBase> thread = mThread.promote();
3820        if (thread != 0) {
3821            MixerThread *mixerThread = (MixerThread *)thread.get();
3822            if (mCblk->frameCount > frames){
3823                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3824                    uint32_t startFrames = (mCblk->frameCount - frames);
3825                    pInBuffer = new Buffer;
3826                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3827                    pInBuffer->frameCount = startFrames;
3828                    pInBuffer->i16 = pInBuffer->mBuffer;
3829                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3830                    mBufferQueue.add(pInBuffer);
3831                } else {
3832                    LOGW ("OutputTrack::write() %p no more buffers in queue", this);
3833                }
3834            }
3835        }
3836    }
3837
3838    while (waitTimeLeftMs) {
3839        // First write pending buffers, then new data
3840        if (mBufferQueue.size()) {
3841            pInBuffer = mBufferQueue.itemAt(0);
3842        } else {
3843            pInBuffer = &inBuffer;
3844        }
3845
3846        if (pInBuffer->frameCount == 0) {
3847            break;
3848        }
3849
3850        if (mOutBuffer.frameCount == 0) {
3851            mOutBuffer.frameCount = pInBuffer->frameCount;
3852            nsecs_t startTime = systemTime();
3853            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
3854                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3855                outputBufferFull = true;
3856                break;
3857            }
3858            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3859            if (waitTimeLeftMs >= waitTimeMs) {
3860                waitTimeLeftMs -= waitTimeMs;
3861            } else {
3862                waitTimeLeftMs = 0;
3863            }
3864        }
3865
3866        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3867        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3868        mCblk->stepUser(outFrames);
3869        pInBuffer->frameCount -= outFrames;
3870        pInBuffer->i16 += outFrames * channelCount;
3871        mOutBuffer.frameCount -= outFrames;
3872        mOutBuffer.i16 += outFrames * channelCount;
3873
3874        if (pInBuffer->frameCount == 0) {
3875            if (mBufferQueue.size()) {
3876                mBufferQueue.removeAt(0);
3877                delete [] pInBuffer->mBuffer;
3878                delete pInBuffer;
3879                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3880            } else {
3881                break;
3882            }
3883        }
3884    }
3885
3886    // If we could not write all frames, allocate a buffer and queue it for next time.
3887    if (inBuffer.frameCount) {
3888        sp<ThreadBase> thread = mThread.promote();
3889        if (thread != 0 && !thread->standby()) {
3890            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3891                pInBuffer = new Buffer;
3892                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3893                pInBuffer->frameCount = inBuffer.frameCount;
3894                pInBuffer->i16 = pInBuffer->mBuffer;
3895                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3896                mBufferQueue.add(pInBuffer);
3897                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3898            } else {
3899                LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3900            }
3901        }
3902    }
3903
3904    // Calling write() with a 0 length buffer, means that no more data will be written:
3905    // If no more buffers are pending, fill output track buffer to make sure it is started
3906    // by output mixer.
3907    if (frames == 0 && mBufferQueue.size() == 0) {
3908        if (mCblk->user < mCblk->frameCount) {
3909            frames = mCblk->frameCount - mCblk->user;
3910            pInBuffer = new Buffer;
3911            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3912            pInBuffer->frameCount = frames;
3913            pInBuffer->i16 = pInBuffer->mBuffer;
3914            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3915            mBufferQueue.add(pInBuffer);
3916        } else if (mActive) {
3917            stop();
3918        }
3919    }
3920
3921    return outputBufferFull;
3922}
3923
3924status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3925{
3926    int active;
3927    status_t result;
3928    audio_track_cblk_t* cblk = mCblk;
3929    uint32_t framesReq = buffer->frameCount;
3930
3931//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3932    buffer->frameCount  = 0;
3933
3934    uint32_t framesAvail = cblk->framesAvailable();
3935
3936
3937    if (framesAvail == 0) {
3938        Mutex::Autolock _l(cblk->lock);
3939        goto start_loop_here;
3940        while (framesAvail == 0) {
3941            active = mActive;
3942            if (UNLIKELY(!active)) {
3943                ALOGV("Not active and NO_MORE_BUFFERS");
3944                return AudioTrack::NO_MORE_BUFFERS;
3945            }
3946            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
3947            if (result != NO_ERROR) {
3948                return AudioTrack::NO_MORE_BUFFERS;
3949            }
3950            // read the server count again
3951        start_loop_here:
3952            framesAvail = cblk->framesAvailable_l();
3953        }
3954    }
3955
3956//    if (framesAvail < framesReq) {
3957//        return AudioTrack::NO_MORE_BUFFERS;
3958//    }
3959
3960    if (framesReq > framesAvail) {
3961        framesReq = framesAvail;
3962    }
3963
3964    uint32_t u = cblk->user;
3965    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
3966
3967    if (u + framesReq > bufferEnd) {
3968        framesReq = bufferEnd - u;
3969    }
3970
3971    buffer->frameCount  = framesReq;
3972    buffer->raw         = (void *)cblk->buffer(u);
3973    return NO_ERROR;
3974}
3975
3976
3977void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
3978{
3979    size_t size = mBufferQueue.size();
3980    Buffer *pBuffer;
3981
3982    for (size_t i = 0; i < size; i++) {
3983        pBuffer = mBufferQueue.itemAt(i);
3984        delete [] pBuffer->mBuffer;
3985        delete pBuffer;
3986    }
3987    mBufferQueue.clear();
3988}
3989
3990// ----------------------------------------------------------------------------
3991
3992AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
3993    :   RefBase(),
3994        mAudioFlinger(audioFlinger),
3995        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
3996        mPid(pid)
3997{
3998    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
3999}
4000
4001// Client destructor must be called with AudioFlinger::mLock held
4002AudioFlinger::Client::~Client()
4003{
4004    mAudioFlinger->removeClient_l(mPid);
4005}
4006
4007const sp<MemoryDealer>& AudioFlinger::Client::heap() const
4008{
4009    return mMemoryDealer;
4010}
4011
4012// ----------------------------------------------------------------------------
4013
4014AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4015                                                     const sp<IAudioFlingerClient>& client,
4016                                                     pid_t pid)
4017    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
4018{
4019}
4020
4021AudioFlinger::NotificationClient::~NotificationClient()
4022{
4023    mClient.clear();
4024}
4025
4026void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4027{
4028    sp<NotificationClient> keep(this);
4029    {
4030        mAudioFlinger->removeNotificationClient(mPid);
4031    }
4032}
4033
4034// ----------------------------------------------------------------------------
4035
4036AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4037    : BnAudioTrack(),
4038      mTrack(track)
4039{
4040}
4041
4042AudioFlinger::TrackHandle::~TrackHandle() {
4043    // just stop the track on deletion, associated resources
4044    // will be freed from the main thread once all pending buffers have
4045    // been played. Unless it's not in the active track list, in which
4046    // case we free everything now...
4047    mTrack->destroy();
4048}
4049
4050status_t AudioFlinger::TrackHandle::start() {
4051    return mTrack->start();
4052}
4053
4054void AudioFlinger::TrackHandle::stop() {
4055    mTrack->stop();
4056}
4057
4058void AudioFlinger::TrackHandle::flush() {
4059    mTrack->flush();
4060}
4061
4062void AudioFlinger::TrackHandle::mute(bool e) {
4063    mTrack->mute(e);
4064}
4065
4066void AudioFlinger::TrackHandle::pause() {
4067    mTrack->pause();
4068}
4069
4070void AudioFlinger::TrackHandle::setVolume(float left, float right) {
4071    mTrack->setVolume(left, right);
4072}
4073
4074sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4075    return mTrack->getCblk();
4076}
4077
4078status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4079{
4080    return mTrack->attachAuxEffect(EffectId);
4081}
4082
4083status_t AudioFlinger::TrackHandle::onTransact(
4084    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4085{
4086    return BnAudioTrack::onTransact(code, data, reply, flags);
4087}
4088
4089// ----------------------------------------------------------------------------
4090
4091sp<IAudioRecord> AudioFlinger::openRecord(
4092        pid_t pid,
4093        int input,
4094        uint32_t sampleRate,
4095        uint32_t format,
4096        uint32_t channelMask,
4097        int frameCount,
4098        uint32_t flags,
4099        int *sessionId,
4100        status_t *status)
4101{
4102    sp<RecordThread::RecordTrack> recordTrack;
4103    sp<RecordHandle> recordHandle;
4104    sp<Client> client;
4105    wp<Client> wclient;
4106    status_t lStatus;
4107    RecordThread *thread;
4108    size_t inFrameCount;
4109    int lSessionId;
4110
4111    // check calling permissions
4112    if (!recordingAllowed()) {
4113        lStatus = PERMISSION_DENIED;
4114        goto Exit;
4115    }
4116
4117    // add client to list
4118    { // scope for mLock
4119        Mutex::Autolock _l(mLock);
4120        thread = checkRecordThread_l(input);
4121        if (thread == NULL) {
4122            lStatus = BAD_VALUE;
4123            goto Exit;
4124        }
4125
4126        wclient = mClients.valueFor(pid);
4127        if (wclient != NULL) {
4128            client = wclient.promote();
4129        } else {
4130            client = new Client(this, pid);
4131            mClients.add(pid, client);
4132        }
4133
4134        // If no audio session id is provided, create one here
4135        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4136            lSessionId = *sessionId;
4137        } else {
4138            lSessionId = nextUniqueId();
4139            if (sessionId != NULL) {
4140                *sessionId = lSessionId;
4141            }
4142        }
4143        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4144        recordTrack = thread->createRecordTrack_l(client,
4145                                                sampleRate,
4146                                                format,
4147                                                channelMask,
4148                                                frameCount,
4149                                                flags,
4150                                                lSessionId,
4151                                                &lStatus);
4152    }
4153    if (lStatus != NO_ERROR) {
4154        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4155        // destructor is called by the TrackBase destructor with mLock held
4156        client.clear();
4157        recordTrack.clear();
4158        goto Exit;
4159    }
4160
4161    // return to handle to client
4162    recordHandle = new RecordHandle(recordTrack);
4163    lStatus = NO_ERROR;
4164
4165Exit:
4166    if (status) {
4167        *status = lStatus;
4168    }
4169    return recordHandle;
4170}
4171
4172// ----------------------------------------------------------------------------
4173
4174AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4175    : BnAudioRecord(),
4176    mRecordTrack(recordTrack)
4177{
4178}
4179
4180AudioFlinger::RecordHandle::~RecordHandle() {
4181    stop();
4182}
4183
4184status_t AudioFlinger::RecordHandle::start() {
4185    ALOGV("RecordHandle::start()");
4186    return mRecordTrack->start();
4187}
4188
4189void AudioFlinger::RecordHandle::stop() {
4190    ALOGV("RecordHandle::stop()");
4191    mRecordTrack->stop();
4192}
4193
4194sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4195    return mRecordTrack->getCblk();
4196}
4197
4198status_t AudioFlinger::RecordHandle::onTransact(
4199    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4200{
4201    return BnAudioRecord::onTransact(code, data, reply, flags);
4202}
4203
4204// ----------------------------------------------------------------------------
4205
4206AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4207                                         AudioStreamIn *input,
4208                                         uint32_t sampleRate,
4209                                         uint32_t channels,
4210                                         int id,
4211                                         uint32_t device) :
4212    ThreadBase(audioFlinger, id, device),
4213    mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
4214{
4215    mType = ThreadBase::RECORD;
4216
4217    snprintf(mName, kNameLength, "AudioIn_%d", id);
4218
4219    mReqChannelCount = popcount(channels);
4220    mReqSampleRate = sampleRate;
4221    readInputParameters();
4222}
4223
4224
4225AudioFlinger::RecordThread::~RecordThread()
4226{
4227    delete[] mRsmpInBuffer;
4228    if (mResampler != 0) {
4229        delete mResampler;
4230        delete[] mRsmpOutBuffer;
4231    }
4232}
4233
4234void AudioFlinger::RecordThread::onFirstRef()
4235{
4236    run(mName, PRIORITY_URGENT_AUDIO);
4237}
4238
4239status_t AudioFlinger::RecordThread::readyToRun()
4240{
4241    status_t status = initCheck();
4242    LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4243    return status;
4244}
4245
4246bool AudioFlinger::RecordThread::threadLoop()
4247{
4248    AudioBufferProvider::Buffer buffer;
4249    sp<RecordTrack> activeTrack;
4250    Vector< sp<EffectChain> > effectChains;
4251
4252    nsecs_t lastWarning = 0;
4253
4254    acquireWakeLock();
4255
4256    // start recording
4257    while (!exitPending()) {
4258
4259        processConfigEvents();
4260
4261        { // scope for mLock
4262            Mutex::Autolock _l(mLock);
4263            checkForNewParameters_l();
4264            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4265                if (!mStandby) {
4266                    mInput->stream->common.standby(&mInput->stream->common);
4267                    mStandby = true;
4268                }
4269
4270                if (exitPending()) break;
4271
4272                releaseWakeLock_l();
4273                ALOGV("RecordThread: loop stopping");
4274                // go to sleep
4275                mWaitWorkCV.wait(mLock);
4276                ALOGV("RecordThread: loop starting");
4277                acquireWakeLock_l();
4278                continue;
4279            }
4280            if (mActiveTrack != 0) {
4281                if (mActiveTrack->mState == TrackBase::PAUSING) {
4282                    if (!mStandby) {
4283                        mInput->stream->common.standby(&mInput->stream->common);
4284                        mStandby = true;
4285                    }
4286                    mActiveTrack.clear();
4287                    mStartStopCond.broadcast();
4288                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4289                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4290                        mActiveTrack.clear();
4291                        mStartStopCond.broadcast();
4292                    } else if (mBytesRead != 0) {
4293                        // record start succeeds only if first read from audio input
4294                        // succeeds
4295                        if (mBytesRead > 0) {
4296                            mActiveTrack->mState = TrackBase::ACTIVE;
4297                        } else {
4298                            mActiveTrack.clear();
4299                        }
4300                        mStartStopCond.broadcast();
4301                    }
4302                    mStandby = false;
4303                }
4304            }
4305            lockEffectChains_l(effectChains);
4306        }
4307
4308        if (mActiveTrack != 0) {
4309            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4310                mActiveTrack->mState != TrackBase::RESUMING) {
4311                unlockEffectChains(effectChains);
4312                usleep(kRecordThreadSleepUs);
4313                continue;
4314            }
4315            for (size_t i = 0; i < effectChains.size(); i ++) {
4316                effectChains[i]->process_l();
4317            }
4318
4319            buffer.frameCount = mFrameCount;
4320            if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4321                size_t framesOut = buffer.frameCount;
4322                if (mResampler == 0) {
4323                    // no resampling
4324                    while (framesOut) {
4325                        size_t framesIn = mFrameCount - mRsmpInIndex;
4326                        if (framesIn) {
4327                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4328                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4329                            if (framesIn > framesOut)
4330                                framesIn = framesOut;
4331                            mRsmpInIndex += framesIn;
4332                            framesOut -= framesIn;
4333                            if ((int)mChannelCount == mReqChannelCount ||
4334                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4335                                memcpy(dst, src, framesIn * mFrameSize);
4336                            } else {
4337                                int16_t *src16 = (int16_t *)src;
4338                                int16_t *dst16 = (int16_t *)dst;
4339                                if (mChannelCount == 1) {
4340                                    while (framesIn--) {
4341                                        *dst16++ = *src16;
4342                                        *dst16++ = *src16++;
4343                                    }
4344                                } else {
4345                                    while (framesIn--) {
4346                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4347                                        src16 += 2;
4348                                    }
4349                                }
4350                            }
4351                        }
4352                        if (framesOut && mFrameCount == mRsmpInIndex) {
4353                            if (framesOut == mFrameCount &&
4354                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4355                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4356                                framesOut = 0;
4357                            } else {
4358                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4359                                mRsmpInIndex = 0;
4360                            }
4361                            if (mBytesRead < 0) {
4362                                LOGE("Error reading audio input");
4363                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4364                                    // Force input into standby so that it tries to
4365                                    // recover at next read attempt
4366                                    mInput->stream->common.standby(&mInput->stream->common);
4367                                    usleep(kRecordThreadSleepUs);
4368                                }
4369                                mRsmpInIndex = mFrameCount;
4370                                framesOut = 0;
4371                                buffer.frameCount = 0;
4372                            }
4373                        }
4374                    }
4375                } else {
4376                    // resampling
4377
4378                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4379                    // alter output frame count as if we were expecting stereo samples
4380                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4381                        framesOut >>= 1;
4382                    }
4383                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4384                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4385                    // are 32 bit aligned which should be always true.
4386                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4387                        AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4388                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4389                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4390                        int16_t *dst = buffer.i16;
4391                        while (framesOut--) {
4392                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4393                            src += 2;
4394                        }
4395                    } else {
4396                        AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4397                    }
4398
4399                }
4400                mActiveTrack->releaseBuffer(&buffer);
4401                mActiveTrack->overflow();
4402            }
4403            // client isn't retrieving buffers fast enough
4404            else {
4405                if (!mActiveTrack->setOverflow()) {
4406                    nsecs_t now = systemTime();
4407                    if ((now - lastWarning) > kWarningThrottle) {
4408                        LOGW("RecordThread: buffer overflow");
4409                        lastWarning = now;
4410                    }
4411                }
4412                // Release the processor for a while before asking for a new buffer.
4413                // This will give the application more chance to read from the buffer and
4414                // clear the overflow.
4415                usleep(kRecordThreadSleepUs);
4416            }
4417        }
4418        // enable changes in effect chain
4419        unlockEffectChains(effectChains);
4420        effectChains.clear();
4421    }
4422
4423    if (!mStandby) {
4424        mInput->stream->common.standby(&mInput->stream->common);
4425    }
4426    mActiveTrack.clear();
4427
4428    mStartStopCond.broadcast();
4429
4430    releaseWakeLock();
4431
4432    ALOGV("RecordThread %p exiting", this);
4433    return false;
4434}
4435
4436
4437sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4438        const sp<AudioFlinger::Client>& client,
4439        uint32_t sampleRate,
4440        int format,
4441        int channelMask,
4442        int frameCount,
4443        uint32_t flags,
4444        int sessionId,
4445        status_t *status)
4446{
4447    sp<RecordTrack> track;
4448    status_t lStatus;
4449
4450    lStatus = initCheck();
4451    if (lStatus != NO_ERROR) {
4452        LOGE("Audio driver not initialized.");
4453        goto Exit;
4454    }
4455
4456    { // scope for mLock
4457        Mutex::Autolock _l(mLock);
4458
4459        track = new RecordTrack(this, client, sampleRate,
4460                      format, channelMask, frameCount, flags, sessionId);
4461
4462        if (track->getCblk() == NULL) {
4463            lStatus = NO_MEMORY;
4464            goto Exit;
4465        }
4466
4467        mTrack = track.get();
4468        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4469        bool suspend = audio_is_bluetooth_sco_device(
4470                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4471        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4472        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4473    }
4474    lStatus = NO_ERROR;
4475
4476Exit:
4477    if (status) {
4478        *status = lStatus;
4479    }
4480    return track;
4481}
4482
4483status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4484{
4485    ALOGV("RecordThread::start");
4486    sp <ThreadBase> strongMe = this;
4487    status_t status = NO_ERROR;
4488    {
4489        AutoMutex lock(&mLock);
4490        if (mActiveTrack != 0) {
4491            if (recordTrack != mActiveTrack.get()) {
4492                status = -EBUSY;
4493            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4494                mActiveTrack->mState = TrackBase::ACTIVE;
4495            }
4496            return status;
4497        }
4498
4499        recordTrack->mState = TrackBase::IDLE;
4500        mActiveTrack = recordTrack;
4501        mLock.unlock();
4502        status_t status = AudioSystem::startInput(mId);
4503        mLock.lock();
4504        if (status != NO_ERROR) {
4505            mActiveTrack.clear();
4506            return status;
4507        }
4508        mRsmpInIndex = mFrameCount;
4509        mBytesRead = 0;
4510        if (mResampler != NULL) {
4511            mResampler->reset();
4512        }
4513        mActiveTrack->mState = TrackBase::RESUMING;
4514        // signal thread to start
4515        ALOGV("Signal record thread");
4516        mWaitWorkCV.signal();
4517        // do not wait for mStartStopCond if exiting
4518        if (mExiting) {
4519            mActiveTrack.clear();
4520            status = INVALID_OPERATION;
4521            goto startError;
4522        }
4523        mStartStopCond.wait(mLock);
4524        if (mActiveTrack == 0) {
4525            ALOGV("Record failed to start");
4526            status = BAD_VALUE;
4527            goto startError;
4528        }
4529        ALOGV("Record started OK");
4530        return status;
4531    }
4532startError:
4533    AudioSystem::stopInput(mId);
4534    return status;
4535}
4536
4537void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4538    ALOGV("RecordThread::stop");
4539    sp <ThreadBase> strongMe = this;
4540    {
4541        AutoMutex lock(&mLock);
4542        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4543            mActiveTrack->mState = TrackBase::PAUSING;
4544            // do not wait for mStartStopCond if exiting
4545            if (mExiting) {
4546                return;
4547            }
4548            mStartStopCond.wait(mLock);
4549            // if we have been restarted, recordTrack == mActiveTrack.get() here
4550            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4551                mLock.unlock();
4552                AudioSystem::stopInput(mId);
4553                mLock.lock();
4554                ALOGV("Record stopped OK");
4555            }
4556        }
4557    }
4558}
4559
4560status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4561{
4562    const size_t SIZE = 256;
4563    char buffer[SIZE];
4564    String8 result;
4565    pid_t pid = 0;
4566
4567    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4568    result.append(buffer);
4569
4570    if (mActiveTrack != 0) {
4571        result.append("Active Track:\n");
4572        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4573        mActiveTrack->dump(buffer, SIZE);
4574        result.append(buffer);
4575
4576        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4577        result.append(buffer);
4578        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4579        result.append(buffer);
4580        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
4581        result.append(buffer);
4582        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4583        result.append(buffer);
4584        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4585        result.append(buffer);
4586
4587
4588    } else {
4589        result.append("No record client\n");
4590    }
4591    write(fd, result.string(), result.size());
4592
4593    dumpBase(fd, args);
4594    dumpEffectChains(fd, args);
4595
4596    return NO_ERROR;
4597}
4598
4599status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4600{
4601    size_t framesReq = buffer->frameCount;
4602    size_t framesReady = mFrameCount - mRsmpInIndex;
4603    int channelCount;
4604
4605    if (framesReady == 0) {
4606        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4607        if (mBytesRead < 0) {
4608            LOGE("RecordThread::getNextBuffer() Error reading audio input");
4609            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4610                // Force input into standby so that it tries to
4611                // recover at next read attempt
4612                mInput->stream->common.standby(&mInput->stream->common);
4613                usleep(kRecordThreadSleepUs);
4614            }
4615            buffer->raw = 0;
4616            buffer->frameCount = 0;
4617            return NOT_ENOUGH_DATA;
4618        }
4619        mRsmpInIndex = 0;
4620        framesReady = mFrameCount;
4621    }
4622
4623    if (framesReq > framesReady) {
4624        framesReq = framesReady;
4625    }
4626
4627    if (mChannelCount == 1 && mReqChannelCount == 2) {
4628        channelCount = 1;
4629    } else {
4630        channelCount = 2;
4631    }
4632    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4633    buffer->frameCount = framesReq;
4634    return NO_ERROR;
4635}
4636
4637void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4638{
4639    mRsmpInIndex += buffer->frameCount;
4640    buffer->frameCount = 0;
4641}
4642
4643bool AudioFlinger::RecordThread::checkForNewParameters_l()
4644{
4645    bool reconfig = false;
4646
4647    while (!mNewParameters.isEmpty()) {
4648        status_t status = NO_ERROR;
4649        String8 keyValuePair = mNewParameters[0];
4650        AudioParameter param = AudioParameter(keyValuePair);
4651        int value;
4652        int reqFormat = mFormat;
4653        int reqSamplingRate = mReqSampleRate;
4654        int reqChannelCount = mReqChannelCount;
4655
4656        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4657            reqSamplingRate = value;
4658            reconfig = true;
4659        }
4660        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4661            reqFormat = value;
4662            reconfig = true;
4663        }
4664        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4665            reqChannelCount = popcount(value);
4666            reconfig = true;
4667        }
4668        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4669            // do not accept frame count changes if tracks are open as the track buffer
4670            // size depends on frame count and correct behavior would not be garantied
4671            // if frame count is changed after track creation
4672            if (mActiveTrack != 0) {
4673                status = INVALID_OPERATION;
4674            } else {
4675                reconfig = true;
4676            }
4677        }
4678        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4679            // forward device change to effects that have requested to be
4680            // aware of attached audio device.
4681            for (size_t i = 0; i < mEffectChains.size(); i++) {
4682                mEffectChains[i]->setDevice_l(value);
4683            }
4684            // store input device and output device but do not forward output device to audio HAL.
4685            // Note that status is ignored by the caller for output device
4686            // (see AudioFlinger::setParameters()
4687            if (value & AUDIO_DEVICE_OUT_ALL) {
4688                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4689                status = BAD_VALUE;
4690            } else {
4691                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4692                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4693                if (mTrack != NULL) {
4694                    bool suspend = audio_is_bluetooth_sco_device(
4695                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4696                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4697                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4698                }
4699            }
4700            mDevice |= (uint32_t)value;
4701        }
4702        if (status == NO_ERROR) {
4703            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4704            if (status == INVALID_OPERATION) {
4705               mInput->stream->common.standby(&mInput->stream->common);
4706               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4707            }
4708            if (reconfig) {
4709                if (status == BAD_VALUE &&
4710                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4711                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4712                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4713                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4714                    (reqChannelCount < 3)) {
4715                    status = NO_ERROR;
4716                }
4717                if (status == NO_ERROR) {
4718                    readInputParameters();
4719                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4720                }
4721            }
4722        }
4723
4724        mNewParameters.removeAt(0);
4725
4726        mParamStatus = status;
4727        mParamCond.signal();
4728        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4729        // already timed out waiting for the status and will never signal the condition.
4730        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
4731    }
4732    return reconfig;
4733}
4734
4735String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4736{
4737    char *s;
4738    String8 out_s8 = String8();
4739
4740    Mutex::Autolock _l(mLock);
4741    if (initCheck() != NO_ERROR) {
4742        return out_s8;
4743    }
4744
4745    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4746    out_s8 = String8(s);
4747    free(s);
4748    return out_s8;
4749}
4750
4751void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4752    AudioSystem::OutputDescriptor desc;
4753    void *param2 = 0;
4754
4755    switch (event) {
4756    case AudioSystem::INPUT_OPENED:
4757    case AudioSystem::INPUT_CONFIG_CHANGED:
4758        desc.channels = mChannelMask;
4759        desc.samplingRate = mSampleRate;
4760        desc.format = mFormat;
4761        desc.frameCount = mFrameCount;
4762        desc.latency = 0;
4763        param2 = &desc;
4764        break;
4765
4766    case AudioSystem::INPUT_CLOSED:
4767    default:
4768        break;
4769    }
4770    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4771}
4772
4773void AudioFlinger::RecordThread::readInputParameters()
4774{
4775    if (mRsmpInBuffer) delete mRsmpInBuffer;
4776    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4777    if (mResampler) delete mResampler;
4778    mResampler = 0;
4779
4780    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4781    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4782    mChannelCount = (uint16_t)popcount(mChannelMask);
4783    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4784    mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
4785    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4786    mFrameCount = mInputBytes / mFrameSize;
4787    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4788
4789    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4790    {
4791        int channelCount;
4792         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4793         // stereo to mono post process as the resampler always outputs stereo.
4794        if (mChannelCount == 1 && mReqChannelCount == 2) {
4795            channelCount = 1;
4796        } else {
4797            channelCount = 2;
4798        }
4799        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4800        mResampler->setSampleRate(mSampleRate);
4801        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4802        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4803
4804        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4805        if (mChannelCount == 1 && mReqChannelCount == 1) {
4806            mFrameCount >>= 1;
4807        }
4808
4809    }
4810    mRsmpInIndex = mFrameCount;
4811}
4812
4813unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4814{
4815    Mutex::Autolock _l(mLock);
4816    if (initCheck() != NO_ERROR) {
4817        return 0;
4818    }
4819
4820    return mInput->stream->get_input_frames_lost(mInput->stream);
4821}
4822
4823uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4824{
4825    Mutex::Autolock _l(mLock);
4826    uint32_t result = 0;
4827    if (getEffectChain_l(sessionId) != 0) {
4828        result = EFFECT_SESSION;
4829    }
4830
4831    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4832        result |= TRACK_SESSION;
4833    }
4834
4835    return result;
4836}
4837
4838AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4839{
4840    Mutex::Autolock _l(mLock);
4841    return mTrack;
4842}
4843
4844AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
4845{
4846    Mutex::Autolock _l(mLock);
4847    return mInput;
4848}
4849
4850AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4851{
4852    Mutex::Autolock _l(mLock);
4853    AudioStreamIn *input = mInput;
4854    mInput = NULL;
4855    return input;
4856}
4857
4858// this method must always be called either with ThreadBase mLock held or inside the thread loop
4859audio_stream_t* AudioFlinger::RecordThread::stream()
4860{
4861    if (mInput == NULL) {
4862        return NULL;
4863    }
4864    return &mInput->stream->common;
4865}
4866
4867
4868// ----------------------------------------------------------------------------
4869
4870int AudioFlinger::openOutput(uint32_t *pDevices,
4871                                uint32_t *pSamplingRate,
4872                                uint32_t *pFormat,
4873                                uint32_t *pChannels,
4874                                uint32_t *pLatencyMs,
4875                                uint32_t flags)
4876{
4877    status_t status;
4878    PlaybackThread *thread = NULL;
4879    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4880    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4881    uint32_t format = pFormat ? *pFormat : 0;
4882    uint32_t channels = pChannels ? *pChannels : 0;
4883    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4884    audio_stream_out_t *outStream;
4885    audio_hw_device_t *outHwDev;
4886
4887    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4888            pDevices ? *pDevices : 0,
4889            samplingRate,
4890            format,
4891            channels,
4892            flags);
4893
4894    if (pDevices == NULL || *pDevices == 0) {
4895        return 0;
4896    }
4897
4898    Mutex::Autolock _l(mLock);
4899
4900    outHwDev = findSuitableHwDev_l(*pDevices);
4901    if (outHwDev == NULL)
4902        return 0;
4903
4904    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
4905                                          &channels, &samplingRate, &outStream);
4906    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4907            outStream,
4908            samplingRate,
4909            format,
4910            channels,
4911            status);
4912
4913    mHardwareStatus = AUDIO_HW_IDLE;
4914    if (outStream != NULL) {
4915        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4916        int id = nextUniqueId();
4917
4918        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4919            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4920            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4921            thread = new DirectOutputThread(this, output, id, *pDevices);
4922            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4923        } else {
4924            thread = new MixerThread(this, output, id, *pDevices);
4925            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4926        }
4927        mPlaybackThreads.add(id, thread);
4928
4929        if (pSamplingRate) *pSamplingRate = samplingRate;
4930        if (pFormat) *pFormat = format;
4931        if (pChannels) *pChannels = channels;
4932        if (pLatencyMs) *pLatencyMs = thread->latency();
4933
4934        // notify client processes of the new output creation
4935        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4936        return id;
4937    }
4938
4939    return 0;
4940}
4941
4942int AudioFlinger::openDuplicateOutput(int output1, int output2)
4943{
4944    Mutex::Autolock _l(mLock);
4945    MixerThread *thread1 = checkMixerThread_l(output1);
4946    MixerThread *thread2 = checkMixerThread_l(output2);
4947
4948    if (thread1 == NULL || thread2 == NULL) {
4949        LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
4950        return 0;
4951    }
4952
4953    int id = nextUniqueId();
4954    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
4955    thread->addOutputTrack(thread2);
4956    mPlaybackThreads.add(id, thread);
4957    // notify client processes of the new output creation
4958    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4959    return id;
4960}
4961
4962status_t AudioFlinger::closeOutput(int output)
4963{
4964    // keep strong reference on the playback thread so that
4965    // it is not destroyed while exit() is executed
4966    sp <PlaybackThread> thread;
4967    {
4968        Mutex::Autolock _l(mLock);
4969        thread = checkPlaybackThread_l(output);
4970        if (thread == NULL) {
4971            return BAD_VALUE;
4972        }
4973
4974        ALOGV("closeOutput() %d", output);
4975
4976        if (thread->type() == ThreadBase::MIXER) {
4977            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4978                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
4979                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
4980                    dupThread->removeOutputTrack((MixerThread *)thread.get());
4981                }
4982            }
4983        }
4984        void *param2 = 0;
4985        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
4986        mPlaybackThreads.removeItem(output);
4987    }
4988    thread->exit();
4989
4990    if (thread->type() != ThreadBase::DUPLICATING) {
4991        AudioStreamOut *out = thread->clearOutput();
4992        // from now on thread->mOutput is NULL
4993        out->hwDev->close_output_stream(out->hwDev, out->stream);
4994        delete out;
4995    }
4996    return NO_ERROR;
4997}
4998
4999status_t AudioFlinger::suspendOutput(int output)
5000{
5001    Mutex::Autolock _l(mLock);
5002    PlaybackThread *thread = checkPlaybackThread_l(output);
5003
5004    if (thread == NULL) {
5005        return BAD_VALUE;
5006    }
5007
5008    ALOGV("suspendOutput() %d", output);
5009    thread->suspend();
5010
5011    return NO_ERROR;
5012}
5013
5014status_t AudioFlinger::restoreOutput(int output)
5015{
5016    Mutex::Autolock _l(mLock);
5017    PlaybackThread *thread = checkPlaybackThread_l(output);
5018
5019    if (thread == NULL) {
5020        return BAD_VALUE;
5021    }
5022
5023    ALOGV("restoreOutput() %d", output);
5024
5025    thread->restore();
5026
5027    return NO_ERROR;
5028}
5029
5030int AudioFlinger::openInput(uint32_t *pDevices,
5031                                uint32_t *pSamplingRate,
5032                                uint32_t *pFormat,
5033                                uint32_t *pChannels,
5034                                uint32_t acoustics)
5035{
5036    status_t status;
5037    RecordThread *thread = NULL;
5038    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5039    uint32_t format = pFormat ? *pFormat : 0;
5040    uint32_t channels = pChannels ? *pChannels : 0;
5041    uint32_t reqSamplingRate = samplingRate;
5042    uint32_t reqFormat = format;
5043    uint32_t reqChannels = channels;
5044    audio_stream_in_t *inStream;
5045    audio_hw_device_t *inHwDev;
5046
5047    if (pDevices == NULL || *pDevices == 0) {
5048        return 0;
5049    }
5050
5051    Mutex::Autolock _l(mLock);
5052
5053    inHwDev = findSuitableHwDev_l(*pDevices);
5054    if (inHwDev == NULL)
5055        return 0;
5056
5057    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5058                                        &channels, &samplingRate,
5059                                        (audio_in_acoustics_t)acoustics,
5060                                        &inStream);
5061    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5062            inStream,
5063            samplingRate,
5064            format,
5065            channels,
5066            acoustics,
5067            status);
5068
5069    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5070    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5071    // or stereo to mono conversions on 16 bit PCM inputs.
5072    if (inStream == NULL && status == BAD_VALUE &&
5073        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5074        (samplingRate <= 2 * reqSamplingRate) &&
5075        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5076        ALOGV("openInput() reopening with proposed sampling rate and channels");
5077        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5078                                            &channels, &samplingRate,
5079                                            (audio_in_acoustics_t)acoustics,
5080                                            &inStream);
5081    }
5082
5083    if (inStream != NULL) {
5084        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5085
5086        int id = nextUniqueId();
5087        // Start record thread
5088        // RecorThread require both input and output device indication to forward to audio
5089        // pre processing modules
5090        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5091        thread = new RecordThread(this,
5092                                  input,
5093                                  reqSamplingRate,
5094                                  reqChannels,
5095                                  id,
5096                                  device);
5097        mRecordThreads.add(id, thread);
5098        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5099        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5100        if (pFormat) *pFormat = format;
5101        if (pChannels) *pChannels = reqChannels;
5102
5103        input->stream->common.standby(&input->stream->common);
5104
5105        // notify client processes of the new input creation
5106        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5107        return id;
5108    }
5109
5110    return 0;
5111}
5112
5113status_t AudioFlinger::closeInput(int input)
5114{
5115    // keep strong reference on the record thread so that
5116    // it is not destroyed while exit() is executed
5117    sp <RecordThread> thread;
5118    {
5119        Mutex::Autolock _l(mLock);
5120        thread = checkRecordThread_l(input);
5121        if (thread == NULL) {
5122            return BAD_VALUE;
5123        }
5124
5125        ALOGV("closeInput() %d", input);
5126        void *param2 = 0;
5127        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5128        mRecordThreads.removeItem(input);
5129    }
5130    thread->exit();
5131
5132    AudioStreamIn *in = thread->clearInput();
5133    // from now on thread->mInput is NULL
5134    in->hwDev->close_input_stream(in->hwDev, in->stream);
5135    delete in;
5136
5137    return NO_ERROR;
5138}
5139
5140status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
5141{
5142    Mutex::Autolock _l(mLock);
5143    MixerThread *dstThread = checkMixerThread_l(output);
5144    if (dstThread == NULL) {
5145        LOGW("setStreamOutput() bad output id %d", output);
5146        return BAD_VALUE;
5147    }
5148
5149    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5150    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5151
5152    dstThread->setStreamValid(stream, true);
5153
5154    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5155        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5156        if (thread != dstThread &&
5157            thread->type() != ThreadBase::DIRECT) {
5158            MixerThread *srcThread = (MixerThread *)thread;
5159            srcThread->setStreamValid(stream, false);
5160            srcThread->invalidateTracks(stream);
5161        }
5162    }
5163
5164    return NO_ERROR;
5165}
5166
5167
5168int AudioFlinger::newAudioSessionId()
5169{
5170    return nextUniqueId();
5171}
5172
5173void AudioFlinger::acquireAudioSessionId(int audioSession)
5174{
5175    Mutex::Autolock _l(mLock);
5176    int caller = IPCThreadState::self()->getCallingPid();
5177    ALOGV("acquiring %d from %d", audioSession, caller);
5178    int num = mAudioSessionRefs.size();
5179    for (int i = 0; i< num; i++) {
5180        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5181        if (ref->sessionid == audioSession && ref->pid == caller) {
5182            ref->cnt++;
5183            ALOGV(" incremented refcount to %d", ref->cnt);
5184            return;
5185        }
5186    }
5187    AudioSessionRef *ref = new AudioSessionRef();
5188    ref->sessionid = audioSession;
5189    ref->pid = caller;
5190    ref->cnt = 1;
5191    mAudioSessionRefs.push(ref);
5192    ALOGV(" added new entry for %d", ref->sessionid);
5193}
5194
5195void AudioFlinger::releaseAudioSessionId(int audioSession)
5196{
5197    Mutex::Autolock _l(mLock);
5198    int caller = IPCThreadState::self()->getCallingPid();
5199    ALOGV("releasing %d from %d", audioSession, caller);
5200    int num = mAudioSessionRefs.size();
5201    for (int i = 0; i< num; i++) {
5202        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5203        if (ref->sessionid == audioSession && ref->pid == caller) {
5204            ref->cnt--;
5205            ALOGV(" decremented refcount to %d", ref->cnt);
5206            if (ref->cnt == 0) {
5207                mAudioSessionRefs.removeAt(i);
5208                delete ref;
5209                purgeStaleEffects_l();
5210            }
5211            return;
5212        }
5213    }
5214    LOGW("session id %d not found for pid %d", audioSession, caller);
5215}
5216
5217void AudioFlinger::purgeStaleEffects_l() {
5218
5219    ALOGV("purging stale effects");
5220
5221    Vector< sp<EffectChain> > chains;
5222
5223    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5224        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5225        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5226            sp<EffectChain> ec = t->mEffectChains[j];
5227            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5228                chains.push(ec);
5229            }
5230        }
5231    }
5232    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5233        sp<RecordThread> t = mRecordThreads.valueAt(i);
5234        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5235            sp<EffectChain> ec = t->mEffectChains[j];
5236            chains.push(ec);
5237        }
5238    }
5239
5240    for (size_t i = 0; i < chains.size(); i++) {
5241        sp<EffectChain> ec = chains[i];
5242        int sessionid = ec->sessionId();
5243        sp<ThreadBase> t = ec->mThread.promote();
5244        if (t == 0) {
5245            continue;
5246        }
5247        size_t numsessionrefs = mAudioSessionRefs.size();
5248        bool found = false;
5249        for (size_t k = 0; k < numsessionrefs; k++) {
5250            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5251            if (ref->sessionid == sessionid) {
5252                ALOGV(" session %d still exists for %d with %d refs",
5253                     sessionid, ref->pid, ref->cnt);
5254                found = true;
5255                break;
5256            }
5257        }
5258        if (!found) {
5259            // remove all effects from the chain
5260            while (ec->mEffects.size()) {
5261                sp<EffectModule> effect = ec->mEffects[0];
5262                effect->unPin();
5263                Mutex::Autolock _l (t->mLock);
5264                t->removeEffect_l(effect);
5265                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5266                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5267                    if (handle != 0) {
5268                        handle->mEffect.clear();
5269                        if (handle->mHasControl && handle->mEnabled) {
5270                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5271                        }
5272                    }
5273                }
5274                AudioSystem::unregisterEffect(effect->id());
5275            }
5276        }
5277    }
5278    return;
5279}
5280
5281// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5282AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5283{
5284    PlaybackThread *thread = NULL;
5285    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5286        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5287    }
5288    return thread;
5289}
5290
5291// checkMixerThread_l() must be called with AudioFlinger::mLock held
5292AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5293{
5294    PlaybackThread *thread = checkPlaybackThread_l(output);
5295    if (thread != NULL) {
5296        if (thread->type() == ThreadBase::DIRECT) {
5297            thread = NULL;
5298        }
5299    }
5300    return (MixerThread *)thread;
5301}
5302
5303// checkRecordThread_l() must be called with AudioFlinger::mLock held
5304AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5305{
5306    RecordThread *thread = NULL;
5307    if (mRecordThreads.indexOfKey(input) >= 0) {
5308        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5309    }
5310    return thread;
5311}
5312
5313uint32_t AudioFlinger::nextUniqueId()
5314{
5315    return android_atomic_inc(&mNextUniqueId);
5316}
5317
5318AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5319{
5320    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5321        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5322        AudioStreamOut *output = thread->getOutput();
5323        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5324            return thread;
5325        }
5326    }
5327    return NULL;
5328}
5329
5330uint32_t AudioFlinger::primaryOutputDevice_l()
5331{
5332    PlaybackThread *thread = primaryPlaybackThread_l();
5333
5334    if (thread == NULL) {
5335        return 0;
5336    }
5337
5338    return thread->device();
5339}
5340
5341
5342// ----------------------------------------------------------------------------
5343//  Effect management
5344// ----------------------------------------------------------------------------
5345
5346
5347status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5348{
5349    Mutex::Autolock _l(mLock);
5350    return EffectQueryNumberEffects(numEffects);
5351}
5352
5353status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5354{
5355    Mutex::Autolock _l(mLock);
5356    return EffectQueryEffect(index, descriptor);
5357}
5358
5359status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5360{
5361    Mutex::Autolock _l(mLock);
5362    return EffectGetDescriptor(pUuid, descriptor);
5363}
5364
5365
5366sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5367        effect_descriptor_t *pDesc,
5368        const sp<IEffectClient>& effectClient,
5369        int32_t priority,
5370        int io,
5371        int sessionId,
5372        status_t *status,
5373        int *id,
5374        int *enabled)
5375{
5376    status_t lStatus = NO_ERROR;
5377    sp<EffectHandle> handle;
5378    effect_descriptor_t desc;
5379    sp<Client> client;
5380    wp<Client> wclient;
5381
5382    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5383            pid, effectClient.get(), priority, sessionId, io);
5384
5385    if (pDesc == NULL) {
5386        lStatus = BAD_VALUE;
5387        goto Exit;
5388    }
5389
5390    // check audio settings permission for global effects
5391    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5392        lStatus = PERMISSION_DENIED;
5393        goto Exit;
5394    }
5395
5396    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5397    // that can only be created by audio policy manager (running in same process)
5398    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5399        lStatus = PERMISSION_DENIED;
5400        goto Exit;
5401    }
5402
5403    if (io == 0) {
5404        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5405            // output must be specified by AudioPolicyManager when using session
5406            // AUDIO_SESSION_OUTPUT_STAGE
5407            lStatus = BAD_VALUE;
5408            goto Exit;
5409        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5410            // if the output returned by getOutputForEffect() is removed before we lock the
5411            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5412            // and we will exit safely
5413            io = AudioSystem::getOutputForEffect(&desc);
5414        }
5415    }
5416
5417    {
5418        Mutex::Autolock _l(mLock);
5419
5420
5421        if (!EffectIsNullUuid(&pDesc->uuid)) {
5422            // if uuid is specified, request effect descriptor
5423            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5424            if (lStatus < 0) {
5425                LOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5426                goto Exit;
5427            }
5428        } else {
5429            // if uuid is not specified, look for an available implementation
5430            // of the required type in effect factory
5431            if (EffectIsNullUuid(&pDesc->type)) {
5432                LOGW("createEffect() no effect type");
5433                lStatus = BAD_VALUE;
5434                goto Exit;
5435            }
5436            uint32_t numEffects = 0;
5437            effect_descriptor_t d;
5438            d.flags = 0; // prevent compiler warning
5439            bool found = false;
5440
5441            lStatus = EffectQueryNumberEffects(&numEffects);
5442            if (lStatus < 0) {
5443                LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5444                goto Exit;
5445            }
5446            for (uint32_t i = 0; i < numEffects; i++) {
5447                lStatus = EffectQueryEffect(i, &desc);
5448                if (lStatus < 0) {
5449                    LOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5450                    continue;
5451                }
5452                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5453                    // If matching type found save effect descriptor. If the session is
5454                    // 0 and the effect is not auxiliary, continue enumeration in case
5455                    // an auxiliary version of this effect type is available
5456                    found = true;
5457                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5458                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5459                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5460                        break;
5461                    }
5462                }
5463            }
5464            if (!found) {
5465                lStatus = BAD_VALUE;
5466                LOGW("createEffect() effect not found");
5467                goto Exit;
5468            }
5469            // For same effect type, chose auxiliary version over insert version if
5470            // connect to output mix (Compliance to OpenSL ES)
5471            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5472                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5473                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5474            }
5475        }
5476
5477        // Do not allow auxiliary effects on a session different from 0 (output mix)
5478        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5479             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5480            lStatus = INVALID_OPERATION;
5481            goto Exit;
5482        }
5483
5484        // check recording permission for visualizer
5485        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5486            !recordingAllowed()) {
5487            lStatus = PERMISSION_DENIED;
5488            goto Exit;
5489        }
5490
5491        // return effect descriptor
5492        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5493
5494        // If output is not specified try to find a matching audio session ID in one of the
5495        // output threads.
5496        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5497        // because of code checking output when entering the function.
5498        // Note: io is never 0 when creating an effect on an input
5499        if (io == 0) {
5500             // look for the thread where the specified audio session is present
5501            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5502                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5503                    io = mPlaybackThreads.keyAt(i);
5504                    break;
5505                }
5506            }
5507            if (io == 0) {
5508               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5509                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5510                       io = mRecordThreads.keyAt(i);
5511                       break;
5512                   }
5513               }
5514            }
5515            // If no output thread contains the requested session ID, default to
5516            // first output. The effect chain will be moved to the correct output
5517            // thread when a track with the same session ID is created
5518            if (io == 0 && mPlaybackThreads.size()) {
5519                io = mPlaybackThreads.keyAt(0);
5520            }
5521            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5522        }
5523        ThreadBase *thread = checkRecordThread_l(io);
5524        if (thread == NULL) {
5525            thread = checkPlaybackThread_l(io);
5526            if (thread == NULL) {
5527                LOGE("createEffect() unknown output thread");
5528                lStatus = BAD_VALUE;
5529                goto Exit;
5530            }
5531        }
5532
5533        wclient = mClients.valueFor(pid);
5534
5535        if (wclient != NULL) {
5536            client = wclient.promote();
5537        } else {
5538            client = new Client(this, pid);
5539            mClients.add(pid, client);
5540        }
5541
5542        // create effect on selected output thread
5543        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5544                &desc, enabled, &lStatus);
5545        if (handle != 0 && id != NULL) {
5546            *id = handle->id();
5547        }
5548    }
5549
5550Exit:
5551    if(status) {
5552        *status = lStatus;
5553    }
5554    return handle;
5555}
5556
5557status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5558{
5559    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5560            sessionId, srcOutput, dstOutput);
5561    Mutex::Autolock _l(mLock);
5562    if (srcOutput == dstOutput) {
5563        LOGW("moveEffects() same dst and src outputs %d", dstOutput);
5564        return NO_ERROR;
5565    }
5566    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5567    if (srcThread == NULL) {
5568        LOGW("moveEffects() bad srcOutput %d", srcOutput);
5569        return BAD_VALUE;
5570    }
5571    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5572    if (dstThread == NULL) {
5573        LOGW("moveEffects() bad dstOutput %d", dstOutput);
5574        return BAD_VALUE;
5575    }
5576
5577    Mutex::Autolock _dl(dstThread->mLock);
5578    Mutex::Autolock _sl(srcThread->mLock);
5579    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5580
5581    return NO_ERROR;
5582}
5583
5584// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5585status_t AudioFlinger::moveEffectChain_l(int sessionId,
5586                                   AudioFlinger::PlaybackThread *srcThread,
5587                                   AudioFlinger::PlaybackThread *dstThread,
5588                                   bool reRegister)
5589{
5590    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5591            sessionId, srcThread, dstThread);
5592
5593    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5594    if (chain == 0) {
5595        LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5596                sessionId, srcThread);
5597        return INVALID_OPERATION;
5598    }
5599
5600    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5601    // so that a new chain is created with correct parameters when first effect is added. This is
5602    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5603    // removed.
5604    srcThread->removeEffectChain_l(chain);
5605
5606    // transfer all effects one by one so that new effect chain is created on new thread with
5607    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5608    int dstOutput = dstThread->id();
5609    sp<EffectChain> dstChain;
5610    uint32_t strategy = 0; // prevent compiler warning
5611    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5612    while (effect != 0) {
5613        srcThread->removeEffect_l(effect);
5614        dstThread->addEffect_l(effect);
5615        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5616        if (effect->state() == EffectModule::ACTIVE ||
5617                effect->state() == EffectModule::STOPPING) {
5618            effect->start();
5619        }
5620        // if the move request is not received from audio policy manager, the effect must be
5621        // re-registered with the new strategy and output
5622        if (dstChain == 0) {
5623            dstChain = effect->chain().promote();
5624            if (dstChain == 0) {
5625                LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5626                srcThread->addEffect_l(effect);
5627                return NO_INIT;
5628            }
5629            strategy = dstChain->strategy();
5630        }
5631        if (reRegister) {
5632            AudioSystem::unregisterEffect(effect->id());
5633            AudioSystem::registerEffect(&effect->desc(),
5634                                        dstOutput,
5635                                        strategy,
5636                                        sessionId,
5637                                        effect->id());
5638        }
5639        effect = chain->getEffectFromId_l(0);
5640    }
5641
5642    return NO_ERROR;
5643}
5644
5645
5646// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5647sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5648        const sp<AudioFlinger::Client>& client,
5649        const sp<IEffectClient>& effectClient,
5650        int32_t priority,
5651        int sessionId,
5652        effect_descriptor_t *desc,
5653        int *enabled,
5654        status_t *status
5655        )
5656{
5657    sp<EffectModule> effect;
5658    sp<EffectHandle> handle;
5659    status_t lStatus;
5660    sp<EffectChain> chain;
5661    bool chainCreated = false;
5662    bool effectCreated = false;
5663    bool effectRegistered = false;
5664
5665    lStatus = initCheck();
5666    if (lStatus != NO_ERROR) {
5667        LOGW("createEffect_l() Audio driver not initialized.");
5668        goto Exit;
5669    }
5670
5671    // Do not allow effects with session ID 0 on direct output or duplicating threads
5672    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5673    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5674        LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5675                desc->name, sessionId);
5676        lStatus = BAD_VALUE;
5677        goto Exit;
5678    }
5679    // Only Pre processor effects are allowed on input threads and only on input threads
5680    if ((mType == RECORD &&
5681            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5682            (mType != RECORD &&
5683                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5684        LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5685                desc->name, desc->flags, mType);
5686        lStatus = BAD_VALUE;
5687        goto Exit;
5688    }
5689
5690    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5691
5692    { // scope for mLock
5693        Mutex::Autolock _l(mLock);
5694
5695        // check for existing effect chain with the requested audio session
5696        chain = getEffectChain_l(sessionId);
5697        if (chain == 0) {
5698            // create a new chain for this session
5699            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5700            chain = new EffectChain(this, sessionId);
5701            addEffectChain_l(chain);
5702            chain->setStrategy(getStrategyForSession_l(sessionId));
5703            chainCreated = true;
5704        } else {
5705            effect = chain->getEffectFromDesc_l(desc);
5706        }
5707
5708        ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
5709
5710        if (effect == 0) {
5711            int id = mAudioFlinger->nextUniqueId();
5712            // Check CPU and memory usage
5713            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5714            if (lStatus != NO_ERROR) {
5715                goto Exit;
5716            }
5717            effectRegistered = true;
5718            // create a new effect module if none present in the chain
5719            effect = new EffectModule(this, chain, desc, id, sessionId);
5720            lStatus = effect->status();
5721            if (lStatus != NO_ERROR) {
5722                goto Exit;
5723            }
5724            lStatus = chain->addEffect_l(effect);
5725            if (lStatus != NO_ERROR) {
5726                goto Exit;
5727            }
5728            effectCreated = true;
5729
5730            effect->setDevice(mDevice);
5731            effect->setMode(mAudioFlinger->getMode());
5732        }
5733        // create effect handle and connect it to effect module
5734        handle = new EffectHandle(effect, client, effectClient, priority);
5735        lStatus = effect->addHandle(handle);
5736        if (enabled) {
5737            *enabled = (int)effect->isEnabled();
5738        }
5739    }
5740
5741Exit:
5742    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5743        Mutex::Autolock _l(mLock);
5744        if (effectCreated) {
5745            chain->removeEffect_l(effect);
5746        }
5747        if (effectRegistered) {
5748            AudioSystem::unregisterEffect(effect->id());
5749        }
5750        if (chainCreated) {
5751            removeEffectChain_l(chain);
5752        }
5753        handle.clear();
5754    }
5755
5756    if(status) {
5757        *status = lStatus;
5758    }
5759    return handle;
5760}
5761
5762sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5763{
5764    sp<EffectModule> effect;
5765
5766    sp<EffectChain> chain = getEffectChain_l(sessionId);
5767    if (chain != 0) {
5768        effect = chain->getEffectFromId_l(effectId);
5769    }
5770    return effect;
5771}
5772
5773// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5774// PlaybackThread::mLock held
5775status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5776{
5777    // check for existing effect chain with the requested audio session
5778    int sessionId = effect->sessionId();
5779    sp<EffectChain> chain = getEffectChain_l(sessionId);
5780    bool chainCreated = false;
5781
5782    if (chain == 0) {
5783        // create a new chain for this session
5784        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5785        chain = new EffectChain(this, sessionId);
5786        addEffectChain_l(chain);
5787        chain->setStrategy(getStrategyForSession_l(sessionId));
5788        chainCreated = true;
5789    }
5790    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5791
5792    if (chain->getEffectFromId_l(effect->id()) != 0) {
5793        LOGW("addEffect_l() %p effect %s already present in chain %p",
5794                this, effect->desc().name, chain.get());
5795        return BAD_VALUE;
5796    }
5797
5798    status_t status = chain->addEffect_l(effect);
5799    if (status != NO_ERROR) {
5800        if (chainCreated) {
5801            removeEffectChain_l(chain);
5802        }
5803        return status;
5804    }
5805
5806    effect->setDevice(mDevice);
5807    effect->setMode(mAudioFlinger->getMode());
5808    return NO_ERROR;
5809}
5810
5811void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5812
5813    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5814    effect_descriptor_t desc = effect->desc();
5815    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5816        detachAuxEffect_l(effect->id());
5817    }
5818
5819    sp<EffectChain> chain = effect->chain().promote();
5820    if (chain != 0) {
5821        // remove effect chain if removing last effect
5822        if (chain->removeEffect_l(effect) == 0) {
5823            removeEffectChain_l(chain);
5824        }
5825    } else {
5826        LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5827    }
5828}
5829
5830void AudioFlinger::ThreadBase::lockEffectChains_l(
5831        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5832{
5833    effectChains = mEffectChains;
5834    for (size_t i = 0; i < mEffectChains.size(); i++) {
5835        mEffectChains[i]->lock();
5836    }
5837}
5838
5839void AudioFlinger::ThreadBase::unlockEffectChains(
5840        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5841{
5842    for (size_t i = 0; i < effectChains.size(); i++) {
5843        effectChains[i]->unlock();
5844    }
5845}
5846
5847sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5848{
5849    Mutex::Autolock _l(mLock);
5850    return getEffectChain_l(sessionId);
5851}
5852
5853sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5854{
5855    sp<EffectChain> chain;
5856
5857    size_t size = mEffectChains.size();
5858    for (size_t i = 0; i < size; i++) {
5859        if (mEffectChains[i]->sessionId() == sessionId) {
5860            chain = mEffectChains[i];
5861            break;
5862        }
5863    }
5864    return chain;
5865}
5866
5867void AudioFlinger::ThreadBase::setMode(uint32_t mode)
5868{
5869    Mutex::Autolock _l(mLock);
5870    size_t size = mEffectChains.size();
5871    for (size_t i = 0; i < size; i++) {
5872        mEffectChains[i]->setMode_l(mode);
5873    }
5874}
5875
5876void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5877                                                    const wp<EffectHandle>& handle,
5878                                                    bool unpiniflast) {
5879
5880    Mutex::Autolock _l(mLock);
5881    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5882    // delete the effect module if removing last handle on it
5883    if (effect->removeHandle(handle) == 0) {
5884        if (!effect->isPinned() || unpiniflast) {
5885            removeEffect_l(effect);
5886            AudioSystem::unregisterEffect(effect->id());
5887        }
5888    }
5889}
5890
5891status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5892{
5893    int session = chain->sessionId();
5894    int16_t *buffer = mMixBuffer;
5895    bool ownsBuffer = false;
5896
5897    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5898    if (session > 0) {
5899        // Only one effect chain can be present in direct output thread and it uses
5900        // the mix buffer as input
5901        if (mType != DIRECT) {
5902            size_t numSamples = mFrameCount * mChannelCount;
5903            buffer = new int16_t[numSamples];
5904            memset(buffer, 0, numSamples * sizeof(int16_t));
5905            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5906            ownsBuffer = true;
5907        }
5908
5909        // Attach all tracks with same session ID to this chain.
5910        for (size_t i = 0; i < mTracks.size(); ++i) {
5911            sp<Track> track = mTracks[i];
5912            if (session == track->sessionId()) {
5913                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5914                track->setMainBuffer(buffer);
5915                chain->incTrackCnt();
5916            }
5917        }
5918
5919        // indicate all active tracks in the chain
5920        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5921            sp<Track> track = mActiveTracks[i].promote();
5922            if (track == 0) continue;
5923            if (session == track->sessionId()) {
5924                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5925                chain->incActiveTrackCnt();
5926            }
5927        }
5928    }
5929
5930    chain->setInBuffer(buffer, ownsBuffer);
5931    chain->setOutBuffer(mMixBuffer);
5932    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5933    // chains list in order to be processed last as it contains output stage effects
5934    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5935    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5936    // after track specific effects and before output stage
5937    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5938    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5939    // Effect chain for other sessions are inserted at beginning of effect
5940    // chains list to be processed before output mix effects. Relative order between other
5941    // sessions is not important
5942    size_t size = mEffectChains.size();
5943    size_t i = 0;
5944    for (i = 0; i < size; i++) {
5945        if (mEffectChains[i]->sessionId() < session) break;
5946    }
5947    mEffectChains.insertAt(chain, i);
5948    checkSuspendOnAddEffectChain_l(chain);
5949
5950    return NO_ERROR;
5951}
5952
5953size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
5954{
5955    int session = chain->sessionId();
5956
5957    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
5958
5959    for (size_t i = 0; i < mEffectChains.size(); i++) {
5960        if (chain == mEffectChains[i]) {
5961            mEffectChains.removeAt(i);
5962            // detach all active tracks from the chain
5963            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5964                sp<Track> track = mActiveTracks[i].promote();
5965                if (track == 0) continue;
5966                if (session == track->sessionId()) {
5967                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
5968                            chain.get(), session);
5969                    chain->decActiveTrackCnt();
5970                }
5971            }
5972
5973            // detach all tracks with same session ID from this chain
5974            for (size_t i = 0; i < mTracks.size(); ++i) {
5975                sp<Track> track = mTracks[i];
5976                if (session == track->sessionId()) {
5977                    track->setMainBuffer(mMixBuffer);
5978                    chain->decTrackCnt();
5979                }
5980            }
5981            break;
5982        }
5983    }
5984    return mEffectChains.size();
5985}
5986
5987status_t AudioFlinger::PlaybackThread::attachAuxEffect(
5988        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
5989{
5990    Mutex::Autolock _l(mLock);
5991    return attachAuxEffect_l(track, EffectId);
5992}
5993
5994status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
5995        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
5996{
5997    status_t status = NO_ERROR;
5998
5999    if (EffectId == 0) {
6000        track->setAuxBuffer(0, NULL);
6001    } else {
6002        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6003        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6004        if (effect != 0) {
6005            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6006                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6007            } else {
6008                status = INVALID_OPERATION;
6009            }
6010        } else {
6011            status = BAD_VALUE;
6012        }
6013    }
6014    return status;
6015}
6016
6017void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6018{
6019     for (size_t i = 0; i < mTracks.size(); ++i) {
6020        sp<Track> track = mTracks[i];
6021        if (track->auxEffectId() == effectId) {
6022            attachAuxEffect_l(track, 0);
6023        }
6024    }
6025}
6026
6027status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6028{
6029    // only one chain per input thread
6030    if (mEffectChains.size() != 0) {
6031        return INVALID_OPERATION;
6032    }
6033    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6034
6035    chain->setInBuffer(NULL);
6036    chain->setOutBuffer(NULL);
6037
6038    checkSuspendOnAddEffectChain_l(chain);
6039
6040    mEffectChains.add(chain);
6041
6042    return NO_ERROR;
6043}
6044
6045size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6046{
6047    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6048    LOGW_IF(mEffectChains.size() != 1,
6049            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6050            chain.get(), mEffectChains.size(), this);
6051    if (mEffectChains.size() == 1) {
6052        mEffectChains.removeAt(0);
6053    }
6054    return 0;
6055}
6056
6057// ----------------------------------------------------------------------------
6058//  EffectModule implementation
6059// ----------------------------------------------------------------------------
6060
6061#undef LOG_TAG
6062#define LOG_TAG "AudioFlinger::EffectModule"
6063
6064AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6065                                        const wp<AudioFlinger::EffectChain>& chain,
6066                                        effect_descriptor_t *desc,
6067                                        int id,
6068                                        int sessionId)
6069    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6070      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6071{
6072    ALOGV("Constructor %p", this);
6073    int lStatus;
6074    sp<ThreadBase> thread = mThread.promote();
6075    if (thread == 0) {
6076        return;
6077    }
6078
6079    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6080
6081    // create effect engine from effect factory
6082    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6083
6084    if (mStatus != NO_ERROR) {
6085        return;
6086    }
6087    lStatus = init();
6088    if (lStatus < 0) {
6089        mStatus = lStatus;
6090        goto Error;
6091    }
6092
6093    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6094        mPinned = true;
6095    }
6096    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6097    return;
6098Error:
6099    EffectRelease(mEffectInterface);
6100    mEffectInterface = NULL;
6101    ALOGV("Constructor Error %d", mStatus);
6102}
6103
6104AudioFlinger::EffectModule::~EffectModule()
6105{
6106    ALOGV("Destructor %p", this);
6107    if (mEffectInterface != NULL) {
6108        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6109                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6110            sp<ThreadBase> thread = mThread.promote();
6111            if (thread != 0) {
6112                audio_stream_t *stream = thread->stream();
6113                if (stream != NULL) {
6114                    stream->remove_audio_effect(stream, mEffectInterface);
6115                }
6116            }
6117        }
6118        // release effect engine
6119        EffectRelease(mEffectInterface);
6120    }
6121}
6122
6123status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6124{
6125    status_t status;
6126
6127    Mutex::Autolock _l(mLock);
6128    // First handle in mHandles has highest priority and controls the effect module
6129    int priority = handle->priority();
6130    size_t size = mHandles.size();
6131    sp<EffectHandle> h;
6132    size_t i;
6133    for (i = 0; i < size; i++) {
6134        h = mHandles[i].promote();
6135        if (h == 0) continue;
6136        if (h->priority() <= priority) break;
6137    }
6138    // if inserted in first place, move effect control from previous owner to this handle
6139    if (i == 0) {
6140        bool enabled = false;
6141        if (h != 0) {
6142            enabled = h->enabled();
6143            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6144        }
6145        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6146        status = NO_ERROR;
6147    } else {
6148        status = ALREADY_EXISTS;
6149    }
6150    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6151    mHandles.insertAt(handle, i);
6152    return status;
6153}
6154
6155size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6156{
6157    Mutex::Autolock _l(mLock);
6158    size_t size = mHandles.size();
6159    size_t i;
6160    for (i = 0; i < size; i++) {
6161        if (mHandles[i] == handle) break;
6162    }
6163    if (i == size) {
6164        return size;
6165    }
6166    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6167
6168    bool enabled = false;
6169    EffectHandle *hdl = handle.unsafe_get();
6170    if (hdl) {
6171        ALOGV("removeHandle() unsafe_get OK");
6172        enabled = hdl->enabled();
6173    }
6174    mHandles.removeAt(i);
6175    size = mHandles.size();
6176    // if removed from first place, move effect control from this handle to next in line
6177    if (i == 0 && size != 0) {
6178        sp<EffectHandle> h = mHandles[0].promote();
6179        if (h != 0) {
6180            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6181        }
6182    }
6183
6184    // Prevent calls to process() and other functions on effect interface from now on.
6185    // The effect engine will be released by the destructor when the last strong reference on
6186    // this object is released which can happen after next process is called.
6187    if (size == 0 && !mPinned) {
6188        mState = DESTROYED;
6189    }
6190
6191    return size;
6192}
6193
6194sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6195{
6196    Mutex::Autolock _l(mLock);
6197    sp<EffectHandle> handle;
6198    if (mHandles.size() != 0) {
6199        handle = mHandles[0].promote();
6200    }
6201    return handle;
6202}
6203
6204void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6205{
6206    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6207    // keep a strong reference on this EffectModule to avoid calling the
6208    // destructor before we exit
6209    sp<EffectModule> keep(this);
6210    {
6211        sp<ThreadBase> thread = mThread.promote();
6212        if (thread != 0) {
6213            thread->disconnectEffect(keep, handle, unpiniflast);
6214        }
6215    }
6216}
6217
6218void AudioFlinger::EffectModule::updateState() {
6219    Mutex::Autolock _l(mLock);
6220
6221    switch (mState) {
6222    case RESTART:
6223        reset_l();
6224        // FALL THROUGH
6225
6226    case STARTING:
6227        // clear auxiliary effect input buffer for next accumulation
6228        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6229            memset(mConfig.inputCfg.buffer.raw,
6230                   0,
6231                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6232        }
6233        start_l();
6234        mState = ACTIVE;
6235        break;
6236    case STOPPING:
6237        stop_l();
6238        mDisableWaitCnt = mMaxDisableWaitCnt;
6239        mState = STOPPED;
6240        break;
6241    case STOPPED:
6242        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6243        // turn off sequence.
6244        if (--mDisableWaitCnt == 0) {
6245            reset_l();
6246            mState = IDLE;
6247        }
6248        break;
6249    default: //IDLE , ACTIVE, DESTROYED
6250        break;
6251    }
6252}
6253
6254void AudioFlinger::EffectModule::process()
6255{
6256    Mutex::Autolock _l(mLock);
6257
6258    if (mState == DESTROYED || mEffectInterface == NULL ||
6259            mConfig.inputCfg.buffer.raw == NULL ||
6260            mConfig.outputCfg.buffer.raw == NULL) {
6261        return;
6262    }
6263
6264    if (isProcessEnabled()) {
6265        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6266        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6267            AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
6268                                        mConfig.inputCfg.buffer.s32,
6269                                        mConfig.inputCfg.buffer.frameCount/2);
6270        }
6271
6272        // do the actual processing in the effect engine
6273        int ret = (*mEffectInterface)->process(mEffectInterface,
6274                                               &mConfig.inputCfg.buffer,
6275                                               &mConfig.outputCfg.buffer);
6276
6277        // force transition to IDLE state when engine is ready
6278        if (mState == STOPPED && ret == -ENODATA) {
6279            mDisableWaitCnt = 1;
6280        }
6281
6282        // clear auxiliary effect input buffer for next accumulation
6283        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6284            memset(mConfig.inputCfg.buffer.raw, 0,
6285                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6286        }
6287    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6288                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6289        // If an insert effect is idle and input buffer is different from output buffer,
6290        // accumulate input onto output
6291        sp<EffectChain> chain = mChain.promote();
6292        if (chain != 0 && chain->activeTrackCnt() != 0) {
6293            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6294            int16_t *in = mConfig.inputCfg.buffer.s16;
6295            int16_t *out = mConfig.outputCfg.buffer.s16;
6296            for (size_t i = 0; i < frameCnt; i++) {
6297                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6298            }
6299        }
6300    }
6301}
6302
6303void AudioFlinger::EffectModule::reset_l()
6304{
6305    if (mEffectInterface == NULL) {
6306        return;
6307    }
6308    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6309}
6310
6311status_t AudioFlinger::EffectModule::configure()
6312{
6313    uint32_t channels;
6314    if (mEffectInterface == NULL) {
6315        return NO_INIT;
6316    }
6317
6318    sp<ThreadBase> thread = mThread.promote();
6319    if (thread == 0) {
6320        return DEAD_OBJECT;
6321    }
6322
6323    // TODO: handle configuration of effects replacing track process
6324    if (thread->channelCount() == 1) {
6325        channels = AUDIO_CHANNEL_OUT_MONO;
6326    } else {
6327        channels = AUDIO_CHANNEL_OUT_STEREO;
6328    }
6329
6330    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6331        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6332    } else {
6333        mConfig.inputCfg.channels = channels;
6334    }
6335    mConfig.outputCfg.channels = channels;
6336    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6337    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6338    mConfig.inputCfg.samplingRate = thread->sampleRate();
6339    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6340    mConfig.inputCfg.bufferProvider.cookie = NULL;
6341    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6342    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6343    mConfig.outputCfg.bufferProvider.cookie = NULL;
6344    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6345    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6346    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6347    // Insert effect:
6348    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6349    // always overwrites output buffer: input buffer == output buffer
6350    // - in other sessions:
6351    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6352    //      other effect: overwrites output buffer: input buffer == output buffer
6353    // Auxiliary effect:
6354    //      accumulates in output buffer: input buffer != output buffer
6355    // Therefore: accumulate <=> input buffer != output buffer
6356    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6357        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6358    } else {
6359        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6360    }
6361    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6362    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6363    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6364    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6365
6366    ALOGV("configure() %p thread %p buffer %p framecount %d",
6367            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6368
6369    status_t cmdStatus;
6370    uint32_t size = sizeof(int);
6371    status_t status = (*mEffectInterface)->command(mEffectInterface,
6372                                                   EFFECT_CMD_CONFIGURE,
6373                                                   sizeof(effect_config_t),
6374                                                   &mConfig,
6375                                                   &size,
6376                                                   &cmdStatus);
6377    if (status == 0) {
6378        status = cmdStatus;
6379    }
6380
6381    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6382            (1000 * mConfig.outputCfg.buffer.frameCount);
6383
6384    return status;
6385}
6386
6387status_t AudioFlinger::EffectModule::init()
6388{
6389    Mutex::Autolock _l(mLock);
6390    if (mEffectInterface == NULL) {
6391        return NO_INIT;
6392    }
6393    status_t cmdStatus;
6394    uint32_t size = sizeof(status_t);
6395    status_t status = (*mEffectInterface)->command(mEffectInterface,
6396                                                   EFFECT_CMD_INIT,
6397                                                   0,
6398                                                   NULL,
6399                                                   &size,
6400                                                   &cmdStatus);
6401    if (status == 0) {
6402        status = cmdStatus;
6403    }
6404    return status;
6405}
6406
6407status_t AudioFlinger::EffectModule::start()
6408{
6409    Mutex::Autolock _l(mLock);
6410    return start_l();
6411}
6412
6413status_t AudioFlinger::EffectModule::start_l()
6414{
6415    if (mEffectInterface == NULL) {
6416        return NO_INIT;
6417    }
6418    status_t cmdStatus;
6419    uint32_t size = sizeof(status_t);
6420    status_t status = (*mEffectInterface)->command(mEffectInterface,
6421                                                   EFFECT_CMD_ENABLE,
6422                                                   0,
6423                                                   NULL,
6424                                                   &size,
6425                                                   &cmdStatus);
6426    if (status == 0) {
6427        status = cmdStatus;
6428    }
6429    if (status == 0 &&
6430            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6431             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6432        sp<ThreadBase> thread = mThread.promote();
6433        if (thread != 0) {
6434            audio_stream_t *stream = thread->stream();
6435            if (stream != NULL) {
6436                stream->add_audio_effect(stream, mEffectInterface);
6437            }
6438        }
6439    }
6440    return status;
6441}
6442
6443status_t AudioFlinger::EffectModule::stop()
6444{
6445    Mutex::Autolock _l(mLock);
6446    return stop_l();
6447}
6448
6449status_t AudioFlinger::EffectModule::stop_l()
6450{
6451    if (mEffectInterface == NULL) {
6452        return NO_INIT;
6453    }
6454    status_t cmdStatus;
6455    uint32_t size = sizeof(status_t);
6456    status_t status = (*mEffectInterface)->command(mEffectInterface,
6457                                                   EFFECT_CMD_DISABLE,
6458                                                   0,
6459                                                   NULL,
6460                                                   &size,
6461                                                   &cmdStatus);
6462    if (status == 0) {
6463        status = cmdStatus;
6464    }
6465    if (status == 0 &&
6466            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6467             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6468        sp<ThreadBase> thread = mThread.promote();
6469        if (thread != 0) {
6470            audio_stream_t *stream = thread->stream();
6471            if (stream != NULL) {
6472                stream->remove_audio_effect(stream, mEffectInterface);
6473            }
6474        }
6475    }
6476    return status;
6477}
6478
6479status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6480                                             uint32_t cmdSize,
6481                                             void *pCmdData,
6482                                             uint32_t *replySize,
6483                                             void *pReplyData)
6484{
6485    Mutex::Autolock _l(mLock);
6486//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6487
6488    if (mState == DESTROYED || mEffectInterface == NULL) {
6489        return NO_INIT;
6490    }
6491    status_t status = (*mEffectInterface)->command(mEffectInterface,
6492                                                   cmdCode,
6493                                                   cmdSize,
6494                                                   pCmdData,
6495                                                   replySize,
6496                                                   pReplyData);
6497    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6498        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6499        for (size_t i = 1; i < mHandles.size(); i++) {
6500            sp<EffectHandle> h = mHandles[i].promote();
6501            if (h != 0) {
6502                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6503            }
6504        }
6505    }
6506    return status;
6507}
6508
6509status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6510{
6511
6512    Mutex::Autolock _l(mLock);
6513    ALOGV("setEnabled %p enabled %d", this, enabled);
6514
6515    if (enabled != isEnabled()) {
6516        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6517        if (enabled && status != NO_ERROR) {
6518            return status;
6519        }
6520
6521        switch (mState) {
6522        // going from disabled to enabled
6523        case IDLE:
6524            mState = STARTING;
6525            break;
6526        case STOPPED:
6527            mState = RESTART;
6528            break;
6529        case STOPPING:
6530            mState = ACTIVE;
6531            break;
6532
6533        // going from enabled to disabled
6534        case RESTART:
6535            mState = STOPPED;
6536            break;
6537        case STARTING:
6538            mState = IDLE;
6539            break;
6540        case ACTIVE:
6541            mState = STOPPING;
6542            break;
6543        case DESTROYED:
6544            return NO_ERROR; // simply ignore as we are being destroyed
6545        }
6546        for (size_t i = 1; i < mHandles.size(); i++) {
6547            sp<EffectHandle> h = mHandles[i].promote();
6548            if (h != 0) {
6549                h->setEnabled(enabled);
6550            }
6551        }
6552    }
6553    return NO_ERROR;
6554}
6555
6556bool AudioFlinger::EffectModule::isEnabled()
6557{
6558    switch (mState) {
6559    case RESTART:
6560    case STARTING:
6561    case ACTIVE:
6562        return true;
6563    case IDLE:
6564    case STOPPING:
6565    case STOPPED:
6566    case DESTROYED:
6567    default:
6568        return false;
6569    }
6570}
6571
6572bool AudioFlinger::EffectModule::isProcessEnabled()
6573{
6574    switch (mState) {
6575    case RESTART:
6576    case ACTIVE:
6577    case STOPPING:
6578    case STOPPED:
6579        return true;
6580    case IDLE:
6581    case STARTING:
6582    case DESTROYED:
6583    default:
6584        return false;
6585    }
6586}
6587
6588status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6589{
6590    Mutex::Autolock _l(mLock);
6591    status_t status = NO_ERROR;
6592
6593    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6594    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6595    if (isProcessEnabled() &&
6596            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6597            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6598        status_t cmdStatus;
6599        uint32_t volume[2];
6600        uint32_t *pVolume = NULL;
6601        uint32_t size = sizeof(volume);
6602        volume[0] = *left;
6603        volume[1] = *right;
6604        if (controller) {
6605            pVolume = volume;
6606        }
6607        status = (*mEffectInterface)->command(mEffectInterface,
6608                                              EFFECT_CMD_SET_VOLUME,
6609                                              size,
6610                                              volume,
6611                                              &size,
6612                                              pVolume);
6613        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6614            *left = volume[0];
6615            *right = volume[1];
6616        }
6617    }
6618    return status;
6619}
6620
6621status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6622{
6623    Mutex::Autolock _l(mLock);
6624    status_t status = NO_ERROR;
6625    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6626        // audio pre processing modules on RecordThread can receive both output and
6627        // input device indication in the same call
6628        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6629        if (dev) {
6630            status_t cmdStatus;
6631            uint32_t size = sizeof(status_t);
6632
6633            status = (*mEffectInterface)->command(mEffectInterface,
6634                                                  EFFECT_CMD_SET_DEVICE,
6635                                                  sizeof(uint32_t),
6636                                                  &dev,
6637                                                  &size,
6638                                                  &cmdStatus);
6639            if (status == NO_ERROR) {
6640                status = cmdStatus;
6641            }
6642        }
6643        dev = device & AUDIO_DEVICE_IN_ALL;
6644        if (dev) {
6645            status_t cmdStatus;
6646            uint32_t size = sizeof(status_t);
6647
6648            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6649                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6650                                                  sizeof(uint32_t),
6651                                                  &dev,
6652                                                  &size,
6653                                                  &cmdStatus);
6654            if (status2 == NO_ERROR) {
6655                status2 = cmdStatus;
6656            }
6657            if (status == NO_ERROR) {
6658                status = status2;
6659            }
6660        }
6661    }
6662    return status;
6663}
6664
6665status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
6666{
6667    Mutex::Autolock _l(mLock);
6668    status_t status = NO_ERROR;
6669    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6670        status_t cmdStatus;
6671        uint32_t size = sizeof(status_t);
6672        status = (*mEffectInterface)->command(mEffectInterface,
6673                                              EFFECT_CMD_SET_AUDIO_MODE,
6674                                              sizeof(int),
6675                                              &mode,
6676                                              &size,
6677                                              &cmdStatus);
6678        if (status == NO_ERROR) {
6679            status = cmdStatus;
6680        }
6681    }
6682    return status;
6683}
6684
6685void AudioFlinger::EffectModule::setSuspended(bool suspended)
6686{
6687    Mutex::Autolock _l(mLock);
6688    mSuspended = suspended;
6689}
6690bool AudioFlinger::EffectModule::suspended()
6691{
6692    Mutex::Autolock _l(mLock);
6693    return mSuspended;
6694}
6695
6696status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6697{
6698    const size_t SIZE = 256;
6699    char buffer[SIZE];
6700    String8 result;
6701
6702    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6703    result.append(buffer);
6704
6705    bool locked = tryLock(mLock);
6706    // failed to lock - AudioFlinger is probably deadlocked
6707    if (!locked) {
6708        result.append("\t\tCould not lock Fx mutex:\n");
6709    }
6710
6711    result.append("\t\tSession Status State Engine:\n");
6712    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6713            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6714    result.append(buffer);
6715
6716    result.append("\t\tDescriptor:\n");
6717    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6718            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6719            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6720            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6721    result.append(buffer);
6722    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6723                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6724                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6725                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6726    result.append(buffer);
6727    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6728            mDescriptor.apiVersion,
6729            mDescriptor.flags);
6730    result.append(buffer);
6731    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6732            mDescriptor.name);
6733    result.append(buffer);
6734    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6735            mDescriptor.implementor);
6736    result.append(buffer);
6737
6738    result.append("\t\t- Input configuration:\n");
6739    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6740    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6741            (uint32_t)mConfig.inputCfg.buffer.raw,
6742            mConfig.inputCfg.buffer.frameCount,
6743            mConfig.inputCfg.samplingRate,
6744            mConfig.inputCfg.channels,
6745            mConfig.inputCfg.format);
6746    result.append(buffer);
6747
6748    result.append("\t\t- Output configuration:\n");
6749    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6750    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6751            (uint32_t)mConfig.outputCfg.buffer.raw,
6752            mConfig.outputCfg.buffer.frameCount,
6753            mConfig.outputCfg.samplingRate,
6754            mConfig.outputCfg.channels,
6755            mConfig.outputCfg.format);
6756    result.append(buffer);
6757
6758    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6759    result.append(buffer);
6760    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6761    for (size_t i = 0; i < mHandles.size(); ++i) {
6762        sp<EffectHandle> handle = mHandles[i].promote();
6763        if (handle != 0) {
6764            handle->dump(buffer, SIZE);
6765            result.append(buffer);
6766        }
6767    }
6768
6769    result.append("\n");
6770
6771    write(fd, result.string(), result.length());
6772
6773    if (locked) {
6774        mLock.unlock();
6775    }
6776
6777    return NO_ERROR;
6778}
6779
6780// ----------------------------------------------------------------------------
6781//  EffectHandle implementation
6782// ----------------------------------------------------------------------------
6783
6784#undef LOG_TAG
6785#define LOG_TAG "AudioFlinger::EffectHandle"
6786
6787AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6788                                        const sp<AudioFlinger::Client>& client,
6789                                        const sp<IEffectClient>& effectClient,
6790                                        int32_t priority)
6791    : BnEffect(),
6792    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6793    mPriority(priority), mHasControl(false), mEnabled(false)
6794{
6795    ALOGV("constructor %p", this);
6796
6797    if (client == 0) {
6798        return;
6799    }
6800    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6801    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6802    if (mCblkMemory != 0) {
6803        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6804
6805        if (mCblk) {
6806            new(mCblk) effect_param_cblk_t();
6807            mBuffer = (uint8_t *)mCblk + bufOffset;
6808         }
6809    } else {
6810        LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6811        return;
6812    }
6813}
6814
6815AudioFlinger::EffectHandle::~EffectHandle()
6816{
6817    ALOGV("Destructor %p", this);
6818    disconnect(false);
6819    ALOGV("Destructor DONE %p", this);
6820}
6821
6822status_t AudioFlinger::EffectHandle::enable()
6823{
6824    ALOGV("enable %p", this);
6825    if (!mHasControl) return INVALID_OPERATION;
6826    if (mEffect == 0) return DEAD_OBJECT;
6827
6828    if (mEnabled) {
6829        return NO_ERROR;
6830    }
6831
6832    mEnabled = true;
6833
6834    sp<ThreadBase> thread = mEffect->thread().promote();
6835    if (thread != 0) {
6836        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6837    }
6838
6839    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6840    if (mEffect->suspended()) {
6841        return NO_ERROR;
6842    }
6843
6844    status_t status = mEffect->setEnabled(true);
6845    if (status != NO_ERROR) {
6846        if (thread != 0) {
6847            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6848        }
6849        mEnabled = false;
6850    }
6851    return status;
6852}
6853
6854status_t AudioFlinger::EffectHandle::disable()
6855{
6856    ALOGV("disable %p", this);
6857    if (!mHasControl) return INVALID_OPERATION;
6858    if (mEffect == 0) return DEAD_OBJECT;
6859
6860    if (!mEnabled) {
6861        return NO_ERROR;
6862    }
6863    mEnabled = false;
6864
6865    if (mEffect->suspended()) {
6866        return NO_ERROR;
6867    }
6868
6869    status_t status = mEffect->setEnabled(false);
6870
6871    sp<ThreadBase> thread = mEffect->thread().promote();
6872    if (thread != 0) {
6873        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6874    }
6875
6876    return status;
6877}
6878
6879void AudioFlinger::EffectHandle::disconnect()
6880{
6881    disconnect(true);
6882}
6883
6884void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6885{
6886    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6887    if (mEffect == 0) {
6888        return;
6889    }
6890    mEffect->disconnect(this, unpiniflast);
6891
6892    if (mHasControl && mEnabled) {
6893        sp<ThreadBase> thread = mEffect->thread().promote();
6894        if (thread != 0) {
6895            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6896        }
6897    }
6898
6899    // release sp on module => module destructor can be called now
6900    mEffect.clear();
6901    if (mClient != 0) {
6902        if (mCblk) {
6903            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6904        }
6905        mCblkMemory.clear();            // and free the shared memory
6906        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6907        mClient.clear();
6908    }
6909}
6910
6911status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6912                                             uint32_t cmdSize,
6913                                             void *pCmdData,
6914                                             uint32_t *replySize,
6915                                             void *pReplyData)
6916{
6917//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6918//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6919
6920    // only get parameter command is permitted for applications not controlling the effect
6921    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6922        return INVALID_OPERATION;
6923    }
6924    if (mEffect == 0) return DEAD_OBJECT;
6925    if (mClient == 0) return INVALID_OPERATION;
6926
6927    // handle commands that are not forwarded transparently to effect engine
6928    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6929        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6930        // no risk to block the whole media server process or mixer threads is we are stuck here
6931        Mutex::Autolock _l(mCblk->lock);
6932        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6933            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6934            mCblk->serverIndex = 0;
6935            mCblk->clientIndex = 0;
6936            return BAD_VALUE;
6937        }
6938        status_t status = NO_ERROR;
6939        while (mCblk->serverIndex < mCblk->clientIndex) {
6940            int reply;
6941            uint32_t rsize = sizeof(int);
6942            int *p = (int *)(mBuffer + mCblk->serverIndex);
6943            int size = *p++;
6944            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
6945                LOGW("command(): invalid parameter block size");
6946                break;
6947            }
6948            effect_param_t *param = (effect_param_t *)p;
6949            if (param->psize == 0 || param->vsize == 0) {
6950                LOGW("command(): null parameter or value size");
6951                mCblk->serverIndex += size;
6952                continue;
6953            }
6954            uint32_t psize = sizeof(effect_param_t) +
6955                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
6956                             param->vsize;
6957            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
6958                                            psize,
6959                                            p,
6960                                            &rsize,
6961                                            &reply);
6962            // stop at first error encountered
6963            if (ret != NO_ERROR) {
6964                status = ret;
6965                *(int *)pReplyData = reply;
6966                break;
6967            } else if (reply != NO_ERROR) {
6968                *(int *)pReplyData = reply;
6969                break;
6970            }
6971            mCblk->serverIndex += size;
6972        }
6973        mCblk->serverIndex = 0;
6974        mCblk->clientIndex = 0;
6975        return status;
6976    } else if (cmdCode == EFFECT_CMD_ENABLE) {
6977        *(int *)pReplyData = NO_ERROR;
6978        return enable();
6979    } else if (cmdCode == EFFECT_CMD_DISABLE) {
6980        *(int *)pReplyData = NO_ERROR;
6981        return disable();
6982    }
6983
6984    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
6985}
6986
6987sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
6988    return mCblkMemory;
6989}
6990
6991void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
6992{
6993    ALOGV("setControl %p control %d", this, hasControl);
6994
6995    mHasControl = hasControl;
6996    mEnabled = enabled;
6997
6998    if (signal && mEffectClient != 0) {
6999        mEffectClient->controlStatusChanged(hasControl);
7000    }
7001}
7002
7003void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7004                                                 uint32_t cmdSize,
7005                                                 void *pCmdData,
7006                                                 uint32_t replySize,
7007                                                 void *pReplyData)
7008{
7009    if (mEffectClient != 0) {
7010        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7011    }
7012}
7013
7014
7015
7016void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7017{
7018    if (mEffectClient != 0) {
7019        mEffectClient->enableStatusChanged(enabled);
7020    }
7021}
7022
7023status_t AudioFlinger::EffectHandle::onTransact(
7024    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7025{
7026    return BnEffect::onTransact(code, data, reply, flags);
7027}
7028
7029
7030void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7031{
7032    bool locked = mCblk ? tryLock(mCblk->lock) : false;
7033
7034    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7035            (mClient == NULL) ? getpid() : mClient->pid(),
7036            mPriority,
7037            mHasControl,
7038            !locked,
7039            mCblk ? mCblk->clientIndex : 0,
7040            mCblk ? mCblk->serverIndex : 0
7041            );
7042
7043    if (locked) {
7044        mCblk->lock.unlock();
7045    }
7046}
7047
7048#undef LOG_TAG
7049#define LOG_TAG "AudioFlinger::EffectChain"
7050
7051AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7052                                        int sessionId)
7053    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7054      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7055      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7056{
7057    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7058    sp<ThreadBase> thread = mThread.promote();
7059    if (thread == 0) {
7060        return;
7061    }
7062    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7063                                    thread->frameCount();
7064}
7065
7066AudioFlinger::EffectChain::~EffectChain()
7067{
7068    if (mOwnInBuffer) {
7069        delete mInBuffer;
7070    }
7071
7072}
7073
7074// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7075sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7076{
7077    sp<EffectModule> effect;
7078    size_t size = mEffects.size();
7079
7080    for (size_t i = 0; i < size; i++) {
7081        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7082            effect = mEffects[i];
7083            break;
7084        }
7085    }
7086    return effect;
7087}
7088
7089// getEffectFromId_l() must be called with ThreadBase::mLock held
7090sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7091{
7092    sp<EffectModule> effect;
7093    size_t size = mEffects.size();
7094
7095    for (size_t i = 0; i < size; i++) {
7096        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7097        if (id == 0 || mEffects[i]->id() == id) {
7098            effect = mEffects[i];
7099            break;
7100        }
7101    }
7102    return effect;
7103}
7104
7105// getEffectFromType_l() must be called with ThreadBase::mLock held
7106sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7107        const effect_uuid_t *type)
7108{
7109    sp<EffectModule> effect;
7110    size_t size = mEffects.size();
7111
7112    for (size_t i = 0; i < size; i++) {
7113        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7114            effect = mEffects[i];
7115            break;
7116        }
7117    }
7118    return effect;
7119}
7120
7121// Must be called with EffectChain::mLock locked
7122void AudioFlinger::EffectChain::process_l()
7123{
7124    sp<ThreadBase> thread = mThread.promote();
7125    if (thread == 0) {
7126        LOGW("process_l(): cannot promote mixer thread");
7127        return;
7128    }
7129    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7130            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7131    // always process effects unless no more tracks are on the session and the effect tail
7132    // has been rendered
7133    bool doProcess = true;
7134    if (!isGlobalSession) {
7135        bool tracksOnSession = (trackCnt() != 0);
7136
7137        if (!tracksOnSession && mTailBufferCount == 0) {
7138            doProcess = false;
7139        }
7140
7141        if (activeTrackCnt() == 0) {
7142            // if no track is active and the effect tail has not been rendered,
7143            // the input buffer must be cleared here as the mixer process will not do it
7144            if (tracksOnSession || mTailBufferCount > 0) {
7145                size_t numSamples = thread->frameCount() * thread->channelCount();
7146                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7147                if (mTailBufferCount > 0) {
7148                    mTailBufferCount--;
7149                }
7150            }
7151        }
7152    }
7153
7154    size_t size = mEffects.size();
7155    if (doProcess) {
7156        for (size_t i = 0; i < size; i++) {
7157            mEffects[i]->process();
7158        }
7159    }
7160    for (size_t i = 0; i < size; i++) {
7161        mEffects[i]->updateState();
7162    }
7163}
7164
7165// addEffect_l() must be called with PlaybackThread::mLock held
7166status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7167{
7168    effect_descriptor_t desc = effect->desc();
7169    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7170
7171    Mutex::Autolock _l(mLock);
7172    effect->setChain(this);
7173    sp<ThreadBase> thread = mThread.promote();
7174    if (thread == 0) {
7175        return NO_INIT;
7176    }
7177    effect->setThread(thread);
7178
7179    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7180        // Auxiliary effects are inserted at the beginning of mEffects vector as
7181        // they are processed first and accumulated in chain input buffer
7182        mEffects.insertAt(effect, 0);
7183
7184        // the input buffer for auxiliary effect contains mono samples in
7185        // 32 bit format. This is to avoid saturation in AudoMixer
7186        // accumulation stage. Saturation is done in EffectModule::process() before
7187        // calling the process in effect engine
7188        size_t numSamples = thread->frameCount();
7189        int32_t *buffer = new int32_t[numSamples];
7190        memset(buffer, 0, numSamples * sizeof(int32_t));
7191        effect->setInBuffer((int16_t *)buffer);
7192        // auxiliary effects output samples to chain input buffer for further processing
7193        // by insert effects
7194        effect->setOutBuffer(mInBuffer);
7195    } else {
7196        // Insert effects are inserted at the end of mEffects vector as they are processed
7197        //  after track and auxiliary effects.
7198        // Insert effect order as a function of indicated preference:
7199        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7200        //  another effect is present
7201        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7202        //  last effect claiming first position
7203        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7204        //  first effect claiming last position
7205        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7206        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7207        // already present
7208
7209        int size = (int)mEffects.size();
7210        int idx_insert = size;
7211        int idx_insert_first = -1;
7212        int idx_insert_last = -1;
7213
7214        for (int i = 0; i < size; i++) {
7215            effect_descriptor_t d = mEffects[i]->desc();
7216            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7217            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7218            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7219                // check invalid effect chaining combinations
7220                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7221                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7222                    LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7223                    return INVALID_OPERATION;
7224                }
7225                // remember position of first insert effect and by default
7226                // select this as insert position for new effect
7227                if (idx_insert == size) {
7228                    idx_insert = i;
7229                }
7230                // remember position of last insert effect claiming
7231                // first position
7232                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7233                    idx_insert_first = i;
7234                }
7235                // remember position of first insert effect claiming
7236                // last position
7237                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7238                    idx_insert_last == -1) {
7239                    idx_insert_last = i;
7240                }
7241            }
7242        }
7243
7244        // modify idx_insert from first position if needed
7245        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7246            if (idx_insert_last != -1) {
7247                idx_insert = idx_insert_last;
7248            } else {
7249                idx_insert = size;
7250            }
7251        } else {
7252            if (idx_insert_first != -1) {
7253                idx_insert = idx_insert_first + 1;
7254            }
7255        }
7256
7257        // always read samples from chain input buffer
7258        effect->setInBuffer(mInBuffer);
7259
7260        // if last effect in the chain, output samples to chain
7261        // output buffer, otherwise to chain input buffer
7262        if (idx_insert == size) {
7263            if (idx_insert != 0) {
7264                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7265                mEffects[idx_insert-1]->configure();
7266            }
7267            effect->setOutBuffer(mOutBuffer);
7268        } else {
7269            effect->setOutBuffer(mInBuffer);
7270        }
7271        mEffects.insertAt(effect, idx_insert);
7272
7273        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7274    }
7275    effect->configure();
7276    return NO_ERROR;
7277}
7278
7279// removeEffect_l() must be called with PlaybackThread::mLock held
7280size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7281{
7282    Mutex::Autolock _l(mLock);
7283    int size = (int)mEffects.size();
7284    int i;
7285    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7286
7287    for (i = 0; i < size; i++) {
7288        if (effect == mEffects[i]) {
7289            // calling stop here will remove pre-processing effect from the audio HAL.
7290            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7291            // the middle of a read from audio HAL
7292            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7293                    mEffects[i]->state() == EffectModule::STOPPING) {
7294                mEffects[i]->stop();
7295            }
7296            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7297                delete[] effect->inBuffer();
7298            } else {
7299                if (i == size - 1 && i != 0) {
7300                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7301                    mEffects[i - 1]->configure();
7302                }
7303            }
7304            mEffects.removeAt(i);
7305            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7306            break;
7307        }
7308    }
7309
7310    return mEffects.size();
7311}
7312
7313// setDevice_l() must be called with PlaybackThread::mLock held
7314void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7315{
7316    size_t size = mEffects.size();
7317    for (size_t i = 0; i < size; i++) {
7318        mEffects[i]->setDevice(device);
7319    }
7320}
7321
7322// setMode_l() must be called with PlaybackThread::mLock held
7323void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
7324{
7325    size_t size = mEffects.size();
7326    for (size_t i = 0; i < size; i++) {
7327        mEffects[i]->setMode(mode);
7328    }
7329}
7330
7331// setVolume_l() must be called with PlaybackThread::mLock held
7332bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7333{
7334    uint32_t newLeft = *left;
7335    uint32_t newRight = *right;
7336    bool hasControl = false;
7337    int ctrlIdx = -1;
7338    size_t size = mEffects.size();
7339
7340    // first update volume controller
7341    for (size_t i = size; i > 0; i--) {
7342        if (mEffects[i - 1]->isProcessEnabled() &&
7343            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7344            ctrlIdx = i - 1;
7345            hasControl = true;
7346            break;
7347        }
7348    }
7349
7350    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7351        if (hasControl) {
7352            *left = mNewLeftVolume;
7353            *right = mNewRightVolume;
7354        }
7355        return hasControl;
7356    }
7357
7358    mVolumeCtrlIdx = ctrlIdx;
7359    mLeftVolume = newLeft;
7360    mRightVolume = newRight;
7361
7362    // second get volume update from volume controller
7363    if (ctrlIdx >= 0) {
7364        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7365        mNewLeftVolume = newLeft;
7366        mNewRightVolume = newRight;
7367    }
7368    // then indicate volume to all other effects in chain.
7369    // Pass altered volume to effects before volume controller
7370    // and requested volume to effects after controller
7371    uint32_t lVol = newLeft;
7372    uint32_t rVol = newRight;
7373
7374    for (size_t i = 0; i < size; i++) {
7375        if ((int)i == ctrlIdx) continue;
7376        // this also works for ctrlIdx == -1 when there is no volume controller
7377        if ((int)i > ctrlIdx) {
7378            lVol = *left;
7379            rVol = *right;
7380        }
7381        mEffects[i]->setVolume(&lVol, &rVol, false);
7382    }
7383    *left = newLeft;
7384    *right = newRight;
7385
7386    return hasControl;
7387}
7388
7389status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7390{
7391    const size_t SIZE = 256;
7392    char buffer[SIZE];
7393    String8 result;
7394
7395    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7396    result.append(buffer);
7397
7398    bool locked = tryLock(mLock);
7399    // failed to lock - AudioFlinger is probably deadlocked
7400    if (!locked) {
7401        result.append("\tCould not lock mutex:\n");
7402    }
7403
7404    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7405    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7406            mEffects.size(),
7407            (uint32_t)mInBuffer,
7408            (uint32_t)mOutBuffer,
7409            mActiveTrackCnt);
7410    result.append(buffer);
7411    write(fd, result.string(), result.size());
7412
7413    for (size_t i = 0; i < mEffects.size(); ++i) {
7414        sp<EffectModule> effect = mEffects[i];
7415        if (effect != 0) {
7416            effect->dump(fd, args);
7417        }
7418    }
7419
7420    if (locked) {
7421        mLock.unlock();
7422    }
7423
7424    return NO_ERROR;
7425}
7426
7427// must be called with ThreadBase::mLock held
7428void AudioFlinger::EffectChain::setEffectSuspended_l(
7429        const effect_uuid_t *type, bool suspend)
7430{
7431    sp<SuspendedEffectDesc> desc;
7432    // use effect type UUID timelow as key as there is no real risk of identical
7433    // timeLow fields among effect type UUIDs.
7434    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7435    if (suspend) {
7436        if (index >= 0) {
7437            desc = mSuspendedEffects.valueAt(index);
7438        } else {
7439            desc = new SuspendedEffectDesc();
7440            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7441            mSuspendedEffects.add(type->timeLow, desc);
7442            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7443        }
7444        if (desc->mRefCount++ == 0) {
7445            sp<EffectModule> effect = getEffectIfEnabled(type);
7446            if (effect != 0) {
7447                desc->mEffect = effect;
7448                effect->setSuspended(true);
7449                effect->setEnabled(false);
7450            }
7451        }
7452    } else {
7453        if (index < 0) {
7454            return;
7455        }
7456        desc = mSuspendedEffects.valueAt(index);
7457        if (desc->mRefCount <= 0) {
7458            LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7459            desc->mRefCount = 1;
7460        }
7461        if (--desc->mRefCount == 0) {
7462            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7463            if (desc->mEffect != 0) {
7464                sp<EffectModule> effect = desc->mEffect.promote();
7465                if (effect != 0) {
7466                    effect->setSuspended(false);
7467                    sp<EffectHandle> handle = effect->controlHandle();
7468                    if (handle != 0) {
7469                        effect->setEnabled(handle->enabled());
7470                    }
7471                }
7472                desc->mEffect.clear();
7473            }
7474            mSuspendedEffects.removeItemsAt(index);
7475        }
7476    }
7477}
7478
7479// must be called with ThreadBase::mLock held
7480void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7481{
7482    sp<SuspendedEffectDesc> desc;
7483
7484    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7485    if (suspend) {
7486        if (index >= 0) {
7487            desc = mSuspendedEffects.valueAt(index);
7488        } else {
7489            desc = new SuspendedEffectDesc();
7490            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7491            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7492        }
7493        if (desc->mRefCount++ == 0) {
7494            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7495            for (size_t i = 0; i < effects.size(); i++) {
7496                setEffectSuspended_l(&effects[i]->desc().type, true);
7497            }
7498        }
7499    } else {
7500        if (index < 0) {
7501            return;
7502        }
7503        desc = mSuspendedEffects.valueAt(index);
7504        if (desc->mRefCount <= 0) {
7505            LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7506            desc->mRefCount = 1;
7507        }
7508        if (--desc->mRefCount == 0) {
7509            Vector<const effect_uuid_t *> types;
7510            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7511                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7512                    continue;
7513                }
7514                types.add(&mSuspendedEffects.valueAt(i)->mType);
7515            }
7516            for (size_t i = 0; i < types.size(); i++) {
7517                setEffectSuspended_l(types[i], false);
7518            }
7519            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7520            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7521        }
7522    }
7523}
7524
7525
7526// The volume effect is used for automated tests only
7527#ifndef OPENSL_ES_H_
7528static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7529                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7530const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7531#endif //OPENSL_ES_H_
7532
7533bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7534{
7535    // auxiliary effects and visualizer are never suspended on output mix
7536    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7537        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7538         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7539         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7540        return false;
7541    }
7542    return true;
7543}
7544
7545Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7546{
7547    Vector< sp<EffectModule> > effects;
7548    for (size_t i = 0; i < mEffects.size(); i++) {
7549        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7550            continue;
7551        }
7552        effects.add(mEffects[i]);
7553    }
7554    return effects;
7555}
7556
7557sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7558                                                            const effect_uuid_t *type)
7559{
7560    sp<EffectModule> effect;
7561    effect = getEffectFromType_l(type);
7562    if (effect != 0 && !effect->isEnabled()) {
7563        effect.clear();
7564    }
7565    return effect;
7566}
7567
7568void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7569                                                            bool enabled)
7570{
7571    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7572    if (enabled) {
7573        if (index < 0) {
7574            // if the effect is not suspend check if all effects are suspended
7575            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7576            if (index < 0) {
7577                return;
7578            }
7579            if (!isEffectEligibleForSuspend(effect->desc())) {
7580                return;
7581            }
7582            setEffectSuspended_l(&effect->desc().type, enabled);
7583            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7584            if (index < 0) {
7585                LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7586                return;
7587            }
7588        }
7589        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7590             effect->desc().type.timeLow);
7591        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7592        // if effect is requested to suspended but was not yet enabled, supend it now.
7593        if (desc->mEffect == 0) {
7594            desc->mEffect = effect;
7595            effect->setEnabled(false);
7596            effect->setSuspended(true);
7597        }
7598    } else {
7599        if (index < 0) {
7600            return;
7601        }
7602        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7603             effect->desc().type.timeLow);
7604        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7605        desc->mEffect.clear();
7606        effect->setSuspended(false);
7607    }
7608}
7609
7610#undef LOG_TAG
7611#define LOG_TAG "AudioFlinger"
7612
7613// ----------------------------------------------------------------------------
7614
7615status_t AudioFlinger::onTransact(
7616        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7617{
7618    return BnAudioFlinger::onTransact(code, data, reply, flags);
7619}
7620
7621}; // namespace android
7622