AudioFlinger.cpp revision 29c23c3aee5ae799b3480dc6876a46c46b019710 
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include <math.h>
23#include <signal.h>
24#include <sys/time.h>
25#include <sys/resource.h>
26
27#include <binder/IPCThreadState.h>
28#include <binder/IServiceManager.h>
29#include <utils/Log.h>
30#include <binder/Parcel.h>
31#include <binder/IPCThreadState.h>
32#include <utils/String16.h>
33#include <utils/threads.h>
34#include <utils/Atomic.h>
35
36#include <cutils/bitops.h>
37#include <cutils/properties.h>
38#include <cutils/compiler.h>
39
40#include <media/IMediaPlayerService.h>
41#include <media/IMediaDeathNotifier.h>
42
43#include <private/media/AudioTrackShared.h>
44#include <private/media/AudioEffectShared.h>
45
46#include <system/audio.h>
47#include <hardware/audio.h>
48
49#include "AudioMixer.h"
50#include "AudioFlinger.h"
51
52#include <media/EffectsFactoryApi.h>
53#include <audio_effects/effect_visualizer.h>
54#include <audio_effects/effect_ns.h>
55#include <audio_effects/effect_aec.h>
56
57#include <audio_utils/primitives.h>
58
59#include <cpustats/ThreadCpuUsage.h>
60#include <powermanager/PowerManager.h>
61// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
62
63// ----------------------------------------------------------------------------
64
65
66namespace android {
67
68static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
69static const char kHardwareLockedString[] = "Hardware lock is taken\n";
70
71//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
72static const float MAX_GAIN = 4096.0f;
73static const uint32_t MAX_GAIN_INT = 0x1000;
74
75// retry counts for buffer fill timeout
76// 50 * ~20msecs = 1 second
77static const int8_t kMaxTrackRetries = 50;
78static const int8_t kMaxTrackStartupRetries = 50;
79// allow less retry attempts on direct output thread.
80// direct outputs can be a scarce resource in audio hardware and should
81// be released as quickly as possible.
82static const int8_t kMaxTrackRetriesDirect = 2;
83
84static const int kDumpLockRetries = 50;
85static const int kDumpLockSleepUs = 20000;
86
87// don't warn about blocked writes or record buffer overflows more often than this
88static const nsecs_t kWarningThrottleNs = seconds(5);
89
90// RecordThread loop sleep time upon application overrun or audio HAL read error
91static const int kRecordThreadSleepUs = 5000;
92
93// maximum time to wait for setParameters to complete
94static const nsecs_t kSetParametersTimeoutNs = seconds(2);
95
96// minimum sleep time for the mixer thread loop when tracks are active but in underrun
97static const uint32_t kMinThreadSleepTimeUs = 5000;
98// maximum divider applied to the active sleep time in the mixer thread loop
99static const uint32_t kMaxThreadSleepTimeShift = 2;
100
101
102// ----------------------------------------------------------------------------
103
104static bool recordingAllowed() {
105    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
106    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
107    if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
108    return ok;
109}
110
111static bool settingsAllowed() {
112    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
113    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
114    if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
115    return ok;
116}
117
118// To collect the amplifier usage
119static void addBatteryData(uint32_t params) {
120    sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
121    if (service == NULL) {
122        // it already logged
123        return;
124    }
125
126    service->addBatteryData(params);
127}
128
129static int load_audio_interface(const char *if_name, const hw_module_t **mod,
130                                audio_hw_device_t **dev)
131{
132    int rc;
133
134    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
135    if (rc)
136        goto out;
137
138    rc = audio_hw_device_open(*mod, dev);
139    ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
140            AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
141    if (rc)
142        goto out;
143
144    return 0;
145
146out:
147    *mod = NULL;
148    *dev = NULL;
149    return rc;
150}
151
152static const char * const audio_interfaces[] = {
153    "primary",
154    "a2dp",
155    "usb",
156};
157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
158
159// ----------------------------------------------------------------------------
160
161AudioFlinger::AudioFlinger()
162    : BnAudioFlinger(),
163        mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
164        mBtNrecIsOff(false)
165{
166}
167
168void AudioFlinger::onFirstRef()
169{
170    int rc = 0;
171
172    Mutex::Autolock _l(mLock);
173
174    /* TODO: move all this work into an Init() function */
175    mHardwareStatus = AUDIO_HW_IDLE;
176
177    for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
178        const hw_module_t *mod;
179        audio_hw_device_t *dev;
180
181        rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
182        if (rc)
183            continue;
184
185        ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
186             mod->name, mod->id);
187        mAudioHwDevs.push(dev);
188
189        if (!mPrimaryHardwareDev) {
190            mPrimaryHardwareDev = dev;
191            ALOGI("Using '%s' (%s.%s) as the primary audio interface",
192                 mod->name, mod->id, audio_interfaces[i]);
193        }
194    }
195
196    mHardwareStatus = AUDIO_HW_INIT;
197
198    if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
199        ALOGE("Primary audio interface not found");
200        return;
201    }
202
203    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
204        audio_hw_device_t *dev = mAudioHwDevs[i];
205
206        mHardwareStatus = AUDIO_HW_INIT;
207        rc = dev->init_check(dev);
208        if (rc == 0) {
209            AutoMutex lock(mHardwareLock);
210
211            mMode = AUDIO_MODE_NORMAL;
212            mHardwareStatus = AUDIO_HW_SET_MODE;
213            dev->set_mode(dev, mMode);
214            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
215            dev->set_master_volume(dev, 1.0f);
216            mHardwareStatus = AUDIO_HW_IDLE;
217        }
218    }
219}
220
221status_t AudioFlinger::initCheck() const
222{
223    Mutex::Autolock _l(mLock);
224    if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
225        return NO_INIT;
226    return NO_ERROR;
227}
228
229AudioFlinger::~AudioFlinger()
230{
231    int num_devs = mAudioHwDevs.size();
232
233    while (!mRecordThreads.isEmpty()) {
234        // closeInput() will remove first entry from mRecordThreads
235        closeInput(mRecordThreads.keyAt(0));
236    }
237    while (!mPlaybackThreads.isEmpty()) {
238        // closeOutput() will remove first entry from mPlaybackThreads
239        closeOutput(mPlaybackThreads.keyAt(0));
240    }
241
242    for (int i = 0; i < num_devs; i++) {
243        audio_hw_device_t *dev = mAudioHwDevs[i];
244        audio_hw_device_close(dev);
245    }
246    mAudioHwDevs.clear();
247}
248
249audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
250{
251    /* first matching HW device is returned */
252    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
253        audio_hw_device_t *dev = mAudioHwDevs[i];
254        if ((dev->get_supported_devices(dev) & devices) == devices)
255            return dev;
256    }
257    return NULL;
258}
259
260status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
261{
262    const size_t SIZE = 256;
263    char buffer[SIZE];
264    String8 result;
265
266    result.append("Clients:\n");
267    for (size_t i = 0; i < mClients.size(); ++i) {
268        wp<Client> wClient = mClients.valueAt(i);
269        if (wClient != 0) {
270            sp<Client> client = wClient.promote();
271            if (client != 0) {
272                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
273                result.append(buffer);
274            }
275        }
276    }
277
278    result.append("Global session refs:\n");
279    result.append(" session pid cnt\n");
280    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
281        AudioSessionRef *r = mAudioSessionRefs[i];
282        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
283        result.append(buffer);
284    }
285    write(fd, result.string(), result.size());
286    return NO_ERROR;
287}
288
289
290status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
291{
292    const size_t SIZE = 256;
293    char buffer[SIZE];
294    String8 result;
295    hardware_call_state hardwareStatus = mHardwareStatus;
296
297    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
298    result.append(buffer);
299    write(fd, result.string(), result.size());
300    return NO_ERROR;
301}
302
303status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
304{
305    const size_t SIZE = 256;
306    char buffer[SIZE];
307    String8 result;
308    snprintf(buffer, SIZE, "Permission Denial: "
309            "can't dump AudioFlinger from pid=%d, uid=%d\n",
310            IPCThreadState::self()->getCallingPid(),
311            IPCThreadState::self()->getCallingUid());
312    result.append(buffer);
313    write(fd, result.string(), result.size());
314    return NO_ERROR;
315}
316
317static bool tryLock(Mutex& mutex)
318{
319    bool locked = false;
320    for (int i = 0; i < kDumpLockRetries; ++i) {
321        if (mutex.tryLock() == NO_ERROR) {
322            locked = true;
323            break;
324        }
325        usleep(kDumpLockSleepUs);
326    }
327    return locked;
328}
329
330status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
331{
332    if (!checkCallingPermission(String16("android.permission.DUMP"))) {
333        dumpPermissionDenial(fd, args);
334    } else {
335        // get state of hardware lock
336        bool hardwareLocked = tryLock(mHardwareLock);
337        if (!hardwareLocked) {
338            String8 result(kHardwareLockedString);
339            write(fd, result.string(), result.size());
340        } else {
341            mHardwareLock.unlock();
342        }
343
344        bool locked = tryLock(mLock);
345
346        // failed to lock - AudioFlinger is probably deadlocked
347        if (!locked) {
348            String8 result(kDeadlockedString);
349            write(fd, result.string(), result.size());
350        }
351
352        dumpClients(fd, args);
353        dumpInternals(fd, args);
354
355        // dump playback threads
356        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
357            mPlaybackThreads.valueAt(i)->dump(fd, args);
358        }
359
360        // dump record threads
361        for (size_t i = 0; i < mRecordThreads.size(); i++) {
362            mRecordThreads.valueAt(i)->dump(fd, args);
363        }
364
365        // dump all hardware devs
366        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
367            audio_hw_device_t *dev = mAudioHwDevs[i];
368            dev->dump(dev, fd);
369        }
370        if (locked) mLock.unlock();
371    }
372    return NO_ERROR;
373}
374
375
376// IAudioFlinger interface
377
378
379sp<IAudioTrack> AudioFlinger::createTrack(
380        pid_t pid,
381        audio_stream_type_t streamType,
382        uint32_t sampleRate,
383        audio_format_t format,
384        uint32_t channelMask,
385        int frameCount,
386        uint32_t flags,
387        const sp<IMemory>& sharedBuffer,
388        int output,
389        int *sessionId,
390        status_t *status)
391{
392    sp<PlaybackThread::Track> track;
393    sp<TrackHandle> trackHandle;
394    sp<Client> client;
395    wp<Client> wclient;
396    status_t lStatus;
397    int lSessionId;
398
399    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
400    // but if someone uses binder directly they could bypass that and cause us to crash
401    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
402        ALOGE("createTrack() invalid stream type %d", streamType);
403        lStatus = BAD_VALUE;
404        goto Exit;
405    }
406
407    {
408        Mutex::Autolock _l(mLock);
409        PlaybackThread *thread = checkPlaybackThread_l(output);
410        PlaybackThread *effectThread = NULL;
411        if (thread == NULL) {
412            ALOGE("unknown output thread");
413            lStatus = BAD_VALUE;
414            goto Exit;
415        }
416
417        wclient = mClients.valueFor(pid);
418
419        if (wclient != NULL) {
420            client = wclient.promote();
421        } else {
422            client = new Client(this, pid);
423            mClients.add(pid, client);
424        }
425
426        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
427        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
428            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
429                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
430                if (mPlaybackThreads.keyAt(i) != output) {
431                    // prevent same audio session on different output threads
432                    uint32_t sessions = t->hasAudioSession(*sessionId);
433                    if (sessions & PlaybackThread::TRACK_SESSION) {
434                        ALOGE("createTrack() session ID %d already in use", *sessionId);
435                        lStatus = BAD_VALUE;
436                        goto Exit;
437                    }
438                    // check if an effect with same session ID is waiting for a track to be created
439                    if (sessions & PlaybackThread::EFFECT_SESSION) {
440                        effectThread = t.get();
441                    }
442                }
443            }
444            lSessionId = *sessionId;
445        } else {
446            // if no audio session id is provided, create one here
447            lSessionId = nextUniqueId();
448            if (sessionId != NULL) {
449                *sessionId = lSessionId;
450            }
451        }
452        ALOGV("createTrack() lSessionId: %d", lSessionId);
453
454        track = thread->createTrack_l(client, streamType, sampleRate, format,
455                channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
456
457        // move effect chain to this output thread if an effect on same session was waiting
458        // for a track to be created
459        if (lStatus == NO_ERROR && effectThread != NULL) {
460            Mutex::Autolock _dl(thread->mLock);
461            Mutex::Autolock _sl(effectThread->mLock);
462            moveEffectChain_l(lSessionId, effectThread, thread, true);
463        }
464    }
465    if (lStatus == NO_ERROR) {
466        trackHandle = new TrackHandle(track);
467    } else {
468        // remove local strong reference to Client before deleting the Track so that the Client
469        // destructor is called by the TrackBase destructor with mLock held
470        client.clear();
471        track.clear();
472    }
473
474Exit:
475    if(status) {
476        *status = lStatus;
477    }
478    return trackHandle;
479}
480
481uint32_t AudioFlinger::sampleRate(int output) const
482{
483    Mutex::Autolock _l(mLock);
484    PlaybackThread *thread = checkPlaybackThread_l(output);
485    if (thread == NULL) {
486        ALOGW("sampleRate() unknown thread %d", output);
487        return 0;
488    }
489    return thread->sampleRate();
490}
491
492int AudioFlinger::channelCount(int output) const
493{
494    Mutex::Autolock _l(mLock);
495    PlaybackThread *thread = checkPlaybackThread_l(output);
496    if (thread == NULL) {
497        ALOGW("channelCount() unknown thread %d", output);
498        return 0;
499    }
500    return thread->channelCount();
501}
502
503audio_format_t AudioFlinger::format(int output) const
504{
505    Mutex::Autolock _l(mLock);
506    PlaybackThread *thread = checkPlaybackThread_l(output);
507    if (thread == NULL) {
508        ALOGW("format() unknown thread %d", output);
509        return AUDIO_FORMAT_INVALID;
510    }
511    return thread->format();
512}
513
514size_t AudioFlinger::frameCount(int output) const
515{
516    Mutex::Autolock _l(mLock);
517    PlaybackThread *thread = checkPlaybackThread_l(output);
518    if (thread == NULL) {
519        ALOGW("frameCount() unknown thread %d", output);
520        return 0;
521    }
522    return thread->frameCount();
523}
524
525uint32_t AudioFlinger::latency(int output) const
526{
527    Mutex::Autolock _l(mLock);
528    PlaybackThread *thread = checkPlaybackThread_l(output);
529    if (thread == NULL) {
530        ALOGW("latency() unknown thread %d", output);
531        return 0;
532    }
533    return thread->latency();
534}
535
536status_t AudioFlinger::setMasterVolume(float value)
537{
538    status_t ret = initCheck();
539    if (ret != NO_ERROR) {
540        return ret;
541    }
542
543    // check calling permissions
544    if (!settingsAllowed()) {
545        return PERMISSION_DENIED;
546    }
547
548    // when hw supports master volume, don't scale in sw mixer
549    { // scope for the lock
550        AutoMutex lock(mHardwareLock);
551        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
552        if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
553            value = 1.0f;
554        }
555        mHardwareStatus = AUDIO_HW_IDLE;
556    }
557
558    Mutex::Autolock _l(mLock);
559    mMasterVolume = value;
560    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
561       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
562
563    return NO_ERROR;
564}
565
566status_t AudioFlinger::setMode(audio_mode_t mode)
567{
568    status_t ret = initCheck();
569    if (ret != NO_ERROR) {
570        return ret;
571    }
572
573    // check calling permissions
574    if (!settingsAllowed()) {
575        return PERMISSION_DENIED;
576    }
577    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
578        ALOGW("Illegal value: setMode(%d)", mode);
579        return BAD_VALUE;
580    }
581
582    { // scope for the lock
583        AutoMutex lock(mHardwareLock);
584        mHardwareStatus = AUDIO_HW_SET_MODE;
585        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
586        mHardwareStatus = AUDIO_HW_IDLE;
587    }
588
589    if (NO_ERROR == ret) {
590        Mutex::Autolock _l(mLock);
591        mMode = mode;
592        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
593           mPlaybackThreads.valueAt(i)->setMode(mode);
594    }
595
596    return ret;
597}
598
599status_t AudioFlinger::setMicMute(bool state)
600{
601    status_t ret = initCheck();
602    if (ret != NO_ERROR) {
603        return ret;
604    }
605
606    // check calling permissions
607    if (!settingsAllowed()) {
608        return PERMISSION_DENIED;
609    }
610
611    AutoMutex lock(mHardwareLock);
612    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
613    ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
614    mHardwareStatus = AUDIO_HW_IDLE;
615    return ret;
616}
617
618bool AudioFlinger::getMicMute() const
619{
620    status_t ret = initCheck();
621    if (ret != NO_ERROR) {
622        return false;
623    }
624
625    bool state = AUDIO_MODE_INVALID;
626    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
627    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
628    mHardwareStatus = AUDIO_HW_IDLE;
629    return state;
630}
631
632status_t AudioFlinger::setMasterMute(bool muted)
633{
634    // check calling permissions
635    if (!settingsAllowed()) {
636        return PERMISSION_DENIED;
637    }
638
639    Mutex::Autolock _l(mLock);
640    mMasterMute = muted;
641    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
642       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
643
644    return NO_ERROR;
645}
646
647float AudioFlinger::masterVolume() const
648{
649    Mutex::Autolock _l(mLock);
650    return masterVolume_l();
651}
652
653bool AudioFlinger::masterMute() const
654{
655    Mutex::Autolock _l(mLock);
656    return masterMute_l();
657}
658
659status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output)
660{
661    // check calling permissions
662    if (!settingsAllowed()) {
663        return PERMISSION_DENIED;
664    }
665
666    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
667        ALOGE("setStreamVolume() invalid stream %d", stream);
668        return BAD_VALUE;
669    }
670
671    AutoMutex lock(mLock);
672    PlaybackThread *thread = NULL;
673    if (output) {
674        thread = checkPlaybackThread_l(output);
675        if (thread == NULL) {
676            return BAD_VALUE;
677        }
678    }
679
680    mStreamTypes[stream].volume = value;
681
682    if (thread == NULL) {
683        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
684           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
685        }
686    } else {
687        thread->setStreamVolume(stream, value);
688    }
689
690    return NO_ERROR;
691}
692
693status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
694{
695    // check calling permissions
696    if (!settingsAllowed()) {
697        return PERMISSION_DENIED;
698    }
699
700    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
701        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
702        ALOGE("setStreamMute() invalid stream %d", stream);
703        return BAD_VALUE;
704    }
705
706    AutoMutex lock(mLock);
707    mStreamTypes[stream].mute = muted;
708    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
709       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
710
711    return NO_ERROR;
712}
713
714float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const
715{
716    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
717        return 0.0f;
718    }
719
720    AutoMutex lock(mLock);
721    float volume;
722    if (output) {
723        PlaybackThread *thread = checkPlaybackThread_l(output);
724        if (thread == NULL) {
725            return 0.0f;
726        }
727        volume = thread->streamVolume(stream);
728    } else {
729        volume = mStreamTypes[stream].volume;
730    }
731
732    return volume;
733}
734
735bool AudioFlinger::streamMute(audio_stream_type_t stream) const
736{
737    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
738        return true;
739    }
740
741    return mStreamTypes[stream].mute;
742}
743
744status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
745{
746    status_t result;
747
748    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
749            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
750    // check calling permissions
751    if (!settingsAllowed()) {
752        return PERMISSION_DENIED;
753    }
754
755    // ioHandle == 0 means the parameters are global to the audio hardware interface
756    if (ioHandle == 0) {
757        AutoMutex lock(mHardwareLock);
758        mHardwareStatus = AUDIO_SET_PARAMETER;
759        status_t final_result = NO_ERROR;
760        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
761            audio_hw_device_t *dev = mAudioHwDevs[i];
762            result = dev->set_parameters(dev, keyValuePairs.string());
763            final_result = result ?: final_result;
764        }
765        mHardwareStatus = AUDIO_HW_IDLE;
766        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
767        AudioParameter param = AudioParameter(keyValuePairs);
768        String8 value;
769        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
770            Mutex::Autolock _l(mLock);
771            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
772            if (mBtNrecIsOff != btNrecIsOff) {
773                for (size_t i = 0; i < mRecordThreads.size(); i++) {
774                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
775                    RecordThread::RecordTrack *track = thread->track();
776                    if (track != NULL) {
777                        audio_devices_t device = (audio_devices_t)(
778                                thread->device() & AUDIO_DEVICE_IN_ALL);
779                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
780                        thread->setEffectSuspended(FX_IID_AEC,
781                                                   suspend,
782                                                   track->sessionId());
783                        thread->setEffectSuspended(FX_IID_NS,
784                                                   suspend,
785                                                   track->sessionId());
786                    }
787                }
788                mBtNrecIsOff = btNrecIsOff;
789            }
790        }
791        return final_result;
792    }
793
794    // hold a strong ref on thread in case closeOutput() or closeInput() is called
795    // and the thread is exited once the lock is released
796    sp<ThreadBase> thread;
797    {
798        Mutex::Autolock _l(mLock);
799        thread = checkPlaybackThread_l(ioHandle);
800        if (thread == NULL) {
801            thread = checkRecordThread_l(ioHandle);
802        } else if (thread == primaryPlaybackThread_l()) {
803            // indicate output device change to all input threads for pre processing
804            AudioParameter param = AudioParameter(keyValuePairs);
805            int value;
806            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
807                for (size_t i = 0; i < mRecordThreads.size(); i++) {
808                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
809                }
810            }
811        }
812    }
813    if (thread != NULL) {
814        result = thread->setParameters(keyValuePairs);
815        return result;
816    }
817    return BAD_VALUE;
818}
819
820String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
821{
822//    ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
823//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
824
825    if (ioHandle == 0) {
826        String8 out_s8;
827
828        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
829            audio_hw_device_t *dev = mAudioHwDevs[i];
830            char *s = dev->get_parameters(dev, keys.string());
831            out_s8 += String8(s);
832            free(s);
833        }
834        return out_s8;
835    }
836
837    Mutex::Autolock _l(mLock);
838
839    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
840    if (playbackThread != NULL) {
841        return playbackThread->getParameters(keys);
842    }
843    RecordThread *recordThread = checkRecordThread_l(ioHandle);
844    if (recordThread != NULL) {
845        return recordThread->getParameters(keys);
846    }
847    return String8("");
848}
849
850size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount)
851{
852    status_t ret = initCheck();
853    if (ret != NO_ERROR) {
854        return 0;
855    }
856
857    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
858}
859
860unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
861{
862    if (ioHandle == 0) {
863        return 0;
864    }
865
866    Mutex::Autolock _l(mLock);
867
868    RecordThread *recordThread = checkRecordThread_l(ioHandle);
869    if (recordThread != NULL) {
870        return recordThread->getInputFramesLost();
871    }
872    return 0;
873}
874
875status_t AudioFlinger::setVoiceVolume(float value)
876{
877    status_t ret = initCheck();
878    if (ret != NO_ERROR) {
879        return ret;
880    }
881
882    // check calling permissions
883    if (!settingsAllowed()) {
884        return PERMISSION_DENIED;
885    }
886
887    AutoMutex lock(mHardwareLock);
888    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
889    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
890    mHardwareStatus = AUDIO_HW_IDLE;
891
892    return ret;
893}
894
895status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
896{
897    status_t status;
898
899    Mutex::Autolock _l(mLock);
900
901    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
902    if (playbackThread != NULL) {
903        return playbackThread->getRenderPosition(halFrames, dspFrames);
904    }
905
906    return BAD_VALUE;
907}
908
909void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
910{
911
912    Mutex::Autolock _l(mLock);
913
914    int pid = IPCThreadState::self()->getCallingPid();
915    if (mNotificationClients.indexOfKey(pid) < 0) {
916        sp<NotificationClient> notificationClient = new NotificationClient(this,
917                                                                            client,
918                                                                            pid);
919        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
920
921        mNotificationClients.add(pid, notificationClient);
922
923        sp<IBinder> binder = client->asBinder();
924        binder->linkToDeath(notificationClient);
925
926        // the config change is always sent from playback or record threads to avoid deadlock
927        // with AudioSystem::gLock
928        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
929            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
930        }
931
932        for (size_t i = 0; i < mRecordThreads.size(); i++) {
933            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
934        }
935    }
936}
937
938void AudioFlinger::removeNotificationClient(pid_t pid)
939{
940    Mutex::Autolock _l(mLock);
941
942    int index = mNotificationClients.indexOfKey(pid);
943    if (index >= 0) {
944        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
945        ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
946        mNotificationClients.removeItem(pid);
947    }
948
949    ALOGV("%d died, releasing its sessions", pid);
950    int num = mAudioSessionRefs.size();
951    bool removed = false;
952    for (int i = 0; i< num; i++) {
953        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
954        ALOGV(" pid %d @ %d", ref->pid, i);
955        if (ref->pid == pid) {
956            ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
957            mAudioSessionRefs.removeAt(i);
958            delete ref;
959            removed = true;
960            i--;
961            num--;
962        }
963    }
964    if (removed) {
965        purgeStaleEffects_l();
966    }
967}
968
969// audioConfigChanged_l() must be called with AudioFlinger::mLock held
970void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
971{
972    size_t size = mNotificationClients.size();
973    for (size_t i = 0; i < size; i++) {
974        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
975    }
976}
977
978// removeClient_l() must be called with AudioFlinger::mLock held
979void AudioFlinger::removeClient_l(pid_t pid)
980{
981    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
982    mClients.removeItem(pid);
983}
984
985
986// ----------------------------------------------------------------------------
987
988AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
989    :   Thread(false),
990        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
991        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false),
992        mDevice(device)
993{
994    mDeathRecipient = new PMDeathRecipient(this);
995}
996
997AudioFlinger::ThreadBase::~ThreadBase()
998{
999    mParamCond.broadcast();
1000    // do not lock the mutex in destructor
1001    releaseWakeLock_l();
1002    if (mPowerManager != 0) {
1003        sp<IBinder> binder = mPowerManager->asBinder();
1004        binder->unlinkToDeath(mDeathRecipient);
1005    }
1006}
1007
1008void AudioFlinger::ThreadBase::exit()
1009{
1010    // keep a strong ref on ourself so that we won't get
1011    // destroyed in the middle of requestExitAndWait()
1012    sp <ThreadBase> strongMe = this;
1013
1014    ALOGV("ThreadBase::exit");
1015    {
1016        AutoMutex lock(mLock);
1017        mExiting = true;
1018        requestExit();
1019        mWaitWorkCV.signal();
1020    }
1021    requestExitAndWait();
1022}
1023
1024uint32_t AudioFlinger::ThreadBase::sampleRate() const
1025{
1026    return mSampleRate;
1027}
1028
1029int AudioFlinger::ThreadBase::channelCount() const
1030{
1031    return (int)mChannelCount;
1032}
1033
1034audio_format_t AudioFlinger::ThreadBase::format() const
1035{
1036    return mFormat;
1037}
1038
1039size_t AudioFlinger::ThreadBase::frameCount() const
1040{
1041    return mFrameCount;
1042}
1043
1044status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1045{
1046    status_t status;
1047
1048    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1049    Mutex::Autolock _l(mLock);
1050
1051    mNewParameters.add(keyValuePairs);
1052    mWaitWorkCV.signal();
1053    // wait condition with timeout in case the thread loop has exited
1054    // before the request could be processed
1055    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
1056        status = mParamStatus;
1057        mWaitWorkCV.signal();
1058    } else {
1059        status = TIMED_OUT;
1060    }
1061    return status;
1062}
1063
1064void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1065{
1066    Mutex::Autolock _l(mLock);
1067    sendConfigEvent_l(event, param);
1068}
1069
1070// sendConfigEvent_l() must be called with ThreadBase::mLock held
1071void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1072{
1073    ConfigEvent configEvent;
1074    configEvent.mEvent = event;
1075    configEvent.mParam = param;
1076    mConfigEvents.add(configEvent);
1077    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1078    mWaitWorkCV.signal();
1079}
1080
1081void AudioFlinger::ThreadBase::processConfigEvents()
1082{
1083    mLock.lock();
1084    while(!mConfigEvents.isEmpty()) {
1085        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1086        ConfigEvent configEvent = mConfigEvents[0];
1087        mConfigEvents.removeAt(0);
1088        // release mLock before locking AudioFlinger mLock: lock order is always
1089        // AudioFlinger then ThreadBase to avoid cross deadlock
1090        mLock.unlock();
1091        mAudioFlinger->mLock.lock();
1092        audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
1093        mAudioFlinger->mLock.unlock();
1094        mLock.lock();
1095    }
1096    mLock.unlock();
1097}
1098
1099status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1100{
1101    const size_t SIZE = 256;
1102    char buffer[SIZE];
1103    String8 result;
1104
1105    bool locked = tryLock(mLock);
1106    if (!locked) {
1107        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1108        write(fd, buffer, strlen(buffer));
1109    }
1110
1111    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1112    result.append(buffer);
1113    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1114    result.append(buffer);
1115    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1116    result.append(buffer);
1117    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1118    result.append(buffer);
1119    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1120    result.append(buffer);
1121    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1122    result.append(buffer);
1123    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
1124    result.append(buffer);
1125
1126    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1127    result.append(buffer);
1128    result.append(" Index Command");
1129    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1130        snprintf(buffer, SIZE, "\n %02d    ", i);
1131        result.append(buffer);
1132        result.append(mNewParameters[i]);
1133    }
1134
1135    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1136    result.append(buffer);
1137    snprintf(buffer, SIZE, " Index event param\n");
1138    result.append(buffer);
1139    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1140        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
1141        result.append(buffer);
1142    }
1143    result.append("\n");
1144
1145    write(fd, result.string(), result.size());
1146
1147    if (locked) {
1148        mLock.unlock();
1149    }
1150    return NO_ERROR;
1151}
1152
1153status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1154{
1155    const size_t SIZE = 256;
1156    char buffer[SIZE];
1157    String8 result;
1158
1159    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1160    write(fd, buffer, strlen(buffer));
1161
1162    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1163        sp<EffectChain> chain = mEffectChains[i];
1164        if (chain != 0) {
1165            chain->dump(fd, args);
1166        }
1167    }
1168    return NO_ERROR;
1169}
1170
1171void AudioFlinger::ThreadBase::acquireWakeLock()
1172{
1173    Mutex::Autolock _l(mLock);
1174    acquireWakeLock_l();
1175}
1176
1177void AudioFlinger::ThreadBase::acquireWakeLock_l()
1178{
1179    if (mPowerManager == 0) {
1180        // use checkService() to avoid blocking if power service is not up yet
1181        sp<IBinder> binder =
1182            defaultServiceManager()->checkService(String16("power"));
1183        if (binder == 0) {
1184            ALOGW("Thread %s cannot connect to the power manager service", mName);
1185        } else {
1186            mPowerManager = interface_cast<IPowerManager>(binder);
1187            binder->linkToDeath(mDeathRecipient);
1188        }
1189    }
1190    if (mPowerManager != 0) {
1191        sp<IBinder> binder = new BBinder();
1192        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
1193                                                         binder,
1194                                                         String16(mName));
1195        if (status == NO_ERROR) {
1196            mWakeLockToken = binder;
1197        }
1198        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
1199    }
1200}
1201
1202void AudioFlinger::ThreadBase::releaseWakeLock()
1203{
1204    Mutex::Autolock _l(mLock);
1205    releaseWakeLock_l();
1206}
1207
1208void AudioFlinger::ThreadBase::releaseWakeLock_l()
1209{
1210    if (mWakeLockToken != 0) {
1211        ALOGV("releaseWakeLock_l() %s", mName);
1212        if (mPowerManager != 0) {
1213            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
1214        }
1215        mWakeLockToken.clear();
1216    }
1217}
1218
1219void AudioFlinger::ThreadBase::clearPowerManager()
1220{
1221    Mutex::Autolock _l(mLock);
1222    releaseWakeLock_l();
1223    mPowerManager.clear();
1224}
1225
1226void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
1227{
1228    sp<ThreadBase> thread = mThread.promote();
1229    if (thread != 0) {
1230        thread->clearPowerManager();
1231    }
1232    ALOGW("power manager service died !!!");
1233}
1234
1235void AudioFlinger::ThreadBase::setEffectSuspended(
1236        const effect_uuid_t *type, bool suspend, int sessionId)
1237{
1238    Mutex::Autolock _l(mLock);
1239    setEffectSuspended_l(type, suspend, sessionId);
1240}
1241
1242void AudioFlinger::ThreadBase::setEffectSuspended_l(
1243        const effect_uuid_t *type, bool suspend, int sessionId)
1244{
1245    sp<EffectChain> chain;
1246    chain = getEffectChain_l(sessionId);
1247    if (chain != 0) {
1248        if (type != NULL) {
1249            chain->setEffectSuspended_l(type, suspend);
1250        } else {
1251            chain->setEffectSuspendedAll_l(suspend);
1252        }
1253    }
1254
1255    updateSuspendedSessions_l(type, suspend, sessionId);
1256}
1257
1258void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1259{
1260    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1261    if (index < 0) {
1262        return;
1263    }
1264
1265    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1266            mSuspendedSessions.editValueAt(index);
1267
1268    for (size_t i = 0; i < sessionEffects.size(); i++) {
1269        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1270        for (int j = 0; j < desc->mRefCount; j++) {
1271            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1272                chain->setEffectSuspendedAll_l(true);
1273            } else {
1274                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1275                     desc->mType.timeLow);
1276                chain->setEffectSuspended_l(&desc->mType, true);
1277            }
1278        }
1279    }
1280}
1281
1282void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1283                                                         bool suspend,
1284                                                         int sessionId)
1285{
1286    int index = mSuspendedSessions.indexOfKey(sessionId);
1287
1288    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1289
1290    if (suspend) {
1291        if (index >= 0) {
1292            sessionEffects = mSuspendedSessions.editValueAt(index);
1293        } else {
1294            mSuspendedSessions.add(sessionId, sessionEffects);
1295        }
1296    } else {
1297        if (index < 0) {
1298            return;
1299        }
1300        sessionEffects = mSuspendedSessions.editValueAt(index);
1301    }
1302
1303
1304    int key = EffectChain::kKeyForSuspendAll;
1305    if (type != NULL) {
1306        key = type->timeLow;
1307    }
1308    index = sessionEffects.indexOfKey(key);
1309
1310    sp <SuspendedSessionDesc> desc;
1311    if (suspend) {
1312        if (index >= 0) {
1313            desc = sessionEffects.valueAt(index);
1314        } else {
1315            desc = new SuspendedSessionDesc();
1316            if (type != NULL) {
1317                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1318            }
1319            sessionEffects.add(key, desc);
1320            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1321        }
1322        desc->mRefCount++;
1323    } else {
1324        if (index < 0) {
1325            return;
1326        }
1327        desc = sessionEffects.valueAt(index);
1328        if (--desc->mRefCount == 0) {
1329            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1330            sessionEffects.removeItemsAt(index);
1331            if (sessionEffects.isEmpty()) {
1332                ALOGV("updateSuspendedSessions_l() restore removing session %d",
1333                                 sessionId);
1334                mSuspendedSessions.removeItem(sessionId);
1335            }
1336        }
1337    }
1338    if (!sessionEffects.isEmpty()) {
1339        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1340    }
1341}
1342
1343void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1344                                                            bool enabled,
1345                                                            int sessionId)
1346{
1347    Mutex::Autolock _l(mLock);
1348    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1349}
1350
1351void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1352                                                            bool enabled,
1353                                                            int sessionId)
1354{
1355    if (mType != RECORD) {
1356        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1357        // another session. This gives the priority to well behaved effect control panels
1358        // and applications not using global effects.
1359        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1360            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1361        }
1362    }
1363
1364    sp<EffectChain> chain = getEffectChain_l(sessionId);
1365    if (chain != 0) {
1366        chain->checkSuspendOnEffectEnabled(effect, enabled);
1367    }
1368}
1369
1370// ----------------------------------------------------------------------------
1371
1372AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1373                                             AudioStreamOut* output,
1374                                             int id,
1375                                             uint32_t device)
1376    :   ThreadBase(audioFlinger, id, device),
1377        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output),
1378        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1379{
1380    snprintf(mName, kNameLength, "AudioOut_%d", id);
1381
1382    readOutputParameters();
1383
1384    // Assumes constructor is called by AudioFlinger with it's mLock held,
1385    // but it would be safer to explicitly pass these as parameters
1386    mMasterVolume = mAudioFlinger->masterVolume_l();
1387    mMasterMute = mAudioFlinger->masterMute_l();
1388
1389    // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
1390    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
1391    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
1392            stream = (audio_stream_type_t) (stream + 1)) {
1393        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1394        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1395        // initialized by stream_type_t default constructor
1396        // mStreamTypes[stream].valid = true;
1397    }
1398}
1399
1400AudioFlinger::PlaybackThread::~PlaybackThread()
1401{
1402    delete [] mMixBuffer;
1403}
1404
1405status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1406{
1407    dumpInternals(fd, args);
1408    dumpTracks(fd, args);
1409    dumpEffectChains(fd, args);
1410    return NO_ERROR;
1411}
1412
1413status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1414{
1415    const size_t SIZE = 256;
1416    char buffer[SIZE];
1417    String8 result;
1418
1419    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1420    result.append(buffer);
1421    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1422    for (size_t i = 0; i < mTracks.size(); ++i) {
1423        sp<Track> track = mTracks[i];
1424        if (track != 0) {
1425            track->dump(buffer, SIZE);
1426            result.append(buffer);
1427        }
1428    }
1429
1430    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1431    result.append(buffer);
1432    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1433    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1434        wp<Track> wTrack = mActiveTracks[i];
1435        if (wTrack != 0) {
1436            sp<Track> track = wTrack.promote();
1437            if (track != 0) {
1438                track->dump(buffer, SIZE);
1439                result.append(buffer);
1440            }
1441        }
1442    }
1443    write(fd, result.string(), result.size());
1444    return NO_ERROR;
1445}
1446
1447status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1448{
1449    const size_t SIZE = 256;
1450    char buffer[SIZE];
1451    String8 result;
1452
1453    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1454    result.append(buffer);
1455    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1456    result.append(buffer);
1457    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1458    result.append(buffer);
1459    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1460    result.append(buffer);
1461    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1462    result.append(buffer);
1463    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1464    result.append(buffer);
1465    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1466    result.append(buffer);
1467    write(fd, result.string(), result.size());
1468
1469    dumpBase(fd, args);
1470
1471    return NO_ERROR;
1472}
1473
1474// Thread virtuals
1475status_t AudioFlinger::PlaybackThread::readyToRun()
1476{
1477    status_t status = initCheck();
1478    if (status == NO_ERROR) {
1479        ALOGI("AudioFlinger's thread %p ready to run", this);
1480    } else {
1481        ALOGE("No working audio driver found.");
1482    }
1483    return status;
1484}
1485
1486void AudioFlinger::PlaybackThread::onFirstRef()
1487{
1488    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1489}
1490
1491// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1492sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1493        const sp<AudioFlinger::Client>& client,
1494        audio_stream_type_t streamType,
1495        uint32_t sampleRate,
1496        audio_format_t format,
1497        uint32_t channelMask,
1498        int frameCount,
1499        const sp<IMemory>& sharedBuffer,
1500        int sessionId,
1501        status_t *status)
1502{
1503    sp<Track> track;
1504    status_t lStatus;
1505
1506    if (mType == DIRECT) {
1507        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1508            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1509                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1510                        "for output %p with format %d",
1511                        sampleRate, format, channelMask, mOutput, mFormat);
1512                lStatus = BAD_VALUE;
1513                goto Exit;
1514            }
1515        }
1516    } else {
1517        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1518        if (sampleRate > mSampleRate*2) {
1519            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1520            lStatus = BAD_VALUE;
1521            goto Exit;
1522        }
1523    }
1524
1525    lStatus = initCheck();
1526    if (lStatus != NO_ERROR) {
1527        ALOGE("Audio driver not initialized.");
1528        goto Exit;
1529    }
1530
1531    { // scope for mLock
1532        Mutex::Autolock _l(mLock);
1533
1534        // all tracks in same audio session must share the same routing strategy otherwise
1535        // conflicts will happen when tracks are moved from one output to another by audio policy
1536        // manager
1537        uint32_t strategy =
1538                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1539        for (size_t i = 0; i < mTracks.size(); ++i) {
1540            sp<Track> t = mTracks[i];
1541            if (t != 0) {
1542                uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
1543                if (sessionId == t->sessionId() && strategy != actual) {
1544                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1545                            strategy, actual);
1546                    lStatus = BAD_VALUE;
1547                    goto Exit;
1548                }
1549            }
1550        }
1551
1552        track = new Track(this, client, streamType, sampleRate, format,
1553                channelMask, frameCount, sharedBuffer, sessionId);
1554        if (track->getCblk() == NULL || track->name() < 0) {
1555            lStatus = NO_MEMORY;
1556            goto Exit;
1557        }
1558        mTracks.add(track);
1559
1560        sp<EffectChain> chain = getEffectChain_l(sessionId);
1561        if (chain != 0) {
1562            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1563            track->setMainBuffer(chain->inBuffer());
1564            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1565            chain->incTrackCnt();
1566        }
1567
1568        // invalidate track immediately if the stream type was moved to another thread since
1569        // createTrack() was called by the client process.
1570        if (!mStreamTypes[streamType].valid) {
1571            ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1572                 this, streamType);
1573            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1574        }
1575    }
1576    lStatus = NO_ERROR;
1577
1578Exit:
1579    if(status) {
1580        *status = lStatus;
1581    }
1582    return track;
1583}
1584
1585uint32_t AudioFlinger::PlaybackThread::latency() const
1586{
1587    Mutex::Autolock _l(mLock);
1588    if (initCheck() == NO_ERROR) {
1589        return mOutput->stream->get_latency(mOutput->stream);
1590    } else {
1591        return 0;
1592    }
1593}
1594
1595status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1596{
1597    mMasterVolume = value;
1598    return NO_ERROR;
1599}
1600
1601status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1602{
1603    mMasterMute = muted;
1604    return NO_ERROR;
1605}
1606
1607float AudioFlinger::PlaybackThread::masterVolume() const
1608{
1609    return mMasterVolume;
1610}
1611
1612bool AudioFlinger::PlaybackThread::masterMute() const
1613{
1614    return mMasterMute;
1615}
1616
1617status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
1618{
1619    mStreamTypes[stream].volume = value;
1620    return NO_ERROR;
1621}
1622
1623status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
1624{
1625    mStreamTypes[stream].mute = muted;
1626    return NO_ERROR;
1627}
1628
1629float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
1630{
1631    return mStreamTypes[stream].volume;
1632}
1633
1634bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const
1635{
1636    return mStreamTypes[stream].mute;
1637}
1638
1639// addTrack_l() must be called with ThreadBase::mLock held
1640status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1641{
1642    status_t status = ALREADY_EXISTS;
1643
1644    // set retry count for buffer fill
1645    track->mRetryCount = kMaxTrackStartupRetries;
1646    if (mActiveTracks.indexOf(track) < 0) {
1647        // the track is newly added, make sure it fills up all its
1648        // buffers before playing. This is to ensure the client will
1649        // effectively get the latency it requested.
1650        track->mFillingUpStatus = Track::FS_FILLING;
1651        track->mResetDone = false;
1652        mActiveTracks.add(track);
1653        if (track->mainBuffer() != mMixBuffer) {
1654            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1655            if (chain != 0) {
1656                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1657                chain->incActiveTrackCnt();
1658            }
1659        }
1660
1661        status = NO_ERROR;
1662    }
1663
1664    ALOGV("mWaitWorkCV.broadcast");
1665    mWaitWorkCV.broadcast();
1666
1667    return status;
1668}
1669
1670// destroyTrack_l() must be called with ThreadBase::mLock held
1671void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1672{
1673    track->mState = TrackBase::TERMINATED;
1674    if (mActiveTracks.indexOf(track) < 0) {
1675        removeTrack_l(track);
1676    }
1677}
1678
1679void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1680{
1681    mTracks.remove(track);
1682    deleteTrackName_l(track->name());
1683    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1684    if (chain != 0) {
1685        chain->decTrackCnt();
1686    }
1687}
1688
1689String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1690{
1691    String8 out_s8 = String8("");
1692    char *s;
1693
1694    Mutex::Autolock _l(mLock);
1695    if (initCheck() != NO_ERROR) {
1696        return out_s8;
1697    }
1698
1699    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1700    out_s8 = String8(s);
1701    free(s);
1702    return out_s8;
1703}
1704
1705// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1706void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1707    AudioSystem::OutputDescriptor desc;
1708    void *param2 = 0;
1709
1710    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1711
1712    switch (event) {
1713    case AudioSystem::OUTPUT_OPENED:
1714    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1715        desc.channels = mChannelMask;
1716        desc.samplingRate = mSampleRate;
1717        desc.format = mFormat;
1718        desc.frameCount = mFrameCount;
1719        desc.latency = latency();
1720        param2 = &desc;
1721        break;
1722
1723    case AudioSystem::STREAM_CONFIG_CHANGED:
1724        param2 = &param;
1725    case AudioSystem::OUTPUT_CLOSED:
1726    default:
1727        break;
1728    }
1729    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1730}
1731
1732void AudioFlinger::PlaybackThread::readOutputParameters()
1733{
1734    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1735    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1736    mChannelCount = (uint16_t)popcount(mChannelMask);
1737    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1738    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
1739    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1740
1741    // FIXME - Current mixer implementation only supports stereo output: Always
1742    // Allocate a stereo buffer even if HW output is mono.
1743    if (mMixBuffer != NULL) delete[] mMixBuffer;
1744    mMixBuffer = new int16_t[mFrameCount * 2];
1745    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1746
1747    // force reconfiguration of effect chains and engines to take new buffer size and audio
1748    // parameters into account
1749    // Note that mLock is not held when readOutputParameters() is called from the constructor
1750    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1751    // matter.
1752    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1753    Vector< sp<EffectChain> > effectChains = mEffectChains;
1754    for (size_t i = 0; i < effectChains.size(); i ++) {
1755        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1756    }
1757}
1758
1759status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1760{
1761    if (halFrames == 0 || dspFrames == 0) {
1762        return BAD_VALUE;
1763    }
1764    Mutex::Autolock _l(mLock);
1765    if (initCheck() != NO_ERROR) {
1766        return INVALID_OPERATION;
1767    }
1768    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1769
1770    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1771}
1772
1773uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1774{
1775    Mutex::Autolock _l(mLock);
1776    uint32_t result = 0;
1777    if (getEffectChain_l(sessionId) != 0) {
1778        result = EFFECT_SESSION;
1779    }
1780
1781    for (size_t i = 0; i < mTracks.size(); ++i) {
1782        sp<Track> track = mTracks[i];
1783        if (sessionId == track->sessionId() &&
1784                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1785            result |= TRACK_SESSION;
1786            break;
1787        }
1788    }
1789
1790    return result;
1791}
1792
1793uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1794{
1795    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1796    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1797    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1798        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1799    }
1800    for (size_t i = 0; i < mTracks.size(); i++) {
1801        sp<Track> track = mTracks[i];
1802        if (sessionId == track->sessionId() &&
1803                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1804            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1805        }
1806    }
1807    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1808}
1809
1810
1811AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1812{
1813    Mutex::Autolock _l(mLock);
1814    return mOutput;
1815}
1816
1817AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1818{
1819    Mutex::Autolock _l(mLock);
1820    AudioStreamOut *output = mOutput;
1821    mOutput = NULL;
1822    return output;
1823}
1824
1825// this method must always be called either with ThreadBase mLock held or inside the thread loop
1826audio_stream_t* AudioFlinger::PlaybackThread::stream()
1827{
1828    if (mOutput == NULL) {
1829        return NULL;
1830    }
1831    return &mOutput->stream->common;
1832}
1833
1834uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
1835{
1836    // A2DP output latency is not due only to buffering capacity. It also reflects encoding,
1837    // decoding and transfer time. So sleeping for half of the latency would likely cause
1838    // underruns
1839    if (audio_is_a2dp_device((audio_devices_t)mDevice)) {
1840        return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000);
1841    } else {
1842        return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
1843    }
1844}
1845
1846// ----------------------------------------------------------------------------
1847
1848AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1849    :   PlaybackThread(audioFlinger, output, id, device),
1850        mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE)
1851{
1852    mType = ThreadBase::MIXER;
1853    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1854
1855    // FIXME - Current mixer implementation only supports stereo output
1856    if (mChannelCount == 1) {
1857        ALOGE("Invalid audio hardware channel count");
1858    }
1859}
1860
1861AudioFlinger::MixerThread::~MixerThread()
1862{
1863    delete mAudioMixer;
1864}
1865
1866bool AudioFlinger::MixerThread::threadLoop()
1867{
1868    Vector< sp<Track> > tracksToRemove;
1869    mixer_state mixerStatus = MIXER_IDLE;
1870    nsecs_t standbyTime = systemTime();
1871    size_t mixBufferSize = mFrameCount * mFrameSize;
1872    // FIXME: Relaxed timing because of a certain device that can't meet latency
1873    // Should be reduced to 2x after the vendor fixes the driver issue
1874    // increase threshold again due to low power audio mode. The way this warning threshold is
1875    // calculated and its usefulness should be reconsidered anyway.
1876    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1877    nsecs_t lastWarning = 0;
1878    bool longStandbyExit = false;
1879    uint32_t activeSleepTime = activeSleepTimeUs();
1880    uint32_t idleSleepTime = idleSleepTimeUs();
1881    uint32_t sleepTime = idleSleepTime;
1882    uint32_t sleepTimeShift = 0;
1883    Vector< sp<EffectChain> > effectChains;
1884#ifdef DEBUG_CPU_USAGE
1885    ThreadCpuUsage cpu;
1886    const CentralTendencyStatistics& stats = cpu.statistics();
1887#endif
1888
1889    acquireWakeLock();
1890
1891    while (!exitPending())
1892    {
1893#ifdef DEBUG_CPU_USAGE
1894        cpu.sampleAndEnable();
1895        unsigned n = stats.n();
1896        // cpu.elapsed() is expensive, so don't call it every loop
1897        if ((n & 127) == 1) {
1898            long long elapsed = cpu.elapsed();
1899            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1900                double perLoop = elapsed / (double) n;
1901                double perLoop100 = perLoop * 0.01;
1902                double mean = stats.mean();
1903                double stddev = stats.stddev();
1904                double minimum = stats.minimum();
1905                double maximum = stats.maximum();
1906                cpu.resetStatistics();
1907                ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
1908                        elapsed * .000000001, n, perLoop * .000001,
1909                        mean * .001,
1910                        stddev * .001,
1911                        minimum * .001,
1912                        maximum * .001,
1913                        mean / perLoop100,
1914                        stddev / perLoop100,
1915                        minimum / perLoop100,
1916                        maximum / perLoop100);
1917            }
1918        }
1919#endif
1920        processConfigEvents();
1921
1922        mixerStatus = MIXER_IDLE;
1923        { // scope for mLock
1924
1925            Mutex::Autolock _l(mLock);
1926
1927            if (checkForNewParameters_l()) {
1928                mixBufferSize = mFrameCount * mFrameSize;
1929                // FIXME: Relaxed timing because of a certain device that can't meet latency
1930                // Should be reduced to 2x after the vendor fixes the driver issue
1931                // increase threshold again due to low power audio mode. The way this warning
1932                // threshold is calculated and its usefulness should be reconsidered anyway.
1933                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1934                activeSleepTime = activeSleepTimeUs();
1935                idleSleepTime = idleSleepTimeUs();
1936            }
1937
1938            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1939
1940            // put audio hardware into standby after short delay
1941            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1942                        mSuspended)) {
1943                if (!mStandby) {
1944                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1945                    mOutput->stream->common.standby(&mOutput->stream->common);
1946                    mStandby = true;
1947                    mBytesWritten = 0;
1948                }
1949
1950                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1951                    // we're about to wait, flush the binder command buffer
1952                    IPCThreadState::self()->flushCommands();
1953
1954                    if (exitPending()) break;
1955
1956                    releaseWakeLock_l();
1957                    // wait until we have something to do...
1958                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1959                    mWaitWorkCV.wait(mLock);
1960                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1961                    acquireWakeLock_l();
1962
1963                    mPrevMixerStatus = MIXER_IDLE;
1964                    if (!mMasterMute) {
1965                        char value[PROPERTY_VALUE_MAX];
1966                        property_get("ro.audio.silent", value, "0");
1967                        if (atoi(value)) {
1968                            ALOGD("Silence is golden");
1969                            setMasterMute(true);
1970                        }
1971                    }
1972
1973                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1974                    sleepTime = idleSleepTime;
1975                    sleepTimeShift = 0;
1976                    continue;
1977                }
1978            }
1979
1980            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1981
1982            // prevent any changes in effect chain list and in each effect chain
1983            // during mixing and effect process as the audio buffers could be deleted
1984            // or modified if an effect is created or deleted
1985            lockEffectChains_l(effectChains);
1986        }
1987
1988        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1989            // mix buffers...
1990            mAudioMixer->process();
1991            // increase sleep time progressively when application underrun condition clears.
1992            // Only increase sleep time if the mixer is ready for two consecutive times to avoid
1993            // that a steady state of alternating ready/not ready conditions keeps the sleep time
1994            // such that we would underrun the audio HAL.
1995            if ((sleepTime == 0) && (sleepTimeShift > 0)) {
1996                sleepTimeShift--;
1997            }
1998            sleepTime = 0;
1999            standbyTime = systemTime() + kStandbyTimeInNsecs;
2000            //TODO: delay standby when effects have a tail
2001        } else {
2002            // If no tracks are ready, sleep once for the duration of an output
2003            // buffer size, then write 0s to the output
2004            if (sleepTime == 0) {
2005                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2006                    sleepTime = activeSleepTime >> sleepTimeShift;
2007                    if (sleepTime < kMinThreadSleepTimeUs) {
2008                        sleepTime = kMinThreadSleepTimeUs;
2009                    }
2010                    // reduce sleep time in case of consecutive application underruns to avoid
2011                    // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
2012                    // duration we would end up writing less data than needed by the audio HAL if
2013                    // the condition persists.
2014                    if (sleepTimeShift < kMaxThreadSleepTimeShift) {
2015                        sleepTimeShift++;
2016                    }
2017                } else {
2018                    sleepTime = idleSleepTime;
2019                }
2020            } else if (mBytesWritten != 0 ||
2021                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
2022                memset (mMixBuffer, 0, mixBufferSize);
2023                sleepTime = 0;
2024                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
2025            }
2026            // TODO add standby time extension fct of effect tail
2027        }
2028
2029        if (mSuspended) {
2030            sleepTime = suspendSleepTimeUs();
2031        }
2032        // sleepTime == 0 means we must write to audio hardware
2033        if (sleepTime == 0) {
2034            for (size_t i = 0; i < effectChains.size(); i ++) {
2035                effectChains[i]->process_l();
2036            }
2037            // enable changes in effect chain
2038            unlockEffectChains(effectChains);
2039            mLastWriteTime = systemTime();
2040            mInWrite = true;
2041            mBytesWritten += mixBufferSize;
2042
2043            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2044            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2045            mNumWrites++;
2046            mInWrite = false;
2047            nsecs_t now = systemTime();
2048            nsecs_t delta = now - mLastWriteTime;
2049            if (!mStandby && delta > maxPeriod) {
2050                mNumDelayedWrites++;
2051                if ((now - lastWarning) > kWarningThrottleNs) {
2052                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2053                            ns2ms(delta), mNumDelayedWrites, this);
2054                    lastWarning = now;
2055                }
2056                if (mStandby) {
2057                    longStandbyExit = true;
2058                }
2059            }
2060            mStandby = false;
2061        } else {
2062            // enable changes in effect chain
2063            unlockEffectChains(effectChains);
2064            usleep(sleepTime);
2065        }
2066
2067        // finally let go of all our tracks, without the lock held
2068        // since we can't guarantee the destructors won't acquire that
2069        // same lock.
2070        tracksToRemove.clear();
2071
2072        // Effect chains will be actually deleted here if they were removed from
2073        // mEffectChains list during mixing or effects processing
2074        effectChains.clear();
2075    }
2076
2077    if (!mStandby) {
2078        mOutput->stream->common.standby(&mOutput->stream->common);
2079    }
2080
2081    releaseWakeLock();
2082
2083    ALOGV("MixerThread %p exiting", this);
2084    return false;
2085}
2086
2087// prepareTracks_l() must be called with ThreadBase::mLock held
2088AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
2089        const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2090{
2091
2092    mixer_state mixerStatus = MIXER_IDLE;
2093    // find out which tracks need to be processed
2094    size_t count = activeTracks.size();
2095    size_t mixedTracks = 0;
2096    size_t tracksWithEffect = 0;
2097
2098    float masterVolume = mMasterVolume;
2099    bool  masterMute = mMasterMute;
2100
2101    if (masterMute) {
2102        masterVolume = 0;
2103    }
2104    // Delegate master volume control to effect in output mix effect chain if needed
2105    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2106    if (chain != 0) {
2107        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2108        chain->setVolume_l(&v, &v);
2109        masterVolume = (float)((v + (1 << 23)) >> 24);
2110        chain.clear();
2111    }
2112
2113    for (size_t i=0 ; i<count ; i++) {
2114        sp<Track> t = activeTracks[i].promote();
2115        if (t == 0) continue;
2116
2117        // this const just means the local variable doesn't change
2118        Track* const track = t.get();
2119        audio_track_cblk_t* cblk = track->cblk();
2120
2121        // The first time a track is added we wait
2122        // for all its buffers to be filled before processing it
2123        int name = track->name();
2124        // make sure that we have enough frames to mix one full buffer.
2125        // enforce this condition only once to enable draining the buffer in case the client
2126        // app does not call stop() and relies on underrun to stop:
2127        // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
2128        // during last round
2129        uint32_t minFrames = 1;
2130        if (!track->isStopped() && !track->isPausing() &&
2131                (mPrevMixerStatus == MIXER_TRACKS_READY)) {
2132            if (t->sampleRate() == (int)mSampleRate) {
2133                minFrames = mFrameCount;
2134            } else {
2135                // +1 for rounding and +1 for additional sample needed for interpolation
2136                minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
2137                // add frames already consumed but not yet released by the resampler
2138                // because cblk->framesReady() will  include these frames
2139                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
2140                // the minimum track buffer size is normally twice the number of frames necessary
2141                // to fill one buffer and the resampler should not leave more than one buffer worth
2142                // of unreleased frames after each pass, but just in case...
2143                ALOG_ASSERT(minFrames <= cblk->frameCount);
2144            }
2145        }
2146        if ((cblk->framesReady() >= minFrames) && track->isReady() &&
2147                !track->isPaused() && !track->isTerminated())
2148        {
2149            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
2150
2151            mixedTracks++;
2152
2153            // track->mainBuffer() != mMixBuffer means there is an effect chain
2154            // connected to the track
2155            chain.clear();
2156            if (track->mainBuffer() != mMixBuffer) {
2157                chain = getEffectChain_l(track->sessionId());
2158                // Delegate volume control to effect in track effect chain if needed
2159                if (chain != 0) {
2160                    tracksWithEffect++;
2161                } else {
2162                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
2163                            name, track->sessionId());
2164                }
2165            }
2166
2167
2168            int param = AudioMixer::VOLUME;
2169            if (track->mFillingUpStatus == Track::FS_FILLED) {
2170                // no ramp for the first volume setting
2171                track->mFillingUpStatus = Track::FS_ACTIVE;
2172                if (track->mState == TrackBase::RESUMING) {
2173                    track->mState = TrackBase::ACTIVE;
2174                    param = AudioMixer::RAMP_VOLUME;
2175                }
2176                mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2177            } else if (cblk->server != 0) {
2178                // If the track is stopped before the first frame was mixed,
2179                // do not apply ramp
2180                param = AudioMixer::RAMP_VOLUME;
2181            }
2182
2183            // compute volume for this track
2184            uint32_t vl, vr, va;
2185            if (track->isMuted() || track->isPausing() ||
2186                mStreamTypes[track->type()].mute) {
2187                vl = vr = va = 0;
2188                if (track->isPausing()) {
2189                    track->setPaused();
2190                }
2191            } else {
2192
2193                // read original volumes with volume control
2194                float typeVolume = mStreamTypes[track->type()].volume;
2195                float v = masterVolume * typeVolume;
2196                uint32_t vlr = cblk->volumeLR;
2197                vl = vlr & 0xFFFF;
2198                vr = vlr >> 16;
2199                // track volumes come from shared memory, so can't be trusted and must be clamped
2200                if (vl > MAX_GAIN_INT) {
2201                    ALOGV("Track left volume out of range: %04X", vl);
2202                    vl = MAX_GAIN_INT;
2203                }
2204                if (vr > MAX_GAIN_INT) {
2205                    ALOGV("Track right volume out of range: %04X", vr);
2206                    vr = MAX_GAIN_INT;
2207                }
2208                // now apply the master volume and stream type volume
2209                vl = (uint32_t)(v * vl) << 12;
2210                vr = (uint32_t)(v * vr) << 12;
2211                // assuming master volume and stream type volume each go up to 1.0,
2212                // vl and vr are now in 8.24 format
2213
2214                uint16_t sendLevel = cblk->getSendLevel_U4_12();
2215                // send level comes from shared memory and so may be corrupt
2216                if (sendLevel >= MAX_GAIN_INT) {
2217                    ALOGV("Track send level out of range: %04X", sendLevel);
2218                    sendLevel = MAX_GAIN_INT;
2219                }
2220                va = (uint32_t)(v * sendLevel);
2221            }
2222            // Delegate volume control to effect in track effect chain if needed
2223            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2224                // Do not ramp volume if volume is controlled by effect
2225                param = AudioMixer::VOLUME;
2226                track->mHasVolumeController = true;
2227            } else {
2228                // force no volume ramp when volume controller was just disabled or removed
2229                // from effect chain to avoid volume spike
2230                if (track->mHasVolumeController) {
2231                    param = AudioMixer::VOLUME;
2232                }
2233                track->mHasVolumeController = false;
2234            }
2235
2236            // Convert volumes from 8.24 to 4.12 format
2237            int16_t left, right, aux;
2238            // This additional clamping is needed in case chain->setVolume_l() overshot
2239            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2240            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2241            left = int16_t(v_clamped);
2242            v_clamped = (vr + (1 << 11)) >> 12;
2243            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2244            right = int16_t(v_clamped);
2245
2246            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2247            aux = int16_t(va);
2248
2249            // XXX: these things DON'T need to be done each time
2250            mAudioMixer->setBufferProvider(name, track);
2251            mAudioMixer->enable(name);
2252
2253            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left);
2254            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right);
2255            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux);
2256            mAudioMixer->setParameter(
2257                name,
2258                AudioMixer::TRACK,
2259                AudioMixer::FORMAT, (void *)track->format());
2260            mAudioMixer->setParameter(
2261                name,
2262                AudioMixer::TRACK,
2263                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2264            mAudioMixer->setParameter(
2265                name,
2266                AudioMixer::RESAMPLE,
2267                AudioMixer::SAMPLE_RATE,
2268                (void *)(cblk->sampleRate));
2269            mAudioMixer->setParameter(
2270                name,
2271                AudioMixer::TRACK,
2272                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2273            mAudioMixer->setParameter(
2274                name,
2275                AudioMixer::TRACK,
2276                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2277
2278            // reset retry count
2279            track->mRetryCount = kMaxTrackRetries;
2280            // If one track is ready, set the mixer ready if:
2281            //  - the mixer was not ready during previous round OR
2282            //  - no other track is not ready
2283            if (mPrevMixerStatus != MIXER_TRACKS_READY ||
2284                    mixerStatus != MIXER_TRACKS_ENABLED) {
2285                mixerStatus = MIXER_TRACKS_READY;
2286            }
2287        } else {
2288            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
2289            if (track->isStopped()) {
2290                track->reset();
2291            }
2292            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2293                // We have consumed all the buffers of this track.
2294                // Remove it from the list of active tracks.
2295                tracksToRemove->add(track);
2296            } else {
2297                // No buffers for this track. Give it a few chances to
2298                // fill a buffer, then remove it from active list.
2299                if (--(track->mRetryCount) <= 0) {
2300                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
2301                    tracksToRemove->add(track);
2302                    // indicate to client process that the track was disabled because of underrun
2303                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2304                // If one track is not ready, mark the mixer also not ready if:
2305                //  - the mixer was ready during previous round OR
2306                //  - no other track is ready
2307                } else if (mPrevMixerStatus == MIXER_TRACKS_READY ||
2308                                mixerStatus != MIXER_TRACKS_READY) {
2309                    mixerStatus = MIXER_TRACKS_ENABLED;
2310                }
2311            }
2312            mAudioMixer->disable(name);
2313        }
2314    }
2315
2316    // remove all the tracks that need to be...
2317    count = tracksToRemove->size();
2318    if (CC_UNLIKELY(count)) {
2319        for (size_t i=0 ; i<count ; i++) {
2320            const sp<Track>& track = tracksToRemove->itemAt(i);
2321            mActiveTracks.remove(track);
2322            if (track->mainBuffer() != mMixBuffer) {
2323                chain = getEffectChain_l(track->sessionId());
2324                if (chain != 0) {
2325                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2326                    chain->decActiveTrackCnt();
2327                }
2328            }
2329            if (track->isTerminated()) {
2330                removeTrack_l(track);
2331            }
2332        }
2333    }
2334
2335    // mix buffer must be cleared if all tracks are connected to an
2336    // effect chain as in this case the mixer will not write to
2337    // mix buffer and track effects will accumulate into it
2338    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2339        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2340    }
2341
2342    mPrevMixerStatus = mixerStatus;
2343    return mixerStatus;
2344}
2345
2346void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType)
2347{
2348    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2349            this,  streamType, mTracks.size());
2350    Mutex::Autolock _l(mLock);
2351
2352    size_t size = mTracks.size();
2353    for (size_t i = 0; i < size; i++) {
2354        sp<Track> t = mTracks[i];
2355        if (t->type() == streamType) {
2356            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2357            t->mCblk->cv.signal();
2358        }
2359    }
2360}
2361
2362void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid)
2363{
2364    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2365            this,  streamType, valid);
2366    Mutex::Autolock _l(mLock);
2367
2368    mStreamTypes[streamType].valid = valid;
2369}
2370
2371// getTrackName_l() must be called with ThreadBase::mLock held
2372int AudioFlinger::MixerThread::getTrackName_l()
2373{
2374    return mAudioMixer->getTrackName();
2375}
2376
2377// deleteTrackName_l() must be called with ThreadBase::mLock held
2378void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2379{
2380    ALOGV("remove track (%d) and delete from mixer", name);
2381    mAudioMixer->deleteTrackName(name);
2382}
2383
2384// checkForNewParameters_l() must be called with ThreadBase::mLock held
2385bool AudioFlinger::MixerThread::checkForNewParameters_l()
2386{
2387    bool reconfig = false;
2388
2389    while (!mNewParameters.isEmpty()) {
2390        status_t status = NO_ERROR;
2391        String8 keyValuePair = mNewParameters[0];
2392        AudioParameter param = AudioParameter(keyValuePair);
2393        int value;
2394
2395        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2396            reconfig = true;
2397        }
2398        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2399            if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
2400                status = BAD_VALUE;
2401            } else {
2402                reconfig = true;
2403            }
2404        }
2405        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2406            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2407                status = BAD_VALUE;
2408            } else {
2409                reconfig = true;
2410            }
2411        }
2412        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2413            // do not accept frame count changes if tracks are open as the track buffer
2414            // size depends on frame count and correct behavior would not be guaranteed
2415            // if frame count is changed after track creation
2416            if (!mTracks.isEmpty()) {
2417                status = INVALID_OPERATION;
2418            } else {
2419                reconfig = true;
2420            }
2421        }
2422        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2423            // when changing the audio output device, call addBatteryData to notify
2424            // the change
2425            if ((int)mDevice != value) {
2426                uint32_t params = 0;
2427                // check whether speaker is on
2428                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2429                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2430                }
2431
2432                int deviceWithoutSpeaker
2433                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2434                // check if any other device (except speaker) is on
2435                if (value & deviceWithoutSpeaker ) {
2436                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2437                }
2438
2439                if (params != 0) {
2440                    addBatteryData(params);
2441                }
2442            }
2443
2444            // forward device change to effects that have requested to be
2445            // aware of attached audio device.
2446            mDevice = (uint32_t)value;
2447            for (size_t i = 0; i < mEffectChains.size(); i++) {
2448                mEffectChains[i]->setDevice_l(mDevice);
2449            }
2450        }
2451
2452        if (status == NO_ERROR) {
2453            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2454                                                    keyValuePair.string());
2455            if (!mStandby && status == INVALID_OPERATION) {
2456               mOutput->stream->common.standby(&mOutput->stream->common);
2457               mStandby = true;
2458               mBytesWritten = 0;
2459               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2460                                                       keyValuePair.string());
2461            }
2462            if (status == NO_ERROR && reconfig) {
2463                delete mAudioMixer;
2464                readOutputParameters();
2465                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2466                for (size_t i = 0; i < mTracks.size() ; i++) {
2467                    int name = getTrackName_l();
2468                    if (name < 0) break;
2469                    mTracks[i]->mName = name;
2470                    // limit track sample rate to 2 x new output sample rate
2471                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2472                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2473                    }
2474                }
2475                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2476            }
2477        }
2478
2479        mNewParameters.removeAt(0);
2480
2481        mParamStatus = status;
2482        mParamCond.signal();
2483        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2484        // already timed out waiting for the status and will never signal the condition.
2485        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2486    }
2487    return reconfig;
2488}
2489
2490status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2491{
2492    const size_t SIZE = 256;
2493    char buffer[SIZE];
2494    String8 result;
2495
2496    PlaybackThread::dumpInternals(fd, args);
2497
2498    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2499    result.append(buffer);
2500    write(fd, result.string(), result.size());
2501    return NO_ERROR;
2502}
2503
2504uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2505{
2506    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2507}
2508
2509uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2510{
2511    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2512}
2513
2514// ----------------------------------------------------------------------------
2515AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2516    :   PlaybackThread(audioFlinger, output, id, device)
2517{
2518    mType = ThreadBase::DIRECT;
2519}
2520
2521AudioFlinger::DirectOutputThread::~DirectOutputThread()
2522{
2523}
2524
2525static inline
2526int32_t mul(int16_t in, int16_t v)
2527{
2528#if defined(__arm__) && !defined(__thumb__)
2529    int32_t out;
2530    asm( "smulbb %[out], %[in], %[v] \n"
2531         : [out]"=r"(out)
2532         : [in]"%r"(in), [v]"r"(v)
2533         : );
2534    return out;
2535#else
2536    return in * int32_t(v);
2537#endif
2538}
2539
2540void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2541{
2542    // Do not apply volume on compressed audio
2543    if (!audio_is_linear_pcm(mFormat)) {
2544        return;
2545    }
2546
2547    // convert to signed 16 bit before volume calculation
2548    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2549        size_t count = mFrameCount * mChannelCount;
2550        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2551        int16_t *dst = mMixBuffer + count-1;
2552        while(count--) {
2553            *dst-- = (int16_t)(*src--^0x80) << 8;
2554        }
2555    }
2556
2557    size_t frameCount = mFrameCount;
2558    int16_t *out = mMixBuffer;
2559    if (ramp) {
2560        if (mChannelCount == 1) {
2561            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2562            int32_t vlInc = d / (int32_t)frameCount;
2563            int32_t vl = ((int32_t)mLeftVolShort << 16);
2564            do {
2565                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2566                out++;
2567                vl += vlInc;
2568            } while (--frameCount);
2569
2570        } else {
2571            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2572            int32_t vlInc = d / (int32_t)frameCount;
2573            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2574            int32_t vrInc = d / (int32_t)frameCount;
2575            int32_t vl = ((int32_t)mLeftVolShort << 16);
2576            int32_t vr = ((int32_t)mRightVolShort << 16);
2577            do {
2578                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2579                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2580                out += 2;
2581                vl += vlInc;
2582                vr += vrInc;
2583            } while (--frameCount);
2584        }
2585    } else {
2586        if (mChannelCount == 1) {
2587            do {
2588                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2589                out++;
2590            } while (--frameCount);
2591        } else {
2592            do {
2593                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2594                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2595                out += 2;
2596            } while (--frameCount);
2597        }
2598    }
2599
2600    // convert back to unsigned 8 bit after volume calculation
2601    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2602        size_t count = mFrameCount * mChannelCount;
2603        int16_t *src = mMixBuffer;
2604        uint8_t *dst = (uint8_t *)mMixBuffer;
2605        while(count--) {
2606            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2607        }
2608    }
2609
2610    mLeftVolShort = leftVol;
2611    mRightVolShort = rightVol;
2612}
2613
2614bool AudioFlinger::DirectOutputThread::threadLoop()
2615{
2616    mixer_state mixerStatus = MIXER_IDLE;
2617    sp<Track> trackToRemove;
2618    sp<Track> activeTrack;
2619    nsecs_t standbyTime = systemTime();
2620    int8_t *curBuf;
2621    size_t mixBufferSize = mFrameCount*mFrameSize;
2622    uint32_t activeSleepTime = activeSleepTimeUs();
2623    uint32_t idleSleepTime = idleSleepTimeUs();
2624    uint32_t sleepTime = idleSleepTime;
2625    // use shorter standby delay as on normal output to release
2626    // hardware resources as soon as possible
2627    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2628
2629    acquireWakeLock();
2630
2631    while (!exitPending())
2632    {
2633        bool rampVolume;
2634        uint16_t leftVol;
2635        uint16_t rightVol;
2636        Vector< sp<EffectChain> > effectChains;
2637
2638        processConfigEvents();
2639
2640        mixerStatus = MIXER_IDLE;
2641
2642        { // scope for the mLock
2643
2644            Mutex::Autolock _l(mLock);
2645
2646            if (checkForNewParameters_l()) {
2647                mixBufferSize = mFrameCount*mFrameSize;
2648                activeSleepTime = activeSleepTimeUs();
2649                idleSleepTime = idleSleepTimeUs();
2650                standbyDelay = microseconds(activeSleepTime*2);
2651            }
2652
2653            // put audio hardware into standby after short delay
2654            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2655                        mSuspended)) {
2656                // wait until we have something to do...
2657                if (!mStandby) {
2658                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2659                    mOutput->stream->common.standby(&mOutput->stream->common);
2660                    mStandby = true;
2661                    mBytesWritten = 0;
2662                }
2663
2664                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2665                    // we're about to wait, flush the binder command buffer
2666                    IPCThreadState::self()->flushCommands();
2667
2668                    if (exitPending()) break;
2669
2670                    releaseWakeLock_l();
2671                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2672                    mWaitWorkCV.wait(mLock);
2673                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2674                    acquireWakeLock_l();
2675
2676                    if (!mMasterMute) {
2677                        char value[PROPERTY_VALUE_MAX];
2678                        property_get("ro.audio.silent", value, "0");
2679                        if (atoi(value)) {
2680                            ALOGD("Silence is golden");
2681                            setMasterMute(true);
2682                        }
2683                    }
2684
2685                    standbyTime = systemTime() + standbyDelay;
2686                    sleepTime = idleSleepTime;
2687                    continue;
2688                }
2689            }
2690
2691            effectChains = mEffectChains;
2692
2693            // find out which tracks need to be processed
2694            if (mActiveTracks.size() != 0) {
2695                sp<Track> t = mActiveTracks[0].promote();
2696                if (t == 0) continue;
2697
2698                Track* const track = t.get();
2699                audio_track_cblk_t* cblk = track->cblk();
2700
2701                // The first time a track is added we wait
2702                // for all its buffers to be filled before processing it
2703                if (cblk->framesReady() && track->isReady() &&
2704                        !track->isPaused() && !track->isTerminated())
2705                {
2706                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2707
2708                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2709                        track->mFillingUpStatus = Track::FS_ACTIVE;
2710                        mLeftVolFloat = mRightVolFloat = 0;
2711                        mLeftVolShort = mRightVolShort = 0;
2712                        if (track->mState == TrackBase::RESUMING) {
2713                            track->mState = TrackBase::ACTIVE;
2714                            rampVolume = true;
2715                        }
2716                    } else if (cblk->server != 0) {
2717                        // If the track is stopped before the first frame was mixed,
2718                        // do not apply ramp
2719                        rampVolume = true;
2720                    }
2721                    // compute volume for this track
2722                    float left, right;
2723                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2724                        mStreamTypes[track->type()].mute) {
2725                        left = right = 0;
2726                        if (track->isPausing()) {
2727                            track->setPaused();
2728                        }
2729                    } else {
2730                        float typeVolume = mStreamTypes[track->type()].volume;
2731                        float v = mMasterVolume * typeVolume;
2732                        uint32_t vlr = cblk->volumeLR;
2733                        float v_clamped = v * (vlr & 0xFFFF);
2734                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2735                        left = v_clamped/MAX_GAIN;
2736                        v_clamped = v * (vlr >> 16);
2737                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2738                        right = v_clamped/MAX_GAIN;
2739                    }
2740
2741                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2742                        mLeftVolFloat = left;
2743                        mRightVolFloat = right;
2744
2745                        // If audio HAL implements volume control,
2746                        // force software volume to nominal value
2747                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2748                            left = 1.0f;
2749                            right = 1.0f;
2750                        }
2751
2752                        // Convert volumes from float to 8.24
2753                        uint32_t vl = (uint32_t)(left * (1 << 24));
2754                        uint32_t vr = (uint32_t)(right * (1 << 24));
2755
2756                        // Delegate volume control to effect in track effect chain if needed
2757                        // only one effect chain can be present on DirectOutputThread, so if
2758                        // there is one, the track is connected to it
2759                        if (!effectChains.isEmpty()) {
2760                            // Do not ramp volume if volume is controlled by effect
2761                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2762                                rampVolume = false;
2763                            }
2764                        }
2765
2766                        // Convert volumes from 8.24 to 4.12 format
2767                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2768                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2769                        leftVol = (uint16_t)v_clamped;
2770                        v_clamped = (vr + (1 << 11)) >> 12;
2771                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2772                        rightVol = (uint16_t)v_clamped;
2773                    } else {
2774                        leftVol = mLeftVolShort;
2775                        rightVol = mRightVolShort;
2776                        rampVolume = false;
2777                    }
2778
2779                    // reset retry count
2780                    track->mRetryCount = kMaxTrackRetriesDirect;
2781                    activeTrack = t;
2782                    mixerStatus = MIXER_TRACKS_READY;
2783                } else {
2784                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2785                    if (track->isStopped()) {
2786                        track->reset();
2787                    }
2788                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2789                        // We have consumed all the buffers of this track.
2790                        // Remove it from the list of active tracks.
2791                        trackToRemove = track;
2792                    } else {
2793                        // No buffers for this track. Give it a few chances to
2794                        // fill a buffer, then remove it from active list.
2795                        if (--(track->mRetryCount) <= 0) {
2796                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2797                            trackToRemove = track;
2798                        } else {
2799                            mixerStatus = MIXER_TRACKS_ENABLED;
2800                        }
2801                    }
2802                }
2803            }
2804
2805            // remove all the tracks that need to be...
2806            if (CC_UNLIKELY(trackToRemove != 0)) {
2807                mActiveTracks.remove(trackToRemove);
2808                if (!effectChains.isEmpty()) {
2809                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2810                            trackToRemove->sessionId());
2811                    effectChains[0]->decActiveTrackCnt();
2812                }
2813                if (trackToRemove->isTerminated()) {
2814                    removeTrack_l(trackToRemove);
2815                }
2816            }
2817
2818            lockEffectChains_l(effectChains);
2819       }
2820
2821        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2822            AudioBufferProvider::Buffer buffer;
2823            size_t frameCount = mFrameCount;
2824            curBuf = (int8_t *)mMixBuffer;
2825            // output audio to hardware
2826            while (frameCount) {
2827                buffer.frameCount = frameCount;
2828                activeTrack->getNextBuffer(&buffer);
2829                if (CC_UNLIKELY(buffer.raw == NULL)) {
2830                    memset(curBuf, 0, frameCount * mFrameSize);
2831                    break;
2832                }
2833                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2834                frameCount -= buffer.frameCount;
2835                curBuf += buffer.frameCount * mFrameSize;
2836                activeTrack->releaseBuffer(&buffer);
2837            }
2838            sleepTime = 0;
2839            standbyTime = systemTime() + standbyDelay;
2840        } else {
2841            if (sleepTime == 0) {
2842                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2843                    sleepTime = activeSleepTime;
2844                } else {
2845                    sleepTime = idleSleepTime;
2846                }
2847            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2848                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2849                sleepTime = 0;
2850            }
2851        }
2852
2853        if (mSuspended) {
2854            sleepTime = suspendSleepTimeUs();
2855        }
2856        // sleepTime == 0 means we must write to audio hardware
2857        if (sleepTime == 0) {
2858            if (mixerStatus == MIXER_TRACKS_READY) {
2859                applyVolume(leftVol, rightVol, rampVolume);
2860            }
2861            for (size_t i = 0; i < effectChains.size(); i ++) {
2862                effectChains[i]->process_l();
2863            }
2864            unlockEffectChains(effectChains);
2865
2866            mLastWriteTime = systemTime();
2867            mInWrite = true;
2868            mBytesWritten += mixBufferSize;
2869            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2870            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2871            mNumWrites++;
2872            mInWrite = false;
2873            mStandby = false;
2874        } else {
2875            unlockEffectChains(effectChains);
2876            usleep(sleepTime);
2877        }
2878
2879        // finally let go of removed track, without the lock held
2880        // since we can't guarantee the destructors won't acquire that
2881        // same lock.
2882        trackToRemove.clear();
2883        activeTrack.clear();
2884
2885        // Effect chains will be actually deleted here if they were removed from
2886        // mEffectChains list during mixing or effects processing
2887        effectChains.clear();
2888    }
2889
2890    if (!mStandby) {
2891        mOutput->stream->common.standby(&mOutput->stream->common);
2892    }
2893
2894    releaseWakeLock();
2895
2896    ALOGV("DirectOutputThread %p exiting", this);
2897    return false;
2898}
2899
2900// getTrackName_l() must be called with ThreadBase::mLock held
2901int AudioFlinger::DirectOutputThread::getTrackName_l()
2902{
2903    return 0;
2904}
2905
2906// deleteTrackName_l() must be called with ThreadBase::mLock held
2907void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2908{
2909}
2910
2911// checkForNewParameters_l() must be called with ThreadBase::mLock held
2912bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2913{
2914    bool reconfig = false;
2915
2916    while (!mNewParameters.isEmpty()) {
2917        status_t status = NO_ERROR;
2918        String8 keyValuePair = mNewParameters[0];
2919        AudioParameter param = AudioParameter(keyValuePair);
2920        int value;
2921
2922        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2923            // do not accept frame count changes if tracks are open as the track buffer
2924            // size depends on frame count and correct behavior would not be garantied
2925            // if frame count is changed after track creation
2926            if (!mTracks.isEmpty()) {
2927                status = INVALID_OPERATION;
2928            } else {
2929                reconfig = true;
2930            }
2931        }
2932        if (status == NO_ERROR) {
2933            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2934                                                    keyValuePair.string());
2935            if (!mStandby && status == INVALID_OPERATION) {
2936               mOutput->stream->common.standby(&mOutput->stream->common);
2937               mStandby = true;
2938               mBytesWritten = 0;
2939               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2940                                                       keyValuePair.string());
2941            }
2942            if (status == NO_ERROR && reconfig) {
2943                readOutputParameters();
2944                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2945            }
2946        }
2947
2948        mNewParameters.removeAt(0);
2949
2950        mParamStatus = status;
2951        mParamCond.signal();
2952        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2953        // already timed out waiting for the status and will never signal the condition.
2954        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2955    }
2956    return reconfig;
2957}
2958
2959uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2960{
2961    uint32_t time;
2962    if (audio_is_linear_pcm(mFormat)) {
2963        time = PlaybackThread::activeSleepTimeUs();
2964    } else {
2965        time = 10000;
2966    }
2967    return time;
2968}
2969
2970uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2971{
2972    uint32_t time;
2973    if (audio_is_linear_pcm(mFormat)) {
2974        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2975    } else {
2976        time = 10000;
2977    }
2978    return time;
2979}
2980
2981uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2982{
2983    uint32_t time;
2984    if (audio_is_linear_pcm(mFormat)) {
2985        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2986    } else {
2987        time = 10000;
2988    }
2989    return time;
2990}
2991
2992
2993// ----------------------------------------------------------------------------
2994
2995AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2996    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2997{
2998    mType = ThreadBase::DUPLICATING;
2999    addOutputTrack(mainThread);
3000}
3001
3002AudioFlinger::DuplicatingThread::~DuplicatingThread()
3003{
3004    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3005        mOutputTracks[i]->destroy();
3006    }
3007    mOutputTracks.clear();
3008}
3009
3010bool AudioFlinger::DuplicatingThread::threadLoop()
3011{
3012    Vector< sp<Track> > tracksToRemove;
3013    mixer_state mixerStatus = MIXER_IDLE;
3014    nsecs_t standbyTime = systemTime();
3015    size_t mixBufferSize = mFrameCount*mFrameSize;
3016    SortedVector< sp<OutputTrack> > outputTracks;
3017    uint32_t writeFrames = 0;
3018    uint32_t activeSleepTime = activeSleepTimeUs();
3019    uint32_t idleSleepTime = idleSleepTimeUs();
3020    uint32_t sleepTime = idleSleepTime;
3021    Vector< sp<EffectChain> > effectChains;
3022
3023    acquireWakeLock();
3024
3025    while (!exitPending())
3026    {
3027        processConfigEvents();
3028
3029        mixerStatus = MIXER_IDLE;
3030        { // scope for the mLock
3031
3032            Mutex::Autolock _l(mLock);
3033
3034            if (checkForNewParameters_l()) {
3035                mixBufferSize = mFrameCount*mFrameSize;
3036                updateWaitTime();
3037                activeSleepTime = activeSleepTimeUs();
3038                idleSleepTime = idleSleepTimeUs();
3039            }
3040
3041            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
3042
3043            for (size_t i = 0; i < mOutputTracks.size(); i++) {
3044                outputTracks.add(mOutputTracks[i]);
3045            }
3046
3047            // put audio hardware into standby after short delay
3048            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
3049                         mSuspended)) {
3050                if (!mStandby) {
3051                    for (size_t i = 0; i < outputTracks.size(); i++) {
3052                        outputTracks[i]->stop();
3053                    }
3054                    mStandby = true;
3055                    mBytesWritten = 0;
3056                }
3057
3058                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
3059                    // we're about to wait, flush the binder command buffer
3060                    IPCThreadState::self()->flushCommands();
3061                    outputTracks.clear();
3062
3063                    if (exitPending()) break;
3064
3065                    releaseWakeLock_l();
3066                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3067                    mWaitWorkCV.wait(mLock);
3068                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3069                    acquireWakeLock_l();
3070
3071                    mPrevMixerStatus = MIXER_IDLE;
3072                    if (!mMasterMute) {
3073                        char value[PROPERTY_VALUE_MAX];
3074                        property_get("ro.audio.silent", value, "0");
3075                        if (atoi(value)) {
3076                            ALOGD("Silence is golden");
3077                            setMasterMute(true);
3078                        }
3079                    }
3080
3081                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3082                    sleepTime = idleSleepTime;
3083                    continue;
3084                }
3085            }
3086
3087            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3088
3089            // prevent any changes in effect chain list and in each effect chain
3090            // during mixing and effect process as the audio buffers could be deleted
3091            // or modified if an effect is created or deleted
3092            lockEffectChains_l(effectChains);
3093        }
3094
3095        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3096            // mix buffers...
3097            if (outputsReady(outputTracks)) {
3098                mAudioMixer->process();
3099            } else {
3100                memset(mMixBuffer, 0, mixBufferSize);
3101            }
3102            sleepTime = 0;
3103            writeFrames = mFrameCount;
3104        } else {
3105            if (sleepTime == 0) {
3106                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3107                    sleepTime = activeSleepTime;
3108                } else {
3109                    sleepTime = idleSleepTime;
3110                }
3111            } else if (mBytesWritten != 0) {
3112                // flush remaining overflow buffers in output tracks
3113                for (size_t i = 0; i < outputTracks.size(); i++) {
3114                    if (outputTracks[i]->isActive()) {
3115                        sleepTime = 0;
3116                        writeFrames = 0;
3117                        memset(mMixBuffer, 0, mixBufferSize);
3118                        break;
3119                    }
3120                }
3121            }
3122        }
3123
3124        if (mSuspended) {
3125            sleepTime = suspendSleepTimeUs();
3126        }
3127        // sleepTime == 0 means we must write to audio hardware
3128        if (sleepTime == 0) {
3129            for (size_t i = 0; i < effectChains.size(); i ++) {
3130                effectChains[i]->process_l();
3131            }
3132            // enable changes in effect chain
3133            unlockEffectChains(effectChains);
3134
3135            standbyTime = systemTime() + kStandbyTimeInNsecs;
3136            for (size_t i = 0; i < outputTracks.size(); i++) {
3137                outputTracks[i]->write(mMixBuffer, writeFrames);
3138            }
3139            mStandby = false;
3140            mBytesWritten += mixBufferSize;
3141        } else {
3142            // enable changes in effect chain
3143            unlockEffectChains(effectChains);
3144            usleep(sleepTime);
3145        }
3146
3147        // finally let go of all our tracks, without the lock held
3148        // since we can't guarantee the destructors won't acquire that
3149        // same lock.
3150        tracksToRemove.clear();
3151        outputTracks.clear();
3152
3153        // Effect chains will be actually deleted here if they were removed from
3154        // mEffectChains list during mixing or effects processing
3155        effectChains.clear();
3156    }
3157
3158    releaseWakeLock();
3159
3160    return false;
3161}
3162
3163void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3164{
3165    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3166    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3167                                            this,
3168                                            mSampleRate,
3169                                            mFormat,
3170                                            mChannelMask,
3171                                            frameCount);
3172    if (outputTrack->cblk() != NULL) {
3173        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3174        mOutputTracks.add(outputTrack);
3175        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3176        updateWaitTime();
3177    }
3178}
3179
3180void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3181{
3182    Mutex::Autolock _l(mLock);
3183    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3184        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3185            mOutputTracks[i]->destroy();
3186            mOutputTracks.removeAt(i);
3187            updateWaitTime();
3188            return;
3189        }
3190    }
3191    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3192}
3193
3194void AudioFlinger::DuplicatingThread::updateWaitTime()
3195{
3196    mWaitTimeMs = UINT_MAX;
3197    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3198        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3199        if (strong != NULL) {
3200            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3201            if (waitTimeMs < mWaitTimeMs) {
3202                mWaitTimeMs = waitTimeMs;
3203            }
3204        }
3205    }
3206}
3207
3208
3209bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3210{
3211    for (size_t i = 0; i < outputTracks.size(); i++) {
3212        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3213        if (thread == 0) {
3214            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3215            return false;
3216        }
3217        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3218        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3219            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3220            return false;
3221        }
3222    }
3223    return true;
3224}
3225
3226uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3227{
3228    return (mWaitTimeMs * 1000) / 2;
3229}
3230
3231// ----------------------------------------------------------------------------
3232
3233// TrackBase constructor must be called with AudioFlinger::mLock held
3234AudioFlinger::ThreadBase::TrackBase::TrackBase(
3235            const wp<ThreadBase>& thread,
3236            const sp<Client>& client,
3237            uint32_t sampleRate,
3238            audio_format_t format,
3239            uint32_t channelMask,
3240            int frameCount,
3241            uint32_t flags,
3242            const sp<IMemory>& sharedBuffer,
3243            int sessionId)
3244    :   RefBase(),
3245        mThread(thread),
3246        mClient(client),
3247        mCblk(0),
3248        mFrameCount(0),
3249        mState(IDLE),
3250        mClientTid(-1),
3251        mFormat(format),
3252        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3253        mSessionId(sessionId)
3254{
3255    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3256
3257    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3258   size_t size = sizeof(audio_track_cblk_t);
3259   uint8_t channelCount = popcount(channelMask);
3260   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3261   if (sharedBuffer == 0) {
3262       size += bufferSize;
3263   }
3264
3265   if (client != NULL) {
3266        mCblkMemory = client->heap()->allocate(size);
3267        if (mCblkMemory != 0) {
3268            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3269            if (mCblk) { // construct the shared structure in-place.
3270                new(mCblk) audio_track_cblk_t();
3271                // clear all buffers
3272                mCblk->frameCount = frameCount;
3273                mCblk->sampleRate = sampleRate;
3274                mChannelCount = channelCount;
3275                mChannelMask = channelMask;
3276                if (sharedBuffer == 0) {
3277                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3278                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3279                    // Force underrun condition to avoid false underrun callback until first data is
3280                    // written to buffer (other flags are cleared)
3281                    mCblk->flags = CBLK_UNDERRUN_ON;
3282                } else {
3283                    mBuffer = sharedBuffer->pointer();
3284                }
3285                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3286            }
3287        } else {
3288            ALOGE("not enough memory for AudioTrack size=%u", size);
3289            client->heap()->dump("AudioTrack");
3290            return;
3291        }
3292   } else {
3293       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3294           // construct the shared structure in-place.
3295           new(mCblk) audio_track_cblk_t();
3296           // clear all buffers
3297           mCblk->frameCount = frameCount;
3298           mCblk->sampleRate = sampleRate;
3299           mChannelCount = channelCount;
3300           mChannelMask = channelMask;
3301           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3302           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3303           // Force underrun condition to avoid false underrun callback until first data is
3304           // written to buffer (other flags are cleared)
3305           mCblk->flags = CBLK_UNDERRUN_ON;
3306           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3307   }
3308}
3309
3310AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3311{
3312    if (mCblk) {
3313        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3314        if (mClient == NULL) {
3315            delete mCblk;
3316        }
3317    }
3318    mCblkMemory.clear();            // and free the shared memory
3319    if (mClient != NULL) {
3320        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3321        mClient.clear();
3322    }
3323}
3324
3325void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3326{
3327    buffer->raw = NULL;
3328    mFrameCount = buffer->frameCount;
3329    step();
3330    buffer->frameCount = 0;
3331}
3332
3333bool AudioFlinger::ThreadBase::TrackBase::step() {
3334    bool result;
3335    audio_track_cblk_t* cblk = this->cblk();
3336
3337    result = cblk->stepServer(mFrameCount);
3338    if (!result) {
3339        ALOGV("stepServer failed acquiring cblk mutex");
3340        mFlags |= STEPSERVER_FAILED;
3341    }
3342    return result;
3343}
3344
3345void AudioFlinger::ThreadBase::TrackBase::reset() {
3346    audio_track_cblk_t* cblk = this->cblk();
3347
3348    cblk->user = 0;
3349    cblk->server = 0;
3350    cblk->userBase = 0;
3351    cblk->serverBase = 0;
3352    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3353    ALOGV("TrackBase::reset");
3354}
3355
3356sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3357{
3358    return mCblkMemory;
3359}
3360
3361int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3362    return (int)mCblk->sampleRate;
3363}
3364
3365int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3366    return (const int)mChannelCount;
3367}
3368
3369uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3370    return mChannelMask;
3371}
3372
3373void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3374    audio_track_cblk_t* cblk = this->cblk();
3375    size_t frameSize = cblk->frameSize;
3376    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
3377    int8_t *bufferEnd = bufferStart + frames * frameSize;
3378
3379    // Check validity of returned pointer in case the track control block would have been corrupted.
3380    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3381        ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) {
3382        ALOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3383                server %d, serverBase %d, user %d, userBase %d",
3384                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3385                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3386        return 0;
3387    }
3388
3389    return bufferStart;
3390}
3391
3392// ----------------------------------------------------------------------------
3393
3394// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3395AudioFlinger::PlaybackThread::Track::Track(
3396            const wp<ThreadBase>& thread,
3397            const sp<Client>& client,
3398            audio_stream_type_t streamType,
3399            uint32_t sampleRate,
3400            audio_format_t format,
3401            uint32_t channelMask,
3402            int frameCount,
3403            const sp<IMemory>& sharedBuffer,
3404            int sessionId)
3405    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3406    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3407    mAuxEffectId(0), mHasVolumeController(false)
3408{
3409    if (mCblk != NULL) {
3410        sp<ThreadBase> baseThread = thread.promote();
3411        if (baseThread != 0) {
3412            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3413            mName = playbackThread->getTrackName_l();
3414            mMainBuffer = playbackThread->mixBuffer();
3415        }
3416        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3417        if (mName < 0) {
3418            ALOGE("no more track names available");
3419        }
3420        mStreamType = streamType;
3421        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3422        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3423        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3424    }
3425}
3426
3427AudioFlinger::PlaybackThread::Track::~Track()
3428{
3429    ALOGV("PlaybackThread::Track destructor");
3430    sp<ThreadBase> thread = mThread.promote();
3431    if (thread != 0) {
3432        Mutex::Autolock _l(thread->mLock);
3433        mState = TERMINATED;
3434    }
3435}
3436
3437void AudioFlinger::PlaybackThread::Track::destroy()
3438{
3439    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3440    // by removing it from mTracks vector, so there is a risk that this Tracks's
3441    // desctructor is called. As the destructor needs to lock mLock,
3442    // we must acquire a strong reference on this Track before locking mLock
3443    // here so that the destructor is called only when exiting this function.
3444    // On the other hand, as long as Track::destroy() is only called by
3445    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3446    // this Track with its member mTrack.
3447    sp<Track> keep(this);
3448    { // scope for mLock
3449        sp<ThreadBase> thread = mThread.promote();
3450        if (thread != 0) {
3451            if (!isOutputTrack()) {
3452                if (mState == ACTIVE || mState == RESUMING) {
3453                    AudioSystem::stopOutput(thread->id(),
3454                                            (audio_stream_type_t)mStreamType,
3455                                            mSessionId);
3456
3457                    // to track the speaker usage
3458                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3459                }
3460                AudioSystem::releaseOutput(thread->id());
3461            }
3462            Mutex::Autolock _l(thread->mLock);
3463            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3464            playbackThread->destroyTrack_l(this);
3465        }
3466    }
3467}
3468
3469void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3470{
3471    uint32_t vlr = mCblk->volumeLR;
3472    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",
3473            mName - AudioMixer::TRACK0,
3474            (mClient == NULL) ? getpid() : mClient->pid(),
3475            mStreamType,
3476            mFormat,
3477            mChannelMask,
3478            mSessionId,
3479            mFrameCount,
3480            mState,
3481            mMute,
3482            mFillingUpStatus,
3483            mCblk->sampleRate,
3484            vlr & 0xFFFF,
3485            vlr >> 16,
3486            mCblk->server,
3487            mCblk->user,
3488            (int)mMainBuffer,
3489            (int)mAuxBuffer);
3490}
3491
3492status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3493{
3494     audio_track_cblk_t* cblk = this->cblk();
3495     uint32_t framesReady;
3496     uint32_t framesReq = buffer->frameCount;
3497
3498     // Check if last stepServer failed, try to step now
3499     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3500         if (!step())  goto getNextBuffer_exit;
3501         ALOGV("stepServer recovered");
3502         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3503     }
3504
3505     framesReady = cblk->framesReady();
3506
3507     if (CC_LIKELY(framesReady)) {
3508        uint32_t s = cblk->server;
3509        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3510
3511        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3512        if (framesReq > framesReady) {
3513            framesReq = framesReady;
3514        }
3515        if (s + framesReq > bufferEnd) {
3516            framesReq = bufferEnd - s;
3517        }
3518
3519         buffer->raw = getBuffer(s, framesReq);
3520         if (buffer->raw == NULL) goto getNextBuffer_exit;
3521
3522         buffer->frameCount = framesReq;
3523        return NO_ERROR;
3524     }
3525
3526getNextBuffer_exit:
3527     buffer->raw = NULL;
3528     buffer->frameCount = 0;
3529     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3530     return NOT_ENOUGH_DATA;
3531}
3532
3533bool AudioFlinger::PlaybackThread::Track::isReady() const {
3534    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3535
3536    if (mCblk->framesReady() >= mCblk->frameCount ||
3537            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3538        mFillingUpStatus = FS_FILLED;
3539        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3540        return true;
3541    }
3542    return false;
3543}
3544
3545status_t AudioFlinger::PlaybackThread::Track::start()
3546{
3547    status_t status = NO_ERROR;
3548    ALOGV("start(%d), calling thread %d session %d",
3549            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3550    sp<ThreadBase> thread = mThread.promote();
3551    if (thread != 0) {
3552        Mutex::Autolock _l(thread->mLock);
3553        int state = mState;
3554        // here the track could be either new, or restarted
3555        // in both cases "unstop" the track
3556        if (mState == PAUSED) {
3557            mState = TrackBase::RESUMING;
3558            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3559        } else {
3560            mState = TrackBase::ACTIVE;
3561            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3562        }
3563
3564        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3565            thread->mLock.unlock();
3566            status = AudioSystem::startOutput(thread->id(),
3567                                              (audio_stream_type_t)mStreamType,
3568                                              mSessionId);
3569            thread->mLock.lock();
3570
3571            // to track the speaker usage
3572            if (status == NO_ERROR) {
3573                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3574            }
3575        }
3576        if (status == NO_ERROR) {
3577            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3578            playbackThread->addTrack_l(this);
3579        } else {
3580            mState = state;
3581        }
3582    } else {
3583        status = BAD_VALUE;
3584    }
3585    return status;
3586}
3587
3588void AudioFlinger::PlaybackThread::Track::stop()
3589{
3590    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3591    sp<ThreadBase> thread = mThread.promote();
3592    if (thread != 0) {
3593        Mutex::Autolock _l(thread->mLock);
3594        int state = mState;
3595        if (mState > STOPPED) {
3596            mState = STOPPED;
3597            // If the track is not active (PAUSED and buffers full), flush buffers
3598            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3599            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3600                reset();
3601            }
3602            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3603        }
3604        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3605            thread->mLock.unlock();
3606            AudioSystem::stopOutput(thread->id(),
3607                                    (audio_stream_type_t)mStreamType,
3608                                    mSessionId);
3609            thread->mLock.lock();
3610
3611            // to track the speaker usage
3612            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3613        }
3614    }
3615}
3616
3617void AudioFlinger::PlaybackThread::Track::pause()
3618{
3619    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3620    sp<ThreadBase> thread = mThread.promote();
3621    if (thread != 0) {
3622        Mutex::Autolock _l(thread->mLock);
3623        if (mState == ACTIVE || mState == RESUMING) {
3624            mState = PAUSING;
3625            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3626            if (!isOutputTrack()) {
3627                thread->mLock.unlock();
3628                AudioSystem::stopOutput(thread->id(),
3629                                        (audio_stream_type_t)mStreamType,
3630                                        mSessionId);
3631                thread->mLock.lock();
3632
3633                // to track the speaker usage
3634                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3635            }
3636        }
3637    }
3638}
3639
3640void AudioFlinger::PlaybackThread::Track::flush()
3641{
3642    ALOGV("flush(%d)", mName);
3643    sp<ThreadBase> thread = mThread.promote();
3644    if (thread != 0) {
3645        Mutex::Autolock _l(thread->mLock);
3646        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3647            return;
3648        }
3649        // No point remaining in PAUSED state after a flush => go to
3650        // STOPPED state
3651        mState = STOPPED;
3652
3653        // do not reset the track if it is still in the process of being stopped or paused.
3654        // this will be done by prepareTracks_l() when the track is stopped.
3655        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3656        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3657            reset();
3658        }
3659    }
3660}
3661
3662void AudioFlinger::PlaybackThread::Track::reset()
3663{
3664    // Do not reset twice to avoid discarding data written just after a flush and before
3665    // the audioflinger thread detects the track is stopped.
3666    if (!mResetDone) {
3667        TrackBase::reset();
3668        // Force underrun condition to avoid false underrun callback until first data is
3669        // written to buffer
3670        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3671        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3672        mFillingUpStatus = FS_FILLING;
3673        mResetDone = true;
3674    }
3675}
3676
3677void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3678{
3679    mMute = muted;
3680}
3681
3682status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3683{
3684    status_t status = DEAD_OBJECT;
3685    sp<ThreadBase> thread = mThread.promote();
3686    if (thread != 0) {
3687       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3688       status = playbackThread->attachAuxEffect(this, EffectId);
3689    }
3690    return status;
3691}
3692
3693void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3694{
3695    mAuxEffectId = EffectId;
3696    mAuxBuffer = buffer;
3697}
3698
3699// ----------------------------------------------------------------------------
3700
3701// RecordTrack constructor must be called with AudioFlinger::mLock held
3702AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3703            const wp<ThreadBase>& thread,
3704            const sp<Client>& client,
3705            uint32_t sampleRate,
3706            audio_format_t format,
3707            uint32_t channelMask,
3708            int frameCount,
3709            uint32_t flags,
3710            int sessionId)
3711    :   TrackBase(thread, client, sampleRate, format,
3712                  channelMask, frameCount, flags, 0, sessionId),
3713        mOverflow(false)
3714{
3715    if (mCblk != NULL) {
3716       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3717       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3718           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3719       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3720           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3721       } else {
3722           mCblk->frameSize = sizeof(int8_t);
3723       }
3724    }
3725}
3726
3727AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3728{
3729    sp<ThreadBase> thread = mThread.promote();
3730    if (thread != 0) {
3731        AudioSystem::releaseInput(thread->id());
3732    }
3733}
3734
3735status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3736{
3737    audio_track_cblk_t* cblk = this->cblk();
3738    uint32_t framesAvail;
3739    uint32_t framesReq = buffer->frameCount;
3740
3741     // Check if last stepServer failed, try to step now
3742    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3743        if (!step()) goto getNextBuffer_exit;
3744        ALOGV("stepServer recovered");
3745        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3746    }
3747
3748    framesAvail = cblk->framesAvailable_l();
3749
3750    if (CC_LIKELY(framesAvail)) {
3751        uint32_t s = cblk->server;
3752        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3753
3754        if (framesReq > framesAvail) {
3755            framesReq = framesAvail;
3756        }
3757        if (s + framesReq > bufferEnd) {
3758            framesReq = bufferEnd - s;
3759        }
3760
3761        buffer->raw = getBuffer(s, framesReq);
3762        if (buffer->raw == NULL) goto getNextBuffer_exit;
3763
3764        buffer->frameCount = framesReq;
3765        return NO_ERROR;
3766    }
3767
3768getNextBuffer_exit:
3769    buffer->raw = NULL;
3770    buffer->frameCount = 0;
3771    return NOT_ENOUGH_DATA;
3772}
3773
3774status_t AudioFlinger::RecordThread::RecordTrack::start()
3775{
3776    sp<ThreadBase> thread = mThread.promote();
3777    if (thread != 0) {
3778        RecordThread *recordThread = (RecordThread *)thread.get();
3779        return recordThread->start(this);
3780    } else {
3781        return BAD_VALUE;
3782    }
3783}
3784
3785void AudioFlinger::RecordThread::RecordTrack::stop()
3786{
3787    sp<ThreadBase> thread = mThread.promote();
3788    if (thread != 0) {
3789        RecordThread *recordThread = (RecordThread *)thread.get();
3790        recordThread->stop(this);
3791        TrackBase::reset();
3792        // Force overerrun condition to avoid false overrun callback until first data is
3793        // read from buffer
3794        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3795    }
3796}
3797
3798void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3799{
3800    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3801            (mClient == NULL) ? getpid() : mClient->pid(),
3802            mFormat,
3803            mChannelMask,
3804            mSessionId,
3805            mFrameCount,
3806            mState,
3807            mCblk->sampleRate,
3808            mCblk->server,
3809            mCblk->user);
3810}
3811
3812
3813// ----------------------------------------------------------------------------
3814
3815AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3816            const wp<ThreadBase>& thread,
3817            DuplicatingThread *sourceThread,
3818            uint32_t sampleRate,
3819            audio_format_t format,
3820            uint32_t channelMask,
3821            int frameCount)
3822    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3823    mActive(false), mSourceThread(sourceThread)
3824{
3825
3826    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3827    if (mCblk != NULL) {
3828        mCblk->flags |= CBLK_DIRECTION_OUT;
3829        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3830        mCblk->volumeLR = (MAX_GAIN_INT << 16) | MAX_GAIN_INT;
3831        mOutBuffer.frameCount = 0;
3832        playbackThread->mTracks.add(this);
3833        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3834                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3835                mCblk, mBuffer, mCblk->buffers,
3836                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3837    } else {
3838        ALOGW("Error creating output track on thread %p", playbackThread);
3839    }
3840}
3841
3842AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3843{
3844    clearBufferQueue();
3845}
3846
3847status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3848{
3849    status_t status = Track::start();
3850    if (status != NO_ERROR) {
3851        return status;
3852    }
3853
3854    mActive = true;
3855    mRetryCount = 127;
3856    return status;
3857}
3858
3859void AudioFlinger::PlaybackThread::OutputTrack::stop()
3860{
3861    Track::stop();
3862    clearBufferQueue();
3863    mOutBuffer.frameCount = 0;
3864    mActive = false;
3865}
3866
3867bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3868{
3869    Buffer *pInBuffer;
3870    Buffer inBuffer;
3871    uint32_t channelCount = mChannelCount;
3872    bool outputBufferFull = false;
3873    inBuffer.frameCount = frames;
3874    inBuffer.i16 = data;
3875
3876    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3877
3878    if (!mActive && frames != 0) {
3879        start();
3880        sp<ThreadBase> thread = mThread.promote();
3881        if (thread != 0) {
3882            MixerThread *mixerThread = (MixerThread *)thread.get();
3883            if (mCblk->frameCount > frames){
3884                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3885                    uint32_t startFrames = (mCblk->frameCount - frames);
3886                    pInBuffer = new Buffer;
3887                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3888                    pInBuffer->frameCount = startFrames;
3889                    pInBuffer->i16 = pInBuffer->mBuffer;
3890                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3891                    mBufferQueue.add(pInBuffer);
3892                } else {
3893                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
3894                }
3895            }
3896        }
3897    }
3898
3899    while (waitTimeLeftMs) {
3900        // First write pending buffers, then new data
3901        if (mBufferQueue.size()) {
3902            pInBuffer = mBufferQueue.itemAt(0);
3903        } else {
3904            pInBuffer = &inBuffer;
3905        }
3906
3907        if (pInBuffer->frameCount == 0) {
3908            break;
3909        }
3910
3911        if (mOutBuffer.frameCount == 0) {
3912            mOutBuffer.frameCount = pInBuffer->frameCount;
3913            nsecs_t startTime = systemTime();
3914            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
3915                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3916                outputBufferFull = true;
3917                break;
3918            }
3919            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3920            if (waitTimeLeftMs >= waitTimeMs) {
3921                waitTimeLeftMs -= waitTimeMs;
3922            } else {
3923                waitTimeLeftMs = 0;
3924            }
3925        }
3926
3927        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3928        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3929        mCblk->stepUser(outFrames);
3930        pInBuffer->frameCount -= outFrames;
3931        pInBuffer->i16 += outFrames * channelCount;
3932        mOutBuffer.frameCount -= outFrames;
3933        mOutBuffer.i16 += outFrames * channelCount;
3934
3935        if (pInBuffer->frameCount == 0) {
3936            if (mBufferQueue.size()) {
3937                mBufferQueue.removeAt(0);
3938                delete [] pInBuffer->mBuffer;
3939                delete pInBuffer;
3940                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3941            } else {
3942                break;
3943            }
3944        }
3945    }
3946
3947    // If we could not write all frames, allocate a buffer and queue it for next time.
3948    if (inBuffer.frameCount) {
3949        sp<ThreadBase> thread = mThread.promote();
3950        if (thread != 0 && !thread->standby()) {
3951            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3952                pInBuffer = new Buffer;
3953                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3954                pInBuffer->frameCount = inBuffer.frameCount;
3955                pInBuffer->i16 = pInBuffer->mBuffer;
3956                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3957                mBufferQueue.add(pInBuffer);
3958                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3959            } else {
3960                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3961            }
3962        }
3963    }
3964
3965    // Calling write() with a 0 length buffer, means that no more data will be written:
3966    // If no more buffers are pending, fill output track buffer to make sure it is started
3967    // by output mixer.
3968    if (frames == 0 && mBufferQueue.size() == 0) {
3969        if (mCblk->user < mCblk->frameCount) {
3970            frames = mCblk->frameCount - mCblk->user;
3971            pInBuffer = new Buffer;
3972            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3973            pInBuffer->frameCount = frames;
3974            pInBuffer->i16 = pInBuffer->mBuffer;
3975            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3976            mBufferQueue.add(pInBuffer);
3977        } else if (mActive) {
3978            stop();
3979        }
3980    }
3981
3982    return outputBufferFull;
3983}
3984
3985status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3986{
3987    int active;
3988    status_t result;
3989    audio_track_cblk_t* cblk = mCblk;
3990    uint32_t framesReq = buffer->frameCount;
3991
3992//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3993    buffer->frameCount  = 0;
3994
3995    uint32_t framesAvail = cblk->framesAvailable();
3996
3997
3998    if (framesAvail == 0) {
3999        Mutex::Autolock _l(cblk->lock);
4000        goto start_loop_here;
4001        while (framesAvail == 0) {
4002            active = mActive;
4003            if (CC_UNLIKELY(!active)) {
4004                ALOGV("Not active and NO_MORE_BUFFERS");
4005                return NO_MORE_BUFFERS;
4006            }
4007            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
4008            if (result != NO_ERROR) {
4009                return NO_MORE_BUFFERS;
4010            }
4011            // read the server count again
4012        start_loop_here:
4013            framesAvail = cblk->framesAvailable_l();
4014        }
4015    }
4016
4017//    if (framesAvail < framesReq) {
4018//        return NO_MORE_BUFFERS;
4019//    }
4020
4021    if (framesReq > framesAvail) {
4022        framesReq = framesAvail;
4023    }
4024
4025    uint32_t u = cblk->user;
4026    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
4027
4028    if (u + framesReq > bufferEnd) {
4029        framesReq = bufferEnd - u;
4030    }
4031
4032    buffer->frameCount  = framesReq;
4033    buffer->raw         = (void *)cblk->buffer(u);
4034    return NO_ERROR;
4035}
4036
4037
4038void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
4039{
4040    size_t size = mBufferQueue.size();
4041    Buffer *pBuffer;
4042
4043    for (size_t i = 0; i < size; i++) {
4044        pBuffer = mBufferQueue.itemAt(i);
4045        delete [] pBuffer->mBuffer;
4046        delete pBuffer;
4047    }
4048    mBufferQueue.clear();
4049}
4050
4051// ----------------------------------------------------------------------------
4052
4053AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
4054    :   RefBase(),
4055        mAudioFlinger(audioFlinger),
4056        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
4057        mPid(pid)
4058{
4059    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
4060}
4061
4062// Client destructor must be called with AudioFlinger::mLock held
4063AudioFlinger::Client::~Client()
4064{
4065    mAudioFlinger->removeClient_l(mPid);
4066}
4067
4068const sp<MemoryDealer>& AudioFlinger::Client::heap() const
4069{
4070    return mMemoryDealer;
4071}
4072
4073// ----------------------------------------------------------------------------
4074
4075AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4076                                                     const sp<IAudioFlingerClient>& client,
4077                                                     pid_t pid)
4078    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
4079{
4080}
4081
4082AudioFlinger::NotificationClient::~NotificationClient()
4083{
4084    mClient.clear();
4085}
4086
4087void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4088{
4089    sp<NotificationClient> keep(this);
4090    {
4091        mAudioFlinger->removeNotificationClient(mPid);
4092    }
4093}
4094
4095// ----------------------------------------------------------------------------
4096
4097AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4098    : BnAudioTrack(),
4099      mTrack(track)
4100{
4101}
4102
4103AudioFlinger::TrackHandle::~TrackHandle() {
4104    // just stop the track on deletion, associated resources
4105    // will be freed from the main thread once all pending buffers have
4106    // been played. Unless it's not in the active track list, in which
4107    // case we free everything now...
4108    mTrack->destroy();
4109}
4110
4111status_t AudioFlinger::TrackHandle::start() {
4112    return mTrack->start();
4113}
4114
4115void AudioFlinger::TrackHandle::stop() {
4116    mTrack->stop();
4117}
4118
4119void AudioFlinger::TrackHandle::flush() {
4120    mTrack->flush();
4121}
4122
4123void AudioFlinger::TrackHandle::mute(bool e) {
4124    mTrack->mute(e);
4125}
4126
4127void AudioFlinger::TrackHandle::pause() {
4128    mTrack->pause();
4129}
4130
4131sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4132    return mTrack->getCblk();
4133}
4134
4135status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4136{
4137    return mTrack->attachAuxEffect(EffectId);
4138}
4139
4140status_t AudioFlinger::TrackHandle::onTransact(
4141    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4142{
4143    return BnAudioTrack::onTransact(code, data, reply, flags);
4144}
4145
4146// ----------------------------------------------------------------------------
4147
4148sp<IAudioRecord> AudioFlinger::openRecord(
4149        pid_t pid,
4150        int input,
4151        uint32_t sampleRate,
4152        audio_format_t format,
4153        uint32_t channelMask,
4154        int frameCount,
4155        uint32_t flags,
4156        int *sessionId,
4157        status_t *status)
4158{
4159    sp<RecordThread::RecordTrack> recordTrack;
4160    sp<RecordHandle> recordHandle;
4161    sp<Client> client;
4162    wp<Client> wclient;
4163    status_t lStatus;
4164    RecordThread *thread;
4165    size_t inFrameCount;
4166    int lSessionId;
4167
4168    // check calling permissions
4169    if (!recordingAllowed()) {
4170        lStatus = PERMISSION_DENIED;
4171        goto Exit;
4172    }
4173
4174    // add client to list
4175    { // scope for mLock
4176        Mutex::Autolock _l(mLock);
4177        thread = checkRecordThread_l(input);
4178        if (thread == NULL) {
4179            lStatus = BAD_VALUE;
4180            goto Exit;
4181        }
4182
4183        wclient = mClients.valueFor(pid);
4184        if (wclient != NULL) {
4185            client = wclient.promote();
4186        } else {
4187            client = new Client(this, pid);
4188            mClients.add(pid, client);
4189        }
4190
4191        // If no audio session id is provided, create one here
4192        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4193            lSessionId = *sessionId;
4194        } else {
4195            lSessionId = nextUniqueId();
4196            if (sessionId != NULL) {
4197                *sessionId = lSessionId;
4198            }
4199        }
4200        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4201        recordTrack = thread->createRecordTrack_l(client,
4202                                                sampleRate,
4203                                                format,
4204                                                channelMask,
4205                                                frameCount,
4206                                                flags,
4207                                                lSessionId,
4208                                                &lStatus);
4209    }
4210    if (lStatus != NO_ERROR) {
4211        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4212        // destructor is called by the TrackBase destructor with mLock held
4213        client.clear();
4214        recordTrack.clear();
4215        goto Exit;
4216    }
4217
4218    // return to handle to client
4219    recordHandle = new RecordHandle(recordTrack);
4220    lStatus = NO_ERROR;
4221
4222Exit:
4223    if (status) {
4224        *status = lStatus;
4225    }
4226    return recordHandle;
4227}
4228
4229// ----------------------------------------------------------------------------
4230
4231AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4232    : BnAudioRecord(),
4233    mRecordTrack(recordTrack)
4234{
4235}
4236
4237AudioFlinger::RecordHandle::~RecordHandle() {
4238    stop();
4239}
4240
4241status_t AudioFlinger::RecordHandle::start() {
4242    ALOGV("RecordHandle::start()");
4243    return mRecordTrack->start();
4244}
4245
4246void AudioFlinger::RecordHandle::stop() {
4247    ALOGV("RecordHandle::stop()");
4248    mRecordTrack->stop();
4249}
4250
4251sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4252    return mRecordTrack->getCblk();
4253}
4254
4255status_t AudioFlinger::RecordHandle::onTransact(
4256    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4257{
4258    return BnAudioRecord::onTransact(code, data, reply, flags);
4259}
4260
4261// ----------------------------------------------------------------------------
4262
4263AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4264                                         AudioStreamIn *input,
4265                                         uint32_t sampleRate,
4266                                         uint32_t channels,
4267                                         int id,
4268                                         uint32_t device) :
4269    ThreadBase(audioFlinger, id, device),
4270    mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
4271{
4272    mType = ThreadBase::RECORD;
4273
4274    snprintf(mName, kNameLength, "AudioIn_%d", id);
4275
4276    mReqChannelCount = popcount(channels);
4277    mReqSampleRate = sampleRate;
4278    readInputParameters();
4279}
4280
4281
4282AudioFlinger::RecordThread::~RecordThread()
4283{
4284    delete[] mRsmpInBuffer;
4285    if (mResampler != NULL) {
4286        delete mResampler;
4287        delete[] mRsmpOutBuffer;
4288    }
4289}
4290
4291void AudioFlinger::RecordThread::onFirstRef()
4292{
4293    run(mName, PRIORITY_URGENT_AUDIO);
4294}
4295
4296status_t AudioFlinger::RecordThread::readyToRun()
4297{
4298    status_t status = initCheck();
4299    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4300    return status;
4301}
4302
4303bool AudioFlinger::RecordThread::threadLoop()
4304{
4305    AudioBufferProvider::Buffer buffer;
4306    sp<RecordTrack> activeTrack;
4307    Vector< sp<EffectChain> > effectChains;
4308
4309    nsecs_t lastWarning = 0;
4310
4311    acquireWakeLock();
4312
4313    // start recording
4314    while (!exitPending()) {
4315
4316        processConfigEvents();
4317
4318        { // scope for mLock
4319            Mutex::Autolock _l(mLock);
4320            checkForNewParameters_l();
4321            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4322                if (!mStandby) {
4323                    mInput->stream->common.standby(&mInput->stream->common);
4324                    mStandby = true;
4325                }
4326
4327                if (exitPending()) break;
4328
4329                releaseWakeLock_l();
4330                ALOGV("RecordThread: loop stopping");
4331                // go to sleep
4332                mWaitWorkCV.wait(mLock);
4333                ALOGV("RecordThread: loop starting");
4334                acquireWakeLock_l();
4335                continue;
4336            }
4337            if (mActiveTrack != 0) {
4338                if (mActiveTrack->mState == TrackBase::PAUSING) {
4339                    if (!mStandby) {
4340                        mInput->stream->common.standby(&mInput->stream->common);
4341                        mStandby = true;
4342                    }
4343                    mActiveTrack.clear();
4344                    mStartStopCond.broadcast();
4345                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4346                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4347                        mActiveTrack.clear();
4348                        mStartStopCond.broadcast();
4349                    } else if (mBytesRead != 0) {
4350                        // record start succeeds only if first read from audio input
4351                        // succeeds
4352                        if (mBytesRead > 0) {
4353                            mActiveTrack->mState = TrackBase::ACTIVE;
4354                        } else {
4355                            mActiveTrack.clear();
4356                        }
4357                        mStartStopCond.broadcast();
4358                    }
4359                    mStandby = false;
4360                }
4361            }
4362            lockEffectChains_l(effectChains);
4363        }
4364
4365        if (mActiveTrack != 0) {
4366            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4367                mActiveTrack->mState != TrackBase::RESUMING) {
4368                unlockEffectChains(effectChains);
4369                usleep(kRecordThreadSleepUs);
4370                continue;
4371            }
4372            for (size_t i = 0; i < effectChains.size(); i ++) {
4373                effectChains[i]->process_l();
4374            }
4375
4376            buffer.frameCount = mFrameCount;
4377            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4378                size_t framesOut = buffer.frameCount;
4379                if (mResampler == NULL) {
4380                    // no resampling
4381                    while (framesOut) {
4382                        size_t framesIn = mFrameCount - mRsmpInIndex;
4383                        if (framesIn) {
4384                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4385                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4386                            if (framesIn > framesOut)
4387                                framesIn = framesOut;
4388                            mRsmpInIndex += framesIn;
4389                            framesOut -= framesIn;
4390                            if ((int)mChannelCount == mReqChannelCount ||
4391                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4392                                memcpy(dst, src, framesIn * mFrameSize);
4393                            } else {
4394                                int16_t *src16 = (int16_t *)src;
4395                                int16_t *dst16 = (int16_t *)dst;
4396                                if (mChannelCount == 1) {
4397                                    while (framesIn--) {
4398                                        *dst16++ = *src16;
4399                                        *dst16++ = *src16++;
4400                                    }
4401                                } else {
4402                                    while (framesIn--) {
4403                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4404                                        src16 += 2;
4405                                    }
4406                                }
4407                            }
4408                        }
4409                        if (framesOut && mFrameCount == mRsmpInIndex) {
4410                            if (framesOut == mFrameCount &&
4411                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4412                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4413                                framesOut = 0;
4414                            } else {
4415                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4416                                mRsmpInIndex = 0;
4417                            }
4418                            if (mBytesRead < 0) {
4419                                ALOGE("Error reading audio input");
4420                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4421                                    // Force input into standby so that it tries to
4422                                    // recover at next read attempt
4423                                    mInput->stream->common.standby(&mInput->stream->common);
4424                                    usleep(kRecordThreadSleepUs);
4425                                }
4426                                mRsmpInIndex = mFrameCount;
4427                                framesOut = 0;
4428                                buffer.frameCount = 0;
4429                            }
4430                        }
4431                    }
4432                } else {
4433                    // resampling
4434
4435                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4436                    // alter output frame count as if we were expecting stereo samples
4437                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4438                        framesOut >>= 1;
4439                    }
4440                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4441                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4442                    // are 32 bit aligned which should be always true.
4443                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4444                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4445                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4446                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4447                        int16_t *dst = buffer.i16;
4448                        while (framesOut--) {
4449                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4450                            src += 2;
4451                        }
4452                    } else {
4453                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4454                    }
4455
4456                }
4457                mActiveTrack->releaseBuffer(&buffer);
4458                mActiveTrack->overflow();
4459            }
4460            // client isn't retrieving buffers fast enough
4461            else {
4462                if (!mActiveTrack->setOverflow()) {
4463                    nsecs_t now = systemTime();
4464                    if ((now - lastWarning) > kWarningThrottleNs) {
4465                        ALOGW("RecordThread: buffer overflow");
4466                        lastWarning = now;
4467                    }
4468                }
4469                // Release the processor for a while before asking for a new buffer.
4470                // This will give the application more chance to read from the buffer and
4471                // clear the overflow.
4472                usleep(kRecordThreadSleepUs);
4473            }
4474        }
4475        // enable changes in effect chain
4476        unlockEffectChains(effectChains);
4477        effectChains.clear();
4478    }
4479
4480    if (!mStandby) {
4481        mInput->stream->common.standby(&mInput->stream->common);
4482    }
4483    mActiveTrack.clear();
4484
4485    mStartStopCond.broadcast();
4486
4487    releaseWakeLock();
4488
4489    ALOGV("RecordThread %p exiting", this);
4490    return false;
4491}
4492
4493
4494sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4495        const sp<AudioFlinger::Client>& client,
4496        uint32_t sampleRate,
4497        audio_format_t format,
4498        int channelMask,
4499        int frameCount,
4500        uint32_t flags,
4501        int sessionId,
4502        status_t *status)
4503{
4504    sp<RecordTrack> track;
4505    status_t lStatus;
4506
4507    lStatus = initCheck();
4508    if (lStatus != NO_ERROR) {
4509        ALOGE("Audio driver not initialized.");
4510        goto Exit;
4511    }
4512
4513    { // scope for mLock
4514        Mutex::Autolock _l(mLock);
4515
4516        track = new RecordTrack(this, client, sampleRate,
4517                      format, channelMask, frameCount, flags, sessionId);
4518
4519        if (track->getCblk() == NULL) {
4520            lStatus = NO_MEMORY;
4521            goto Exit;
4522        }
4523
4524        mTrack = track.get();
4525        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4526        bool suspend = audio_is_bluetooth_sco_device(
4527                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4528        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4529        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4530    }
4531    lStatus = NO_ERROR;
4532
4533Exit:
4534    if (status) {
4535        *status = lStatus;
4536    }
4537    return track;
4538}
4539
4540status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4541{
4542    ALOGV("RecordThread::start");
4543    sp <ThreadBase> strongMe = this;
4544    status_t status = NO_ERROR;
4545    {
4546        AutoMutex lock(mLock);
4547        if (mActiveTrack != 0) {
4548            if (recordTrack != mActiveTrack.get()) {
4549                status = -EBUSY;
4550            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4551                mActiveTrack->mState = TrackBase::ACTIVE;
4552            }
4553            return status;
4554        }
4555
4556        recordTrack->mState = TrackBase::IDLE;
4557        mActiveTrack = recordTrack;
4558        mLock.unlock();
4559        status_t status = AudioSystem::startInput(mId);
4560        mLock.lock();
4561        if (status != NO_ERROR) {
4562            mActiveTrack.clear();
4563            return status;
4564        }
4565        mRsmpInIndex = mFrameCount;
4566        mBytesRead = 0;
4567        if (mResampler != NULL) {
4568            mResampler->reset();
4569        }
4570        mActiveTrack->mState = TrackBase::RESUMING;
4571        // signal thread to start
4572        ALOGV("Signal record thread");
4573        mWaitWorkCV.signal();
4574        // do not wait for mStartStopCond if exiting
4575        if (mExiting) {
4576            mActiveTrack.clear();
4577            status = INVALID_OPERATION;
4578            goto startError;
4579        }
4580        mStartStopCond.wait(mLock);
4581        if (mActiveTrack == 0) {
4582            ALOGV("Record failed to start");
4583            status = BAD_VALUE;
4584            goto startError;
4585        }
4586        ALOGV("Record started OK");
4587        return status;
4588    }
4589startError:
4590    AudioSystem::stopInput(mId);
4591    return status;
4592}
4593
4594void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4595    ALOGV("RecordThread::stop");
4596    sp <ThreadBase> strongMe = this;
4597    {
4598        AutoMutex lock(mLock);
4599        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4600            mActiveTrack->mState = TrackBase::PAUSING;
4601            // do not wait for mStartStopCond if exiting
4602            if (mExiting) {
4603                return;
4604            }
4605            mStartStopCond.wait(mLock);
4606            // if we have been restarted, recordTrack == mActiveTrack.get() here
4607            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4608                mLock.unlock();
4609                AudioSystem::stopInput(mId);
4610                mLock.lock();
4611                ALOGV("Record stopped OK");
4612            }
4613        }
4614    }
4615}
4616
4617status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4618{
4619    const size_t SIZE = 256;
4620    char buffer[SIZE];
4621    String8 result;
4622    pid_t pid = 0;
4623
4624    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4625    result.append(buffer);
4626
4627    if (mActiveTrack != 0) {
4628        result.append("Active Track:\n");
4629        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4630        mActiveTrack->dump(buffer, SIZE);
4631        result.append(buffer);
4632
4633        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4634        result.append(buffer);
4635        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4636        result.append(buffer);
4637        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4638        result.append(buffer);
4639        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4640        result.append(buffer);
4641        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4642        result.append(buffer);
4643
4644
4645    } else {
4646        result.append("No record client\n");
4647    }
4648    write(fd, result.string(), result.size());
4649
4650    dumpBase(fd, args);
4651    dumpEffectChains(fd, args);
4652
4653    return NO_ERROR;
4654}
4655
4656status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4657{
4658    size_t framesReq = buffer->frameCount;
4659    size_t framesReady = mFrameCount - mRsmpInIndex;
4660    int channelCount;
4661
4662    if (framesReady == 0) {
4663        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4664        if (mBytesRead < 0) {
4665            ALOGE("RecordThread::getNextBuffer() Error reading audio input");
4666            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4667                // Force input into standby so that it tries to
4668                // recover at next read attempt
4669                mInput->stream->common.standby(&mInput->stream->common);
4670                usleep(kRecordThreadSleepUs);
4671            }
4672            buffer->raw = NULL;
4673            buffer->frameCount = 0;
4674            return NOT_ENOUGH_DATA;
4675        }
4676        mRsmpInIndex = 0;
4677        framesReady = mFrameCount;
4678    }
4679
4680    if (framesReq > framesReady) {
4681        framesReq = framesReady;
4682    }
4683
4684    if (mChannelCount == 1 && mReqChannelCount == 2) {
4685        channelCount = 1;
4686    } else {
4687        channelCount = 2;
4688    }
4689    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4690    buffer->frameCount = framesReq;
4691    return NO_ERROR;
4692}
4693
4694void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4695{
4696    mRsmpInIndex += buffer->frameCount;
4697    buffer->frameCount = 0;
4698}
4699
4700bool AudioFlinger::RecordThread::checkForNewParameters_l()
4701{
4702    bool reconfig = false;
4703
4704    while (!mNewParameters.isEmpty()) {
4705        status_t status = NO_ERROR;
4706        String8 keyValuePair = mNewParameters[0];
4707        AudioParameter param = AudioParameter(keyValuePair);
4708        int value;
4709        audio_format_t reqFormat = mFormat;
4710        int reqSamplingRate = mReqSampleRate;
4711        int reqChannelCount = mReqChannelCount;
4712
4713        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4714            reqSamplingRate = value;
4715            reconfig = true;
4716        }
4717        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4718            reqFormat = (audio_format_t) value;
4719            reconfig = true;
4720        }
4721        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4722            reqChannelCount = popcount(value);
4723            reconfig = true;
4724        }
4725        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4726            // do not accept frame count changes if tracks are open as the track buffer
4727            // size depends on frame count and correct behavior would not be garantied
4728            // if frame count is changed after track creation
4729            if (mActiveTrack != 0) {
4730                status = INVALID_OPERATION;
4731            } else {
4732                reconfig = true;
4733            }
4734        }
4735        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4736            // forward device change to effects that have requested to be
4737            // aware of attached audio device.
4738            for (size_t i = 0; i < mEffectChains.size(); i++) {
4739                mEffectChains[i]->setDevice_l(value);
4740            }
4741            // store input device and output device but do not forward output device to audio HAL.
4742            // Note that status is ignored by the caller for output device
4743            // (see AudioFlinger::setParameters()
4744            if (value & AUDIO_DEVICE_OUT_ALL) {
4745                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4746                status = BAD_VALUE;
4747            } else {
4748                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4749                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4750                if (mTrack != NULL) {
4751                    bool suspend = audio_is_bluetooth_sco_device(
4752                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4753                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4754                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4755                }
4756            }
4757            mDevice |= (uint32_t)value;
4758        }
4759        if (status == NO_ERROR) {
4760            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4761            if (status == INVALID_OPERATION) {
4762               mInput->stream->common.standby(&mInput->stream->common);
4763               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4764            }
4765            if (reconfig) {
4766                if (status == BAD_VALUE &&
4767                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4768                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4769                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4770                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4771                    (reqChannelCount < 3)) {
4772                    status = NO_ERROR;
4773                }
4774                if (status == NO_ERROR) {
4775                    readInputParameters();
4776                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4777                }
4778            }
4779        }
4780
4781        mNewParameters.removeAt(0);
4782
4783        mParamStatus = status;
4784        mParamCond.signal();
4785        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4786        // already timed out waiting for the status and will never signal the condition.
4787        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4788    }
4789    return reconfig;
4790}
4791
4792String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4793{
4794    char *s;
4795    String8 out_s8 = String8();
4796
4797    Mutex::Autolock _l(mLock);
4798    if (initCheck() != NO_ERROR) {
4799        return out_s8;
4800    }
4801
4802    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4803    out_s8 = String8(s);
4804    free(s);
4805    return out_s8;
4806}
4807
4808void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4809    AudioSystem::OutputDescriptor desc;
4810    void *param2 = 0;
4811
4812    switch (event) {
4813    case AudioSystem::INPUT_OPENED:
4814    case AudioSystem::INPUT_CONFIG_CHANGED:
4815        desc.channels = mChannelMask;
4816        desc.samplingRate = mSampleRate;
4817        desc.format = mFormat;
4818        desc.frameCount = mFrameCount;
4819        desc.latency = 0;
4820        param2 = &desc;
4821        break;
4822
4823    case AudioSystem::INPUT_CLOSED:
4824    default:
4825        break;
4826    }
4827    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4828}
4829
4830void AudioFlinger::RecordThread::readInputParameters()
4831{
4832    if (mRsmpInBuffer) delete mRsmpInBuffer;
4833    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4834    if (mResampler) delete mResampler;
4835    mResampler = NULL;
4836
4837    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4838    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4839    mChannelCount = (uint16_t)popcount(mChannelMask);
4840    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4841    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
4842    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4843    mFrameCount = mInputBytes / mFrameSize;
4844    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4845
4846    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4847    {
4848        int channelCount;
4849         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4850         // stereo to mono post process as the resampler always outputs stereo.
4851        if (mChannelCount == 1 && mReqChannelCount == 2) {
4852            channelCount = 1;
4853        } else {
4854            channelCount = 2;
4855        }
4856        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4857        mResampler->setSampleRate(mSampleRate);
4858        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4859        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4860
4861        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4862        if (mChannelCount == 1 && mReqChannelCount == 1) {
4863            mFrameCount >>= 1;
4864        }
4865
4866    }
4867    mRsmpInIndex = mFrameCount;
4868}
4869
4870unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4871{
4872    Mutex::Autolock _l(mLock);
4873    if (initCheck() != NO_ERROR) {
4874        return 0;
4875    }
4876
4877    return mInput->stream->get_input_frames_lost(mInput->stream);
4878}
4879
4880uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4881{
4882    Mutex::Autolock _l(mLock);
4883    uint32_t result = 0;
4884    if (getEffectChain_l(sessionId) != 0) {
4885        result = EFFECT_SESSION;
4886    }
4887
4888    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4889        result |= TRACK_SESSION;
4890    }
4891
4892    return result;
4893}
4894
4895AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4896{
4897    Mutex::Autolock _l(mLock);
4898    return mTrack;
4899}
4900
4901AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
4902{
4903    Mutex::Autolock _l(mLock);
4904    return mInput;
4905}
4906
4907AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4908{
4909    Mutex::Autolock _l(mLock);
4910    AudioStreamIn *input = mInput;
4911    mInput = NULL;
4912    return input;
4913}
4914
4915// this method must always be called either with ThreadBase mLock held or inside the thread loop
4916audio_stream_t* AudioFlinger::RecordThread::stream()
4917{
4918    if (mInput == NULL) {
4919        return NULL;
4920    }
4921    return &mInput->stream->common;
4922}
4923
4924
4925// ----------------------------------------------------------------------------
4926
4927int AudioFlinger::openOutput(uint32_t *pDevices,
4928                                uint32_t *pSamplingRate,
4929                                audio_format_t *pFormat,
4930                                uint32_t *pChannels,
4931                                uint32_t *pLatencyMs,
4932                                uint32_t flags)
4933{
4934    status_t status;
4935    PlaybackThread *thread = NULL;
4936    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4937    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4938    audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
4939    uint32_t channels = pChannels ? *pChannels : 0;
4940    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4941    audio_stream_out_t *outStream;
4942    audio_hw_device_t *outHwDev;
4943
4944    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4945            pDevices ? *pDevices : 0,
4946            samplingRate,
4947            format,
4948            channels,
4949            flags);
4950
4951    if (pDevices == NULL || *pDevices == 0) {
4952        return 0;
4953    }
4954
4955    Mutex::Autolock _l(mLock);
4956
4957    outHwDev = findSuitableHwDev_l(*pDevices);
4958    if (outHwDev == NULL)
4959        return 0;
4960
4961    status = outHwDev->open_output_stream(outHwDev, *pDevices, &format,
4962                                          &channels, &samplingRate, &outStream);
4963    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4964            outStream,
4965            samplingRate,
4966            format,
4967            channels,
4968            status);
4969
4970    mHardwareStatus = AUDIO_HW_IDLE;
4971    if (outStream != NULL) {
4972        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4973        int id = nextUniqueId();
4974
4975        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4976            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4977            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4978            thread = new DirectOutputThread(this, output, id, *pDevices);
4979            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4980        } else {
4981            thread = new MixerThread(this, output, id, *pDevices);
4982            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4983        }
4984        mPlaybackThreads.add(id, thread);
4985
4986        if (pSamplingRate) *pSamplingRate = samplingRate;
4987        if (pFormat) *pFormat = format;
4988        if (pChannels) *pChannels = channels;
4989        if (pLatencyMs) *pLatencyMs = thread->latency();
4990
4991        // notify client processes of the new output creation
4992        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4993        return id;
4994    }
4995
4996    return 0;
4997}
4998
4999int AudioFlinger::openDuplicateOutput(int output1, int output2)
5000{
5001    Mutex::Autolock _l(mLock);
5002    MixerThread *thread1 = checkMixerThread_l(output1);
5003    MixerThread *thread2 = checkMixerThread_l(output2);
5004
5005    if (thread1 == NULL || thread2 == NULL) {
5006        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
5007        return 0;
5008    }
5009
5010    int id = nextUniqueId();
5011    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
5012    thread->addOutputTrack(thread2);
5013    mPlaybackThreads.add(id, thread);
5014    // notify client processes of the new output creation
5015    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
5016    return id;
5017}
5018
5019status_t AudioFlinger::closeOutput(int output)
5020{
5021    // keep strong reference on the playback thread so that
5022    // it is not destroyed while exit() is executed
5023    sp <PlaybackThread> thread;
5024    {
5025        Mutex::Autolock _l(mLock);
5026        thread = checkPlaybackThread_l(output);
5027        if (thread == NULL) {
5028            return BAD_VALUE;
5029        }
5030
5031        ALOGV("closeOutput() %d", output);
5032
5033        if (thread->type() == ThreadBase::MIXER) {
5034            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5035                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
5036                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
5037                    dupThread->removeOutputTrack((MixerThread *)thread.get());
5038                }
5039            }
5040        }
5041        void *param2 = 0;
5042        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
5043        mPlaybackThreads.removeItem(output);
5044    }
5045    thread->exit();
5046
5047    if (thread->type() != ThreadBase::DUPLICATING) {
5048        AudioStreamOut *out = thread->clearOutput();
5049        // from now on thread->mOutput is NULL
5050        out->hwDev->close_output_stream(out->hwDev, out->stream);
5051        delete out;
5052    }
5053    return NO_ERROR;
5054}
5055
5056status_t AudioFlinger::suspendOutput(int output)
5057{
5058    Mutex::Autolock _l(mLock);
5059    PlaybackThread *thread = checkPlaybackThread_l(output);
5060
5061    if (thread == NULL) {
5062        return BAD_VALUE;
5063    }
5064
5065    ALOGV("suspendOutput() %d", output);
5066    thread->suspend();
5067
5068    return NO_ERROR;
5069}
5070
5071status_t AudioFlinger::restoreOutput(int output)
5072{
5073    Mutex::Autolock _l(mLock);
5074    PlaybackThread *thread = checkPlaybackThread_l(output);
5075
5076    if (thread == NULL) {
5077        return BAD_VALUE;
5078    }
5079
5080    ALOGV("restoreOutput() %d", output);
5081
5082    thread->restore();
5083
5084    return NO_ERROR;
5085}
5086
5087int AudioFlinger::openInput(uint32_t *pDevices,
5088                                uint32_t *pSamplingRate,
5089                                audio_format_t *pFormat,
5090                                uint32_t *pChannels,
5091                                uint32_t acoustics)
5092{
5093    status_t status;
5094    RecordThread *thread = NULL;
5095    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5096    audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
5097    uint32_t channels = pChannels ? *pChannels : 0;
5098    uint32_t reqSamplingRate = samplingRate;
5099    audio_format_t reqFormat = format;
5100    uint32_t reqChannels = channels;
5101    audio_stream_in_t *inStream;
5102    audio_hw_device_t *inHwDev;
5103
5104    if (pDevices == NULL || *pDevices == 0) {
5105        return 0;
5106    }
5107
5108    Mutex::Autolock _l(mLock);
5109
5110    inHwDev = findSuitableHwDev_l(*pDevices);
5111    if (inHwDev == NULL)
5112        return 0;
5113
5114    status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
5115                                        &channels, &samplingRate,
5116                                        (audio_in_acoustics_t)acoustics,
5117                                        &inStream);
5118    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5119            inStream,
5120            samplingRate,
5121            format,
5122            channels,
5123            acoustics,
5124            status);
5125
5126    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5127    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5128    // or stereo to mono conversions on 16 bit PCM inputs.
5129    if (inStream == NULL && status == BAD_VALUE &&
5130        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5131        (samplingRate <= 2 * reqSamplingRate) &&
5132        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5133        ALOGV("openInput() reopening with proposed sampling rate and channels");
5134        status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
5135                                            &channels, &samplingRate,
5136                                            (audio_in_acoustics_t)acoustics,
5137                                            &inStream);
5138    }
5139
5140    if (inStream != NULL) {
5141        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5142
5143        int id = nextUniqueId();
5144        // Start record thread
5145        // RecorThread require both input and output device indication to forward to audio
5146        // pre processing modules
5147        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5148        thread = new RecordThread(this,
5149                                  input,
5150                                  reqSamplingRate,
5151                                  reqChannels,
5152                                  id,
5153                                  device);
5154        mRecordThreads.add(id, thread);
5155        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5156        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5157        if (pFormat) *pFormat = format;
5158        if (pChannels) *pChannels = reqChannels;
5159
5160        input->stream->common.standby(&input->stream->common);
5161
5162        // notify client processes of the new input creation
5163        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5164        return id;
5165    }
5166
5167    return 0;
5168}
5169
5170status_t AudioFlinger::closeInput(int input)
5171{
5172    // keep strong reference on the record thread so that
5173    // it is not destroyed while exit() is executed
5174    sp <RecordThread> thread;
5175    {
5176        Mutex::Autolock _l(mLock);
5177        thread = checkRecordThread_l(input);
5178        if (thread == NULL) {
5179            return BAD_VALUE;
5180        }
5181
5182        ALOGV("closeInput() %d", input);
5183        void *param2 = 0;
5184        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5185        mRecordThreads.removeItem(input);
5186    }
5187    thread->exit();
5188
5189    AudioStreamIn *in = thread->clearInput();
5190    // from now on thread->mInput is NULL
5191    in->hwDev->close_input_stream(in->hwDev, in->stream);
5192    delete in;
5193
5194    return NO_ERROR;
5195}
5196
5197status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output)
5198{
5199    Mutex::Autolock _l(mLock);
5200    MixerThread *dstThread = checkMixerThread_l(output);
5201    if (dstThread == NULL) {
5202        ALOGW("setStreamOutput() bad output id %d", output);
5203        return BAD_VALUE;
5204    }
5205
5206    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5207    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5208
5209    dstThread->setStreamValid(stream, true);
5210
5211    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5212        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5213        if (thread != dstThread &&
5214            thread->type() != ThreadBase::DIRECT) {
5215            MixerThread *srcThread = (MixerThread *)thread;
5216            srcThread->setStreamValid(stream, false);
5217            srcThread->invalidateTracks(stream);
5218        }
5219    }
5220
5221    return NO_ERROR;
5222}
5223
5224
5225int AudioFlinger::newAudioSessionId()
5226{
5227    return nextUniqueId();
5228}
5229
5230void AudioFlinger::acquireAudioSessionId(int audioSession)
5231{
5232    Mutex::Autolock _l(mLock);
5233    int caller = IPCThreadState::self()->getCallingPid();
5234    ALOGV("acquiring %d from %d", audioSession, caller);
5235    int num = mAudioSessionRefs.size();
5236    for (int i = 0; i< num; i++) {
5237        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5238        if (ref->sessionid == audioSession && ref->pid == caller) {
5239            ref->cnt++;
5240            ALOGV(" incremented refcount to %d", ref->cnt);
5241            return;
5242        }
5243    }
5244    AudioSessionRef *ref = new AudioSessionRef();
5245    ref->sessionid = audioSession;
5246    ref->pid = caller;
5247    ref->cnt = 1;
5248    mAudioSessionRefs.push(ref);
5249    ALOGV(" added new entry for %d", ref->sessionid);
5250}
5251
5252void AudioFlinger::releaseAudioSessionId(int audioSession)
5253{
5254    Mutex::Autolock _l(mLock);
5255    int caller = IPCThreadState::self()->getCallingPid();
5256    ALOGV("releasing %d from %d", audioSession, caller);
5257    int num = mAudioSessionRefs.size();
5258    for (int i = 0; i< num; i++) {
5259        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5260        if (ref->sessionid == audioSession && ref->pid == caller) {
5261            ref->cnt--;
5262            ALOGV(" decremented refcount to %d", ref->cnt);
5263            if (ref->cnt == 0) {
5264                mAudioSessionRefs.removeAt(i);
5265                delete ref;
5266                purgeStaleEffects_l();
5267            }
5268            return;
5269        }
5270    }
5271    ALOGW("session id %d not found for pid %d", audioSession, caller);
5272}
5273
5274void AudioFlinger::purgeStaleEffects_l() {
5275
5276    ALOGV("purging stale effects");
5277
5278    Vector< sp<EffectChain> > chains;
5279
5280    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5281        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5282        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5283            sp<EffectChain> ec = t->mEffectChains[j];
5284            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5285                chains.push(ec);
5286            }
5287        }
5288    }
5289    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5290        sp<RecordThread> t = mRecordThreads.valueAt(i);
5291        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5292            sp<EffectChain> ec = t->mEffectChains[j];
5293            chains.push(ec);
5294        }
5295    }
5296
5297    for (size_t i = 0; i < chains.size(); i++) {
5298        sp<EffectChain> ec = chains[i];
5299        int sessionid = ec->sessionId();
5300        sp<ThreadBase> t = ec->mThread.promote();
5301        if (t == 0) {
5302            continue;
5303        }
5304        size_t numsessionrefs = mAudioSessionRefs.size();
5305        bool found = false;
5306        for (size_t k = 0; k < numsessionrefs; k++) {
5307            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5308            if (ref->sessionid == sessionid) {
5309                ALOGV(" session %d still exists for %d with %d refs",
5310                     sessionid, ref->pid, ref->cnt);
5311                found = true;
5312                break;
5313            }
5314        }
5315        if (!found) {
5316            // remove all effects from the chain
5317            while (ec->mEffects.size()) {
5318                sp<EffectModule> effect = ec->mEffects[0];
5319                effect->unPin();
5320                Mutex::Autolock _l (t->mLock);
5321                t->removeEffect_l(effect);
5322                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5323                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5324                    if (handle != 0) {
5325                        handle->mEffect.clear();
5326                        if (handle->mHasControl && handle->mEnabled) {
5327                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5328                        }
5329                    }
5330                }
5331                AudioSystem::unregisterEffect(effect->id());
5332            }
5333        }
5334    }
5335    return;
5336}
5337
5338// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5339AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5340{
5341    PlaybackThread *thread = NULL;
5342    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5343        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5344    }
5345    return thread;
5346}
5347
5348// checkMixerThread_l() must be called with AudioFlinger::mLock held
5349AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5350{
5351    PlaybackThread *thread = checkPlaybackThread_l(output);
5352    if (thread != NULL) {
5353        if (thread->type() == ThreadBase::DIRECT) {
5354            thread = NULL;
5355        }
5356    }
5357    return (MixerThread *)thread;
5358}
5359
5360// checkRecordThread_l() must be called with AudioFlinger::mLock held
5361AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5362{
5363    RecordThread *thread = NULL;
5364    if (mRecordThreads.indexOfKey(input) >= 0) {
5365        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5366    }
5367    return thread;
5368}
5369
5370uint32_t AudioFlinger::nextUniqueId()
5371{
5372    return android_atomic_inc(&mNextUniqueId);
5373}
5374
5375AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5376{
5377    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5378        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5379        AudioStreamOut *output = thread->getOutput();
5380        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5381            return thread;
5382        }
5383    }
5384    return NULL;
5385}
5386
5387uint32_t AudioFlinger::primaryOutputDevice_l()
5388{
5389    PlaybackThread *thread = primaryPlaybackThread_l();
5390
5391    if (thread == NULL) {
5392        return 0;
5393    }
5394
5395    return thread->device();
5396}
5397
5398
5399// ----------------------------------------------------------------------------
5400//  Effect management
5401// ----------------------------------------------------------------------------
5402
5403
5404status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5405{
5406    Mutex::Autolock _l(mLock);
5407    return EffectQueryNumberEffects(numEffects);
5408}
5409
5410status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5411{
5412    Mutex::Autolock _l(mLock);
5413    return EffectQueryEffect(index, descriptor);
5414}
5415
5416status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5417{
5418    Mutex::Autolock _l(mLock);
5419    return EffectGetDescriptor(pUuid, descriptor);
5420}
5421
5422
5423sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5424        effect_descriptor_t *pDesc,
5425        const sp<IEffectClient>& effectClient,
5426        int32_t priority,
5427        int io,
5428        int sessionId,
5429        status_t *status,
5430        int *id,
5431        int *enabled)
5432{
5433    status_t lStatus = NO_ERROR;
5434    sp<EffectHandle> handle;
5435    effect_descriptor_t desc;
5436    sp<Client> client;
5437    wp<Client> wclient;
5438
5439    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5440            pid, effectClient.get(), priority, sessionId, io);
5441
5442    if (pDesc == NULL) {
5443        lStatus = BAD_VALUE;
5444        goto Exit;
5445    }
5446
5447    // check audio settings permission for global effects
5448    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5449        lStatus = PERMISSION_DENIED;
5450        goto Exit;
5451    }
5452
5453    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5454    // that can only be created by audio policy manager (running in same process)
5455    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5456        lStatus = PERMISSION_DENIED;
5457        goto Exit;
5458    }
5459
5460    if (io == 0) {
5461        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5462            // output must be specified by AudioPolicyManager when using session
5463            // AUDIO_SESSION_OUTPUT_STAGE
5464            lStatus = BAD_VALUE;
5465            goto Exit;
5466        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5467            // if the output returned by getOutputForEffect() is removed before we lock the
5468            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5469            // and we will exit safely
5470            io = AudioSystem::getOutputForEffect(&desc);
5471        }
5472    }
5473
5474    {
5475        Mutex::Autolock _l(mLock);
5476
5477
5478        if (!EffectIsNullUuid(&pDesc->uuid)) {
5479            // if uuid is specified, request effect descriptor
5480            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5481            if (lStatus < 0) {
5482                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5483                goto Exit;
5484            }
5485        } else {
5486            // if uuid is not specified, look for an available implementation
5487            // of the required type in effect factory
5488            if (EffectIsNullUuid(&pDesc->type)) {
5489                ALOGW("createEffect() no effect type");
5490                lStatus = BAD_VALUE;
5491                goto Exit;
5492            }
5493            uint32_t numEffects = 0;
5494            effect_descriptor_t d;
5495            d.flags = 0; // prevent compiler warning
5496            bool found = false;
5497
5498            lStatus = EffectQueryNumberEffects(&numEffects);
5499            if (lStatus < 0) {
5500                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5501                goto Exit;
5502            }
5503            for (uint32_t i = 0; i < numEffects; i++) {
5504                lStatus = EffectQueryEffect(i, &desc);
5505                if (lStatus < 0) {
5506                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5507                    continue;
5508                }
5509                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5510                    // If matching type found save effect descriptor. If the session is
5511                    // 0 and the effect is not auxiliary, continue enumeration in case
5512                    // an auxiliary version of this effect type is available
5513                    found = true;
5514                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5515                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5516                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5517                        break;
5518                    }
5519                }
5520            }
5521            if (!found) {
5522                lStatus = BAD_VALUE;
5523                ALOGW("createEffect() effect not found");
5524                goto Exit;
5525            }
5526            // For same effect type, chose auxiliary version over insert version if
5527            // connect to output mix (Compliance to OpenSL ES)
5528            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5529                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5530                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5531            }
5532        }
5533
5534        // Do not allow auxiliary effects on a session different from 0 (output mix)
5535        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5536             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5537            lStatus = INVALID_OPERATION;
5538            goto Exit;
5539        }
5540
5541        // check recording permission for visualizer
5542        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5543            !recordingAllowed()) {
5544            lStatus = PERMISSION_DENIED;
5545            goto Exit;
5546        }
5547
5548        // return effect descriptor
5549        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5550
5551        // If output is not specified try to find a matching audio session ID in one of the
5552        // output threads.
5553        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5554        // because of code checking output when entering the function.
5555        // Note: io is never 0 when creating an effect on an input
5556        if (io == 0) {
5557             // look for the thread where the specified audio session is present
5558            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5559                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5560                    io = mPlaybackThreads.keyAt(i);
5561                    break;
5562                }
5563            }
5564            if (io == 0) {
5565               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5566                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5567                       io = mRecordThreads.keyAt(i);
5568                       break;
5569                   }
5570               }
5571            }
5572            // If no output thread contains the requested session ID, default to
5573            // first output. The effect chain will be moved to the correct output
5574            // thread when a track with the same session ID is created
5575            if (io == 0 && mPlaybackThreads.size()) {
5576                io = mPlaybackThreads.keyAt(0);
5577            }
5578            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5579        }
5580        ThreadBase *thread = checkRecordThread_l(io);
5581        if (thread == NULL) {
5582            thread = checkPlaybackThread_l(io);
5583            if (thread == NULL) {
5584                ALOGE("createEffect() unknown output thread");
5585                lStatus = BAD_VALUE;
5586                goto Exit;
5587            }
5588        }
5589
5590        wclient = mClients.valueFor(pid);
5591
5592        if (wclient != NULL) {
5593            client = wclient.promote();
5594        } else {
5595            client = new Client(this, pid);
5596            mClients.add(pid, client);
5597        }
5598
5599        // create effect on selected output thread
5600        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5601                &desc, enabled, &lStatus);
5602        if (handle != 0 && id != NULL) {
5603            *id = handle->id();
5604        }
5605    }
5606
5607Exit:
5608    if(status) {
5609        *status = lStatus;
5610    }
5611    return handle;
5612}
5613
5614status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5615{
5616    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5617            sessionId, srcOutput, dstOutput);
5618    Mutex::Autolock _l(mLock);
5619    if (srcOutput == dstOutput) {
5620        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
5621        return NO_ERROR;
5622    }
5623    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5624    if (srcThread == NULL) {
5625        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
5626        return BAD_VALUE;
5627    }
5628    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5629    if (dstThread == NULL) {
5630        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
5631        return BAD_VALUE;
5632    }
5633
5634    Mutex::Autolock _dl(dstThread->mLock);
5635    Mutex::Autolock _sl(srcThread->mLock);
5636    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5637
5638    return NO_ERROR;
5639}
5640
5641// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5642status_t AudioFlinger::moveEffectChain_l(int sessionId,
5643                                   AudioFlinger::PlaybackThread *srcThread,
5644                                   AudioFlinger::PlaybackThread *dstThread,
5645                                   bool reRegister)
5646{
5647    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5648            sessionId, srcThread, dstThread);
5649
5650    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5651    if (chain == 0) {
5652        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5653                sessionId, srcThread);
5654        return INVALID_OPERATION;
5655    }
5656
5657    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5658    // so that a new chain is created with correct parameters when first effect is added. This is
5659    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5660    // removed.
5661    srcThread->removeEffectChain_l(chain);
5662
5663    // transfer all effects one by one so that new effect chain is created on new thread with
5664    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5665    int dstOutput = dstThread->id();
5666    sp<EffectChain> dstChain;
5667    uint32_t strategy = 0; // prevent compiler warning
5668    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5669    while (effect != 0) {
5670        srcThread->removeEffect_l(effect);
5671        dstThread->addEffect_l(effect);
5672        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5673        if (effect->state() == EffectModule::ACTIVE ||
5674                effect->state() == EffectModule::STOPPING) {
5675            effect->start();
5676        }
5677        // if the move request is not received from audio policy manager, the effect must be
5678        // re-registered with the new strategy and output
5679        if (dstChain == 0) {
5680            dstChain = effect->chain().promote();
5681            if (dstChain == 0) {
5682                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5683                srcThread->addEffect_l(effect);
5684                return NO_INIT;
5685            }
5686            strategy = dstChain->strategy();
5687        }
5688        if (reRegister) {
5689            AudioSystem::unregisterEffect(effect->id());
5690            AudioSystem::registerEffect(&effect->desc(),
5691                                        dstOutput,
5692                                        strategy,
5693                                        sessionId,
5694                                        effect->id());
5695        }
5696        effect = chain->getEffectFromId_l(0);
5697    }
5698
5699    return NO_ERROR;
5700}
5701
5702
5703// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5704sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5705        const sp<AudioFlinger::Client>& client,
5706        const sp<IEffectClient>& effectClient,
5707        int32_t priority,
5708        int sessionId,
5709        effect_descriptor_t *desc,
5710        int *enabled,
5711        status_t *status
5712        )
5713{
5714    sp<EffectModule> effect;
5715    sp<EffectHandle> handle;
5716    status_t lStatus;
5717    sp<EffectChain> chain;
5718    bool chainCreated = false;
5719    bool effectCreated = false;
5720    bool effectRegistered = false;
5721
5722    lStatus = initCheck();
5723    if (lStatus != NO_ERROR) {
5724        ALOGW("createEffect_l() Audio driver not initialized.");
5725        goto Exit;
5726    }
5727
5728    // Do not allow effects with session ID 0 on direct output or duplicating threads
5729    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5730    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5731        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5732                desc->name, sessionId);
5733        lStatus = BAD_VALUE;
5734        goto Exit;
5735    }
5736    // Only Pre processor effects are allowed on input threads and only on input threads
5737    if ((mType == RECORD &&
5738            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5739            (mType != RECORD &&
5740                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5741        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5742                desc->name, desc->flags, mType);
5743        lStatus = BAD_VALUE;
5744        goto Exit;
5745    }
5746
5747    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5748
5749    { // scope for mLock
5750        Mutex::Autolock _l(mLock);
5751
5752        // check for existing effect chain with the requested audio session
5753        chain = getEffectChain_l(sessionId);
5754        if (chain == 0) {
5755            // create a new chain for this session
5756            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5757            chain = new EffectChain(this, sessionId);
5758            addEffectChain_l(chain);
5759            chain->setStrategy(getStrategyForSession_l(sessionId));
5760            chainCreated = true;
5761        } else {
5762            effect = chain->getEffectFromDesc_l(desc);
5763        }
5764
5765        ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
5766
5767        if (effect == 0) {
5768            int id = mAudioFlinger->nextUniqueId();
5769            // Check CPU and memory usage
5770            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5771            if (lStatus != NO_ERROR) {
5772                goto Exit;
5773            }
5774            effectRegistered = true;
5775            // create a new effect module if none present in the chain
5776            effect = new EffectModule(this, chain, desc, id, sessionId);
5777            lStatus = effect->status();
5778            if (lStatus != NO_ERROR) {
5779                goto Exit;
5780            }
5781            lStatus = chain->addEffect_l(effect);
5782            if (lStatus != NO_ERROR) {
5783                goto Exit;
5784            }
5785            effectCreated = true;
5786
5787            effect->setDevice(mDevice);
5788            effect->setMode(mAudioFlinger->getMode());
5789        }
5790        // create effect handle and connect it to effect module
5791        handle = new EffectHandle(effect, client, effectClient, priority);
5792        lStatus = effect->addHandle(handle);
5793        if (enabled) {
5794            *enabled = (int)effect->isEnabled();
5795        }
5796    }
5797
5798Exit:
5799    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5800        Mutex::Autolock _l(mLock);
5801        if (effectCreated) {
5802            chain->removeEffect_l(effect);
5803        }
5804        if (effectRegistered) {
5805            AudioSystem::unregisterEffect(effect->id());
5806        }
5807        if (chainCreated) {
5808            removeEffectChain_l(chain);
5809        }
5810        handle.clear();
5811    }
5812
5813    if(status) {
5814        *status = lStatus;
5815    }
5816    return handle;
5817}
5818
5819sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5820{
5821    sp<EffectModule> effect;
5822
5823    sp<EffectChain> chain = getEffectChain_l(sessionId);
5824    if (chain != 0) {
5825        effect = chain->getEffectFromId_l(effectId);
5826    }
5827    return effect;
5828}
5829
5830// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5831// PlaybackThread::mLock held
5832status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5833{
5834    // check for existing effect chain with the requested audio session
5835    int sessionId = effect->sessionId();
5836    sp<EffectChain> chain = getEffectChain_l(sessionId);
5837    bool chainCreated = false;
5838
5839    if (chain == 0) {
5840        // create a new chain for this session
5841        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5842        chain = new EffectChain(this, sessionId);
5843        addEffectChain_l(chain);
5844        chain->setStrategy(getStrategyForSession_l(sessionId));
5845        chainCreated = true;
5846    }
5847    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5848
5849    if (chain->getEffectFromId_l(effect->id()) != 0) {
5850        ALOGW("addEffect_l() %p effect %s already present in chain %p",
5851                this, effect->desc().name, chain.get());
5852        return BAD_VALUE;
5853    }
5854
5855    status_t status = chain->addEffect_l(effect);
5856    if (status != NO_ERROR) {
5857        if (chainCreated) {
5858            removeEffectChain_l(chain);
5859        }
5860        return status;
5861    }
5862
5863    effect->setDevice(mDevice);
5864    effect->setMode(mAudioFlinger->getMode());
5865    return NO_ERROR;
5866}
5867
5868void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5869
5870    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5871    effect_descriptor_t desc = effect->desc();
5872    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5873        detachAuxEffect_l(effect->id());
5874    }
5875
5876    sp<EffectChain> chain = effect->chain().promote();
5877    if (chain != 0) {
5878        // remove effect chain if removing last effect
5879        if (chain->removeEffect_l(effect) == 0) {
5880            removeEffectChain_l(chain);
5881        }
5882    } else {
5883        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5884    }
5885}
5886
5887void AudioFlinger::ThreadBase::lockEffectChains_l(
5888        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5889{
5890    effectChains = mEffectChains;
5891    for (size_t i = 0; i < mEffectChains.size(); i++) {
5892        mEffectChains[i]->lock();
5893    }
5894}
5895
5896void AudioFlinger::ThreadBase::unlockEffectChains(
5897        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5898{
5899    for (size_t i = 0; i < effectChains.size(); i++) {
5900        effectChains[i]->unlock();
5901    }
5902}
5903
5904sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5905{
5906    Mutex::Autolock _l(mLock);
5907    return getEffectChain_l(sessionId);
5908}
5909
5910sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5911{
5912    sp<EffectChain> chain;
5913
5914    size_t size = mEffectChains.size();
5915    for (size_t i = 0; i < size; i++) {
5916        if (mEffectChains[i]->sessionId() == sessionId) {
5917            chain = mEffectChains[i];
5918            break;
5919        }
5920    }
5921    return chain;
5922}
5923
5924void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
5925{
5926    Mutex::Autolock _l(mLock);
5927    size_t size = mEffectChains.size();
5928    for (size_t i = 0; i < size; i++) {
5929        mEffectChains[i]->setMode_l(mode);
5930    }
5931}
5932
5933void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5934                                                    const wp<EffectHandle>& handle,
5935                                                    bool unpiniflast) {
5936
5937    Mutex::Autolock _l(mLock);
5938    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5939    // delete the effect module if removing last handle on it
5940    if (effect->removeHandle(handle) == 0) {
5941        if (!effect->isPinned() || unpiniflast) {
5942            removeEffect_l(effect);
5943            AudioSystem::unregisterEffect(effect->id());
5944        }
5945    }
5946}
5947
5948status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5949{
5950    int session = chain->sessionId();
5951    int16_t *buffer = mMixBuffer;
5952    bool ownsBuffer = false;
5953
5954    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5955    if (session > 0) {
5956        // Only one effect chain can be present in direct output thread and it uses
5957        // the mix buffer as input
5958        if (mType != DIRECT) {
5959            size_t numSamples = mFrameCount * mChannelCount;
5960            buffer = new int16_t[numSamples];
5961            memset(buffer, 0, numSamples * sizeof(int16_t));
5962            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5963            ownsBuffer = true;
5964        }
5965
5966        // Attach all tracks with same session ID to this chain.
5967        for (size_t i = 0; i < mTracks.size(); ++i) {
5968            sp<Track> track = mTracks[i];
5969            if (session == track->sessionId()) {
5970                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5971                track->setMainBuffer(buffer);
5972                chain->incTrackCnt();
5973            }
5974        }
5975
5976        // indicate all active tracks in the chain
5977        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5978            sp<Track> track = mActiveTracks[i].promote();
5979            if (track == 0) continue;
5980            if (session == track->sessionId()) {
5981                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5982                chain->incActiveTrackCnt();
5983            }
5984        }
5985    }
5986
5987    chain->setInBuffer(buffer, ownsBuffer);
5988    chain->setOutBuffer(mMixBuffer);
5989    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5990    // chains list in order to be processed last as it contains output stage effects
5991    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5992    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5993    // after track specific effects and before output stage
5994    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5995    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5996    // Effect chain for other sessions are inserted at beginning of effect
5997    // chains list to be processed before output mix effects. Relative order between other
5998    // sessions is not important
5999    size_t size = mEffectChains.size();
6000    size_t i = 0;
6001    for (i = 0; i < size; i++) {
6002        if (mEffectChains[i]->sessionId() < session) break;
6003    }
6004    mEffectChains.insertAt(chain, i);
6005    checkSuspendOnAddEffectChain_l(chain);
6006
6007    return NO_ERROR;
6008}
6009
6010size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
6011{
6012    int session = chain->sessionId();
6013
6014    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
6015
6016    for (size_t i = 0; i < mEffectChains.size(); i++) {
6017        if (chain == mEffectChains[i]) {
6018            mEffectChains.removeAt(i);
6019            // detach all active tracks from the chain
6020            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
6021                sp<Track> track = mActiveTracks[i].promote();
6022                if (track == 0) continue;
6023                if (session == track->sessionId()) {
6024                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
6025                            chain.get(), session);
6026                    chain->decActiveTrackCnt();
6027                }
6028            }
6029
6030            // detach all tracks with same session ID from this chain
6031            for (size_t i = 0; i < mTracks.size(); ++i) {
6032                sp<Track> track = mTracks[i];
6033                if (session == track->sessionId()) {
6034                    track->setMainBuffer(mMixBuffer);
6035                    chain->decTrackCnt();
6036                }
6037            }
6038            break;
6039        }
6040    }
6041    return mEffectChains.size();
6042}
6043
6044status_t AudioFlinger::PlaybackThread::attachAuxEffect(
6045        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6046{
6047    Mutex::Autolock _l(mLock);
6048    return attachAuxEffect_l(track, EffectId);
6049}
6050
6051status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6052        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6053{
6054    status_t status = NO_ERROR;
6055
6056    if (EffectId == 0) {
6057        track->setAuxBuffer(0, NULL);
6058    } else {
6059        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6060        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6061        if (effect != 0) {
6062            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6063                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6064            } else {
6065                status = INVALID_OPERATION;
6066            }
6067        } else {
6068            status = BAD_VALUE;
6069        }
6070    }
6071    return status;
6072}
6073
6074void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6075{
6076     for (size_t i = 0; i < mTracks.size(); ++i) {
6077        sp<Track> track = mTracks[i];
6078        if (track->auxEffectId() == effectId) {
6079            attachAuxEffect_l(track, 0);
6080        }
6081    }
6082}
6083
6084status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6085{
6086    // only one chain per input thread
6087    if (mEffectChains.size() != 0) {
6088        return INVALID_OPERATION;
6089    }
6090    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6091
6092    chain->setInBuffer(NULL);
6093    chain->setOutBuffer(NULL);
6094
6095    checkSuspendOnAddEffectChain_l(chain);
6096
6097    mEffectChains.add(chain);
6098
6099    return NO_ERROR;
6100}
6101
6102size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6103{
6104    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6105    ALOGW_IF(mEffectChains.size() != 1,
6106            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6107            chain.get(), mEffectChains.size(), this);
6108    if (mEffectChains.size() == 1) {
6109        mEffectChains.removeAt(0);
6110    }
6111    return 0;
6112}
6113
6114// ----------------------------------------------------------------------------
6115//  EffectModule implementation
6116// ----------------------------------------------------------------------------
6117
6118#undef LOG_TAG
6119#define LOG_TAG "AudioFlinger::EffectModule"
6120
6121AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6122                                        const wp<AudioFlinger::EffectChain>& chain,
6123                                        effect_descriptor_t *desc,
6124                                        int id,
6125                                        int sessionId)
6126    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6127      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6128{
6129    ALOGV("Constructor %p", this);
6130    int lStatus;
6131    sp<ThreadBase> thread = mThread.promote();
6132    if (thread == 0) {
6133        return;
6134    }
6135
6136    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6137
6138    // create effect engine from effect factory
6139    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6140
6141    if (mStatus != NO_ERROR) {
6142        return;
6143    }
6144    lStatus = init();
6145    if (lStatus < 0) {
6146        mStatus = lStatus;
6147        goto Error;
6148    }
6149
6150    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6151        mPinned = true;
6152    }
6153    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6154    return;
6155Error:
6156    EffectRelease(mEffectInterface);
6157    mEffectInterface = NULL;
6158    ALOGV("Constructor Error %d", mStatus);
6159}
6160
6161AudioFlinger::EffectModule::~EffectModule()
6162{
6163    ALOGV("Destructor %p", this);
6164    if (mEffectInterface != NULL) {
6165        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6166                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6167            sp<ThreadBase> thread = mThread.promote();
6168            if (thread != 0) {
6169                audio_stream_t *stream = thread->stream();
6170                if (stream != NULL) {
6171                    stream->remove_audio_effect(stream, mEffectInterface);
6172                }
6173            }
6174        }
6175        // release effect engine
6176        EffectRelease(mEffectInterface);
6177    }
6178}
6179
6180status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6181{
6182    status_t status;
6183
6184    Mutex::Autolock _l(mLock);
6185    // First handle in mHandles has highest priority and controls the effect module
6186    int priority = handle->priority();
6187    size_t size = mHandles.size();
6188    sp<EffectHandle> h;
6189    size_t i;
6190    for (i = 0; i < size; i++) {
6191        h = mHandles[i].promote();
6192        if (h == 0) continue;
6193        if (h->priority() <= priority) break;
6194    }
6195    // if inserted in first place, move effect control from previous owner to this handle
6196    if (i == 0) {
6197        bool enabled = false;
6198        if (h != 0) {
6199            enabled = h->enabled();
6200            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6201        }
6202        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6203        status = NO_ERROR;
6204    } else {
6205        status = ALREADY_EXISTS;
6206    }
6207    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6208    mHandles.insertAt(handle, i);
6209    return status;
6210}
6211
6212size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6213{
6214    Mutex::Autolock _l(mLock);
6215    size_t size = mHandles.size();
6216    size_t i;
6217    for (i = 0; i < size; i++) {
6218        if (mHandles[i] == handle) break;
6219    }
6220    if (i == size) {
6221        return size;
6222    }
6223    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6224
6225    bool enabled = false;
6226    EffectHandle *hdl = handle.unsafe_get();
6227    if (hdl) {
6228        ALOGV("removeHandle() unsafe_get OK");
6229        enabled = hdl->enabled();
6230    }
6231    mHandles.removeAt(i);
6232    size = mHandles.size();
6233    // if removed from first place, move effect control from this handle to next in line
6234    if (i == 0 && size != 0) {
6235        sp<EffectHandle> h = mHandles[0].promote();
6236        if (h != 0) {
6237            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6238        }
6239    }
6240
6241    // Prevent calls to process() and other functions on effect interface from now on.
6242    // The effect engine will be released by the destructor when the last strong reference on
6243    // this object is released which can happen after next process is called.
6244    if (size == 0 && !mPinned) {
6245        mState = DESTROYED;
6246    }
6247
6248    return size;
6249}
6250
6251sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6252{
6253    Mutex::Autolock _l(mLock);
6254    sp<EffectHandle> handle;
6255    if (mHandles.size() != 0) {
6256        handle = mHandles[0].promote();
6257    }
6258    return handle;
6259}
6260
6261void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6262{
6263    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6264    // keep a strong reference on this EffectModule to avoid calling the
6265    // destructor before we exit
6266    sp<EffectModule> keep(this);
6267    {
6268        sp<ThreadBase> thread = mThread.promote();
6269        if (thread != 0) {
6270            thread->disconnectEffect(keep, handle, unpiniflast);
6271        }
6272    }
6273}
6274
6275void AudioFlinger::EffectModule::updateState() {
6276    Mutex::Autolock _l(mLock);
6277
6278    switch (mState) {
6279    case RESTART:
6280        reset_l();
6281        // FALL THROUGH
6282
6283    case STARTING:
6284        // clear auxiliary effect input buffer for next accumulation
6285        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6286            memset(mConfig.inputCfg.buffer.raw,
6287                   0,
6288                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6289        }
6290        start_l();
6291        mState = ACTIVE;
6292        break;
6293    case STOPPING:
6294        stop_l();
6295        mDisableWaitCnt = mMaxDisableWaitCnt;
6296        mState = STOPPED;
6297        break;
6298    case STOPPED:
6299        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6300        // turn off sequence.
6301        if (--mDisableWaitCnt == 0) {
6302            reset_l();
6303            mState = IDLE;
6304        }
6305        break;
6306    default: //IDLE , ACTIVE, DESTROYED
6307        break;
6308    }
6309}
6310
6311void AudioFlinger::EffectModule::process()
6312{
6313    Mutex::Autolock _l(mLock);
6314
6315    if (mState == DESTROYED || mEffectInterface == NULL ||
6316            mConfig.inputCfg.buffer.raw == NULL ||
6317            mConfig.outputCfg.buffer.raw == NULL) {
6318        return;
6319    }
6320
6321    if (isProcessEnabled()) {
6322        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6323        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6324            ditherAndClamp(mConfig.inputCfg.buffer.s32,
6325                                        mConfig.inputCfg.buffer.s32,
6326                                        mConfig.inputCfg.buffer.frameCount/2);
6327        }
6328
6329        // do the actual processing in the effect engine
6330        int ret = (*mEffectInterface)->process(mEffectInterface,
6331                                               &mConfig.inputCfg.buffer,
6332                                               &mConfig.outputCfg.buffer);
6333
6334        // force transition to IDLE state when engine is ready
6335        if (mState == STOPPED && ret == -ENODATA) {
6336            mDisableWaitCnt = 1;
6337        }
6338
6339        // clear auxiliary effect input buffer for next accumulation
6340        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6341            memset(mConfig.inputCfg.buffer.raw, 0,
6342                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6343        }
6344    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6345                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6346        // If an insert effect is idle and input buffer is different from output buffer,
6347        // accumulate input onto output
6348        sp<EffectChain> chain = mChain.promote();
6349        if (chain != 0 && chain->activeTrackCnt() != 0) {
6350            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6351            int16_t *in = mConfig.inputCfg.buffer.s16;
6352            int16_t *out = mConfig.outputCfg.buffer.s16;
6353            for (size_t i = 0; i < frameCnt; i++) {
6354                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6355            }
6356        }
6357    }
6358}
6359
6360void AudioFlinger::EffectModule::reset_l()
6361{
6362    if (mEffectInterface == NULL) {
6363        return;
6364    }
6365    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6366}
6367
6368status_t AudioFlinger::EffectModule::configure()
6369{
6370    uint32_t channels;
6371    if (mEffectInterface == NULL) {
6372        return NO_INIT;
6373    }
6374
6375    sp<ThreadBase> thread = mThread.promote();
6376    if (thread == 0) {
6377        return DEAD_OBJECT;
6378    }
6379
6380    // TODO: handle configuration of effects replacing track process
6381    if (thread->channelCount() == 1) {
6382        channels = AUDIO_CHANNEL_OUT_MONO;
6383    } else {
6384        channels = AUDIO_CHANNEL_OUT_STEREO;
6385    }
6386
6387    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6388        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6389    } else {
6390        mConfig.inputCfg.channels = channels;
6391    }
6392    mConfig.outputCfg.channels = channels;
6393    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6394    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6395    mConfig.inputCfg.samplingRate = thread->sampleRate();
6396    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6397    mConfig.inputCfg.bufferProvider.cookie = NULL;
6398    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6399    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6400    mConfig.outputCfg.bufferProvider.cookie = NULL;
6401    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6402    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6403    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6404    // Insert effect:
6405    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6406    // always overwrites output buffer: input buffer == output buffer
6407    // - in other sessions:
6408    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6409    //      other effect: overwrites output buffer: input buffer == output buffer
6410    // Auxiliary effect:
6411    //      accumulates in output buffer: input buffer != output buffer
6412    // Therefore: accumulate <=> input buffer != output buffer
6413    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6414        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6415    } else {
6416        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6417    }
6418    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6419    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6420    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6421    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6422
6423    ALOGV("configure() %p thread %p buffer %p framecount %d",
6424            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6425
6426    status_t cmdStatus;
6427    uint32_t size = sizeof(int);
6428    status_t status = (*mEffectInterface)->command(mEffectInterface,
6429                                                   EFFECT_CMD_SET_CONFIG,
6430                                                   sizeof(effect_config_t),
6431                                                   &mConfig,
6432                                                   &size,
6433                                                   &cmdStatus);
6434    if (status == 0) {
6435        status = cmdStatus;
6436    }
6437
6438    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6439            (1000 * mConfig.outputCfg.buffer.frameCount);
6440
6441    return status;
6442}
6443
6444status_t AudioFlinger::EffectModule::init()
6445{
6446    Mutex::Autolock _l(mLock);
6447    if (mEffectInterface == NULL) {
6448        return NO_INIT;
6449    }
6450    status_t cmdStatus;
6451    uint32_t size = sizeof(status_t);
6452    status_t status = (*mEffectInterface)->command(mEffectInterface,
6453                                                   EFFECT_CMD_INIT,
6454                                                   0,
6455                                                   NULL,
6456                                                   &size,
6457                                                   &cmdStatus);
6458    if (status == 0) {
6459        status = cmdStatus;
6460    }
6461    return status;
6462}
6463
6464status_t AudioFlinger::EffectModule::start()
6465{
6466    Mutex::Autolock _l(mLock);
6467    return start_l();
6468}
6469
6470status_t AudioFlinger::EffectModule::start_l()
6471{
6472    if (mEffectInterface == NULL) {
6473        return NO_INIT;
6474    }
6475    status_t cmdStatus;
6476    uint32_t size = sizeof(status_t);
6477    status_t status = (*mEffectInterface)->command(mEffectInterface,
6478                                                   EFFECT_CMD_ENABLE,
6479                                                   0,
6480                                                   NULL,
6481                                                   &size,
6482                                                   &cmdStatus);
6483    if (status == 0) {
6484        status = cmdStatus;
6485    }
6486    if (status == 0 &&
6487            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6488             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6489        sp<ThreadBase> thread = mThread.promote();
6490        if (thread != 0) {
6491            audio_stream_t *stream = thread->stream();
6492            if (stream != NULL) {
6493                stream->add_audio_effect(stream, mEffectInterface);
6494            }
6495        }
6496    }
6497    return status;
6498}
6499
6500status_t AudioFlinger::EffectModule::stop()
6501{
6502    Mutex::Autolock _l(mLock);
6503    return stop_l();
6504}
6505
6506status_t AudioFlinger::EffectModule::stop_l()
6507{
6508    if (mEffectInterface == NULL) {
6509        return NO_INIT;
6510    }
6511    status_t cmdStatus;
6512    uint32_t size = sizeof(status_t);
6513    status_t status = (*mEffectInterface)->command(mEffectInterface,
6514                                                   EFFECT_CMD_DISABLE,
6515                                                   0,
6516                                                   NULL,
6517                                                   &size,
6518                                                   &cmdStatus);
6519    if (status == 0) {
6520        status = cmdStatus;
6521    }
6522    if (status == 0 &&
6523            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6524             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6525        sp<ThreadBase> thread = mThread.promote();
6526        if (thread != 0) {
6527            audio_stream_t *stream = thread->stream();
6528            if (stream != NULL) {
6529                stream->remove_audio_effect(stream, mEffectInterface);
6530            }
6531        }
6532    }
6533    return status;
6534}
6535
6536status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6537                                             uint32_t cmdSize,
6538                                             void *pCmdData,
6539                                             uint32_t *replySize,
6540                                             void *pReplyData)
6541{
6542    Mutex::Autolock _l(mLock);
6543//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6544
6545    if (mState == DESTROYED || mEffectInterface == NULL) {
6546        return NO_INIT;
6547    }
6548    status_t status = (*mEffectInterface)->command(mEffectInterface,
6549                                                   cmdCode,
6550                                                   cmdSize,
6551                                                   pCmdData,
6552                                                   replySize,
6553                                                   pReplyData);
6554    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6555        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6556        for (size_t i = 1; i < mHandles.size(); i++) {
6557            sp<EffectHandle> h = mHandles[i].promote();
6558            if (h != 0) {
6559                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6560            }
6561        }
6562    }
6563    return status;
6564}
6565
6566status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6567{
6568
6569    Mutex::Autolock _l(mLock);
6570    ALOGV("setEnabled %p enabled %d", this, enabled);
6571
6572    if (enabled != isEnabled()) {
6573        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6574        if (enabled && status != NO_ERROR) {
6575            return status;
6576        }
6577
6578        switch (mState) {
6579        // going from disabled to enabled
6580        case IDLE:
6581            mState = STARTING;
6582            break;
6583        case STOPPED:
6584            mState = RESTART;
6585            break;
6586        case STOPPING:
6587            mState = ACTIVE;
6588            break;
6589
6590        // going from enabled to disabled
6591        case RESTART:
6592            mState = STOPPED;
6593            break;
6594        case STARTING:
6595            mState = IDLE;
6596            break;
6597        case ACTIVE:
6598            mState = STOPPING;
6599            break;
6600        case DESTROYED:
6601            return NO_ERROR; // simply ignore as we are being destroyed
6602        }
6603        for (size_t i = 1; i < mHandles.size(); i++) {
6604            sp<EffectHandle> h = mHandles[i].promote();
6605            if (h != 0) {
6606                h->setEnabled(enabled);
6607            }
6608        }
6609    }
6610    return NO_ERROR;
6611}
6612
6613bool AudioFlinger::EffectModule::isEnabled()
6614{
6615    switch (mState) {
6616    case RESTART:
6617    case STARTING:
6618    case ACTIVE:
6619        return true;
6620    case IDLE:
6621    case STOPPING:
6622    case STOPPED:
6623    case DESTROYED:
6624    default:
6625        return false;
6626    }
6627}
6628
6629bool AudioFlinger::EffectModule::isProcessEnabled()
6630{
6631    switch (mState) {
6632    case RESTART:
6633    case ACTIVE:
6634    case STOPPING:
6635    case STOPPED:
6636        return true;
6637    case IDLE:
6638    case STARTING:
6639    case DESTROYED:
6640    default:
6641        return false;
6642    }
6643}
6644
6645status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6646{
6647    Mutex::Autolock _l(mLock);
6648    status_t status = NO_ERROR;
6649
6650    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6651    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6652    if (isProcessEnabled() &&
6653            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6654            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6655        status_t cmdStatus;
6656        uint32_t volume[2];
6657        uint32_t *pVolume = NULL;
6658        uint32_t size = sizeof(volume);
6659        volume[0] = *left;
6660        volume[1] = *right;
6661        if (controller) {
6662            pVolume = volume;
6663        }
6664        status = (*mEffectInterface)->command(mEffectInterface,
6665                                              EFFECT_CMD_SET_VOLUME,
6666                                              size,
6667                                              volume,
6668                                              &size,
6669                                              pVolume);
6670        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6671            *left = volume[0];
6672            *right = volume[1];
6673        }
6674    }
6675    return status;
6676}
6677
6678status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6679{
6680    Mutex::Autolock _l(mLock);
6681    status_t status = NO_ERROR;
6682    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6683        // audio pre processing modules on RecordThread can receive both output and
6684        // input device indication in the same call
6685        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6686        if (dev) {
6687            status_t cmdStatus;
6688            uint32_t size = sizeof(status_t);
6689
6690            status = (*mEffectInterface)->command(mEffectInterface,
6691                                                  EFFECT_CMD_SET_DEVICE,
6692                                                  sizeof(uint32_t),
6693                                                  &dev,
6694                                                  &size,
6695                                                  &cmdStatus);
6696            if (status == NO_ERROR) {
6697                status = cmdStatus;
6698            }
6699        }
6700        dev = device & AUDIO_DEVICE_IN_ALL;
6701        if (dev) {
6702            status_t cmdStatus;
6703            uint32_t size = sizeof(status_t);
6704
6705            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6706                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6707                                                  sizeof(uint32_t),
6708                                                  &dev,
6709                                                  &size,
6710                                                  &cmdStatus);
6711            if (status2 == NO_ERROR) {
6712                status2 = cmdStatus;
6713            }
6714            if (status == NO_ERROR) {
6715                status = status2;
6716            }
6717        }
6718    }
6719    return status;
6720}
6721
6722status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
6723{
6724    Mutex::Autolock _l(mLock);
6725    status_t status = NO_ERROR;
6726    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6727        status_t cmdStatus;
6728        uint32_t size = sizeof(status_t);
6729        status = (*mEffectInterface)->command(mEffectInterface,
6730                                              EFFECT_CMD_SET_AUDIO_MODE,
6731                                              sizeof(audio_mode_t),
6732                                              &mode,
6733                                              &size,
6734                                              &cmdStatus);
6735        if (status == NO_ERROR) {
6736            status = cmdStatus;
6737        }
6738    }
6739    return status;
6740}
6741
6742void AudioFlinger::EffectModule::setSuspended(bool suspended)
6743{
6744    Mutex::Autolock _l(mLock);
6745    mSuspended = suspended;
6746}
6747
6748bool AudioFlinger::EffectModule::suspended() const
6749{
6750    Mutex::Autolock _l(mLock);
6751    return mSuspended;
6752}
6753
6754status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6755{
6756    const size_t SIZE = 256;
6757    char buffer[SIZE];
6758    String8 result;
6759
6760    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6761    result.append(buffer);
6762
6763    bool locked = tryLock(mLock);
6764    // failed to lock - AudioFlinger is probably deadlocked
6765    if (!locked) {
6766        result.append("\t\tCould not lock Fx mutex:\n");
6767    }
6768
6769    result.append("\t\tSession Status State Engine:\n");
6770    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6771            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6772    result.append(buffer);
6773
6774    result.append("\t\tDescriptor:\n");
6775    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6776            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6777            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6778            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6779    result.append(buffer);
6780    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6781                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6782                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6783                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6784    result.append(buffer);
6785    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6786            mDescriptor.apiVersion,
6787            mDescriptor.flags);
6788    result.append(buffer);
6789    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6790            mDescriptor.name);
6791    result.append(buffer);
6792    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6793            mDescriptor.implementor);
6794    result.append(buffer);
6795
6796    result.append("\t\t- Input configuration:\n");
6797    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6798    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6799            (uint32_t)mConfig.inputCfg.buffer.raw,
6800            mConfig.inputCfg.buffer.frameCount,
6801            mConfig.inputCfg.samplingRate,
6802            mConfig.inputCfg.channels,
6803            mConfig.inputCfg.format);
6804    result.append(buffer);
6805
6806    result.append("\t\t- Output configuration:\n");
6807    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6808    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6809            (uint32_t)mConfig.outputCfg.buffer.raw,
6810            mConfig.outputCfg.buffer.frameCount,
6811            mConfig.outputCfg.samplingRate,
6812            mConfig.outputCfg.channels,
6813            mConfig.outputCfg.format);
6814    result.append(buffer);
6815
6816    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6817    result.append(buffer);
6818    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6819    for (size_t i = 0; i < mHandles.size(); ++i) {
6820        sp<EffectHandle> handle = mHandles[i].promote();
6821        if (handle != 0) {
6822            handle->dump(buffer, SIZE);
6823            result.append(buffer);
6824        }
6825    }
6826
6827    result.append("\n");
6828
6829    write(fd, result.string(), result.length());
6830
6831    if (locked) {
6832        mLock.unlock();
6833    }
6834
6835    return NO_ERROR;
6836}
6837
6838// ----------------------------------------------------------------------------
6839//  EffectHandle implementation
6840// ----------------------------------------------------------------------------
6841
6842#undef LOG_TAG
6843#define LOG_TAG "AudioFlinger::EffectHandle"
6844
6845AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6846                                        const sp<AudioFlinger::Client>& client,
6847                                        const sp<IEffectClient>& effectClient,
6848                                        int32_t priority)
6849    : BnEffect(),
6850    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6851    mPriority(priority), mHasControl(false), mEnabled(false)
6852{
6853    ALOGV("constructor %p", this);
6854
6855    if (client == 0) {
6856        return;
6857    }
6858    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6859    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6860    if (mCblkMemory != 0) {
6861        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6862
6863        if (mCblk) {
6864            new(mCblk) effect_param_cblk_t();
6865            mBuffer = (uint8_t *)mCblk + bufOffset;
6866         }
6867    } else {
6868        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6869        return;
6870    }
6871}
6872
6873AudioFlinger::EffectHandle::~EffectHandle()
6874{
6875    ALOGV("Destructor %p", this);
6876    disconnect(false);
6877    ALOGV("Destructor DONE %p", this);
6878}
6879
6880status_t AudioFlinger::EffectHandle::enable()
6881{
6882    ALOGV("enable %p", this);
6883    if (!mHasControl) return INVALID_OPERATION;
6884    if (mEffect == 0) return DEAD_OBJECT;
6885
6886    if (mEnabled) {
6887        return NO_ERROR;
6888    }
6889
6890    mEnabled = true;
6891
6892    sp<ThreadBase> thread = mEffect->thread().promote();
6893    if (thread != 0) {
6894        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6895    }
6896
6897    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6898    if (mEffect->suspended()) {
6899        return NO_ERROR;
6900    }
6901
6902    status_t status = mEffect->setEnabled(true);
6903    if (status != NO_ERROR) {
6904        if (thread != 0) {
6905            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6906        }
6907        mEnabled = false;
6908    }
6909    return status;
6910}
6911
6912status_t AudioFlinger::EffectHandle::disable()
6913{
6914    ALOGV("disable %p", this);
6915    if (!mHasControl) return INVALID_OPERATION;
6916    if (mEffect == 0) return DEAD_OBJECT;
6917
6918    if (!mEnabled) {
6919        return NO_ERROR;
6920    }
6921    mEnabled = false;
6922
6923    if (mEffect->suspended()) {
6924        return NO_ERROR;
6925    }
6926
6927    status_t status = mEffect->setEnabled(false);
6928
6929    sp<ThreadBase> thread = mEffect->thread().promote();
6930    if (thread != 0) {
6931        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6932    }
6933
6934    return status;
6935}
6936
6937void AudioFlinger::EffectHandle::disconnect()
6938{
6939    disconnect(true);
6940}
6941
6942void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6943{
6944    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6945    if (mEffect == 0) {
6946        return;
6947    }
6948    mEffect->disconnect(this, unpiniflast);
6949
6950    if (mHasControl && mEnabled) {
6951        sp<ThreadBase> thread = mEffect->thread().promote();
6952        if (thread != 0) {
6953            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6954        }
6955    }
6956
6957    // release sp on module => module destructor can be called now
6958    mEffect.clear();
6959    if (mClient != 0) {
6960        if (mCblk) {
6961            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6962        }
6963        mCblkMemory.clear();            // and free the shared memory
6964        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6965        mClient.clear();
6966    }
6967}
6968
6969status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6970                                             uint32_t cmdSize,
6971                                             void *pCmdData,
6972                                             uint32_t *replySize,
6973                                             void *pReplyData)
6974{
6975//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6976//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6977
6978    // only get parameter command is permitted for applications not controlling the effect
6979    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6980        return INVALID_OPERATION;
6981    }
6982    if (mEffect == 0) return DEAD_OBJECT;
6983    if (mClient == 0) return INVALID_OPERATION;
6984
6985    // handle commands that are not forwarded transparently to effect engine
6986    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6987        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6988        // no risk to block the whole media server process or mixer threads is we are stuck here
6989        Mutex::Autolock _l(mCblk->lock);
6990        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6991            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6992            mCblk->serverIndex = 0;
6993            mCblk->clientIndex = 0;
6994            return BAD_VALUE;
6995        }
6996        status_t status = NO_ERROR;
6997        while (mCblk->serverIndex < mCblk->clientIndex) {
6998            int reply;
6999            uint32_t rsize = sizeof(int);
7000            int *p = (int *)(mBuffer + mCblk->serverIndex);
7001            int size = *p++;
7002            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
7003                ALOGW("command(): invalid parameter block size");
7004                break;
7005            }
7006            effect_param_t *param = (effect_param_t *)p;
7007            if (param->psize == 0 || param->vsize == 0) {
7008                ALOGW("command(): null parameter or value size");
7009                mCblk->serverIndex += size;
7010                continue;
7011            }
7012            uint32_t psize = sizeof(effect_param_t) +
7013                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
7014                             param->vsize;
7015            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
7016                                            psize,
7017                                            p,
7018                                            &rsize,
7019                                            &reply);
7020            // stop at first error encountered
7021            if (ret != NO_ERROR) {
7022                status = ret;
7023                *(int *)pReplyData = reply;
7024                break;
7025            } else if (reply != NO_ERROR) {
7026                *(int *)pReplyData = reply;
7027                break;
7028            }
7029            mCblk->serverIndex += size;
7030        }
7031        mCblk->serverIndex = 0;
7032        mCblk->clientIndex = 0;
7033        return status;
7034    } else if (cmdCode == EFFECT_CMD_ENABLE) {
7035        *(int *)pReplyData = NO_ERROR;
7036        return enable();
7037    } else if (cmdCode == EFFECT_CMD_DISABLE) {
7038        *(int *)pReplyData = NO_ERROR;
7039        return disable();
7040    }
7041
7042    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7043}
7044
7045sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
7046    return mCblkMemory;
7047}
7048
7049void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
7050{
7051    ALOGV("setControl %p control %d", this, hasControl);
7052
7053    mHasControl = hasControl;
7054    mEnabled = enabled;
7055
7056    if (signal && mEffectClient != 0) {
7057        mEffectClient->controlStatusChanged(hasControl);
7058    }
7059}
7060
7061void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7062                                                 uint32_t cmdSize,
7063                                                 void *pCmdData,
7064                                                 uint32_t replySize,
7065                                                 void *pReplyData)
7066{
7067    if (mEffectClient != 0) {
7068        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7069    }
7070}
7071
7072
7073
7074void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7075{
7076    if (mEffectClient != 0) {
7077        mEffectClient->enableStatusChanged(enabled);
7078    }
7079}
7080
7081status_t AudioFlinger::EffectHandle::onTransact(
7082    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7083{
7084    return BnEffect::onTransact(code, data, reply, flags);
7085}
7086
7087
7088void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7089{
7090    bool locked = mCblk ? tryLock(mCblk->lock) : false;
7091
7092    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7093            (mClient == NULL) ? getpid() : mClient->pid(),
7094            mPriority,
7095            mHasControl,
7096            !locked,
7097            mCblk ? mCblk->clientIndex : 0,
7098            mCblk ? mCblk->serverIndex : 0
7099            );
7100
7101    if (locked) {
7102        mCblk->lock.unlock();
7103    }
7104}
7105
7106#undef LOG_TAG
7107#define LOG_TAG "AudioFlinger::EffectChain"
7108
7109AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7110                                        int sessionId)
7111    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7112      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7113      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7114{
7115    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7116    sp<ThreadBase> thread = mThread.promote();
7117    if (thread == 0) {
7118        return;
7119    }
7120    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7121                                    thread->frameCount();
7122}
7123
7124AudioFlinger::EffectChain::~EffectChain()
7125{
7126    if (mOwnInBuffer) {
7127        delete mInBuffer;
7128    }
7129
7130}
7131
7132// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7133sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7134{
7135    sp<EffectModule> effect;
7136    size_t size = mEffects.size();
7137
7138    for (size_t i = 0; i < size; i++) {
7139        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7140            effect = mEffects[i];
7141            break;
7142        }
7143    }
7144    return effect;
7145}
7146
7147// getEffectFromId_l() must be called with ThreadBase::mLock held
7148sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7149{
7150    sp<EffectModule> effect;
7151    size_t size = mEffects.size();
7152
7153    for (size_t i = 0; i < size; i++) {
7154        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7155        if (id == 0 || mEffects[i]->id() == id) {
7156            effect = mEffects[i];
7157            break;
7158        }
7159    }
7160    return effect;
7161}
7162
7163// getEffectFromType_l() must be called with ThreadBase::mLock held
7164sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7165        const effect_uuid_t *type)
7166{
7167    sp<EffectModule> effect;
7168    size_t size = mEffects.size();
7169
7170    for (size_t i = 0; i < size; i++) {
7171        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7172            effect = mEffects[i];
7173            break;
7174        }
7175    }
7176    return effect;
7177}
7178
7179// Must be called with EffectChain::mLock locked
7180void AudioFlinger::EffectChain::process_l()
7181{
7182    sp<ThreadBase> thread = mThread.promote();
7183    if (thread == 0) {
7184        ALOGW("process_l(): cannot promote mixer thread");
7185        return;
7186    }
7187    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7188            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7189    // always process effects unless no more tracks are on the session and the effect tail
7190    // has been rendered
7191    bool doProcess = true;
7192    if (!isGlobalSession) {
7193        bool tracksOnSession = (trackCnt() != 0);
7194
7195        if (!tracksOnSession && mTailBufferCount == 0) {
7196            doProcess = false;
7197        }
7198
7199        if (activeTrackCnt() == 0) {
7200            // if no track is active and the effect tail has not been rendered,
7201            // the input buffer must be cleared here as the mixer process will not do it
7202            if (tracksOnSession || mTailBufferCount > 0) {
7203                size_t numSamples = thread->frameCount() * thread->channelCount();
7204                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7205                if (mTailBufferCount > 0) {
7206                    mTailBufferCount--;
7207                }
7208            }
7209        }
7210    }
7211
7212    size_t size = mEffects.size();
7213    if (doProcess) {
7214        for (size_t i = 0; i < size; i++) {
7215            mEffects[i]->process();
7216        }
7217    }
7218    for (size_t i = 0; i < size; i++) {
7219        mEffects[i]->updateState();
7220    }
7221}
7222
7223// addEffect_l() must be called with PlaybackThread::mLock held
7224status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7225{
7226    effect_descriptor_t desc = effect->desc();
7227    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7228
7229    Mutex::Autolock _l(mLock);
7230    effect->setChain(this);
7231    sp<ThreadBase> thread = mThread.promote();
7232    if (thread == 0) {
7233        return NO_INIT;
7234    }
7235    effect->setThread(thread);
7236
7237    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7238        // Auxiliary effects are inserted at the beginning of mEffects vector as
7239        // they are processed first and accumulated in chain input buffer
7240        mEffects.insertAt(effect, 0);
7241
7242        // the input buffer for auxiliary effect contains mono samples in
7243        // 32 bit format. This is to avoid saturation in AudoMixer
7244        // accumulation stage. Saturation is done in EffectModule::process() before
7245        // calling the process in effect engine
7246        size_t numSamples = thread->frameCount();
7247        int32_t *buffer = new int32_t[numSamples];
7248        memset(buffer, 0, numSamples * sizeof(int32_t));
7249        effect->setInBuffer((int16_t *)buffer);
7250        // auxiliary effects output samples to chain input buffer for further processing
7251        // by insert effects
7252        effect->setOutBuffer(mInBuffer);
7253    } else {
7254        // Insert effects are inserted at the end of mEffects vector as they are processed
7255        //  after track and auxiliary effects.
7256        // Insert effect order as a function of indicated preference:
7257        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7258        //  another effect is present
7259        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7260        //  last effect claiming first position
7261        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7262        //  first effect claiming last position
7263        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7264        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7265        // already present
7266
7267        int size = (int)mEffects.size();
7268        int idx_insert = size;
7269        int idx_insert_first = -1;
7270        int idx_insert_last = -1;
7271
7272        for (int i = 0; i < size; i++) {
7273            effect_descriptor_t d = mEffects[i]->desc();
7274            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7275            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7276            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7277                // check invalid effect chaining combinations
7278                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7279                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7280                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7281                    return INVALID_OPERATION;
7282                }
7283                // remember position of first insert effect and by default
7284                // select this as insert position for new effect
7285                if (idx_insert == size) {
7286                    idx_insert = i;
7287                }
7288                // remember position of last insert effect claiming
7289                // first position
7290                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7291                    idx_insert_first = i;
7292                }
7293                // remember position of first insert effect claiming
7294                // last position
7295                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7296                    idx_insert_last == -1) {
7297                    idx_insert_last = i;
7298                }
7299            }
7300        }
7301
7302        // modify idx_insert from first position if needed
7303        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7304            if (idx_insert_last != -1) {
7305                idx_insert = idx_insert_last;
7306            } else {
7307                idx_insert = size;
7308            }
7309        } else {
7310            if (idx_insert_first != -1) {
7311                idx_insert = idx_insert_first + 1;
7312            }
7313        }
7314
7315        // always read samples from chain input buffer
7316        effect->setInBuffer(mInBuffer);
7317
7318        // if last effect in the chain, output samples to chain
7319        // output buffer, otherwise to chain input buffer
7320        if (idx_insert == size) {
7321            if (idx_insert != 0) {
7322                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7323                mEffects[idx_insert-1]->configure();
7324            }
7325            effect->setOutBuffer(mOutBuffer);
7326        } else {
7327            effect->setOutBuffer(mInBuffer);
7328        }
7329        mEffects.insertAt(effect, idx_insert);
7330
7331        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7332    }
7333    effect->configure();
7334    return NO_ERROR;
7335}
7336
7337// removeEffect_l() must be called with PlaybackThread::mLock held
7338size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7339{
7340    Mutex::Autolock _l(mLock);
7341    int size = (int)mEffects.size();
7342    int i;
7343    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7344
7345    for (i = 0; i < size; i++) {
7346        if (effect == mEffects[i]) {
7347            // calling stop here will remove pre-processing effect from the audio HAL.
7348            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7349            // the middle of a read from audio HAL
7350            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7351                    mEffects[i]->state() == EffectModule::STOPPING) {
7352                mEffects[i]->stop();
7353            }
7354            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7355                delete[] effect->inBuffer();
7356            } else {
7357                if (i == size - 1 && i != 0) {
7358                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7359                    mEffects[i - 1]->configure();
7360                }
7361            }
7362            mEffects.removeAt(i);
7363            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7364            break;
7365        }
7366    }
7367
7368    return mEffects.size();
7369}
7370
7371// setDevice_l() must be called with PlaybackThread::mLock held
7372void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7373{
7374    size_t size = mEffects.size();
7375    for (size_t i = 0; i < size; i++) {
7376        mEffects[i]->setDevice(device);
7377    }
7378}
7379
7380// setMode_l() must be called with PlaybackThread::mLock held
7381void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode)
7382{
7383    size_t size = mEffects.size();
7384    for (size_t i = 0; i < size; i++) {
7385        mEffects[i]->setMode(mode);
7386    }
7387}
7388
7389// setVolume_l() must be called with PlaybackThread::mLock held
7390bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7391{
7392    uint32_t newLeft = *left;
7393    uint32_t newRight = *right;
7394    bool hasControl = false;
7395    int ctrlIdx = -1;
7396    size_t size = mEffects.size();
7397
7398    // first update volume controller
7399    for (size_t i = size; i > 0; i--) {
7400        if (mEffects[i - 1]->isProcessEnabled() &&
7401            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7402            ctrlIdx = i - 1;
7403            hasControl = true;
7404            break;
7405        }
7406    }
7407
7408    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7409        if (hasControl) {
7410            *left = mNewLeftVolume;
7411            *right = mNewRightVolume;
7412        }
7413        return hasControl;
7414    }
7415
7416    mVolumeCtrlIdx = ctrlIdx;
7417    mLeftVolume = newLeft;
7418    mRightVolume = newRight;
7419
7420    // second get volume update from volume controller
7421    if (ctrlIdx >= 0) {
7422        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7423        mNewLeftVolume = newLeft;
7424        mNewRightVolume = newRight;
7425    }
7426    // then indicate volume to all other effects in chain.
7427    // Pass altered volume to effects before volume controller
7428    // and requested volume to effects after controller
7429    uint32_t lVol = newLeft;
7430    uint32_t rVol = newRight;
7431
7432    for (size_t i = 0; i < size; i++) {
7433        if ((int)i == ctrlIdx) continue;
7434        // this also works for ctrlIdx == -1 when there is no volume controller
7435        if ((int)i > ctrlIdx) {
7436            lVol = *left;
7437            rVol = *right;
7438        }
7439        mEffects[i]->setVolume(&lVol, &rVol, false);
7440    }
7441    *left = newLeft;
7442    *right = newRight;
7443
7444    return hasControl;
7445}
7446
7447status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7448{
7449    const size_t SIZE = 256;
7450    char buffer[SIZE];
7451    String8 result;
7452
7453    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7454    result.append(buffer);
7455
7456    bool locked = tryLock(mLock);
7457    // failed to lock - AudioFlinger is probably deadlocked
7458    if (!locked) {
7459        result.append("\tCould not lock mutex:\n");
7460    }
7461
7462    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7463    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7464            mEffects.size(),
7465            (uint32_t)mInBuffer,
7466            (uint32_t)mOutBuffer,
7467            mActiveTrackCnt);
7468    result.append(buffer);
7469    write(fd, result.string(), result.size());
7470
7471    for (size_t i = 0; i < mEffects.size(); ++i) {
7472        sp<EffectModule> effect = mEffects[i];
7473        if (effect != 0) {
7474            effect->dump(fd, args);
7475        }
7476    }
7477
7478    if (locked) {
7479        mLock.unlock();
7480    }
7481
7482    return NO_ERROR;
7483}
7484
7485// must be called with ThreadBase::mLock held
7486void AudioFlinger::EffectChain::setEffectSuspended_l(
7487        const effect_uuid_t *type, bool suspend)
7488{
7489    sp<SuspendedEffectDesc> desc;
7490    // use effect type UUID timelow as key as there is no real risk of identical
7491    // timeLow fields among effect type UUIDs.
7492    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7493    if (suspend) {
7494        if (index >= 0) {
7495            desc = mSuspendedEffects.valueAt(index);
7496        } else {
7497            desc = new SuspendedEffectDesc();
7498            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7499            mSuspendedEffects.add(type->timeLow, desc);
7500            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7501        }
7502        if (desc->mRefCount++ == 0) {
7503            sp<EffectModule> effect = getEffectIfEnabled(type);
7504            if (effect != 0) {
7505                desc->mEffect = effect;
7506                effect->setSuspended(true);
7507                effect->setEnabled(false);
7508            }
7509        }
7510    } else {
7511        if (index < 0) {
7512            return;
7513        }
7514        desc = mSuspendedEffects.valueAt(index);
7515        if (desc->mRefCount <= 0) {
7516            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7517            desc->mRefCount = 1;
7518        }
7519        if (--desc->mRefCount == 0) {
7520            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7521            if (desc->mEffect != 0) {
7522                sp<EffectModule> effect = desc->mEffect.promote();
7523                if (effect != 0) {
7524                    effect->setSuspended(false);
7525                    sp<EffectHandle> handle = effect->controlHandle();
7526                    if (handle != 0) {
7527                        effect->setEnabled(handle->enabled());
7528                    }
7529                }
7530                desc->mEffect.clear();
7531            }
7532            mSuspendedEffects.removeItemsAt(index);
7533        }
7534    }
7535}
7536
7537// must be called with ThreadBase::mLock held
7538void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7539{
7540    sp<SuspendedEffectDesc> desc;
7541
7542    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7543    if (suspend) {
7544        if (index >= 0) {
7545            desc = mSuspendedEffects.valueAt(index);
7546        } else {
7547            desc = new SuspendedEffectDesc();
7548            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7549            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7550        }
7551        if (desc->mRefCount++ == 0) {
7552            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7553            for (size_t i = 0; i < effects.size(); i++) {
7554                setEffectSuspended_l(&effects[i]->desc().type, true);
7555            }
7556        }
7557    } else {
7558        if (index < 0) {
7559            return;
7560        }
7561        desc = mSuspendedEffects.valueAt(index);
7562        if (desc->mRefCount <= 0) {
7563            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7564            desc->mRefCount = 1;
7565        }
7566        if (--desc->mRefCount == 0) {
7567            Vector<const effect_uuid_t *> types;
7568            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7569                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7570                    continue;
7571                }
7572                types.add(&mSuspendedEffects.valueAt(i)->mType);
7573            }
7574            for (size_t i = 0; i < types.size(); i++) {
7575                setEffectSuspended_l(types[i], false);
7576            }
7577            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7578            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7579        }
7580    }
7581}
7582
7583
7584// The volume effect is used for automated tests only
7585#ifndef OPENSL_ES_H_
7586static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7587                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7588const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7589#endif //OPENSL_ES_H_
7590
7591bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7592{
7593    // auxiliary effects and visualizer are never suspended on output mix
7594    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7595        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7596         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7597         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7598        return false;
7599    }
7600    return true;
7601}
7602
7603Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7604{
7605    Vector< sp<EffectModule> > effects;
7606    for (size_t i = 0; i < mEffects.size(); i++) {
7607        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7608            continue;
7609        }
7610        effects.add(mEffects[i]);
7611    }
7612    return effects;
7613}
7614
7615sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7616                                                            const effect_uuid_t *type)
7617{
7618    sp<EffectModule> effect;
7619    effect = getEffectFromType_l(type);
7620    if (effect != 0 && !effect->isEnabled()) {
7621        effect.clear();
7622    }
7623    return effect;
7624}
7625
7626void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7627                                                            bool enabled)
7628{
7629    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7630    if (enabled) {
7631        if (index < 0) {
7632            // if the effect is not suspend check if all effects are suspended
7633            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7634            if (index < 0) {
7635                return;
7636            }
7637            if (!isEffectEligibleForSuspend(effect->desc())) {
7638                return;
7639            }
7640            setEffectSuspended_l(&effect->desc().type, enabled);
7641            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7642            if (index < 0) {
7643                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7644                return;
7645            }
7646        }
7647        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7648             effect->desc().type.timeLow);
7649        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7650        // if effect is requested to suspended but was not yet enabled, supend it now.
7651        if (desc->mEffect == 0) {
7652            desc->mEffect = effect;
7653            effect->setEnabled(false);
7654            effect->setSuspended(true);
7655        }
7656    } else {
7657        if (index < 0) {
7658            return;
7659        }
7660        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7661             effect->desc().type.timeLow);
7662        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7663        desc->mEffect.clear();
7664        effect->setSuspended(false);
7665    }
7666}
7667
7668#undef LOG_TAG
7669#define LOG_TAG "AudioFlinger"
7670
7671// ----------------------------------------------------------------------------
7672
7673status_t AudioFlinger::onTransact(
7674        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7675{
7676    return BnAudioFlinger::onTransact(code, data, reply, flags);
7677}
7678
7679}; // namespace android
7680