AudioFlinger.cpp revision 62da7fbd60bee2dd57f503126266e9f04311d400 
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    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() const
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->getVolumeLR();
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                // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
2465                mAudioMixer = NULL;
2466                readOutputParameters();
2467                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2468                for (size_t i = 0; i < mTracks.size() ; i++) {
2469                    int name = getTrackName_l();
2470                    if (name < 0) break;
2471                    mTracks[i]->mName = name;
2472                    // limit track sample rate to 2 x new output sample rate
2473                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2474                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2475                    }
2476                }
2477                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2478            }
2479        }
2480
2481        mNewParameters.removeAt(0);
2482
2483        mParamStatus = status;
2484        mParamCond.signal();
2485        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2486        // already timed out waiting for the status and will never signal the condition.
2487        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2488    }
2489    return reconfig;
2490}
2491
2492status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2493{
2494    const size_t SIZE = 256;
2495    char buffer[SIZE];
2496    String8 result;
2497
2498    PlaybackThread::dumpInternals(fd, args);
2499
2500    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2501    result.append(buffer);
2502    write(fd, result.string(), result.size());
2503    return NO_ERROR;
2504}
2505
2506uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2507{
2508    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2509}
2510
2511uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2512{
2513    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2514}
2515
2516// ----------------------------------------------------------------------------
2517AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2518    :   PlaybackThread(audioFlinger, output, id, device)
2519{
2520    mType = ThreadBase::DIRECT;
2521}
2522
2523AudioFlinger::DirectOutputThread::~DirectOutputThread()
2524{
2525}
2526
2527static inline
2528int32_t mul(int16_t in, int16_t v)
2529{
2530#if defined(__arm__) && !defined(__thumb__)
2531    int32_t out;
2532    asm( "smulbb %[out], %[in], %[v] \n"
2533         : [out]"=r"(out)
2534         : [in]"%r"(in), [v]"r"(v)
2535         : );
2536    return out;
2537#else
2538    return in * int32_t(v);
2539#endif
2540}
2541
2542void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2543{
2544    // Do not apply volume on compressed audio
2545    if (!audio_is_linear_pcm(mFormat)) {
2546        return;
2547    }
2548
2549    // convert to signed 16 bit before volume calculation
2550    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2551        size_t count = mFrameCount * mChannelCount;
2552        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2553        int16_t *dst = mMixBuffer + count-1;
2554        while(count--) {
2555            *dst-- = (int16_t)(*src--^0x80) << 8;
2556        }
2557    }
2558
2559    size_t frameCount = mFrameCount;
2560    int16_t *out = mMixBuffer;
2561    if (ramp) {
2562        if (mChannelCount == 1) {
2563            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2564            int32_t vlInc = d / (int32_t)frameCount;
2565            int32_t vl = ((int32_t)mLeftVolShort << 16);
2566            do {
2567                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2568                out++;
2569                vl += vlInc;
2570            } while (--frameCount);
2571
2572        } else {
2573            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2574            int32_t vlInc = d / (int32_t)frameCount;
2575            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2576            int32_t vrInc = d / (int32_t)frameCount;
2577            int32_t vl = ((int32_t)mLeftVolShort << 16);
2578            int32_t vr = ((int32_t)mRightVolShort << 16);
2579            do {
2580                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2581                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2582                out += 2;
2583                vl += vlInc;
2584                vr += vrInc;
2585            } while (--frameCount);
2586        }
2587    } else {
2588        if (mChannelCount == 1) {
2589            do {
2590                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2591                out++;
2592            } while (--frameCount);
2593        } else {
2594            do {
2595                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2596                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2597                out += 2;
2598            } while (--frameCount);
2599        }
2600    }
2601
2602    // convert back to unsigned 8 bit after volume calculation
2603    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2604        size_t count = mFrameCount * mChannelCount;
2605        int16_t *src = mMixBuffer;
2606        uint8_t *dst = (uint8_t *)mMixBuffer;
2607        while(count--) {
2608            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2609        }
2610    }
2611
2612    mLeftVolShort = leftVol;
2613    mRightVolShort = rightVol;
2614}
2615
2616bool AudioFlinger::DirectOutputThread::threadLoop()
2617{
2618    mixer_state mixerStatus = MIXER_IDLE;
2619    sp<Track> trackToRemove;
2620    sp<Track> activeTrack;
2621    nsecs_t standbyTime = systemTime();
2622    int8_t *curBuf;
2623    size_t mixBufferSize = mFrameCount*mFrameSize;
2624    uint32_t activeSleepTime = activeSleepTimeUs();
2625    uint32_t idleSleepTime = idleSleepTimeUs();
2626    uint32_t sleepTime = idleSleepTime;
2627    // use shorter standby delay as on normal output to release
2628    // hardware resources as soon as possible
2629    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2630
2631    acquireWakeLock();
2632
2633    while (!exitPending())
2634    {
2635        bool rampVolume;
2636        uint16_t leftVol;
2637        uint16_t rightVol;
2638        Vector< sp<EffectChain> > effectChains;
2639
2640        processConfigEvents();
2641
2642        mixerStatus = MIXER_IDLE;
2643
2644        { // scope for the mLock
2645
2646            Mutex::Autolock _l(mLock);
2647
2648            if (checkForNewParameters_l()) {
2649                mixBufferSize = mFrameCount*mFrameSize;
2650                activeSleepTime = activeSleepTimeUs();
2651                idleSleepTime = idleSleepTimeUs();
2652                standbyDelay = microseconds(activeSleepTime*2);
2653            }
2654
2655            // put audio hardware into standby after short delay
2656            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2657                        mSuspended)) {
2658                // wait until we have something to do...
2659                if (!mStandby) {
2660                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2661                    mOutput->stream->common.standby(&mOutput->stream->common);
2662                    mStandby = true;
2663                    mBytesWritten = 0;
2664                }
2665
2666                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2667                    // we're about to wait, flush the binder command buffer
2668                    IPCThreadState::self()->flushCommands();
2669
2670                    if (exitPending()) break;
2671
2672                    releaseWakeLock_l();
2673                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2674                    mWaitWorkCV.wait(mLock);
2675                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2676                    acquireWakeLock_l();
2677
2678                    if (!mMasterMute) {
2679                        char value[PROPERTY_VALUE_MAX];
2680                        property_get("ro.audio.silent", value, "0");
2681                        if (atoi(value)) {
2682                            ALOGD("Silence is golden");
2683                            setMasterMute(true);
2684                        }
2685                    }
2686
2687                    standbyTime = systemTime() + standbyDelay;
2688                    sleepTime = idleSleepTime;
2689                    continue;
2690                }
2691            }
2692
2693            effectChains = mEffectChains;
2694
2695            // find out which tracks need to be processed
2696            if (mActiveTracks.size() != 0) {
2697                sp<Track> t = mActiveTracks[0].promote();
2698                if (t == 0) continue;
2699
2700                Track* const track = t.get();
2701                audio_track_cblk_t* cblk = track->cblk();
2702
2703                // The first time a track is added we wait
2704                // for all its buffers to be filled before processing it
2705                if (cblk->framesReady() && track->isReady() &&
2706                        !track->isPaused() && !track->isTerminated())
2707                {
2708                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2709
2710                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2711                        track->mFillingUpStatus = Track::FS_ACTIVE;
2712                        mLeftVolFloat = mRightVolFloat = 0;
2713                        mLeftVolShort = mRightVolShort = 0;
2714                        if (track->mState == TrackBase::RESUMING) {
2715                            track->mState = TrackBase::ACTIVE;
2716                            rampVolume = true;
2717                        }
2718                    } else if (cblk->server != 0) {
2719                        // If the track is stopped before the first frame was mixed,
2720                        // do not apply ramp
2721                        rampVolume = true;
2722                    }
2723                    // compute volume for this track
2724                    float left, right;
2725                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2726                        mStreamTypes[track->type()].mute) {
2727                        left = right = 0;
2728                        if (track->isPausing()) {
2729                            track->setPaused();
2730                        }
2731                    } else {
2732                        float typeVolume = mStreamTypes[track->type()].volume;
2733                        float v = mMasterVolume * typeVolume;
2734                        uint32_t vlr = cblk->getVolumeLR();
2735                        float v_clamped = v * (vlr & 0xFFFF);
2736                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2737                        left = v_clamped/MAX_GAIN;
2738                        v_clamped = v * (vlr >> 16);
2739                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2740                        right = v_clamped/MAX_GAIN;
2741                    }
2742
2743                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2744                        mLeftVolFloat = left;
2745                        mRightVolFloat = right;
2746
2747                        // If audio HAL implements volume control,
2748                        // force software volume to nominal value
2749                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2750                            left = 1.0f;
2751                            right = 1.0f;
2752                        }
2753
2754                        // Convert volumes from float to 8.24
2755                        uint32_t vl = (uint32_t)(left * (1 << 24));
2756                        uint32_t vr = (uint32_t)(right * (1 << 24));
2757
2758                        // Delegate volume control to effect in track effect chain if needed
2759                        // only one effect chain can be present on DirectOutputThread, so if
2760                        // there is one, the track is connected to it
2761                        if (!effectChains.isEmpty()) {
2762                            // Do not ramp volume if volume is controlled by effect
2763                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2764                                rampVolume = false;
2765                            }
2766                        }
2767
2768                        // Convert volumes from 8.24 to 4.12 format
2769                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2770                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2771                        leftVol = (uint16_t)v_clamped;
2772                        v_clamped = (vr + (1 << 11)) >> 12;
2773                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2774                        rightVol = (uint16_t)v_clamped;
2775                    } else {
2776                        leftVol = mLeftVolShort;
2777                        rightVol = mRightVolShort;
2778                        rampVolume = false;
2779                    }
2780
2781                    // reset retry count
2782                    track->mRetryCount = kMaxTrackRetriesDirect;
2783                    activeTrack = t;
2784                    mixerStatus = MIXER_TRACKS_READY;
2785                } else {
2786                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2787                    if (track->isStopped()) {
2788                        track->reset();
2789                    }
2790                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2791                        // We have consumed all the buffers of this track.
2792                        // Remove it from the list of active tracks.
2793                        trackToRemove = track;
2794                    } else {
2795                        // No buffers for this track. Give it a few chances to
2796                        // fill a buffer, then remove it from active list.
2797                        if (--(track->mRetryCount) <= 0) {
2798                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2799                            trackToRemove = track;
2800                        } else {
2801                            mixerStatus = MIXER_TRACKS_ENABLED;
2802                        }
2803                    }
2804                }
2805            }
2806
2807            // remove all the tracks that need to be...
2808            if (CC_UNLIKELY(trackToRemove != 0)) {
2809                mActiveTracks.remove(trackToRemove);
2810                if (!effectChains.isEmpty()) {
2811                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2812                            trackToRemove->sessionId());
2813                    effectChains[0]->decActiveTrackCnt();
2814                }
2815                if (trackToRemove->isTerminated()) {
2816                    removeTrack_l(trackToRemove);
2817                }
2818            }
2819
2820            lockEffectChains_l(effectChains);
2821       }
2822
2823        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2824            AudioBufferProvider::Buffer buffer;
2825            size_t frameCount = mFrameCount;
2826            curBuf = (int8_t *)mMixBuffer;
2827            // output audio to hardware
2828            while (frameCount) {
2829                buffer.frameCount = frameCount;
2830                activeTrack->getNextBuffer(&buffer);
2831                if (CC_UNLIKELY(buffer.raw == NULL)) {
2832                    memset(curBuf, 0, frameCount * mFrameSize);
2833                    break;
2834                }
2835                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2836                frameCount -= buffer.frameCount;
2837                curBuf += buffer.frameCount * mFrameSize;
2838                activeTrack->releaseBuffer(&buffer);
2839            }
2840            sleepTime = 0;
2841            standbyTime = systemTime() + standbyDelay;
2842        } else {
2843            if (sleepTime == 0) {
2844                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2845                    sleepTime = activeSleepTime;
2846                } else {
2847                    sleepTime = idleSleepTime;
2848                }
2849            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2850                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2851                sleepTime = 0;
2852            }
2853        }
2854
2855        if (mSuspended) {
2856            sleepTime = suspendSleepTimeUs();
2857        }
2858        // sleepTime == 0 means we must write to audio hardware
2859        if (sleepTime == 0) {
2860            if (mixerStatus == MIXER_TRACKS_READY) {
2861                applyVolume(leftVol, rightVol, rampVolume);
2862            }
2863            for (size_t i = 0; i < effectChains.size(); i ++) {
2864                effectChains[i]->process_l();
2865            }
2866            unlockEffectChains(effectChains);
2867
2868            mLastWriteTime = systemTime();
2869            mInWrite = true;
2870            mBytesWritten += mixBufferSize;
2871            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2872            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2873            mNumWrites++;
2874            mInWrite = false;
2875            mStandby = false;
2876        } else {
2877            unlockEffectChains(effectChains);
2878            usleep(sleepTime);
2879        }
2880
2881        // finally let go of removed track, without the lock held
2882        // since we can't guarantee the destructors won't acquire that
2883        // same lock.
2884        trackToRemove.clear();
2885        activeTrack.clear();
2886
2887        // Effect chains will be actually deleted here if they were removed from
2888        // mEffectChains list during mixing or effects processing
2889        effectChains.clear();
2890    }
2891
2892    if (!mStandby) {
2893        mOutput->stream->common.standby(&mOutput->stream->common);
2894    }
2895
2896    releaseWakeLock();
2897
2898    ALOGV("DirectOutputThread %p exiting", this);
2899    return false;
2900}
2901
2902// getTrackName_l() must be called with ThreadBase::mLock held
2903int AudioFlinger::DirectOutputThread::getTrackName_l()
2904{
2905    return 0;
2906}
2907
2908// deleteTrackName_l() must be called with ThreadBase::mLock held
2909void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2910{
2911}
2912
2913// checkForNewParameters_l() must be called with ThreadBase::mLock held
2914bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2915{
2916    bool reconfig = false;
2917
2918    while (!mNewParameters.isEmpty()) {
2919        status_t status = NO_ERROR;
2920        String8 keyValuePair = mNewParameters[0];
2921        AudioParameter param = AudioParameter(keyValuePair);
2922        int value;
2923
2924        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2925            // do not accept frame count changes if tracks are open as the track buffer
2926            // size depends on frame count and correct behavior would not be garantied
2927            // if frame count is changed after track creation
2928            if (!mTracks.isEmpty()) {
2929                status = INVALID_OPERATION;
2930            } else {
2931                reconfig = true;
2932            }
2933        }
2934        if (status == NO_ERROR) {
2935            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2936                                                    keyValuePair.string());
2937            if (!mStandby && status == INVALID_OPERATION) {
2938               mOutput->stream->common.standby(&mOutput->stream->common);
2939               mStandby = true;
2940               mBytesWritten = 0;
2941               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2942                                                       keyValuePair.string());
2943            }
2944            if (status == NO_ERROR && reconfig) {
2945                readOutputParameters();
2946                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2947            }
2948        }
2949
2950        mNewParameters.removeAt(0);
2951
2952        mParamStatus = status;
2953        mParamCond.signal();
2954        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2955        // already timed out waiting for the status and will never signal the condition.
2956        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2957    }
2958    return reconfig;
2959}
2960
2961uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2962{
2963    uint32_t time;
2964    if (audio_is_linear_pcm(mFormat)) {
2965        time = PlaybackThread::activeSleepTimeUs();
2966    } else {
2967        time = 10000;
2968    }
2969    return time;
2970}
2971
2972uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2973{
2974    uint32_t time;
2975    if (audio_is_linear_pcm(mFormat)) {
2976        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2977    } else {
2978        time = 10000;
2979    }
2980    return time;
2981}
2982
2983uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2984{
2985    uint32_t time;
2986    if (audio_is_linear_pcm(mFormat)) {
2987        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2988    } else {
2989        time = 10000;
2990    }
2991    return time;
2992}
2993
2994
2995// ----------------------------------------------------------------------------
2996
2997AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2998    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2999{
3000    mType = ThreadBase::DUPLICATING;
3001    addOutputTrack(mainThread);
3002}
3003
3004AudioFlinger::DuplicatingThread::~DuplicatingThread()
3005{
3006    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3007        mOutputTracks[i]->destroy();
3008    }
3009    mOutputTracks.clear();
3010}
3011
3012bool AudioFlinger::DuplicatingThread::threadLoop()
3013{
3014    Vector< sp<Track> > tracksToRemove;
3015    mixer_state mixerStatus = MIXER_IDLE;
3016    nsecs_t standbyTime = systemTime();
3017    size_t mixBufferSize = mFrameCount*mFrameSize;
3018    SortedVector< sp<OutputTrack> > outputTracks;
3019    uint32_t writeFrames = 0;
3020    uint32_t activeSleepTime = activeSleepTimeUs();
3021    uint32_t idleSleepTime = idleSleepTimeUs();
3022    uint32_t sleepTime = idleSleepTime;
3023    Vector< sp<EffectChain> > effectChains;
3024
3025    acquireWakeLock();
3026
3027    while (!exitPending())
3028    {
3029        processConfigEvents();
3030
3031        mixerStatus = MIXER_IDLE;
3032        { // scope for the mLock
3033
3034            Mutex::Autolock _l(mLock);
3035
3036            if (checkForNewParameters_l()) {
3037                mixBufferSize = mFrameCount*mFrameSize;
3038                updateWaitTime();
3039                activeSleepTime = activeSleepTimeUs();
3040                idleSleepTime = idleSleepTimeUs();
3041            }
3042
3043            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
3044
3045            for (size_t i = 0; i < mOutputTracks.size(); i++) {
3046                outputTracks.add(mOutputTracks[i]);
3047            }
3048
3049            // put audio hardware into standby after short delay
3050            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
3051                         mSuspended)) {
3052                if (!mStandby) {
3053                    for (size_t i = 0; i < outputTracks.size(); i++) {
3054                        outputTracks[i]->stop();
3055                    }
3056                    mStandby = true;
3057                    mBytesWritten = 0;
3058                }
3059
3060                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
3061                    // we're about to wait, flush the binder command buffer
3062                    IPCThreadState::self()->flushCommands();
3063                    outputTracks.clear();
3064
3065                    if (exitPending()) break;
3066
3067                    releaseWakeLock_l();
3068                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3069                    mWaitWorkCV.wait(mLock);
3070                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3071                    acquireWakeLock_l();
3072
3073                    mPrevMixerStatus = MIXER_IDLE;
3074                    if (!mMasterMute) {
3075                        char value[PROPERTY_VALUE_MAX];
3076                        property_get("ro.audio.silent", value, "0");
3077                        if (atoi(value)) {
3078                            ALOGD("Silence is golden");
3079                            setMasterMute(true);
3080                        }
3081                    }
3082
3083                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3084                    sleepTime = idleSleepTime;
3085                    continue;
3086                }
3087            }
3088
3089            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3090
3091            // prevent any changes in effect chain list and in each effect chain
3092            // during mixing and effect process as the audio buffers could be deleted
3093            // or modified if an effect is created or deleted
3094            lockEffectChains_l(effectChains);
3095        }
3096
3097        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3098            // mix buffers...
3099            if (outputsReady(outputTracks)) {
3100                mAudioMixer->process();
3101            } else {
3102                memset(mMixBuffer, 0, mixBufferSize);
3103            }
3104            sleepTime = 0;
3105            writeFrames = mFrameCount;
3106        } else {
3107            if (sleepTime == 0) {
3108                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3109                    sleepTime = activeSleepTime;
3110                } else {
3111                    sleepTime = idleSleepTime;
3112                }
3113            } else if (mBytesWritten != 0) {
3114                // flush remaining overflow buffers in output tracks
3115                for (size_t i = 0; i < outputTracks.size(); i++) {
3116                    if (outputTracks[i]->isActive()) {
3117                        sleepTime = 0;
3118                        writeFrames = 0;
3119                        memset(mMixBuffer, 0, mixBufferSize);
3120                        break;
3121                    }
3122                }
3123            }
3124        }
3125
3126        if (mSuspended) {
3127            sleepTime = suspendSleepTimeUs();
3128        }
3129        // sleepTime == 0 means we must write to audio hardware
3130        if (sleepTime == 0) {
3131            for (size_t i = 0; i < effectChains.size(); i ++) {
3132                effectChains[i]->process_l();
3133            }
3134            // enable changes in effect chain
3135            unlockEffectChains(effectChains);
3136
3137            standbyTime = systemTime() + kStandbyTimeInNsecs;
3138            for (size_t i = 0; i < outputTracks.size(); i++) {
3139                outputTracks[i]->write(mMixBuffer, writeFrames);
3140            }
3141            mStandby = false;
3142            mBytesWritten += mixBufferSize;
3143        } else {
3144            // enable changes in effect chain
3145            unlockEffectChains(effectChains);
3146            usleep(sleepTime);
3147        }
3148
3149        // finally let go of all our tracks, without the lock held
3150        // since we can't guarantee the destructors won't acquire that
3151        // same lock.
3152        tracksToRemove.clear();
3153        outputTracks.clear();
3154
3155        // Effect chains will be actually deleted here if they were removed from
3156        // mEffectChains list during mixing or effects processing
3157        effectChains.clear();
3158    }
3159
3160    releaseWakeLock();
3161
3162    return false;
3163}
3164
3165void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3166{
3167    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3168    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3169                                            this,
3170                                            mSampleRate,
3171                                            mFormat,
3172                                            mChannelMask,
3173                                            frameCount);
3174    if (outputTrack->cblk() != NULL) {
3175        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3176        mOutputTracks.add(outputTrack);
3177        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3178        updateWaitTime();
3179    }
3180}
3181
3182void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3183{
3184    Mutex::Autolock _l(mLock);
3185    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3186        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3187            mOutputTracks[i]->destroy();
3188            mOutputTracks.removeAt(i);
3189            updateWaitTime();
3190            return;
3191        }
3192    }
3193    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3194}
3195
3196void AudioFlinger::DuplicatingThread::updateWaitTime()
3197{
3198    mWaitTimeMs = UINT_MAX;
3199    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3200        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3201        if (strong != NULL) {
3202            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3203            if (waitTimeMs < mWaitTimeMs) {
3204                mWaitTimeMs = waitTimeMs;
3205            }
3206        }
3207    }
3208}
3209
3210
3211bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3212{
3213    for (size_t i = 0; i < outputTracks.size(); i++) {
3214        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3215        if (thread == 0) {
3216            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3217            return false;
3218        }
3219        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3220        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3221            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3222            return false;
3223        }
3224    }
3225    return true;
3226}
3227
3228uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3229{
3230    return (mWaitTimeMs * 1000) / 2;
3231}
3232
3233// ----------------------------------------------------------------------------
3234
3235// TrackBase constructor must be called with AudioFlinger::mLock held
3236AudioFlinger::ThreadBase::TrackBase::TrackBase(
3237            const wp<ThreadBase>& thread,
3238            const sp<Client>& client,
3239            uint32_t sampleRate,
3240            audio_format_t format,
3241            uint32_t channelMask,
3242            int frameCount,
3243            uint32_t flags,
3244            const sp<IMemory>& sharedBuffer,
3245            int sessionId)
3246    :   RefBase(),
3247        mThread(thread),
3248        mClient(client),
3249        mCblk(0),
3250        mFrameCount(0),
3251        mState(IDLE),
3252        mClientTid(-1),
3253        mFormat(format),
3254        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3255        mSessionId(sessionId)
3256{
3257    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3258
3259    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3260   size_t size = sizeof(audio_track_cblk_t);
3261   uint8_t channelCount = popcount(channelMask);
3262   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3263   if (sharedBuffer == 0) {
3264       size += bufferSize;
3265   }
3266
3267   if (client != NULL) {
3268        mCblkMemory = client->heap()->allocate(size);
3269        if (mCblkMemory != 0) {
3270            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3271            if (mCblk) { // construct the shared structure in-place.
3272                new(mCblk) audio_track_cblk_t();
3273                // clear all buffers
3274                mCblk->frameCount = frameCount;
3275                mCblk->sampleRate = sampleRate;
3276                mChannelCount = channelCount;
3277                mChannelMask = channelMask;
3278                if (sharedBuffer == 0) {
3279                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3280                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3281                    // Force underrun condition to avoid false underrun callback until first data is
3282                    // written to buffer (other flags are cleared)
3283                    mCblk->flags = CBLK_UNDERRUN_ON;
3284                } else {
3285                    mBuffer = sharedBuffer->pointer();
3286                }
3287                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3288            }
3289        } else {
3290            ALOGE("not enough memory for AudioTrack size=%u", size);
3291            client->heap()->dump("AudioTrack");
3292            return;
3293        }
3294   } else {
3295       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3296           // construct the shared structure in-place.
3297           new(mCblk) audio_track_cblk_t();
3298           // clear all buffers
3299           mCblk->frameCount = frameCount;
3300           mCblk->sampleRate = sampleRate;
3301           mChannelCount = channelCount;
3302           mChannelMask = channelMask;
3303           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3304           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3305           // Force underrun condition to avoid false underrun callback until first data is
3306           // written to buffer (other flags are cleared)
3307           mCblk->flags = CBLK_UNDERRUN_ON;
3308           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3309   }
3310}
3311
3312AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3313{
3314    if (mCblk) {
3315        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3316        if (mClient == NULL) {
3317            delete mCblk;
3318        }
3319    }
3320    mCblkMemory.clear();            // and free the shared memory
3321    if (mClient != NULL) {
3322        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3323        mClient.clear();
3324    }
3325}
3326
3327void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3328{
3329    buffer->raw = NULL;
3330    mFrameCount = buffer->frameCount;
3331    step();
3332    buffer->frameCount = 0;
3333}
3334
3335bool AudioFlinger::ThreadBase::TrackBase::step() {
3336    bool result;
3337    audio_track_cblk_t* cblk = this->cblk();
3338
3339    result = cblk->stepServer(mFrameCount);
3340    if (!result) {
3341        ALOGV("stepServer failed acquiring cblk mutex");
3342        mFlags |= STEPSERVER_FAILED;
3343    }
3344    return result;
3345}
3346
3347void AudioFlinger::ThreadBase::TrackBase::reset() {
3348    audio_track_cblk_t* cblk = this->cblk();
3349
3350    cblk->user = 0;
3351    cblk->server = 0;
3352    cblk->userBase = 0;
3353    cblk->serverBase = 0;
3354    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3355    ALOGV("TrackBase::reset");
3356}
3357
3358sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3359{
3360    return mCblkMemory;
3361}
3362
3363int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3364    return (int)mCblk->sampleRate;
3365}
3366
3367int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3368    return (const int)mChannelCount;
3369}
3370
3371uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3372    return mChannelMask;
3373}
3374
3375void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3376    audio_track_cblk_t* cblk = this->cblk();
3377    size_t frameSize = cblk->frameSize;
3378    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
3379    int8_t *bufferEnd = bufferStart + frames * frameSize;
3380
3381    // Check validity of returned pointer in case the track control block would have been corrupted.
3382    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3383        ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) {
3384        ALOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3385                server %d, serverBase %d, user %d, userBase %d",
3386                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3387                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3388        return 0;
3389    }
3390
3391    return bufferStart;
3392}
3393
3394// ----------------------------------------------------------------------------
3395
3396// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3397AudioFlinger::PlaybackThread::Track::Track(
3398            const wp<ThreadBase>& thread,
3399            const sp<Client>& client,
3400            audio_stream_type_t streamType,
3401            uint32_t sampleRate,
3402            audio_format_t format,
3403            uint32_t channelMask,
3404            int frameCount,
3405            const sp<IMemory>& sharedBuffer,
3406            int sessionId)
3407    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3408    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3409    mAuxEffectId(0), mHasVolumeController(false)
3410{
3411    if (mCblk != NULL) {
3412        sp<ThreadBase> baseThread = thread.promote();
3413        if (baseThread != 0) {
3414            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3415            mName = playbackThread->getTrackName_l();
3416            mMainBuffer = playbackThread->mixBuffer();
3417        }
3418        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3419        if (mName < 0) {
3420            ALOGE("no more track names available");
3421        }
3422        mStreamType = streamType;
3423        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3424        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3425        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3426    }
3427}
3428
3429AudioFlinger::PlaybackThread::Track::~Track()
3430{
3431    ALOGV("PlaybackThread::Track destructor");
3432    sp<ThreadBase> thread = mThread.promote();
3433    if (thread != 0) {
3434        Mutex::Autolock _l(thread->mLock);
3435        mState = TERMINATED;
3436    }
3437}
3438
3439void AudioFlinger::PlaybackThread::Track::destroy()
3440{
3441    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3442    // by removing it from mTracks vector, so there is a risk that this Tracks's
3443    // desctructor is called. As the destructor needs to lock mLock,
3444    // we must acquire a strong reference on this Track before locking mLock
3445    // here so that the destructor is called only when exiting this function.
3446    // On the other hand, as long as Track::destroy() is only called by
3447    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3448    // this Track with its member mTrack.
3449    sp<Track> keep(this);
3450    { // scope for mLock
3451        sp<ThreadBase> thread = mThread.promote();
3452        if (thread != 0) {
3453            if (!isOutputTrack()) {
3454                if (mState == ACTIVE || mState == RESUMING) {
3455                    AudioSystem::stopOutput(thread->id(),
3456                                            (audio_stream_type_t)mStreamType,
3457                                            mSessionId);
3458
3459                    // to track the speaker usage
3460                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3461                }
3462                AudioSystem::releaseOutput(thread->id());
3463            }
3464            Mutex::Autolock _l(thread->mLock);
3465            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3466            playbackThread->destroyTrack_l(this);
3467        }
3468    }
3469}
3470
3471void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3472{
3473    uint32_t vlr = mCblk->getVolumeLR();
3474    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",
3475            mName - AudioMixer::TRACK0,
3476            (mClient == NULL) ? getpid() : mClient->pid(),
3477            mStreamType,
3478            mFormat,
3479            mChannelMask,
3480            mSessionId,
3481            mFrameCount,
3482            mState,
3483            mMute,
3484            mFillingUpStatus,
3485            mCblk->sampleRate,
3486            vlr & 0xFFFF,
3487            vlr >> 16,
3488            mCblk->server,
3489            mCblk->user,
3490            (int)mMainBuffer,
3491            (int)mAuxBuffer);
3492}
3493
3494status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3495{
3496     audio_track_cblk_t* cblk = this->cblk();
3497     uint32_t framesReady;
3498     uint32_t framesReq = buffer->frameCount;
3499
3500     // Check if last stepServer failed, try to step now
3501     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3502         if (!step())  goto getNextBuffer_exit;
3503         ALOGV("stepServer recovered");
3504         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3505     }
3506
3507     framesReady = cblk->framesReady();
3508
3509     if (CC_LIKELY(framesReady)) {
3510        uint32_t s = cblk->server;
3511        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3512
3513        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3514        if (framesReq > framesReady) {
3515            framesReq = framesReady;
3516        }
3517        if (s + framesReq > bufferEnd) {
3518            framesReq = bufferEnd - s;
3519        }
3520
3521         buffer->raw = getBuffer(s, framesReq);
3522         if (buffer->raw == NULL) goto getNextBuffer_exit;
3523
3524         buffer->frameCount = framesReq;
3525        return NO_ERROR;
3526     }
3527
3528getNextBuffer_exit:
3529     buffer->raw = NULL;
3530     buffer->frameCount = 0;
3531     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3532     return NOT_ENOUGH_DATA;
3533}
3534
3535bool AudioFlinger::PlaybackThread::Track::isReady() const {
3536    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3537
3538    if (mCblk->framesReady() >= mCblk->frameCount ||
3539            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3540        mFillingUpStatus = FS_FILLED;
3541        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3542        return true;
3543    }
3544    return false;
3545}
3546
3547status_t AudioFlinger::PlaybackThread::Track::start()
3548{
3549    status_t status = NO_ERROR;
3550    ALOGV("start(%d), calling thread %d session %d",
3551            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3552    sp<ThreadBase> thread = mThread.promote();
3553    if (thread != 0) {
3554        Mutex::Autolock _l(thread->mLock);
3555        track_state state = mState;
3556        // here the track could be either new, or restarted
3557        // in both cases "unstop" the track
3558        if (mState == PAUSED) {
3559            mState = TrackBase::RESUMING;
3560            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3561        } else {
3562            mState = TrackBase::ACTIVE;
3563            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3564        }
3565
3566        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3567            thread->mLock.unlock();
3568            status = AudioSystem::startOutput(thread->id(),
3569                                              (audio_stream_type_t)mStreamType,
3570                                              mSessionId);
3571            thread->mLock.lock();
3572
3573            // to track the speaker usage
3574            if (status == NO_ERROR) {
3575                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3576            }
3577        }
3578        if (status == NO_ERROR) {
3579            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3580            playbackThread->addTrack_l(this);
3581        } else {
3582            mState = state;
3583        }
3584    } else {
3585        status = BAD_VALUE;
3586    }
3587    return status;
3588}
3589
3590void AudioFlinger::PlaybackThread::Track::stop()
3591{
3592    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3593    sp<ThreadBase> thread = mThread.promote();
3594    if (thread != 0) {
3595        Mutex::Autolock _l(thread->mLock);
3596        track_state state = mState;
3597        if (mState > STOPPED) {
3598            mState = STOPPED;
3599            // If the track is not active (PAUSED and buffers full), flush buffers
3600            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3601            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3602                reset();
3603            }
3604            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3605        }
3606        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3607            thread->mLock.unlock();
3608            AudioSystem::stopOutput(thread->id(),
3609                                    (audio_stream_type_t)mStreamType,
3610                                    mSessionId);
3611            thread->mLock.lock();
3612
3613            // to track the speaker usage
3614            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3615        }
3616    }
3617}
3618
3619void AudioFlinger::PlaybackThread::Track::pause()
3620{
3621    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3622    sp<ThreadBase> thread = mThread.promote();
3623    if (thread != 0) {
3624        Mutex::Autolock _l(thread->mLock);
3625        if (mState == ACTIVE || mState == RESUMING) {
3626            mState = PAUSING;
3627            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3628            if (!isOutputTrack()) {
3629                thread->mLock.unlock();
3630                AudioSystem::stopOutput(thread->id(),
3631                                        (audio_stream_type_t)mStreamType,
3632                                        mSessionId);
3633                thread->mLock.lock();
3634
3635                // to track the speaker usage
3636                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3637            }
3638        }
3639    }
3640}
3641
3642void AudioFlinger::PlaybackThread::Track::flush()
3643{
3644    ALOGV("flush(%d)", mName);
3645    sp<ThreadBase> thread = mThread.promote();
3646    if (thread != 0) {
3647        Mutex::Autolock _l(thread->mLock);
3648        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3649            return;
3650        }
3651        // No point remaining in PAUSED state after a flush => go to
3652        // STOPPED state
3653        mState = STOPPED;
3654
3655        // do not reset the track if it is still in the process of being stopped or paused.
3656        // this will be done by prepareTracks_l() when the track is stopped.
3657        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3658        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3659            reset();
3660        }
3661    }
3662}
3663
3664void AudioFlinger::PlaybackThread::Track::reset()
3665{
3666    // Do not reset twice to avoid discarding data written just after a flush and before
3667    // the audioflinger thread detects the track is stopped.
3668    if (!mResetDone) {
3669        TrackBase::reset();
3670        // Force underrun condition to avoid false underrun callback until first data is
3671        // written to buffer
3672        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3673        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3674        mFillingUpStatus = FS_FILLING;
3675        mResetDone = true;
3676    }
3677}
3678
3679void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3680{
3681    mMute = muted;
3682}
3683
3684status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3685{
3686    status_t status = DEAD_OBJECT;
3687    sp<ThreadBase> thread = mThread.promote();
3688    if (thread != 0) {
3689       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3690       status = playbackThread->attachAuxEffect(this, EffectId);
3691    }
3692    return status;
3693}
3694
3695void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3696{
3697    mAuxEffectId = EffectId;
3698    mAuxBuffer = buffer;
3699}
3700
3701// ----------------------------------------------------------------------------
3702
3703// RecordTrack constructor must be called with AudioFlinger::mLock held
3704AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3705            const wp<ThreadBase>& thread,
3706            const sp<Client>& client,
3707            uint32_t sampleRate,
3708            audio_format_t format,
3709            uint32_t channelMask,
3710            int frameCount,
3711            uint32_t flags,
3712            int sessionId)
3713    :   TrackBase(thread, client, sampleRate, format,
3714                  channelMask, frameCount, flags, 0, sessionId),
3715        mOverflow(false)
3716{
3717    if (mCblk != NULL) {
3718       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3719       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3720           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3721       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3722           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3723       } else {
3724           mCblk->frameSize = sizeof(int8_t);
3725       }
3726    }
3727}
3728
3729AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3730{
3731    sp<ThreadBase> thread = mThread.promote();
3732    if (thread != 0) {
3733        AudioSystem::releaseInput(thread->id());
3734    }
3735}
3736
3737status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3738{
3739    audio_track_cblk_t* cblk = this->cblk();
3740    uint32_t framesAvail;
3741    uint32_t framesReq = buffer->frameCount;
3742
3743     // Check if last stepServer failed, try to step now
3744    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3745        if (!step()) goto getNextBuffer_exit;
3746        ALOGV("stepServer recovered");
3747        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3748    }
3749
3750    framesAvail = cblk->framesAvailable_l();
3751
3752    if (CC_LIKELY(framesAvail)) {
3753        uint32_t s = cblk->server;
3754        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3755
3756        if (framesReq > framesAvail) {
3757            framesReq = framesAvail;
3758        }
3759        if (s + framesReq > bufferEnd) {
3760            framesReq = bufferEnd - s;
3761        }
3762
3763        buffer->raw = getBuffer(s, framesReq);
3764        if (buffer->raw == NULL) goto getNextBuffer_exit;
3765
3766        buffer->frameCount = framesReq;
3767        return NO_ERROR;
3768    }
3769
3770getNextBuffer_exit:
3771    buffer->raw = NULL;
3772    buffer->frameCount = 0;
3773    return NOT_ENOUGH_DATA;
3774}
3775
3776status_t AudioFlinger::RecordThread::RecordTrack::start()
3777{
3778    sp<ThreadBase> thread = mThread.promote();
3779    if (thread != 0) {
3780        RecordThread *recordThread = (RecordThread *)thread.get();
3781        return recordThread->start(this);
3782    } else {
3783        return BAD_VALUE;
3784    }
3785}
3786
3787void AudioFlinger::RecordThread::RecordTrack::stop()
3788{
3789    sp<ThreadBase> thread = mThread.promote();
3790    if (thread != 0) {
3791        RecordThread *recordThread = (RecordThread *)thread.get();
3792        recordThread->stop(this);
3793        TrackBase::reset();
3794        // Force overerrun condition to avoid false overrun callback until first data is
3795        // read from buffer
3796        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3797    }
3798}
3799
3800void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3801{
3802    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3803            (mClient == NULL) ? getpid() : mClient->pid(),
3804            mFormat,
3805            mChannelMask,
3806            mSessionId,
3807            mFrameCount,
3808            mState,
3809            mCblk->sampleRate,
3810            mCblk->server,
3811            mCblk->user);
3812}
3813
3814
3815// ----------------------------------------------------------------------------
3816
3817AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3818            const wp<ThreadBase>& thread,
3819            DuplicatingThread *sourceThread,
3820            uint32_t sampleRate,
3821            audio_format_t format,
3822            uint32_t channelMask,
3823            int frameCount)
3824    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3825    mActive(false), mSourceThread(sourceThread)
3826{
3827
3828    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3829    if (mCblk != NULL) {
3830        mCblk->flags |= CBLK_DIRECTION_OUT;
3831        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3832        mOutBuffer.frameCount = 0;
3833        playbackThread->mTracks.add(this);
3834        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3835                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3836                mCblk, mBuffer, mCblk->buffers,
3837                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3838    } else {
3839        ALOGW("Error creating output track on thread %p", playbackThread);
3840    }
3841}
3842
3843AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3844{
3845    clearBufferQueue();
3846}
3847
3848status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3849{
3850    status_t status = Track::start();
3851    if (status != NO_ERROR) {
3852        return status;
3853    }
3854
3855    mActive = true;
3856    mRetryCount = 127;
3857    return status;
3858}
3859
3860void AudioFlinger::PlaybackThread::OutputTrack::stop()
3861{
3862    Track::stop();
3863    clearBufferQueue();
3864    mOutBuffer.frameCount = 0;
3865    mActive = false;
3866}
3867
3868bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3869{
3870    Buffer *pInBuffer;
3871    Buffer inBuffer;
3872    uint32_t channelCount = mChannelCount;
3873    bool outputBufferFull = false;
3874    inBuffer.frameCount = frames;
3875    inBuffer.i16 = data;
3876
3877    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3878
3879    if (!mActive && frames != 0) {
3880        start();
3881        sp<ThreadBase> thread = mThread.promote();
3882        if (thread != 0) {
3883            MixerThread *mixerThread = (MixerThread *)thread.get();
3884            if (mCblk->frameCount > frames){
3885                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3886                    uint32_t startFrames = (mCblk->frameCount - frames);
3887                    pInBuffer = new Buffer;
3888                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3889                    pInBuffer->frameCount = startFrames;
3890                    pInBuffer->i16 = pInBuffer->mBuffer;
3891                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3892                    mBufferQueue.add(pInBuffer);
3893                } else {
3894                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
3895                }
3896            }
3897        }
3898    }
3899
3900    while (waitTimeLeftMs) {
3901        // First write pending buffers, then new data
3902        if (mBufferQueue.size()) {
3903            pInBuffer = mBufferQueue.itemAt(0);
3904        } else {
3905            pInBuffer = &inBuffer;
3906        }
3907
3908        if (pInBuffer->frameCount == 0) {
3909            break;
3910        }
3911
3912        if (mOutBuffer.frameCount == 0) {
3913            mOutBuffer.frameCount = pInBuffer->frameCount;
3914            nsecs_t startTime = systemTime();
3915            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
3916                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3917                outputBufferFull = true;
3918                break;
3919            }
3920            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3921            if (waitTimeLeftMs >= waitTimeMs) {
3922                waitTimeLeftMs -= waitTimeMs;
3923            } else {
3924                waitTimeLeftMs = 0;
3925            }
3926        }
3927
3928        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3929        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3930        mCblk->stepUser(outFrames);
3931        pInBuffer->frameCount -= outFrames;
3932        pInBuffer->i16 += outFrames * channelCount;
3933        mOutBuffer.frameCount -= outFrames;
3934        mOutBuffer.i16 += outFrames * channelCount;
3935
3936        if (pInBuffer->frameCount == 0) {
3937            if (mBufferQueue.size()) {
3938                mBufferQueue.removeAt(0);
3939                delete [] pInBuffer->mBuffer;
3940                delete pInBuffer;
3941                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3942            } else {
3943                break;
3944            }
3945        }
3946    }
3947
3948    // If we could not write all frames, allocate a buffer and queue it for next time.
3949    if (inBuffer.frameCount) {
3950        sp<ThreadBase> thread = mThread.promote();
3951        if (thread != 0 && !thread->standby()) {
3952            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3953                pInBuffer = new Buffer;
3954                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3955                pInBuffer->frameCount = inBuffer.frameCount;
3956                pInBuffer->i16 = pInBuffer->mBuffer;
3957                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3958                mBufferQueue.add(pInBuffer);
3959                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3960            } else {
3961                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3962            }
3963        }
3964    }
3965
3966    // Calling write() with a 0 length buffer, means that no more data will be written:
3967    // If no more buffers are pending, fill output track buffer to make sure it is started
3968    // by output mixer.
3969    if (frames == 0 && mBufferQueue.size() == 0) {
3970        if (mCblk->user < mCblk->frameCount) {
3971            frames = mCblk->frameCount - mCblk->user;
3972            pInBuffer = new Buffer;
3973            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3974            pInBuffer->frameCount = frames;
3975            pInBuffer->i16 = pInBuffer->mBuffer;
3976            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3977            mBufferQueue.add(pInBuffer);
3978        } else if (mActive) {
3979            stop();
3980        }
3981    }
3982
3983    return outputBufferFull;
3984}
3985
3986status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3987{
3988    int active;
3989    status_t result;
3990    audio_track_cblk_t* cblk = mCblk;
3991    uint32_t framesReq = buffer->frameCount;
3992
3993//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3994    buffer->frameCount  = 0;
3995
3996    uint32_t framesAvail = cblk->framesAvailable();
3997
3998
3999    if (framesAvail == 0) {
4000        Mutex::Autolock _l(cblk->lock);
4001        goto start_loop_here;
4002        while (framesAvail == 0) {
4003            active = mActive;
4004            if (CC_UNLIKELY(!active)) {
4005                ALOGV("Not active and NO_MORE_BUFFERS");
4006                return NO_MORE_BUFFERS;
4007            }
4008            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
4009            if (result != NO_ERROR) {
4010                return NO_MORE_BUFFERS;
4011            }
4012            // read the server count again
4013        start_loop_here:
4014            framesAvail = cblk->framesAvailable_l();
4015        }
4016    }
4017
4018//    if (framesAvail < framesReq) {
4019//        return NO_MORE_BUFFERS;
4020//    }
4021
4022    if (framesReq > framesAvail) {
4023        framesReq = framesAvail;
4024    }
4025
4026    uint32_t u = cblk->user;
4027    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
4028
4029    if (u + framesReq > bufferEnd) {
4030        framesReq = bufferEnd - u;
4031    }
4032
4033    buffer->frameCount  = framesReq;
4034    buffer->raw         = (void *)cblk->buffer(u);
4035    return NO_ERROR;
4036}
4037
4038
4039void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
4040{
4041    size_t size = mBufferQueue.size();
4042    Buffer *pBuffer;
4043
4044    for (size_t i = 0; i < size; i++) {
4045        pBuffer = mBufferQueue.itemAt(i);
4046        delete [] pBuffer->mBuffer;
4047        delete pBuffer;
4048    }
4049    mBufferQueue.clear();
4050}
4051
4052// ----------------------------------------------------------------------------
4053
4054AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
4055    :   RefBase(),
4056        mAudioFlinger(audioFlinger),
4057        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
4058        mPid(pid)
4059{
4060    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
4061}
4062
4063// Client destructor must be called with AudioFlinger::mLock held
4064AudioFlinger::Client::~Client()
4065{
4066    mAudioFlinger->removeClient_l(mPid);
4067}
4068
4069sp<MemoryDealer> AudioFlinger::Client::heap() const
4070{
4071    return mMemoryDealer;
4072}
4073
4074// ----------------------------------------------------------------------------
4075
4076AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4077                                                     const sp<IAudioFlingerClient>& client,
4078                                                     pid_t pid)
4079    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
4080{
4081}
4082
4083AudioFlinger::NotificationClient::~NotificationClient()
4084{
4085    mClient.clear();
4086}
4087
4088void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4089{
4090    sp<NotificationClient> keep(this);
4091    {
4092        mAudioFlinger->removeNotificationClient(mPid);
4093    }
4094}
4095
4096// ----------------------------------------------------------------------------
4097
4098AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4099    : BnAudioTrack(),
4100      mTrack(track)
4101{
4102}
4103
4104AudioFlinger::TrackHandle::~TrackHandle() {
4105    // just stop the track on deletion, associated resources
4106    // will be freed from the main thread once all pending buffers have
4107    // been played. Unless it's not in the active track list, in which
4108    // case we free everything now...
4109    mTrack->destroy();
4110}
4111
4112sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4113    return mTrack->getCblk();
4114}
4115
4116status_t AudioFlinger::TrackHandle::start() {
4117    return mTrack->start();
4118}
4119
4120void AudioFlinger::TrackHandle::stop() {
4121    mTrack->stop();
4122}
4123
4124void AudioFlinger::TrackHandle::flush() {
4125    mTrack->flush();
4126}
4127
4128void AudioFlinger::TrackHandle::mute(bool e) {
4129    mTrack->mute(e);
4130}
4131
4132void AudioFlinger::TrackHandle::pause() {
4133    mTrack->pause();
4134}
4135
4136status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4137{
4138    return mTrack->attachAuxEffect(EffectId);
4139}
4140
4141status_t AudioFlinger::TrackHandle::onTransact(
4142    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4143{
4144    return BnAudioTrack::onTransact(code, data, reply, flags);
4145}
4146
4147// ----------------------------------------------------------------------------
4148
4149sp<IAudioRecord> AudioFlinger::openRecord(
4150        pid_t pid,
4151        int input,
4152        uint32_t sampleRate,
4153        audio_format_t format,
4154        uint32_t channelMask,
4155        int frameCount,
4156        uint32_t flags,
4157        int *sessionId,
4158        status_t *status)
4159{
4160    sp<RecordThread::RecordTrack> recordTrack;
4161    sp<RecordHandle> recordHandle;
4162    sp<Client> client;
4163    wp<Client> wclient;
4164    status_t lStatus;
4165    RecordThread *thread;
4166    size_t inFrameCount;
4167    int lSessionId;
4168
4169    // check calling permissions
4170    if (!recordingAllowed()) {
4171        lStatus = PERMISSION_DENIED;
4172        goto Exit;
4173    }
4174
4175    // add client to list
4176    { // scope for mLock
4177        Mutex::Autolock _l(mLock);
4178        thread = checkRecordThread_l(input);
4179        if (thread == NULL) {
4180            lStatus = BAD_VALUE;
4181            goto Exit;
4182        }
4183
4184        wclient = mClients.valueFor(pid);
4185        if (wclient != NULL) {
4186            client = wclient.promote();
4187        } else {
4188            client = new Client(this, pid);
4189            mClients.add(pid, client);
4190        }
4191
4192        // If no audio session id is provided, create one here
4193        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4194            lSessionId = *sessionId;
4195        } else {
4196            lSessionId = nextUniqueId();
4197            if (sessionId != NULL) {
4198                *sessionId = lSessionId;
4199            }
4200        }
4201        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4202        recordTrack = thread->createRecordTrack_l(client,
4203                                                sampleRate,
4204                                                format,
4205                                                channelMask,
4206                                                frameCount,
4207                                                flags,
4208                                                lSessionId,
4209                                                &lStatus);
4210    }
4211    if (lStatus != NO_ERROR) {
4212        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4213        // destructor is called by the TrackBase destructor with mLock held
4214        client.clear();
4215        recordTrack.clear();
4216        goto Exit;
4217    }
4218
4219    // return to handle to client
4220    recordHandle = new RecordHandle(recordTrack);
4221    lStatus = NO_ERROR;
4222
4223Exit:
4224    if (status) {
4225        *status = lStatus;
4226    }
4227    return recordHandle;
4228}
4229
4230// ----------------------------------------------------------------------------
4231
4232AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4233    : BnAudioRecord(),
4234    mRecordTrack(recordTrack)
4235{
4236}
4237
4238AudioFlinger::RecordHandle::~RecordHandle() {
4239    stop();
4240}
4241
4242sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4243    return mRecordTrack->getCblk();
4244}
4245
4246status_t AudioFlinger::RecordHandle::start() {
4247    ALOGV("RecordHandle::start()");
4248    return mRecordTrack->start();
4249}
4250
4251void AudioFlinger::RecordHandle::stop() {
4252    ALOGV("RecordHandle::stop()");
4253    mRecordTrack->stop();
4254}
4255
4256status_t AudioFlinger::RecordHandle::onTransact(
4257    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4258{
4259    return BnAudioRecord::onTransact(code, data, reply, flags);
4260}
4261
4262// ----------------------------------------------------------------------------
4263
4264AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4265                                         AudioStreamIn *input,
4266                                         uint32_t sampleRate,
4267                                         uint32_t channels,
4268                                         int id,
4269                                         uint32_t device) :
4270    ThreadBase(audioFlinger, id, device),
4271    mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
4272{
4273    mType = ThreadBase::RECORD;
4274
4275    snprintf(mName, kNameLength, "AudioIn_%d", id);
4276
4277    mReqChannelCount = popcount(channels);
4278    mReqSampleRate = sampleRate;
4279    readInputParameters();
4280}
4281
4282
4283AudioFlinger::RecordThread::~RecordThread()
4284{
4285    delete[] mRsmpInBuffer;
4286    delete mResampler;
4287    delete[] mRsmpOutBuffer;
4288}
4289
4290void AudioFlinger::RecordThread::onFirstRef()
4291{
4292    run(mName, PRIORITY_URGENT_AUDIO);
4293}
4294
4295status_t AudioFlinger::RecordThread::readyToRun()
4296{
4297    status_t status = initCheck();
4298    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4299    return status;
4300}
4301
4302bool AudioFlinger::RecordThread::threadLoop()
4303{
4304    AudioBufferProvider::Buffer buffer;
4305    sp<RecordTrack> activeTrack;
4306    Vector< sp<EffectChain> > effectChains;
4307
4308    nsecs_t lastWarning = 0;
4309
4310    acquireWakeLock();
4311
4312    // start recording
4313    while (!exitPending()) {
4314
4315        processConfigEvents();
4316
4317        { // scope for mLock
4318            Mutex::Autolock _l(mLock);
4319            checkForNewParameters_l();
4320            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4321                if (!mStandby) {
4322                    mInput->stream->common.standby(&mInput->stream->common);
4323                    mStandby = true;
4324                }
4325
4326                if (exitPending()) break;
4327
4328                releaseWakeLock_l();
4329                ALOGV("RecordThread: loop stopping");
4330                // go to sleep
4331                mWaitWorkCV.wait(mLock);
4332                ALOGV("RecordThread: loop starting");
4333                acquireWakeLock_l();
4334                continue;
4335            }
4336            if (mActiveTrack != 0) {
4337                if (mActiveTrack->mState == TrackBase::PAUSING) {
4338                    if (!mStandby) {
4339                        mInput->stream->common.standby(&mInput->stream->common);
4340                        mStandby = true;
4341                    }
4342                    mActiveTrack.clear();
4343                    mStartStopCond.broadcast();
4344                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4345                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4346                        mActiveTrack.clear();
4347                        mStartStopCond.broadcast();
4348                    } else if (mBytesRead != 0) {
4349                        // record start succeeds only if first read from audio input
4350                        // succeeds
4351                        if (mBytesRead > 0) {
4352                            mActiveTrack->mState = TrackBase::ACTIVE;
4353                        } else {
4354                            mActiveTrack.clear();
4355                        }
4356                        mStartStopCond.broadcast();
4357                    }
4358                    mStandby = false;
4359                }
4360            }
4361            lockEffectChains_l(effectChains);
4362        }
4363
4364        if (mActiveTrack != 0) {
4365            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4366                mActiveTrack->mState != TrackBase::RESUMING) {
4367                unlockEffectChains(effectChains);
4368                usleep(kRecordThreadSleepUs);
4369                continue;
4370            }
4371            for (size_t i = 0; i < effectChains.size(); i ++) {
4372                effectChains[i]->process_l();
4373            }
4374
4375            buffer.frameCount = mFrameCount;
4376            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4377                size_t framesOut = buffer.frameCount;
4378                if (mResampler == NULL) {
4379                    // no resampling
4380                    while (framesOut) {
4381                        size_t framesIn = mFrameCount - mRsmpInIndex;
4382                        if (framesIn) {
4383                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4384                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4385                            if (framesIn > framesOut)
4386                                framesIn = framesOut;
4387                            mRsmpInIndex += framesIn;
4388                            framesOut -= framesIn;
4389                            if ((int)mChannelCount == mReqChannelCount ||
4390                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4391                                memcpy(dst, src, framesIn * mFrameSize);
4392                            } else {
4393                                int16_t *src16 = (int16_t *)src;
4394                                int16_t *dst16 = (int16_t *)dst;
4395                                if (mChannelCount == 1) {
4396                                    while (framesIn--) {
4397                                        *dst16++ = *src16;
4398                                        *dst16++ = *src16++;
4399                                    }
4400                                } else {
4401                                    while (framesIn--) {
4402                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4403                                        src16 += 2;
4404                                    }
4405                                }
4406                            }
4407                        }
4408                        if (framesOut && mFrameCount == mRsmpInIndex) {
4409                            if (framesOut == mFrameCount &&
4410                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4411                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4412                                framesOut = 0;
4413                            } else {
4414                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4415                                mRsmpInIndex = 0;
4416                            }
4417                            if (mBytesRead < 0) {
4418                                ALOGE("Error reading audio input");
4419                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4420                                    // Force input into standby so that it tries to
4421                                    // recover at next read attempt
4422                                    mInput->stream->common.standby(&mInput->stream->common);
4423                                    usleep(kRecordThreadSleepUs);
4424                                }
4425                                mRsmpInIndex = mFrameCount;
4426                                framesOut = 0;
4427                                buffer.frameCount = 0;
4428                            }
4429                        }
4430                    }
4431                } else {
4432                    // resampling
4433
4434                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4435                    // alter output frame count as if we were expecting stereo samples
4436                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4437                        framesOut >>= 1;
4438                    }
4439                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4440                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4441                    // are 32 bit aligned which should be always true.
4442                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4443                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4444                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4445                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4446                        int16_t *dst = buffer.i16;
4447                        while (framesOut--) {
4448                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4449                            src += 2;
4450                        }
4451                    } else {
4452                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4453                    }
4454
4455                }
4456                mActiveTrack->releaseBuffer(&buffer);
4457                mActiveTrack->overflow();
4458            }
4459            // client isn't retrieving buffers fast enough
4460            else {
4461                if (!mActiveTrack->setOverflow()) {
4462                    nsecs_t now = systemTime();
4463                    if ((now - lastWarning) > kWarningThrottleNs) {
4464                        ALOGW("RecordThread: buffer overflow");
4465                        lastWarning = now;
4466                    }
4467                }
4468                // Release the processor for a while before asking for a new buffer.
4469                // This will give the application more chance to read from the buffer and
4470                // clear the overflow.
4471                usleep(kRecordThreadSleepUs);
4472            }
4473        }
4474        // enable changes in effect chain
4475        unlockEffectChains(effectChains);
4476        effectChains.clear();
4477    }
4478
4479    if (!mStandby) {
4480        mInput->stream->common.standby(&mInput->stream->common);
4481    }
4482    mActiveTrack.clear();
4483
4484    mStartStopCond.broadcast();
4485
4486    releaseWakeLock();
4487
4488    ALOGV("RecordThread %p exiting", this);
4489    return false;
4490}
4491
4492
4493sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4494        const sp<AudioFlinger::Client>& client,
4495        uint32_t sampleRate,
4496        audio_format_t format,
4497        int channelMask,
4498        int frameCount,
4499        uint32_t flags,
4500        int sessionId,
4501        status_t *status)
4502{
4503    sp<RecordTrack> track;
4504    status_t lStatus;
4505
4506    lStatus = initCheck();
4507    if (lStatus != NO_ERROR) {
4508        ALOGE("Audio driver not initialized.");
4509        goto Exit;
4510    }
4511
4512    { // scope for mLock
4513        Mutex::Autolock _l(mLock);
4514
4515        track = new RecordTrack(this, client, sampleRate,
4516                      format, channelMask, frameCount, flags, sessionId);
4517
4518        if (track->getCblk() == NULL) {
4519            lStatus = NO_MEMORY;
4520            goto Exit;
4521        }
4522
4523        mTrack = track.get();
4524        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4525        bool suspend = audio_is_bluetooth_sco_device(
4526                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4527        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4528        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4529    }
4530    lStatus = NO_ERROR;
4531
4532Exit:
4533    if (status) {
4534        *status = lStatus;
4535    }
4536    return track;
4537}
4538
4539status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4540{
4541    ALOGV("RecordThread::start");
4542    sp <ThreadBase> strongMe = this;
4543    status_t status = NO_ERROR;
4544    {
4545        AutoMutex lock(mLock);
4546        if (mActiveTrack != 0) {
4547            if (recordTrack != mActiveTrack.get()) {
4548                status = -EBUSY;
4549            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4550                mActiveTrack->mState = TrackBase::ACTIVE;
4551            }
4552            return status;
4553        }
4554
4555        recordTrack->mState = TrackBase::IDLE;
4556        mActiveTrack = recordTrack;
4557        mLock.unlock();
4558        status_t status = AudioSystem::startInput(mId);
4559        mLock.lock();
4560        if (status != NO_ERROR) {
4561            mActiveTrack.clear();
4562            return status;
4563        }
4564        mRsmpInIndex = mFrameCount;
4565        mBytesRead = 0;
4566        if (mResampler != NULL) {
4567            mResampler->reset();
4568        }
4569        mActiveTrack->mState = TrackBase::RESUMING;
4570        // signal thread to start
4571        ALOGV("Signal record thread");
4572        mWaitWorkCV.signal();
4573        // do not wait for mStartStopCond if exiting
4574        if (mExiting) {
4575            mActiveTrack.clear();
4576            status = INVALID_OPERATION;
4577            goto startError;
4578        }
4579        mStartStopCond.wait(mLock);
4580        if (mActiveTrack == 0) {
4581            ALOGV("Record failed to start");
4582            status = BAD_VALUE;
4583            goto startError;
4584        }
4585        ALOGV("Record started OK");
4586        return status;
4587    }
4588startError:
4589    AudioSystem::stopInput(mId);
4590    return status;
4591}
4592
4593void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4594    ALOGV("RecordThread::stop");
4595    sp <ThreadBase> strongMe = this;
4596    {
4597        AutoMutex lock(mLock);
4598        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4599            mActiveTrack->mState = TrackBase::PAUSING;
4600            // do not wait for mStartStopCond if exiting
4601            if (mExiting) {
4602                return;
4603            }
4604            mStartStopCond.wait(mLock);
4605            // if we have been restarted, recordTrack == mActiveTrack.get() here
4606            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4607                mLock.unlock();
4608                AudioSystem::stopInput(mId);
4609                mLock.lock();
4610                ALOGV("Record stopped OK");
4611            }
4612        }
4613    }
4614}
4615
4616status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4617{
4618    const size_t SIZE = 256;
4619    char buffer[SIZE];
4620    String8 result;
4621    pid_t pid = 0;
4622
4623    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4624    result.append(buffer);
4625
4626    if (mActiveTrack != 0) {
4627        result.append("Active Track:\n");
4628        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4629        mActiveTrack->dump(buffer, SIZE);
4630        result.append(buffer);
4631
4632        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4633        result.append(buffer);
4634        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4635        result.append(buffer);
4636        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4637        result.append(buffer);
4638        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4639        result.append(buffer);
4640        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4641        result.append(buffer);
4642
4643
4644    } else {
4645        result.append("No record client\n");
4646    }
4647    write(fd, result.string(), result.size());
4648
4649    dumpBase(fd, args);
4650    dumpEffectChains(fd, args);
4651
4652    return NO_ERROR;
4653}
4654
4655status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4656{
4657    size_t framesReq = buffer->frameCount;
4658    size_t framesReady = mFrameCount - mRsmpInIndex;
4659    int channelCount;
4660
4661    if (framesReady == 0) {
4662        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4663        if (mBytesRead < 0) {
4664            ALOGE("RecordThread::getNextBuffer() Error reading audio input");
4665            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4666                // Force input into standby so that it tries to
4667                // recover at next read attempt
4668                mInput->stream->common.standby(&mInput->stream->common);
4669                usleep(kRecordThreadSleepUs);
4670            }
4671            buffer->raw = NULL;
4672            buffer->frameCount = 0;
4673            return NOT_ENOUGH_DATA;
4674        }
4675        mRsmpInIndex = 0;
4676        framesReady = mFrameCount;
4677    }
4678
4679    if (framesReq > framesReady) {
4680        framesReq = framesReady;
4681    }
4682
4683    if (mChannelCount == 1 && mReqChannelCount == 2) {
4684        channelCount = 1;
4685    } else {
4686        channelCount = 2;
4687    }
4688    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4689    buffer->frameCount = framesReq;
4690    return NO_ERROR;
4691}
4692
4693void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4694{
4695    mRsmpInIndex += buffer->frameCount;
4696    buffer->frameCount = 0;
4697}
4698
4699bool AudioFlinger::RecordThread::checkForNewParameters_l()
4700{
4701    bool reconfig = false;
4702
4703    while (!mNewParameters.isEmpty()) {
4704        status_t status = NO_ERROR;
4705        String8 keyValuePair = mNewParameters[0];
4706        AudioParameter param = AudioParameter(keyValuePair);
4707        int value;
4708        audio_format_t reqFormat = mFormat;
4709        int reqSamplingRate = mReqSampleRate;
4710        int reqChannelCount = mReqChannelCount;
4711
4712        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4713            reqSamplingRate = value;
4714            reconfig = true;
4715        }
4716        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4717            reqFormat = (audio_format_t) value;
4718            reconfig = true;
4719        }
4720        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4721            reqChannelCount = popcount(value);
4722            reconfig = true;
4723        }
4724        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4725            // do not accept frame count changes if tracks are open as the track buffer
4726            // size depends on frame count and correct behavior would not be garantied
4727            // if frame count is changed after track creation
4728            if (mActiveTrack != 0) {
4729                status = INVALID_OPERATION;
4730            } else {
4731                reconfig = true;
4732            }
4733        }
4734        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4735            // forward device change to effects that have requested to be
4736            // aware of attached audio device.
4737            for (size_t i = 0; i < mEffectChains.size(); i++) {
4738                mEffectChains[i]->setDevice_l(value);
4739            }
4740            // store input device and output device but do not forward output device to audio HAL.
4741            // Note that status is ignored by the caller for output device
4742            // (see AudioFlinger::setParameters()
4743            if (value & AUDIO_DEVICE_OUT_ALL) {
4744                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4745                status = BAD_VALUE;
4746            } else {
4747                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4748                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4749                if (mTrack != NULL) {
4750                    bool suspend = audio_is_bluetooth_sco_device(
4751                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4752                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4753                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4754                }
4755            }
4756            mDevice |= (uint32_t)value;
4757        }
4758        if (status == NO_ERROR) {
4759            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4760            if (status == INVALID_OPERATION) {
4761               mInput->stream->common.standby(&mInput->stream->common);
4762               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4763            }
4764            if (reconfig) {
4765                if (status == BAD_VALUE &&
4766                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4767                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4768                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4769                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4770                    (reqChannelCount < 3)) {
4771                    status = NO_ERROR;
4772                }
4773                if (status == NO_ERROR) {
4774                    readInputParameters();
4775                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4776                }
4777            }
4778        }
4779
4780        mNewParameters.removeAt(0);
4781
4782        mParamStatus = status;
4783        mParamCond.signal();
4784        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4785        // already timed out waiting for the status and will never signal the condition.
4786        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4787    }
4788    return reconfig;
4789}
4790
4791String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4792{
4793    char *s;
4794    String8 out_s8 = String8();
4795
4796    Mutex::Autolock _l(mLock);
4797    if (initCheck() != NO_ERROR) {
4798        return out_s8;
4799    }
4800
4801    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4802    out_s8 = String8(s);
4803    free(s);
4804    return out_s8;
4805}
4806
4807void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4808    AudioSystem::OutputDescriptor desc;
4809    void *param2 = 0;
4810
4811    switch (event) {
4812    case AudioSystem::INPUT_OPENED:
4813    case AudioSystem::INPUT_CONFIG_CHANGED:
4814        desc.channels = mChannelMask;
4815        desc.samplingRate = mSampleRate;
4816        desc.format = mFormat;
4817        desc.frameCount = mFrameCount;
4818        desc.latency = 0;
4819        param2 = &desc;
4820        break;
4821
4822    case AudioSystem::INPUT_CLOSED:
4823    default:
4824        break;
4825    }
4826    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4827}
4828
4829void AudioFlinger::RecordThread::readInputParameters()
4830{
4831    delete mRsmpInBuffer;
4832    // mRsmpInBuffer is always assigned a new[] below
4833    delete mRsmpOutBuffer;
4834    mRsmpOutBuffer = NULL;
4835    delete mResampler;
4836    mResampler = NULL;
4837
4838    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4839    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4840    mChannelCount = (uint16_t)popcount(mChannelMask);
4841    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4842    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
4843    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4844    mFrameCount = mInputBytes / mFrameSize;
4845    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4846
4847    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4848    {
4849        int channelCount;
4850         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4851         // stereo to mono post process as the resampler always outputs stereo.
4852        if (mChannelCount == 1 && mReqChannelCount == 2) {
4853            channelCount = 1;
4854        } else {
4855            channelCount = 2;
4856        }
4857        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4858        mResampler->setSampleRate(mSampleRate);
4859        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4860        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4861
4862        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4863        if (mChannelCount == 1 && mReqChannelCount == 1) {
4864            mFrameCount >>= 1;
4865        }
4866
4867    }
4868    mRsmpInIndex = mFrameCount;
4869}
4870
4871unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4872{
4873    Mutex::Autolock _l(mLock);
4874    if (initCheck() != NO_ERROR) {
4875        return 0;
4876    }
4877
4878    return mInput->stream->get_input_frames_lost(mInput->stream);
4879}
4880
4881uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4882{
4883    Mutex::Autolock _l(mLock);
4884    uint32_t result = 0;
4885    if (getEffectChain_l(sessionId) != 0) {
4886        result = EFFECT_SESSION;
4887    }
4888
4889    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4890        result |= TRACK_SESSION;
4891    }
4892
4893    return result;
4894}
4895
4896AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4897{
4898    Mutex::Autolock _l(mLock);
4899    return mTrack;
4900}
4901
4902AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const
4903{
4904    Mutex::Autolock _l(mLock);
4905    return mInput;
4906}
4907
4908AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4909{
4910    Mutex::Autolock _l(mLock);
4911    AudioStreamIn *input = mInput;
4912    mInput = NULL;
4913    return input;
4914}
4915
4916// this method must always be called either with ThreadBase mLock held or inside the thread loop
4917audio_stream_t* AudioFlinger::RecordThread::stream()
4918{
4919    if (mInput == NULL) {
4920        return NULL;
4921    }
4922    return &mInput->stream->common;
4923}
4924
4925
4926// ----------------------------------------------------------------------------
4927
4928int AudioFlinger::openOutput(uint32_t *pDevices,
4929                                uint32_t *pSamplingRate,
4930                                audio_format_t *pFormat,
4931                                uint32_t *pChannels,
4932                                uint32_t *pLatencyMs,
4933                                uint32_t flags)
4934{
4935    status_t status;
4936    PlaybackThread *thread = NULL;
4937    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4938    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4939    audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
4940    uint32_t channels = pChannels ? *pChannels : 0;
4941    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4942    audio_stream_out_t *outStream;
4943    audio_hw_device_t *outHwDev;
4944
4945    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4946            pDevices ? *pDevices : 0,
4947            samplingRate,
4948            format,
4949            channels,
4950            flags);
4951
4952    if (pDevices == NULL || *pDevices == 0) {
4953        return 0;
4954    }
4955
4956    Mutex::Autolock _l(mLock);
4957
4958    outHwDev = findSuitableHwDev_l(*pDevices);
4959    if (outHwDev == NULL)
4960        return 0;
4961
4962    status = outHwDev->open_output_stream(outHwDev, *pDevices, &format,
4963                                          &channels, &samplingRate, &outStream);
4964    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4965            outStream,
4966            samplingRate,
4967            format,
4968            channels,
4969            status);
4970
4971    mHardwareStatus = AUDIO_HW_IDLE;
4972    if (outStream != NULL) {
4973        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4974        int id = nextUniqueId();
4975
4976        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4977            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4978            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4979            thread = new DirectOutputThread(this, output, id, *pDevices);
4980            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4981        } else {
4982            thread = new MixerThread(this, output, id, *pDevices);
4983            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4984        }
4985        mPlaybackThreads.add(id, thread);
4986
4987        if (pSamplingRate) *pSamplingRate = samplingRate;
4988        if (pFormat) *pFormat = format;
4989        if (pChannels) *pChannels = channels;
4990        if (pLatencyMs) *pLatencyMs = thread->latency();
4991
4992        // notify client processes of the new output creation
4993        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4994        return id;
4995    }
4996
4997    return 0;
4998}
4999
5000int AudioFlinger::openDuplicateOutput(int output1, int output2)
5001{
5002    Mutex::Autolock _l(mLock);
5003    MixerThread *thread1 = checkMixerThread_l(output1);
5004    MixerThread *thread2 = checkMixerThread_l(output2);
5005
5006    if (thread1 == NULL || thread2 == NULL) {
5007        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
5008        return 0;
5009    }
5010
5011    int id = nextUniqueId();
5012    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
5013    thread->addOutputTrack(thread2);
5014    mPlaybackThreads.add(id, thread);
5015    // notify client processes of the new output creation
5016    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
5017    return id;
5018}
5019
5020status_t AudioFlinger::closeOutput(int output)
5021{
5022    // keep strong reference on the playback thread so that
5023    // it is not destroyed while exit() is executed
5024    sp <PlaybackThread> thread;
5025    {
5026        Mutex::Autolock _l(mLock);
5027        thread = checkPlaybackThread_l(output);
5028        if (thread == NULL) {
5029            return BAD_VALUE;
5030        }
5031
5032        ALOGV("closeOutput() %d", output);
5033
5034        if (thread->type() == ThreadBase::MIXER) {
5035            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5036                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
5037                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
5038                    dupThread->removeOutputTrack((MixerThread *)thread.get());
5039                }
5040            }
5041        }
5042        void *param2 = 0;
5043        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
5044        mPlaybackThreads.removeItem(output);
5045    }
5046    thread->exit();
5047
5048    if (thread->type() != ThreadBase::DUPLICATING) {
5049        AudioStreamOut *out = thread->clearOutput();
5050        assert(out != NULL);
5051        // from now on thread->mOutput is NULL
5052        out->hwDev->close_output_stream(out->hwDev, out->stream);
5053        delete out;
5054    }
5055    return NO_ERROR;
5056}
5057
5058status_t AudioFlinger::suspendOutput(int output)
5059{
5060    Mutex::Autolock _l(mLock);
5061    PlaybackThread *thread = checkPlaybackThread_l(output);
5062
5063    if (thread == NULL) {
5064        return BAD_VALUE;
5065    }
5066
5067    ALOGV("suspendOutput() %d", output);
5068    thread->suspend();
5069
5070    return NO_ERROR;
5071}
5072
5073status_t AudioFlinger::restoreOutput(int output)
5074{
5075    Mutex::Autolock _l(mLock);
5076    PlaybackThread *thread = checkPlaybackThread_l(output);
5077
5078    if (thread == NULL) {
5079        return BAD_VALUE;
5080    }
5081
5082    ALOGV("restoreOutput() %d", output);
5083
5084    thread->restore();
5085
5086    return NO_ERROR;
5087}
5088
5089int AudioFlinger::openInput(uint32_t *pDevices,
5090                                uint32_t *pSamplingRate,
5091                                audio_format_t *pFormat,
5092                                uint32_t *pChannels,
5093                                uint32_t acoustics)
5094{
5095    status_t status;
5096    RecordThread *thread = NULL;
5097    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5098    audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
5099    uint32_t channels = pChannels ? *pChannels : 0;
5100    uint32_t reqSamplingRate = samplingRate;
5101    audio_format_t reqFormat = format;
5102    uint32_t reqChannels = channels;
5103    audio_stream_in_t *inStream;
5104    audio_hw_device_t *inHwDev;
5105
5106    if (pDevices == NULL || *pDevices == 0) {
5107        return 0;
5108    }
5109
5110    Mutex::Autolock _l(mLock);
5111
5112    inHwDev = findSuitableHwDev_l(*pDevices);
5113    if (inHwDev == NULL)
5114        return 0;
5115
5116    status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
5117                                        &channels, &samplingRate,
5118                                        (audio_in_acoustics_t)acoustics,
5119                                        &inStream);
5120    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5121            inStream,
5122            samplingRate,
5123            format,
5124            channels,
5125            acoustics,
5126            status);
5127
5128    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5129    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5130    // or stereo to mono conversions on 16 bit PCM inputs.
5131    if (inStream == NULL && status == BAD_VALUE &&
5132        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5133        (samplingRate <= 2 * reqSamplingRate) &&
5134        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5135        ALOGV("openInput() reopening with proposed sampling rate and channels");
5136        status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
5137                                            &channels, &samplingRate,
5138                                            (audio_in_acoustics_t)acoustics,
5139                                            &inStream);
5140    }
5141
5142    if (inStream != NULL) {
5143        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5144
5145        int id = nextUniqueId();
5146        // Start record thread
5147        // RecorThread require both input and output device indication to forward to audio
5148        // pre processing modules
5149        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5150        thread = new RecordThread(this,
5151                                  input,
5152                                  reqSamplingRate,
5153                                  reqChannels,
5154                                  id,
5155                                  device);
5156        mRecordThreads.add(id, thread);
5157        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5158        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5159        if (pFormat) *pFormat = format;
5160        if (pChannels) *pChannels = reqChannels;
5161
5162        input->stream->common.standby(&input->stream->common);
5163
5164        // notify client processes of the new input creation
5165        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5166        return id;
5167    }
5168
5169    return 0;
5170}
5171
5172status_t AudioFlinger::closeInput(int input)
5173{
5174    // keep strong reference on the record thread so that
5175    // it is not destroyed while exit() is executed
5176    sp <RecordThread> thread;
5177    {
5178        Mutex::Autolock _l(mLock);
5179        thread = checkRecordThread_l(input);
5180        if (thread == NULL) {
5181            return BAD_VALUE;
5182        }
5183
5184        ALOGV("closeInput() %d", input);
5185        void *param2 = 0;
5186        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5187        mRecordThreads.removeItem(input);
5188    }
5189    thread->exit();
5190
5191    AudioStreamIn *in = thread->clearInput();
5192    assert(in != NULL);
5193    // from now on thread->mInput is NULL
5194    in->hwDev->close_input_stream(in->hwDev, in->stream);
5195    delete in;
5196
5197    return NO_ERROR;
5198}
5199
5200status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output)
5201{
5202    Mutex::Autolock _l(mLock);
5203    MixerThread *dstThread = checkMixerThread_l(output);
5204    if (dstThread == NULL) {
5205        ALOGW("setStreamOutput() bad output id %d", output);
5206        return BAD_VALUE;
5207    }
5208
5209    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5210    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5211
5212    dstThread->setStreamValid(stream, true);
5213
5214    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5215        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5216        if (thread != dstThread &&
5217            thread->type() != ThreadBase::DIRECT) {
5218            MixerThread *srcThread = (MixerThread *)thread;
5219            srcThread->setStreamValid(stream, false);
5220            srcThread->invalidateTracks(stream);
5221        }
5222    }
5223
5224    return NO_ERROR;
5225}
5226
5227
5228int AudioFlinger::newAudioSessionId()
5229{
5230    return nextUniqueId();
5231}
5232
5233void AudioFlinger::acquireAudioSessionId(int audioSession)
5234{
5235    Mutex::Autolock _l(mLock);
5236    int caller = IPCThreadState::self()->getCallingPid();
5237    ALOGV("acquiring %d from %d", audioSession, caller);
5238    int num = mAudioSessionRefs.size();
5239    for (int i = 0; i< num; i++) {
5240        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5241        if (ref->sessionid == audioSession && ref->pid == caller) {
5242            ref->cnt++;
5243            ALOGV(" incremented refcount to %d", ref->cnt);
5244            return;
5245        }
5246    }
5247    AudioSessionRef *ref = new AudioSessionRef();
5248    ref->sessionid = audioSession;
5249    ref->pid = caller;
5250    ref->cnt = 1;
5251    mAudioSessionRefs.push(ref);
5252    ALOGV(" added new entry for %d", ref->sessionid);
5253}
5254
5255void AudioFlinger::releaseAudioSessionId(int audioSession)
5256{
5257    Mutex::Autolock _l(mLock);
5258    int caller = IPCThreadState::self()->getCallingPid();
5259    ALOGV("releasing %d from %d", audioSession, caller);
5260    int num = mAudioSessionRefs.size();
5261    for (int i = 0; i< num; i++) {
5262        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5263        if (ref->sessionid == audioSession && ref->pid == caller) {
5264            ref->cnt--;
5265            ALOGV(" decremented refcount to %d", ref->cnt);
5266            if (ref->cnt == 0) {
5267                mAudioSessionRefs.removeAt(i);
5268                delete ref;
5269                purgeStaleEffects_l();
5270            }
5271            return;
5272        }
5273    }
5274    ALOGW("session id %d not found for pid %d", audioSession, caller);
5275}
5276
5277void AudioFlinger::purgeStaleEffects_l() {
5278
5279    ALOGV("purging stale effects");
5280
5281    Vector< sp<EffectChain> > chains;
5282
5283    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5284        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5285        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5286            sp<EffectChain> ec = t->mEffectChains[j];
5287            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5288                chains.push(ec);
5289            }
5290        }
5291    }
5292    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5293        sp<RecordThread> t = mRecordThreads.valueAt(i);
5294        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5295            sp<EffectChain> ec = t->mEffectChains[j];
5296            chains.push(ec);
5297        }
5298    }
5299
5300    for (size_t i = 0; i < chains.size(); i++) {
5301        sp<EffectChain> ec = chains[i];
5302        int sessionid = ec->sessionId();
5303        sp<ThreadBase> t = ec->mThread.promote();
5304        if (t == 0) {
5305            continue;
5306        }
5307        size_t numsessionrefs = mAudioSessionRefs.size();
5308        bool found = false;
5309        for (size_t k = 0; k < numsessionrefs; k++) {
5310            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5311            if (ref->sessionid == sessionid) {
5312                ALOGV(" session %d still exists for %d with %d refs",
5313                     sessionid, ref->pid, ref->cnt);
5314                found = true;
5315                break;
5316            }
5317        }
5318        if (!found) {
5319            // remove all effects from the chain
5320            while (ec->mEffects.size()) {
5321                sp<EffectModule> effect = ec->mEffects[0];
5322                effect->unPin();
5323                Mutex::Autolock _l (t->mLock);
5324                t->removeEffect_l(effect);
5325                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5326                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5327                    if (handle != 0) {
5328                        handle->mEffect.clear();
5329                        if (handle->mHasControl && handle->mEnabled) {
5330                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5331                        }
5332                    }
5333                }
5334                AudioSystem::unregisterEffect(effect->id());
5335            }
5336        }
5337    }
5338    return;
5339}
5340
5341// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5342AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5343{
5344    PlaybackThread *thread = NULL;
5345    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5346        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5347    }
5348    return thread;
5349}
5350
5351// checkMixerThread_l() must be called with AudioFlinger::mLock held
5352AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5353{
5354    PlaybackThread *thread = checkPlaybackThread_l(output);
5355    if (thread != NULL) {
5356        if (thread->type() == ThreadBase::DIRECT) {
5357            thread = NULL;
5358        }
5359    }
5360    return (MixerThread *)thread;
5361}
5362
5363// checkRecordThread_l() must be called with AudioFlinger::mLock held
5364AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5365{
5366    RecordThread *thread = NULL;
5367    if (mRecordThreads.indexOfKey(input) >= 0) {
5368        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5369    }
5370    return thread;
5371}
5372
5373uint32_t AudioFlinger::nextUniqueId()
5374{
5375    return android_atomic_inc(&mNextUniqueId);
5376}
5377
5378AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5379{
5380    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5381        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5382        AudioStreamOut *output = thread->getOutput();
5383        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5384            return thread;
5385        }
5386    }
5387    return NULL;
5388}
5389
5390uint32_t AudioFlinger::primaryOutputDevice_l()
5391{
5392    PlaybackThread *thread = primaryPlaybackThread_l();
5393
5394    if (thread == NULL) {
5395        return 0;
5396    }
5397
5398    return thread->device();
5399}
5400
5401
5402// ----------------------------------------------------------------------------
5403//  Effect management
5404// ----------------------------------------------------------------------------
5405
5406
5407status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5408{
5409    Mutex::Autolock _l(mLock);
5410    return EffectQueryNumberEffects(numEffects);
5411}
5412
5413status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5414{
5415    Mutex::Autolock _l(mLock);
5416    return EffectQueryEffect(index, descriptor);
5417}
5418
5419status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5420{
5421    Mutex::Autolock _l(mLock);
5422    return EffectGetDescriptor(pUuid, descriptor);
5423}
5424
5425
5426sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5427        effect_descriptor_t *pDesc,
5428        const sp<IEffectClient>& effectClient,
5429        int32_t priority,
5430        int io,
5431        int sessionId,
5432        status_t *status,
5433        int *id,
5434        int *enabled)
5435{
5436    status_t lStatus = NO_ERROR;
5437    sp<EffectHandle> handle;
5438    effect_descriptor_t desc;
5439    sp<Client> client;
5440    wp<Client> wclient;
5441
5442    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5443            pid, effectClient.get(), priority, sessionId, io);
5444
5445    if (pDesc == NULL) {
5446        lStatus = BAD_VALUE;
5447        goto Exit;
5448    }
5449
5450    // check audio settings permission for global effects
5451    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5452        lStatus = PERMISSION_DENIED;
5453        goto Exit;
5454    }
5455
5456    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5457    // that can only be created by audio policy manager (running in same process)
5458    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5459        lStatus = PERMISSION_DENIED;
5460        goto Exit;
5461    }
5462
5463    if (io == 0) {
5464        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5465            // output must be specified by AudioPolicyManager when using session
5466            // AUDIO_SESSION_OUTPUT_STAGE
5467            lStatus = BAD_VALUE;
5468            goto Exit;
5469        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5470            // if the output returned by getOutputForEffect() is removed before we lock the
5471            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5472            // and we will exit safely
5473            io = AudioSystem::getOutputForEffect(&desc);
5474        }
5475    }
5476
5477    {
5478        Mutex::Autolock _l(mLock);
5479
5480
5481        if (!EffectIsNullUuid(&pDesc->uuid)) {
5482            // if uuid is specified, request effect descriptor
5483            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5484            if (lStatus < 0) {
5485                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5486                goto Exit;
5487            }
5488        } else {
5489            // if uuid is not specified, look for an available implementation
5490            // of the required type in effect factory
5491            if (EffectIsNullUuid(&pDesc->type)) {
5492                ALOGW("createEffect() no effect type");
5493                lStatus = BAD_VALUE;
5494                goto Exit;
5495            }
5496            uint32_t numEffects = 0;
5497            effect_descriptor_t d;
5498            d.flags = 0; // prevent compiler warning
5499            bool found = false;
5500
5501            lStatus = EffectQueryNumberEffects(&numEffects);
5502            if (lStatus < 0) {
5503                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5504                goto Exit;
5505            }
5506            for (uint32_t i = 0; i < numEffects; i++) {
5507                lStatus = EffectQueryEffect(i, &desc);
5508                if (lStatus < 0) {
5509                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5510                    continue;
5511                }
5512                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5513                    // If matching type found save effect descriptor. If the session is
5514                    // 0 and the effect is not auxiliary, continue enumeration in case
5515                    // an auxiliary version of this effect type is available
5516                    found = true;
5517                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5518                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5519                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5520                        break;
5521                    }
5522                }
5523            }
5524            if (!found) {
5525                lStatus = BAD_VALUE;
5526                ALOGW("createEffect() effect not found");
5527                goto Exit;
5528            }
5529            // For same effect type, chose auxiliary version over insert version if
5530            // connect to output mix (Compliance to OpenSL ES)
5531            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5532                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5533                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5534            }
5535        }
5536
5537        // Do not allow auxiliary effects on a session different from 0 (output mix)
5538        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5539             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5540            lStatus = INVALID_OPERATION;
5541            goto Exit;
5542        }
5543
5544        // check recording permission for visualizer
5545        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5546            !recordingAllowed()) {
5547            lStatus = PERMISSION_DENIED;
5548            goto Exit;
5549        }
5550
5551        // return effect descriptor
5552        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5553
5554        // If output is not specified try to find a matching audio session ID in one of the
5555        // output threads.
5556        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5557        // because of code checking output when entering the function.
5558        // Note: io is never 0 when creating an effect on an input
5559        if (io == 0) {
5560             // look for the thread where the specified audio session is present
5561            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5562                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5563                    io = mPlaybackThreads.keyAt(i);
5564                    break;
5565                }
5566            }
5567            if (io == 0) {
5568               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5569                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5570                       io = mRecordThreads.keyAt(i);
5571                       break;
5572                   }
5573               }
5574            }
5575            // If no output thread contains the requested session ID, default to
5576            // first output. The effect chain will be moved to the correct output
5577            // thread when a track with the same session ID is created
5578            if (io == 0 && mPlaybackThreads.size()) {
5579                io = mPlaybackThreads.keyAt(0);
5580            }
5581            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5582        }
5583        ThreadBase *thread = checkRecordThread_l(io);
5584        if (thread == NULL) {
5585            thread = checkPlaybackThread_l(io);
5586            if (thread == NULL) {
5587                ALOGE("createEffect() unknown output thread");
5588                lStatus = BAD_VALUE;
5589                goto Exit;
5590            }
5591        }
5592
5593        wclient = mClients.valueFor(pid);
5594
5595        if (wclient != NULL) {
5596            client = wclient.promote();
5597        } else {
5598            client = new Client(this, pid);
5599            mClients.add(pid, client);
5600        }
5601
5602        // create effect on selected output thread
5603        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5604                &desc, enabled, &lStatus);
5605        if (handle != 0 && id != NULL) {
5606            *id = handle->id();
5607        }
5608    }
5609
5610Exit:
5611    if(status) {
5612        *status = lStatus;
5613    }
5614    return handle;
5615}
5616
5617status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5618{
5619    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5620            sessionId, srcOutput, dstOutput);
5621    Mutex::Autolock _l(mLock);
5622    if (srcOutput == dstOutput) {
5623        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
5624        return NO_ERROR;
5625    }
5626    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5627    if (srcThread == NULL) {
5628        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
5629        return BAD_VALUE;
5630    }
5631    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5632    if (dstThread == NULL) {
5633        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
5634        return BAD_VALUE;
5635    }
5636
5637    Mutex::Autolock _dl(dstThread->mLock);
5638    Mutex::Autolock _sl(srcThread->mLock);
5639    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5640
5641    return NO_ERROR;
5642}
5643
5644// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5645status_t AudioFlinger::moveEffectChain_l(int sessionId,
5646                                   AudioFlinger::PlaybackThread *srcThread,
5647                                   AudioFlinger::PlaybackThread *dstThread,
5648                                   bool reRegister)
5649{
5650    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5651            sessionId, srcThread, dstThread);
5652
5653    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5654    if (chain == 0) {
5655        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5656                sessionId, srcThread);
5657        return INVALID_OPERATION;
5658    }
5659
5660    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5661    // so that a new chain is created with correct parameters when first effect is added. This is
5662    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5663    // removed.
5664    srcThread->removeEffectChain_l(chain);
5665
5666    // transfer all effects one by one so that new effect chain is created on new thread with
5667    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5668    int dstOutput = dstThread->id();
5669    sp<EffectChain> dstChain;
5670    uint32_t strategy = 0; // prevent compiler warning
5671    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5672    while (effect != 0) {
5673        srcThread->removeEffect_l(effect);
5674        dstThread->addEffect_l(effect);
5675        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5676        if (effect->state() == EffectModule::ACTIVE ||
5677                effect->state() == EffectModule::STOPPING) {
5678            effect->start();
5679        }
5680        // if the move request is not received from audio policy manager, the effect must be
5681        // re-registered with the new strategy and output
5682        if (dstChain == 0) {
5683            dstChain = effect->chain().promote();
5684            if (dstChain == 0) {
5685                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5686                srcThread->addEffect_l(effect);
5687                return NO_INIT;
5688            }
5689            strategy = dstChain->strategy();
5690        }
5691        if (reRegister) {
5692            AudioSystem::unregisterEffect(effect->id());
5693            AudioSystem::registerEffect(&effect->desc(),
5694                                        dstOutput,
5695                                        strategy,
5696                                        sessionId,
5697                                        effect->id());
5698        }
5699        effect = chain->getEffectFromId_l(0);
5700    }
5701
5702    return NO_ERROR;
5703}
5704
5705
5706// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5707sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5708        const sp<AudioFlinger::Client>& client,
5709        const sp<IEffectClient>& effectClient,
5710        int32_t priority,
5711        int sessionId,
5712        effect_descriptor_t *desc,
5713        int *enabled,
5714        status_t *status
5715        )
5716{
5717    sp<EffectModule> effect;
5718    sp<EffectHandle> handle;
5719    status_t lStatus;
5720    sp<EffectChain> chain;
5721    bool chainCreated = false;
5722    bool effectCreated = false;
5723    bool effectRegistered = false;
5724
5725    lStatus = initCheck();
5726    if (lStatus != NO_ERROR) {
5727        ALOGW("createEffect_l() Audio driver not initialized.");
5728        goto Exit;
5729    }
5730
5731    // Do not allow effects with session ID 0 on direct output or duplicating threads
5732    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5733    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5734        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5735                desc->name, sessionId);
5736        lStatus = BAD_VALUE;
5737        goto Exit;
5738    }
5739    // Only Pre processor effects are allowed on input threads and only on input threads
5740    if ((mType == RECORD &&
5741            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5742            (mType != RECORD &&
5743                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5744        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5745                desc->name, desc->flags, mType);
5746        lStatus = BAD_VALUE;
5747        goto Exit;
5748    }
5749
5750    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5751
5752    { // scope for mLock
5753        Mutex::Autolock _l(mLock);
5754
5755        // check for existing effect chain with the requested audio session
5756        chain = getEffectChain_l(sessionId);
5757        if (chain == 0) {
5758            // create a new chain for this session
5759            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5760            chain = new EffectChain(this, sessionId);
5761            addEffectChain_l(chain);
5762            chain->setStrategy(getStrategyForSession_l(sessionId));
5763            chainCreated = true;
5764        } else {
5765            effect = chain->getEffectFromDesc_l(desc);
5766        }
5767
5768        ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
5769
5770        if (effect == 0) {
5771            int id = mAudioFlinger->nextUniqueId();
5772            // Check CPU and memory usage
5773            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5774            if (lStatus != NO_ERROR) {
5775                goto Exit;
5776            }
5777            effectRegistered = true;
5778            // create a new effect module if none present in the chain
5779            effect = new EffectModule(this, chain, desc, id, sessionId);
5780            lStatus = effect->status();
5781            if (lStatus != NO_ERROR) {
5782                goto Exit;
5783            }
5784            lStatus = chain->addEffect_l(effect);
5785            if (lStatus != NO_ERROR) {
5786                goto Exit;
5787            }
5788            effectCreated = true;
5789
5790            effect->setDevice(mDevice);
5791            effect->setMode(mAudioFlinger->getMode());
5792        }
5793        // create effect handle and connect it to effect module
5794        handle = new EffectHandle(effect, client, effectClient, priority);
5795        lStatus = effect->addHandle(handle);
5796        if (enabled) {
5797            *enabled = (int)effect->isEnabled();
5798        }
5799    }
5800
5801Exit:
5802    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5803        Mutex::Autolock _l(mLock);
5804        if (effectCreated) {
5805            chain->removeEffect_l(effect);
5806        }
5807        if (effectRegistered) {
5808            AudioSystem::unregisterEffect(effect->id());
5809        }
5810        if (chainCreated) {
5811            removeEffectChain_l(chain);
5812        }
5813        handle.clear();
5814    }
5815
5816    if(status) {
5817        *status = lStatus;
5818    }
5819    return handle;
5820}
5821
5822sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5823{
5824    sp<EffectModule> effect;
5825
5826    sp<EffectChain> chain = getEffectChain_l(sessionId);
5827    if (chain != 0) {
5828        effect = chain->getEffectFromId_l(effectId);
5829    }
5830    return effect;
5831}
5832
5833// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5834// PlaybackThread::mLock held
5835status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5836{
5837    // check for existing effect chain with the requested audio session
5838    int sessionId = effect->sessionId();
5839    sp<EffectChain> chain = getEffectChain_l(sessionId);
5840    bool chainCreated = false;
5841
5842    if (chain == 0) {
5843        // create a new chain for this session
5844        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5845        chain = new EffectChain(this, sessionId);
5846        addEffectChain_l(chain);
5847        chain->setStrategy(getStrategyForSession_l(sessionId));
5848        chainCreated = true;
5849    }
5850    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5851
5852    if (chain->getEffectFromId_l(effect->id()) != 0) {
5853        ALOGW("addEffect_l() %p effect %s already present in chain %p",
5854                this, effect->desc().name, chain.get());
5855        return BAD_VALUE;
5856    }
5857
5858    status_t status = chain->addEffect_l(effect);
5859    if (status != NO_ERROR) {
5860        if (chainCreated) {
5861            removeEffectChain_l(chain);
5862        }
5863        return status;
5864    }
5865
5866    effect->setDevice(mDevice);
5867    effect->setMode(mAudioFlinger->getMode());
5868    return NO_ERROR;
5869}
5870
5871void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5872
5873    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5874    effect_descriptor_t desc = effect->desc();
5875    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5876        detachAuxEffect_l(effect->id());
5877    }
5878
5879    sp<EffectChain> chain = effect->chain().promote();
5880    if (chain != 0) {
5881        // remove effect chain if removing last effect
5882        if (chain->removeEffect_l(effect) == 0) {
5883            removeEffectChain_l(chain);
5884        }
5885    } else {
5886        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5887    }
5888}
5889
5890void AudioFlinger::ThreadBase::lockEffectChains_l(
5891        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5892{
5893    effectChains = mEffectChains;
5894    for (size_t i = 0; i < mEffectChains.size(); i++) {
5895        mEffectChains[i]->lock();
5896    }
5897}
5898
5899void AudioFlinger::ThreadBase::unlockEffectChains(
5900        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5901{
5902    for (size_t i = 0; i < effectChains.size(); i++) {
5903        effectChains[i]->unlock();
5904    }
5905}
5906
5907sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5908{
5909    Mutex::Autolock _l(mLock);
5910    return getEffectChain_l(sessionId);
5911}
5912
5913sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5914{
5915    sp<EffectChain> chain;
5916
5917    size_t size = mEffectChains.size();
5918    for (size_t i = 0; i < size; i++) {
5919        if (mEffectChains[i]->sessionId() == sessionId) {
5920            chain = mEffectChains[i];
5921            break;
5922        }
5923    }
5924    return chain;
5925}
5926
5927void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
5928{
5929    Mutex::Autolock _l(mLock);
5930    size_t size = mEffectChains.size();
5931    for (size_t i = 0; i < size; i++) {
5932        mEffectChains[i]->setMode_l(mode);
5933    }
5934}
5935
5936void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5937                                                    const wp<EffectHandle>& handle,
5938                                                    bool unpiniflast) {
5939
5940    Mutex::Autolock _l(mLock);
5941    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5942    // delete the effect module if removing last handle on it
5943    if (effect->removeHandle(handle) == 0) {
5944        if (!effect->isPinned() || unpiniflast) {
5945            removeEffect_l(effect);
5946            AudioSystem::unregisterEffect(effect->id());
5947        }
5948    }
5949}
5950
5951status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5952{
5953    int session = chain->sessionId();
5954    int16_t *buffer = mMixBuffer;
5955    bool ownsBuffer = false;
5956
5957    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5958    if (session > 0) {
5959        // Only one effect chain can be present in direct output thread and it uses
5960        // the mix buffer as input
5961        if (mType != DIRECT) {
5962            size_t numSamples = mFrameCount * mChannelCount;
5963            buffer = new int16_t[numSamples];
5964            memset(buffer, 0, numSamples * sizeof(int16_t));
5965            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5966            ownsBuffer = true;
5967        }
5968
5969        // Attach all tracks with same session ID to this chain.
5970        for (size_t i = 0; i < mTracks.size(); ++i) {
5971            sp<Track> track = mTracks[i];
5972            if (session == track->sessionId()) {
5973                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5974                track->setMainBuffer(buffer);
5975                chain->incTrackCnt();
5976            }
5977        }
5978
5979        // indicate all active tracks in the chain
5980        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5981            sp<Track> track = mActiveTracks[i].promote();
5982            if (track == 0) continue;
5983            if (session == track->sessionId()) {
5984                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5985                chain->incActiveTrackCnt();
5986            }
5987        }
5988    }
5989
5990    chain->setInBuffer(buffer, ownsBuffer);
5991    chain->setOutBuffer(mMixBuffer);
5992    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5993    // chains list in order to be processed last as it contains output stage effects
5994    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5995    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5996    // after track specific effects and before output stage
5997    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5998    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5999    // Effect chain for other sessions are inserted at beginning of effect
6000    // chains list to be processed before output mix effects. Relative order between other
6001    // sessions is not important
6002    size_t size = mEffectChains.size();
6003    size_t i = 0;
6004    for (i = 0; i < size; i++) {
6005        if (mEffectChains[i]->sessionId() < session) break;
6006    }
6007    mEffectChains.insertAt(chain, i);
6008    checkSuspendOnAddEffectChain_l(chain);
6009
6010    return NO_ERROR;
6011}
6012
6013size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
6014{
6015    int session = chain->sessionId();
6016
6017    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
6018
6019    for (size_t i = 0; i < mEffectChains.size(); i++) {
6020        if (chain == mEffectChains[i]) {
6021            mEffectChains.removeAt(i);
6022            // detach all active tracks from the chain
6023            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
6024                sp<Track> track = mActiveTracks[i].promote();
6025                if (track == 0) continue;
6026                if (session == track->sessionId()) {
6027                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
6028                            chain.get(), session);
6029                    chain->decActiveTrackCnt();
6030                }
6031            }
6032
6033            // detach all tracks with same session ID from this chain
6034            for (size_t i = 0; i < mTracks.size(); ++i) {
6035                sp<Track> track = mTracks[i];
6036                if (session == track->sessionId()) {
6037                    track->setMainBuffer(mMixBuffer);
6038                    chain->decTrackCnt();
6039                }
6040            }
6041            break;
6042        }
6043    }
6044    return mEffectChains.size();
6045}
6046
6047status_t AudioFlinger::PlaybackThread::attachAuxEffect(
6048        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6049{
6050    Mutex::Autolock _l(mLock);
6051    return attachAuxEffect_l(track, EffectId);
6052}
6053
6054status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6055        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6056{
6057    status_t status = NO_ERROR;
6058
6059    if (EffectId == 0) {
6060        track->setAuxBuffer(0, NULL);
6061    } else {
6062        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6063        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6064        if (effect != 0) {
6065            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6066                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6067            } else {
6068                status = INVALID_OPERATION;
6069            }
6070        } else {
6071            status = BAD_VALUE;
6072        }
6073    }
6074    return status;
6075}
6076
6077void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6078{
6079     for (size_t i = 0; i < mTracks.size(); ++i) {
6080        sp<Track> track = mTracks[i];
6081        if (track->auxEffectId() == effectId) {
6082            attachAuxEffect_l(track, 0);
6083        }
6084    }
6085}
6086
6087status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6088{
6089    // only one chain per input thread
6090    if (mEffectChains.size() != 0) {
6091        return INVALID_OPERATION;
6092    }
6093    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6094
6095    chain->setInBuffer(NULL);
6096    chain->setOutBuffer(NULL);
6097
6098    checkSuspendOnAddEffectChain_l(chain);
6099
6100    mEffectChains.add(chain);
6101
6102    return NO_ERROR;
6103}
6104
6105size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6106{
6107    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6108    ALOGW_IF(mEffectChains.size() != 1,
6109            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6110            chain.get(), mEffectChains.size(), this);
6111    if (mEffectChains.size() == 1) {
6112        mEffectChains.removeAt(0);
6113    }
6114    return 0;
6115}
6116
6117// ----------------------------------------------------------------------------
6118//  EffectModule implementation
6119// ----------------------------------------------------------------------------
6120
6121#undef LOG_TAG
6122#define LOG_TAG "AudioFlinger::EffectModule"
6123
6124AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6125                                        const wp<AudioFlinger::EffectChain>& chain,
6126                                        effect_descriptor_t *desc,
6127                                        int id,
6128                                        int sessionId)
6129    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6130      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6131{
6132    ALOGV("Constructor %p", this);
6133    int lStatus;
6134    sp<ThreadBase> thread = mThread.promote();
6135    if (thread == 0) {
6136        return;
6137    }
6138
6139    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6140
6141    // create effect engine from effect factory
6142    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6143
6144    if (mStatus != NO_ERROR) {
6145        return;
6146    }
6147    lStatus = init();
6148    if (lStatus < 0) {
6149        mStatus = lStatus;
6150        goto Error;
6151    }
6152
6153    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6154        mPinned = true;
6155    }
6156    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6157    return;
6158Error:
6159    EffectRelease(mEffectInterface);
6160    mEffectInterface = NULL;
6161    ALOGV("Constructor Error %d", mStatus);
6162}
6163
6164AudioFlinger::EffectModule::~EffectModule()
6165{
6166    ALOGV("Destructor %p", this);
6167    if (mEffectInterface != NULL) {
6168        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6169                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6170            sp<ThreadBase> thread = mThread.promote();
6171            if (thread != 0) {
6172                audio_stream_t *stream = thread->stream();
6173                if (stream != NULL) {
6174                    stream->remove_audio_effect(stream, mEffectInterface);
6175                }
6176            }
6177        }
6178        // release effect engine
6179        EffectRelease(mEffectInterface);
6180    }
6181}
6182
6183status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle)
6184{
6185    status_t status;
6186
6187    Mutex::Autolock _l(mLock);
6188    // First handle in mHandles has highest priority and controls the effect module
6189    int priority = handle->priority();
6190    size_t size = mHandles.size();
6191    sp<EffectHandle> h;
6192    size_t i;
6193    for (i = 0; i < size; i++) {
6194        h = mHandles[i].promote();
6195        if (h == 0) continue;
6196        if (h->priority() <= priority) break;
6197    }
6198    // if inserted in first place, move effect control from previous owner to this handle
6199    if (i == 0) {
6200        bool enabled = false;
6201        if (h != 0) {
6202            enabled = h->enabled();
6203            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6204        }
6205        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6206        status = NO_ERROR;
6207    } else {
6208        status = ALREADY_EXISTS;
6209    }
6210    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6211    mHandles.insertAt(handle, i);
6212    return status;
6213}
6214
6215size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6216{
6217    Mutex::Autolock _l(mLock);
6218    size_t size = mHandles.size();
6219    size_t i;
6220    for (i = 0; i < size; i++) {
6221        if (mHandles[i] == handle) break;
6222    }
6223    if (i == size) {
6224        return size;
6225    }
6226    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6227
6228    bool enabled = false;
6229    EffectHandle *hdl = handle.unsafe_get();
6230    if (hdl) {
6231        ALOGV("removeHandle() unsafe_get OK");
6232        enabled = hdl->enabled();
6233    }
6234    mHandles.removeAt(i);
6235    size = mHandles.size();
6236    // if removed from first place, move effect control from this handle to next in line
6237    if (i == 0 && size != 0) {
6238        sp<EffectHandle> h = mHandles[0].promote();
6239        if (h != 0) {
6240            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6241        }
6242    }
6243
6244    // Prevent calls to process() and other functions on effect interface from now on.
6245    // The effect engine will be released by the destructor when the last strong reference on
6246    // this object is released which can happen after next process is called.
6247    if (size == 0 && !mPinned) {
6248        mState = DESTROYED;
6249    }
6250
6251    return size;
6252}
6253
6254sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6255{
6256    Mutex::Autolock _l(mLock);
6257    sp<EffectHandle> handle;
6258    if (mHandles.size() != 0) {
6259        handle = mHandles[0].promote();
6260    }
6261    return handle;
6262}
6263
6264void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6265{
6266    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6267    // keep a strong reference on this EffectModule to avoid calling the
6268    // destructor before we exit
6269    sp<EffectModule> keep(this);
6270    {
6271        sp<ThreadBase> thread = mThread.promote();
6272        if (thread != 0) {
6273            thread->disconnectEffect(keep, handle, unpiniflast);
6274        }
6275    }
6276}
6277
6278void AudioFlinger::EffectModule::updateState() {
6279    Mutex::Autolock _l(mLock);
6280
6281    switch (mState) {
6282    case RESTART:
6283        reset_l();
6284        // FALL THROUGH
6285
6286    case STARTING:
6287        // clear auxiliary effect input buffer for next accumulation
6288        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6289            memset(mConfig.inputCfg.buffer.raw,
6290                   0,
6291                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6292        }
6293        start_l();
6294        mState = ACTIVE;
6295        break;
6296    case STOPPING:
6297        stop_l();
6298        mDisableWaitCnt = mMaxDisableWaitCnt;
6299        mState = STOPPED;
6300        break;
6301    case STOPPED:
6302        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6303        // turn off sequence.
6304        if (--mDisableWaitCnt == 0) {
6305            reset_l();
6306            mState = IDLE;
6307        }
6308        break;
6309    default: //IDLE , ACTIVE, DESTROYED
6310        break;
6311    }
6312}
6313
6314void AudioFlinger::EffectModule::process()
6315{
6316    Mutex::Autolock _l(mLock);
6317
6318    if (mState == DESTROYED || mEffectInterface == NULL ||
6319            mConfig.inputCfg.buffer.raw == NULL ||
6320            mConfig.outputCfg.buffer.raw == NULL) {
6321        return;
6322    }
6323
6324    if (isProcessEnabled()) {
6325        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6326        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6327            ditherAndClamp(mConfig.inputCfg.buffer.s32,
6328                                        mConfig.inputCfg.buffer.s32,
6329                                        mConfig.inputCfg.buffer.frameCount/2);
6330        }
6331
6332        // do the actual processing in the effect engine
6333        int ret = (*mEffectInterface)->process(mEffectInterface,
6334                                               &mConfig.inputCfg.buffer,
6335                                               &mConfig.outputCfg.buffer);
6336
6337        // force transition to IDLE state when engine is ready
6338        if (mState == STOPPED && ret == -ENODATA) {
6339            mDisableWaitCnt = 1;
6340        }
6341
6342        // clear auxiliary effect input buffer for next accumulation
6343        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6344            memset(mConfig.inputCfg.buffer.raw, 0,
6345                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6346        }
6347    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6348                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6349        // If an insert effect is idle and input buffer is different from output buffer,
6350        // accumulate input onto output
6351        sp<EffectChain> chain = mChain.promote();
6352        if (chain != 0 && chain->activeTrackCnt() != 0) {
6353            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6354            int16_t *in = mConfig.inputCfg.buffer.s16;
6355            int16_t *out = mConfig.outputCfg.buffer.s16;
6356            for (size_t i = 0; i < frameCnt; i++) {
6357                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6358            }
6359        }
6360    }
6361}
6362
6363void AudioFlinger::EffectModule::reset_l()
6364{
6365    if (mEffectInterface == NULL) {
6366        return;
6367    }
6368    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6369}
6370
6371status_t AudioFlinger::EffectModule::configure()
6372{
6373    uint32_t channels;
6374    if (mEffectInterface == NULL) {
6375        return NO_INIT;
6376    }
6377
6378    sp<ThreadBase> thread = mThread.promote();
6379    if (thread == 0) {
6380        return DEAD_OBJECT;
6381    }
6382
6383    // TODO: handle configuration of effects replacing track process
6384    if (thread->channelCount() == 1) {
6385        channels = AUDIO_CHANNEL_OUT_MONO;
6386    } else {
6387        channels = AUDIO_CHANNEL_OUT_STEREO;
6388    }
6389
6390    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6391        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6392    } else {
6393        mConfig.inputCfg.channels = channels;
6394    }
6395    mConfig.outputCfg.channels = channels;
6396    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6397    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6398    mConfig.inputCfg.samplingRate = thread->sampleRate();
6399    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6400    mConfig.inputCfg.bufferProvider.cookie = NULL;
6401    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6402    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6403    mConfig.outputCfg.bufferProvider.cookie = NULL;
6404    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6405    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6406    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6407    // Insert effect:
6408    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6409    // always overwrites output buffer: input buffer == output buffer
6410    // - in other sessions:
6411    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6412    //      other effect: overwrites output buffer: input buffer == output buffer
6413    // Auxiliary effect:
6414    //      accumulates in output buffer: input buffer != output buffer
6415    // Therefore: accumulate <=> input buffer != output buffer
6416    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6417        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6418    } else {
6419        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6420    }
6421    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6422    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6423    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6424    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6425
6426    ALOGV("configure() %p thread %p buffer %p framecount %d",
6427            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6428
6429    status_t cmdStatus;
6430    uint32_t size = sizeof(int);
6431    status_t status = (*mEffectInterface)->command(mEffectInterface,
6432                                                   EFFECT_CMD_SET_CONFIG,
6433                                                   sizeof(effect_config_t),
6434                                                   &mConfig,
6435                                                   &size,
6436                                                   &cmdStatus);
6437    if (status == 0) {
6438        status = cmdStatus;
6439    }
6440
6441    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6442            (1000 * mConfig.outputCfg.buffer.frameCount);
6443
6444    return status;
6445}
6446
6447status_t AudioFlinger::EffectModule::init()
6448{
6449    Mutex::Autolock _l(mLock);
6450    if (mEffectInterface == NULL) {
6451        return NO_INIT;
6452    }
6453    status_t cmdStatus;
6454    uint32_t size = sizeof(status_t);
6455    status_t status = (*mEffectInterface)->command(mEffectInterface,
6456                                                   EFFECT_CMD_INIT,
6457                                                   0,
6458                                                   NULL,
6459                                                   &size,
6460                                                   &cmdStatus);
6461    if (status == 0) {
6462        status = cmdStatus;
6463    }
6464    return status;
6465}
6466
6467status_t AudioFlinger::EffectModule::start()
6468{
6469    Mutex::Autolock _l(mLock);
6470    return start_l();
6471}
6472
6473status_t AudioFlinger::EffectModule::start_l()
6474{
6475    if (mEffectInterface == NULL) {
6476        return NO_INIT;
6477    }
6478    status_t cmdStatus;
6479    uint32_t size = sizeof(status_t);
6480    status_t status = (*mEffectInterface)->command(mEffectInterface,
6481                                                   EFFECT_CMD_ENABLE,
6482                                                   0,
6483                                                   NULL,
6484                                                   &size,
6485                                                   &cmdStatus);
6486    if (status == 0) {
6487        status = cmdStatus;
6488    }
6489    if (status == 0 &&
6490            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6491             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6492        sp<ThreadBase> thread = mThread.promote();
6493        if (thread != 0) {
6494            audio_stream_t *stream = thread->stream();
6495            if (stream != NULL) {
6496                stream->add_audio_effect(stream, mEffectInterface);
6497            }
6498        }
6499    }
6500    return status;
6501}
6502
6503status_t AudioFlinger::EffectModule::stop()
6504{
6505    Mutex::Autolock _l(mLock);
6506    return stop_l();
6507}
6508
6509status_t AudioFlinger::EffectModule::stop_l()
6510{
6511    if (mEffectInterface == NULL) {
6512        return NO_INIT;
6513    }
6514    status_t cmdStatus;
6515    uint32_t size = sizeof(status_t);
6516    status_t status = (*mEffectInterface)->command(mEffectInterface,
6517                                                   EFFECT_CMD_DISABLE,
6518                                                   0,
6519                                                   NULL,
6520                                                   &size,
6521                                                   &cmdStatus);
6522    if (status == 0) {
6523        status = cmdStatus;
6524    }
6525    if (status == 0 &&
6526            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6527             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6528        sp<ThreadBase> thread = mThread.promote();
6529        if (thread != 0) {
6530            audio_stream_t *stream = thread->stream();
6531            if (stream != NULL) {
6532                stream->remove_audio_effect(stream, mEffectInterface);
6533            }
6534        }
6535    }
6536    return status;
6537}
6538
6539status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6540                                             uint32_t cmdSize,
6541                                             void *pCmdData,
6542                                             uint32_t *replySize,
6543                                             void *pReplyData)
6544{
6545    Mutex::Autolock _l(mLock);
6546//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6547
6548    if (mState == DESTROYED || mEffectInterface == NULL) {
6549        return NO_INIT;
6550    }
6551    status_t status = (*mEffectInterface)->command(mEffectInterface,
6552                                                   cmdCode,
6553                                                   cmdSize,
6554                                                   pCmdData,
6555                                                   replySize,
6556                                                   pReplyData);
6557    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6558        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6559        for (size_t i = 1; i < mHandles.size(); i++) {
6560            sp<EffectHandle> h = mHandles[i].promote();
6561            if (h != 0) {
6562                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6563            }
6564        }
6565    }
6566    return status;
6567}
6568
6569status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6570{
6571
6572    Mutex::Autolock _l(mLock);
6573    ALOGV("setEnabled %p enabled %d", this, enabled);
6574
6575    if (enabled != isEnabled()) {
6576        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6577        if (enabled && status != NO_ERROR) {
6578            return status;
6579        }
6580
6581        switch (mState) {
6582        // going from disabled to enabled
6583        case IDLE:
6584            mState = STARTING;
6585            break;
6586        case STOPPED:
6587            mState = RESTART;
6588            break;
6589        case STOPPING:
6590            mState = ACTIVE;
6591            break;
6592
6593        // going from enabled to disabled
6594        case RESTART:
6595            mState = STOPPED;
6596            break;
6597        case STARTING:
6598            mState = IDLE;
6599            break;
6600        case ACTIVE:
6601            mState = STOPPING;
6602            break;
6603        case DESTROYED:
6604            return NO_ERROR; // simply ignore as we are being destroyed
6605        }
6606        for (size_t i = 1; i < mHandles.size(); i++) {
6607            sp<EffectHandle> h = mHandles[i].promote();
6608            if (h != 0) {
6609                h->setEnabled(enabled);
6610            }
6611        }
6612    }
6613    return NO_ERROR;
6614}
6615
6616bool AudioFlinger::EffectModule::isEnabled()
6617{
6618    switch (mState) {
6619    case RESTART:
6620    case STARTING:
6621    case ACTIVE:
6622        return true;
6623    case IDLE:
6624    case STOPPING:
6625    case STOPPED:
6626    case DESTROYED:
6627    default:
6628        return false;
6629    }
6630}
6631
6632bool AudioFlinger::EffectModule::isProcessEnabled()
6633{
6634    switch (mState) {
6635    case RESTART:
6636    case ACTIVE:
6637    case STOPPING:
6638    case STOPPED:
6639        return true;
6640    case IDLE:
6641    case STARTING:
6642    case DESTROYED:
6643    default:
6644        return false;
6645    }
6646}
6647
6648status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6649{
6650    Mutex::Autolock _l(mLock);
6651    status_t status = NO_ERROR;
6652
6653    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6654    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6655    if (isProcessEnabled() &&
6656            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6657            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6658        status_t cmdStatus;
6659        uint32_t volume[2];
6660        uint32_t *pVolume = NULL;
6661        uint32_t size = sizeof(volume);
6662        volume[0] = *left;
6663        volume[1] = *right;
6664        if (controller) {
6665            pVolume = volume;
6666        }
6667        status = (*mEffectInterface)->command(mEffectInterface,
6668                                              EFFECT_CMD_SET_VOLUME,
6669                                              size,
6670                                              volume,
6671                                              &size,
6672                                              pVolume);
6673        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6674            *left = volume[0];
6675            *right = volume[1];
6676        }
6677    }
6678    return status;
6679}
6680
6681status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6682{
6683    Mutex::Autolock _l(mLock);
6684    status_t status = NO_ERROR;
6685    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6686        // audio pre processing modules on RecordThread can receive both output and
6687        // input device indication in the same call
6688        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6689        if (dev) {
6690            status_t cmdStatus;
6691            uint32_t size = sizeof(status_t);
6692
6693            status = (*mEffectInterface)->command(mEffectInterface,
6694                                                  EFFECT_CMD_SET_DEVICE,
6695                                                  sizeof(uint32_t),
6696                                                  &dev,
6697                                                  &size,
6698                                                  &cmdStatus);
6699            if (status == NO_ERROR) {
6700                status = cmdStatus;
6701            }
6702        }
6703        dev = device & AUDIO_DEVICE_IN_ALL;
6704        if (dev) {
6705            status_t cmdStatus;
6706            uint32_t size = sizeof(status_t);
6707
6708            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6709                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6710                                                  sizeof(uint32_t),
6711                                                  &dev,
6712                                                  &size,
6713                                                  &cmdStatus);
6714            if (status2 == NO_ERROR) {
6715                status2 = cmdStatus;
6716            }
6717            if (status == NO_ERROR) {
6718                status = status2;
6719            }
6720        }
6721    }
6722    return status;
6723}
6724
6725status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
6726{
6727    Mutex::Autolock _l(mLock);
6728    status_t status = NO_ERROR;
6729    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6730        status_t cmdStatus;
6731        uint32_t size = sizeof(status_t);
6732        status = (*mEffectInterface)->command(mEffectInterface,
6733                                              EFFECT_CMD_SET_AUDIO_MODE,
6734                                              sizeof(audio_mode_t),
6735                                              &mode,
6736                                              &size,
6737                                              &cmdStatus);
6738        if (status == NO_ERROR) {
6739            status = cmdStatus;
6740        }
6741    }
6742    return status;
6743}
6744
6745void AudioFlinger::EffectModule::setSuspended(bool suspended)
6746{
6747    Mutex::Autolock _l(mLock);
6748    mSuspended = suspended;
6749}
6750
6751bool AudioFlinger::EffectModule::suspended() const
6752{
6753    Mutex::Autolock _l(mLock);
6754    return mSuspended;
6755}
6756
6757status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6758{
6759    const size_t SIZE = 256;
6760    char buffer[SIZE];
6761    String8 result;
6762
6763    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6764    result.append(buffer);
6765
6766    bool locked = tryLock(mLock);
6767    // failed to lock - AudioFlinger is probably deadlocked
6768    if (!locked) {
6769        result.append("\t\tCould not lock Fx mutex:\n");
6770    }
6771
6772    result.append("\t\tSession Status State Engine:\n");
6773    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6774            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6775    result.append(buffer);
6776
6777    result.append("\t\tDescriptor:\n");
6778    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6779            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6780            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6781            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6782    result.append(buffer);
6783    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6784                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6785                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6786                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6787    result.append(buffer);
6788    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6789            mDescriptor.apiVersion,
6790            mDescriptor.flags);
6791    result.append(buffer);
6792    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6793            mDescriptor.name);
6794    result.append(buffer);
6795    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6796            mDescriptor.implementor);
6797    result.append(buffer);
6798
6799    result.append("\t\t- Input configuration:\n");
6800    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6801    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6802            (uint32_t)mConfig.inputCfg.buffer.raw,
6803            mConfig.inputCfg.buffer.frameCount,
6804            mConfig.inputCfg.samplingRate,
6805            mConfig.inputCfg.channels,
6806            mConfig.inputCfg.format);
6807    result.append(buffer);
6808
6809    result.append("\t\t- Output configuration:\n");
6810    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6811    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6812            (uint32_t)mConfig.outputCfg.buffer.raw,
6813            mConfig.outputCfg.buffer.frameCount,
6814            mConfig.outputCfg.samplingRate,
6815            mConfig.outputCfg.channels,
6816            mConfig.outputCfg.format);
6817    result.append(buffer);
6818
6819    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6820    result.append(buffer);
6821    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6822    for (size_t i = 0; i < mHandles.size(); ++i) {
6823        sp<EffectHandle> handle = mHandles[i].promote();
6824        if (handle != 0) {
6825            handle->dump(buffer, SIZE);
6826            result.append(buffer);
6827        }
6828    }
6829
6830    result.append("\n");
6831
6832    write(fd, result.string(), result.length());
6833
6834    if (locked) {
6835        mLock.unlock();
6836    }
6837
6838    return NO_ERROR;
6839}
6840
6841// ----------------------------------------------------------------------------
6842//  EffectHandle implementation
6843// ----------------------------------------------------------------------------
6844
6845#undef LOG_TAG
6846#define LOG_TAG "AudioFlinger::EffectHandle"
6847
6848AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6849                                        const sp<AudioFlinger::Client>& client,
6850                                        const sp<IEffectClient>& effectClient,
6851                                        int32_t priority)
6852    : BnEffect(),
6853    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6854    mPriority(priority), mHasControl(false), mEnabled(false)
6855{
6856    ALOGV("constructor %p", this);
6857
6858    if (client == 0) {
6859        return;
6860    }
6861    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6862    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6863    if (mCblkMemory != 0) {
6864        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6865
6866        if (mCblk) {
6867            new(mCblk) effect_param_cblk_t();
6868            mBuffer = (uint8_t *)mCblk + bufOffset;
6869         }
6870    } else {
6871        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6872        return;
6873    }
6874}
6875
6876AudioFlinger::EffectHandle::~EffectHandle()
6877{
6878    ALOGV("Destructor %p", this);
6879    disconnect(false);
6880    ALOGV("Destructor DONE %p", this);
6881}
6882
6883status_t AudioFlinger::EffectHandle::enable()
6884{
6885    ALOGV("enable %p", this);
6886    if (!mHasControl) return INVALID_OPERATION;
6887    if (mEffect == 0) return DEAD_OBJECT;
6888
6889    if (mEnabled) {
6890        return NO_ERROR;
6891    }
6892
6893    mEnabled = true;
6894
6895    sp<ThreadBase> thread = mEffect->thread().promote();
6896    if (thread != 0) {
6897        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6898    }
6899
6900    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6901    if (mEffect->suspended()) {
6902        return NO_ERROR;
6903    }
6904
6905    status_t status = mEffect->setEnabled(true);
6906    if (status != NO_ERROR) {
6907        if (thread != 0) {
6908            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6909        }
6910        mEnabled = false;
6911    }
6912    return status;
6913}
6914
6915status_t AudioFlinger::EffectHandle::disable()
6916{
6917    ALOGV("disable %p", this);
6918    if (!mHasControl) return INVALID_OPERATION;
6919    if (mEffect == 0) return DEAD_OBJECT;
6920
6921    if (!mEnabled) {
6922        return NO_ERROR;
6923    }
6924    mEnabled = false;
6925
6926    if (mEffect->suspended()) {
6927        return NO_ERROR;
6928    }
6929
6930    status_t status = mEffect->setEnabled(false);
6931
6932    sp<ThreadBase> thread = mEffect->thread().promote();
6933    if (thread != 0) {
6934        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6935    }
6936
6937    return status;
6938}
6939
6940void AudioFlinger::EffectHandle::disconnect()
6941{
6942    disconnect(true);
6943}
6944
6945void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6946{
6947    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6948    if (mEffect == 0) {
6949        return;
6950    }
6951    mEffect->disconnect(this, unpiniflast);
6952
6953    if (mHasControl && mEnabled) {
6954        sp<ThreadBase> thread = mEffect->thread().promote();
6955        if (thread != 0) {
6956            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6957        }
6958    }
6959
6960    // release sp on module => module destructor can be called now
6961    mEffect.clear();
6962    if (mClient != 0) {
6963        if (mCblk) {
6964            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6965        }
6966        mCblkMemory.clear();            // and free the shared memory
6967        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6968        mClient.clear();
6969    }
6970}
6971
6972status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6973                                             uint32_t cmdSize,
6974                                             void *pCmdData,
6975                                             uint32_t *replySize,
6976                                             void *pReplyData)
6977{
6978//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6979//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6980
6981    // only get parameter command is permitted for applications not controlling the effect
6982    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6983        return INVALID_OPERATION;
6984    }
6985    if (mEffect == 0) return DEAD_OBJECT;
6986    if (mClient == 0) return INVALID_OPERATION;
6987
6988    // handle commands that are not forwarded transparently to effect engine
6989    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6990        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6991        // no risk to block the whole media server process or mixer threads is we are stuck here
6992        Mutex::Autolock _l(mCblk->lock);
6993        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6994            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6995            mCblk->serverIndex = 0;
6996            mCblk->clientIndex = 0;
6997            return BAD_VALUE;
6998        }
6999        status_t status = NO_ERROR;
7000        while (mCblk->serverIndex < mCblk->clientIndex) {
7001            int reply;
7002            uint32_t rsize = sizeof(int);
7003            int *p = (int *)(mBuffer + mCblk->serverIndex);
7004            int size = *p++;
7005            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
7006                ALOGW("command(): invalid parameter block size");
7007                break;
7008            }
7009            effect_param_t *param = (effect_param_t *)p;
7010            if (param->psize == 0 || param->vsize == 0) {
7011                ALOGW("command(): null parameter or value size");
7012                mCblk->serverIndex += size;
7013                continue;
7014            }
7015            uint32_t psize = sizeof(effect_param_t) +
7016                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
7017                             param->vsize;
7018            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
7019                                            psize,
7020                                            p,
7021                                            &rsize,
7022                                            &reply);
7023            // stop at first error encountered
7024            if (ret != NO_ERROR) {
7025                status = ret;
7026                *(int *)pReplyData = reply;
7027                break;
7028            } else if (reply != NO_ERROR) {
7029                *(int *)pReplyData = reply;
7030                break;
7031            }
7032            mCblk->serverIndex += size;
7033        }
7034        mCblk->serverIndex = 0;
7035        mCblk->clientIndex = 0;
7036        return status;
7037    } else if (cmdCode == EFFECT_CMD_ENABLE) {
7038        *(int *)pReplyData = NO_ERROR;
7039        return enable();
7040    } else if (cmdCode == EFFECT_CMD_DISABLE) {
7041        *(int *)pReplyData = NO_ERROR;
7042        return disable();
7043    }
7044
7045    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7046}
7047
7048sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
7049    return mCblkMemory;
7050}
7051
7052void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
7053{
7054    ALOGV("setControl %p control %d", this, hasControl);
7055
7056    mHasControl = hasControl;
7057    mEnabled = enabled;
7058
7059    if (signal && mEffectClient != 0) {
7060        mEffectClient->controlStatusChanged(hasControl);
7061    }
7062}
7063
7064void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7065                                                 uint32_t cmdSize,
7066                                                 void *pCmdData,
7067                                                 uint32_t replySize,
7068                                                 void *pReplyData)
7069{
7070    if (mEffectClient != 0) {
7071        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7072    }
7073}
7074
7075
7076
7077void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7078{
7079    if (mEffectClient != 0) {
7080        mEffectClient->enableStatusChanged(enabled);
7081    }
7082}
7083
7084status_t AudioFlinger::EffectHandle::onTransact(
7085    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7086{
7087    return BnEffect::onTransact(code, data, reply, flags);
7088}
7089
7090
7091void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7092{
7093    bool locked = mCblk ? tryLock(mCblk->lock) : false;
7094
7095    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7096            (mClient == NULL) ? getpid() : mClient->pid(),
7097            mPriority,
7098            mHasControl,
7099            !locked,
7100            mCblk ? mCblk->clientIndex : 0,
7101            mCblk ? mCblk->serverIndex : 0
7102            );
7103
7104    if (locked) {
7105        mCblk->lock.unlock();
7106    }
7107}
7108
7109#undef LOG_TAG
7110#define LOG_TAG "AudioFlinger::EffectChain"
7111
7112AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7113                                        int sessionId)
7114    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7115      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7116      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7117{
7118    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7119    sp<ThreadBase> thread = mThread.promote();
7120    if (thread == 0) {
7121        return;
7122    }
7123    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7124                                    thread->frameCount();
7125}
7126
7127AudioFlinger::EffectChain::~EffectChain()
7128{
7129    if (mOwnInBuffer) {
7130        delete mInBuffer;
7131    }
7132
7133}
7134
7135// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7136sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7137{
7138    sp<EffectModule> effect;
7139    size_t size = mEffects.size();
7140
7141    for (size_t i = 0; i < size; i++) {
7142        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7143            effect = mEffects[i];
7144            break;
7145        }
7146    }
7147    return effect;
7148}
7149
7150// getEffectFromId_l() must be called with ThreadBase::mLock held
7151sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7152{
7153    sp<EffectModule> effect;
7154    size_t size = mEffects.size();
7155
7156    for (size_t i = 0; i < size; i++) {
7157        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7158        if (id == 0 || mEffects[i]->id() == id) {
7159            effect = mEffects[i];
7160            break;
7161        }
7162    }
7163    return effect;
7164}
7165
7166// getEffectFromType_l() must be called with ThreadBase::mLock held
7167sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7168        const effect_uuid_t *type)
7169{
7170    sp<EffectModule> effect;
7171    size_t size = mEffects.size();
7172
7173    for (size_t i = 0; i < size; i++) {
7174        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7175            effect = mEffects[i];
7176            break;
7177        }
7178    }
7179    return effect;
7180}
7181
7182// Must be called with EffectChain::mLock locked
7183void AudioFlinger::EffectChain::process_l()
7184{
7185    sp<ThreadBase> thread = mThread.promote();
7186    if (thread == 0) {
7187        ALOGW("process_l(): cannot promote mixer thread");
7188        return;
7189    }
7190    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7191            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7192    // always process effects unless no more tracks are on the session and the effect tail
7193    // has been rendered
7194    bool doProcess = true;
7195    if (!isGlobalSession) {
7196        bool tracksOnSession = (trackCnt() != 0);
7197
7198        if (!tracksOnSession && mTailBufferCount == 0) {
7199            doProcess = false;
7200        }
7201
7202        if (activeTrackCnt() == 0) {
7203            // if no track is active and the effect tail has not been rendered,
7204            // the input buffer must be cleared here as the mixer process will not do it
7205            if (tracksOnSession || mTailBufferCount > 0) {
7206                size_t numSamples = thread->frameCount() * thread->channelCount();
7207                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7208                if (mTailBufferCount > 0) {
7209                    mTailBufferCount--;
7210                }
7211            }
7212        }
7213    }
7214
7215    size_t size = mEffects.size();
7216    if (doProcess) {
7217        for (size_t i = 0; i < size; i++) {
7218            mEffects[i]->process();
7219        }
7220    }
7221    for (size_t i = 0; i < size; i++) {
7222        mEffects[i]->updateState();
7223    }
7224}
7225
7226// addEffect_l() must be called with PlaybackThread::mLock held
7227status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7228{
7229    effect_descriptor_t desc = effect->desc();
7230    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7231
7232    Mutex::Autolock _l(mLock);
7233    effect->setChain(this);
7234    sp<ThreadBase> thread = mThread.promote();
7235    if (thread == 0) {
7236        return NO_INIT;
7237    }
7238    effect->setThread(thread);
7239
7240    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7241        // Auxiliary effects are inserted at the beginning of mEffects vector as
7242        // they are processed first and accumulated in chain input buffer
7243        mEffects.insertAt(effect, 0);
7244
7245        // the input buffer for auxiliary effect contains mono samples in
7246        // 32 bit format. This is to avoid saturation in AudoMixer
7247        // accumulation stage. Saturation is done in EffectModule::process() before
7248        // calling the process in effect engine
7249        size_t numSamples = thread->frameCount();
7250        int32_t *buffer = new int32_t[numSamples];
7251        memset(buffer, 0, numSamples * sizeof(int32_t));
7252        effect->setInBuffer((int16_t *)buffer);
7253        // auxiliary effects output samples to chain input buffer for further processing
7254        // by insert effects
7255        effect->setOutBuffer(mInBuffer);
7256    } else {
7257        // Insert effects are inserted at the end of mEffects vector as they are processed
7258        //  after track and auxiliary effects.
7259        // Insert effect order as a function of indicated preference:
7260        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7261        //  another effect is present
7262        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7263        //  last effect claiming first position
7264        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7265        //  first effect claiming last position
7266        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7267        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7268        // already present
7269
7270        int size = (int)mEffects.size();
7271        int idx_insert = size;
7272        int idx_insert_first = -1;
7273        int idx_insert_last = -1;
7274
7275        for (int i = 0; i < size; i++) {
7276            effect_descriptor_t d = mEffects[i]->desc();
7277            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7278            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7279            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7280                // check invalid effect chaining combinations
7281                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7282                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7283                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7284                    return INVALID_OPERATION;
7285                }
7286                // remember position of first insert effect and by default
7287                // select this as insert position for new effect
7288                if (idx_insert == size) {
7289                    idx_insert = i;
7290                }
7291                // remember position of last insert effect claiming
7292                // first position
7293                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7294                    idx_insert_first = i;
7295                }
7296                // remember position of first insert effect claiming
7297                // last position
7298                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7299                    idx_insert_last == -1) {
7300                    idx_insert_last = i;
7301                }
7302            }
7303        }
7304
7305        // modify idx_insert from first position if needed
7306        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7307            if (idx_insert_last != -1) {
7308                idx_insert = idx_insert_last;
7309            } else {
7310                idx_insert = size;
7311            }
7312        } else {
7313            if (idx_insert_first != -1) {
7314                idx_insert = idx_insert_first + 1;
7315            }
7316        }
7317
7318        // always read samples from chain input buffer
7319        effect->setInBuffer(mInBuffer);
7320
7321        // if last effect in the chain, output samples to chain
7322        // output buffer, otherwise to chain input buffer
7323        if (idx_insert == size) {
7324            if (idx_insert != 0) {
7325                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7326                mEffects[idx_insert-1]->configure();
7327            }
7328            effect->setOutBuffer(mOutBuffer);
7329        } else {
7330            effect->setOutBuffer(mInBuffer);
7331        }
7332        mEffects.insertAt(effect, idx_insert);
7333
7334        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7335    }
7336    effect->configure();
7337    return NO_ERROR;
7338}
7339
7340// removeEffect_l() must be called with PlaybackThread::mLock held
7341size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7342{
7343    Mutex::Autolock _l(mLock);
7344    int size = (int)mEffects.size();
7345    int i;
7346    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7347
7348    for (i = 0; i < size; i++) {
7349        if (effect == mEffects[i]) {
7350            // calling stop here will remove pre-processing effect from the audio HAL.
7351            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7352            // the middle of a read from audio HAL
7353            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7354                    mEffects[i]->state() == EffectModule::STOPPING) {
7355                mEffects[i]->stop();
7356            }
7357            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7358                delete[] effect->inBuffer();
7359            } else {
7360                if (i == size - 1 && i != 0) {
7361                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7362                    mEffects[i - 1]->configure();
7363                }
7364            }
7365            mEffects.removeAt(i);
7366            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7367            break;
7368        }
7369    }
7370
7371    return mEffects.size();
7372}
7373
7374// setDevice_l() must be called with PlaybackThread::mLock held
7375void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7376{
7377    size_t size = mEffects.size();
7378    for (size_t i = 0; i < size; i++) {
7379        mEffects[i]->setDevice(device);
7380    }
7381}
7382
7383// setMode_l() must be called with PlaybackThread::mLock held
7384void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode)
7385{
7386    size_t size = mEffects.size();
7387    for (size_t i = 0; i < size; i++) {
7388        mEffects[i]->setMode(mode);
7389    }
7390}
7391
7392// setVolume_l() must be called with PlaybackThread::mLock held
7393bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7394{
7395    uint32_t newLeft = *left;
7396    uint32_t newRight = *right;
7397    bool hasControl = false;
7398    int ctrlIdx = -1;
7399    size_t size = mEffects.size();
7400
7401    // first update volume controller
7402    for (size_t i = size; i > 0; i--) {
7403        if (mEffects[i - 1]->isProcessEnabled() &&
7404            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7405            ctrlIdx = i - 1;
7406            hasControl = true;
7407            break;
7408        }
7409    }
7410
7411    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7412        if (hasControl) {
7413            *left = mNewLeftVolume;
7414            *right = mNewRightVolume;
7415        }
7416        return hasControl;
7417    }
7418
7419    mVolumeCtrlIdx = ctrlIdx;
7420    mLeftVolume = newLeft;
7421    mRightVolume = newRight;
7422
7423    // second get volume update from volume controller
7424    if (ctrlIdx >= 0) {
7425        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7426        mNewLeftVolume = newLeft;
7427        mNewRightVolume = newRight;
7428    }
7429    // then indicate volume to all other effects in chain.
7430    // Pass altered volume to effects before volume controller
7431    // and requested volume to effects after controller
7432    uint32_t lVol = newLeft;
7433    uint32_t rVol = newRight;
7434
7435    for (size_t i = 0; i < size; i++) {
7436        if ((int)i == ctrlIdx) continue;
7437        // this also works for ctrlIdx == -1 when there is no volume controller
7438        if ((int)i > ctrlIdx) {
7439            lVol = *left;
7440            rVol = *right;
7441        }
7442        mEffects[i]->setVolume(&lVol, &rVol, false);
7443    }
7444    *left = newLeft;
7445    *right = newRight;
7446
7447    return hasControl;
7448}
7449
7450status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7451{
7452    const size_t SIZE = 256;
7453    char buffer[SIZE];
7454    String8 result;
7455
7456    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7457    result.append(buffer);
7458
7459    bool locked = tryLock(mLock);
7460    // failed to lock - AudioFlinger is probably deadlocked
7461    if (!locked) {
7462        result.append("\tCould not lock mutex:\n");
7463    }
7464
7465    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7466    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7467            mEffects.size(),
7468            (uint32_t)mInBuffer,
7469            (uint32_t)mOutBuffer,
7470            mActiveTrackCnt);
7471    result.append(buffer);
7472    write(fd, result.string(), result.size());
7473
7474    for (size_t i = 0; i < mEffects.size(); ++i) {
7475        sp<EffectModule> effect = mEffects[i];
7476        if (effect != 0) {
7477            effect->dump(fd, args);
7478        }
7479    }
7480
7481    if (locked) {
7482        mLock.unlock();
7483    }
7484
7485    return NO_ERROR;
7486}
7487
7488// must be called with ThreadBase::mLock held
7489void AudioFlinger::EffectChain::setEffectSuspended_l(
7490        const effect_uuid_t *type, bool suspend)
7491{
7492    sp<SuspendedEffectDesc> desc;
7493    // use effect type UUID timelow as key as there is no real risk of identical
7494    // timeLow fields among effect type UUIDs.
7495    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7496    if (suspend) {
7497        if (index >= 0) {
7498            desc = mSuspendedEffects.valueAt(index);
7499        } else {
7500            desc = new SuspendedEffectDesc();
7501            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7502            mSuspendedEffects.add(type->timeLow, desc);
7503            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7504        }
7505        if (desc->mRefCount++ == 0) {
7506            sp<EffectModule> effect = getEffectIfEnabled(type);
7507            if (effect != 0) {
7508                desc->mEffect = effect;
7509                effect->setSuspended(true);
7510                effect->setEnabled(false);
7511            }
7512        }
7513    } else {
7514        if (index < 0) {
7515            return;
7516        }
7517        desc = mSuspendedEffects.valueAt(index);
7518        if (desc->mRefCount <= 0) {
7519            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7520            desc->mRefCount = 1;
7521        }
7522        if (--desc->mRefCount == 0) {
7523            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7524            if (desc->mEffect != 0) {
7525                sp<EffectModule> effect = desc->mEffect.promote();
7526                if (effect != 0) {
7527                    effect->setSuspended(false);
7528                    sp<EffectHandle> handle = effect->controlHandle();
7529                    if (handle != 0) {
7530                        effect->setEnabled(handle->enabled());
7531                    }
7532                }
7533                desc->mEffect.clear();
7534            }
7535            mSuspendedEffects.removeItemsAt(index);
7536        }
7537    }
7538}
7539
7540// must be called with ThreadBase::mLock held
7541void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7542{
7543    sp<SuspendedEffectDesc> desc;
7544
7545    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7546    if (suspend) {
7547        if (index >= 0) {
7548            desc = mSuspendedEffects.valueAt(index);
7549        } else {
7550            desc = new SuspendedEffectDesc();
7551            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7552            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7553        }
7554        if (desc->mRefCount++ == 0) {
7555            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7556            for (size_t i = 0; i < effects.size(); i++) {
7557                setEffectSuspended_l(&effects[i]->desc().type, true);
7558            }
7559        }
7560    } else {
7561        if (index < 0) {
7562            return;
7563        }
7564        desc = mSuspendedEffects.valueAt(index);
7565        if (desc->mRefCount <= 0) {
7566            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7567            desc->mRefCount = 1;
7568        }
7569        if (--desc->mRefCount == 0) {
7570            Vector<const effect_uuid_t *> types;
7571            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7572                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7573                    continue;
7574                }
7575                types.add(&mSuspendedEffects.valueAt(i)->mType);
7576            }
7577            for (size_t i = 0; i < types.size(); i++) {
7578                setEffectSuspended_l(types[i], false);
7579            }
7580            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7581            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7582        }
7583    }
7584}
7585
7586
7587// The volume effect is used for automated tests only
7588#ifndef OPENSL_ES_H_
7589static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7590                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7591const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7592#endif //OPENSL_ES_H_
7593
7594bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7595{
7596    // auxiliary effects and visualizer are never suspended on output mix
7597    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7598        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7599         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7600         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7601        return false;
7602    }
7603    return true;
7604}
7605
7606Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7607{
7608    Vector< sp<EffectModule> > effects;
7609    for (size_t i = 0; i < mEffects.size(); i++) {
7610        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7611            continue;
7612        }
7613        effects.add(mEffects[i]);
7614    }
7615    return effects;
7616}
7617
7618sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7619                                                            const effect_uuid_t *type)
7620{
7621    sp<EffectModule> effect;
7622    effect = getEffectFromType_l(type);
7623    if (effect != 0 && !effect->isEnabled()) {
7624        effect.clear();
7625    }
7626    return effect;
7627}
7628
7629void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7630                                                            bool enabled)
7631{
7632    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7633    if (enabled) {
7634        if (index < 0) {
7635            // if the effect is not suspend check if all effects are suspended
7636            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7637            if (index < 0) {
7638                return;
7639            }
7640            if (!isEffectEligibleForSuspend(effect->desc())) {
7641                return;
7642            }
7643            setEffectSuspended_l(&effect->desc().type, enabled);
7644            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7645            if (index < 0) {
7646                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7647                return;
7648            }
7649        }
7650        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7651             effect->desc().type.timeLow);
7652        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7653        // if effect is requested to suspended but was not yet enabled, supend it now.
7654        if (desc->mEffect == 0) {
7655            desc->mEffect = effect;
7656            effect->setEnabled(false);
7657            effect->setSuspended(true);
7658        }
7659    } else {
7660        if (index < 0) {
7661            return;
7662        }
7663        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7664             effect->desc().type.timeLow);
7665        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7666        desc->mEffect.clear();
7667        effect->setSuspended(false);
7668    }
7669}
7670
7671#undef LOG_TAG
7672#define LOG_TAG "AudioFlinger"
7673
7674// ----------------------------------------------------------------------------
7675
7676status_t AudioFlinger::onTransact(
7677        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7678{
7679    return BnAudioFlinger::onTransact(code, data, reply, flags);
7680}
7681
7682}; // namespace android
7683