AudioFlinger.cpp revision e616d4e6de6d53ddebbc3d7fb381af94589c2232 
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),
164        mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef()
165        mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
166        mMode(AUDIO_MODE_INVALID),
167        mBtNrecIsOff(false)
168{
169}
170
171void AudioFlinger::onFirstRef()
172{
173    int rc = 0;
174
175    Mutex::Autolock _l(mLock);
176
177    /* TODO: move all this work into an Init() function */
178
179    for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
180        const hw_module_t *mod;
181        audio_hw_device_t *dev;
182
183        rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
184        if (rc)
185            continue;
186
187        ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
188             mod->name, mod->id);
189        mAudioHwDevs.push(dev);
190
191        if (!mPrimaryHardwareDev) {
192            mPrimaryHardwareDev = dev;
193            ALOGI("Using '%s' (%s.%s) as the primary audio interface",
194                 mod->name, mod->id, audio_interfaces[i]);
195        }
196    }
197
198    mHardwareStatus = AUDIO_HW_INIT;
199
200    if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
201        ALOGE("Primary audio interface not found");
202        return;
203    }
204
205    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
206        audio_hw_device_t *dev = mAudioHwDevs[i];
207
208        mHardwareStatus = AUDIO_HW_INIT;
209        rc = dev->init_check(dev);
210        if (rc == 0) {
211            AutoMutex lock(mHardwareLock);
212
213            mMode = AUDIO_MODE_NORMAL;
214            mHardwareStatus = AUDIO_HW_SET_MODE;
215            dev->set_mode(dev, mMode);
216            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
217            dev->set_master_volume(dev, 1.0f);
218            mHardwareStatus = AUDIO_HW_IDLE;
219        }
220    }
221}
222
223status_t AudioFlinger::initCheck() const
224{
225    Mutex::Autolock _l(mLock);
226    if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
227        return NO_INIT;
228    return NO_ERROR;
229}
230
231AudioFlinger::~AudioFlinger()
232{
233    int num_devs = mAudioHwDevs.size();
234
235    while (!mRecordThreads.isEmpty()) {
236        // closeInput() will remove first entry from mRecordThreads
237        closeInput(mRecordThreads.keyAt(0));
238    }
239    while (!mPlaybackThreads.isEmpty()) {
240        // closeOutput() will remove first entry from mPlaybackThreads
241        closeOutput(mPlaybackThreads.keyAt(0));
242    }
243
244    for (int i = 0; i < num_devs; i++) {
245        audio_hw_device_t *dev = mAudioHwDevs[i];
246        audio_hw_device_close(dev);
247    }
248    mAudioHwDevs.clear();
249}
250
251audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
252{
253    /* first matching HW device is returned */
254    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
255        audio_hw_device_t *dev = mAudioHwDevs[i];
256        if ((dev->get_supported_devices(dev) & devices) == devices)
257            return dev;
258    }
259    return NULL;
260}
261
262status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
263{
264    const size_t SIZE = 256;
265    char buffer[SIZE];
266    String8 result;
267
268    result.append("Clients:\n");
269    for (size_t i = 0; i < mClients.size(); ++i) {
270        sp<Client> client = mClients.valueAt(i).promote();
271        if (client != 0) {
272            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
273            result.append(buffer);
274        }
275    }
276
277    result.append("Global session refs:\n");
278    result.append(" session pid cnt\n");
279    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
280        AudioSessionRef *r = mAudioSessionRefs[i];
281        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
282        result.append(buffer);
283    }
284    write(fd, result.string(), result.size());
285    return NO_ERROR;
286}
287
288
289status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
290{
291    const size_t SIZE = 256;
292    char buffer[SIZE];
293    String8 result;
294    hardware_call_state hardwareStatus = mHardwareStatus;
295
296    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
297    result.append(buffer);
298    write(fd, result.string(), result.size());
299    return NO_ERROR;
300}
301
302status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
303{
304    const size_t SIZE = 256;
305    char buffer[SIZE];
306    String8 result;
307    snprintf(buffer, SIZE, "Permission Denial: "
308            "can't dump AudioFlinger from pid=%d, uid=%d\n",
309            IPCThreadState::self()->getCallingPid(),
310            IPCThreadState::self()->getCallingUid());
311    result.append(buffer);
312    write(fd, result.string(), result.size());
313    return NO_ERROR;
314}
315
316static bool tryLock(Mutex& mutex)
317{
318    bool locked = false;
319    for (int i = 0; i < kDumpLockRetries; ++i) {
320        if (mutex.tryLock() == NO_ERROR) {
321            locked = true;
322            break;
323        }
324        usleep(kDumpLockSleepUs);
325    }
326    return locked;
327}
328
329status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
330{
331    if (!checkCallingPermission(String16("android.permission.DUMP"))) {
332        dumpPermissionDenial(fd, args);
333    } else {
334        // get state of hardware lock
335        bool hardwareLocked = tryLock(mHardwareLock);
336        if (!hardwareLocked) {
337            String8 result(kHardwareLockedString);
338            write(fd, result.string(), result.size());
339        } else {
340            mHardwareLock.unlock();
341        }
342
343        bool locked = tryLock(mLock);
344
345        // failed to lock - AudioFlinger is probably deadlocked
346        if (!locked) {
347            String8 result(kDeadlockedString);
348            write(fd, result.string(), result.size());
349        }
350
351        dumpClients(fd, args);
352        dumpInternals(fd, args);
353
354        // dump playback threads
355        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
356            mPlaybackThreads.valueAt(i)->dump(fd, args);
357        }
358
359        // dump record threads
360        for (size_t i = 0; i < mRecordThreads.size(); i++) {
361            mRecordThreads.valueAt(i)->dump(fd, args);
362        }
363
364        // dump all hardware devs
365        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
366            audio_hw_device_t *dev = mAudioHwDevs[i];
367            dev->dump(dev, fd);
368        }
369        if (locked) mLock.unlock();
370    }
371    return NO_ERROR;
372}
373
374
375// IAudioFlinger interface
376
377
378sp<IAudioTrack> AudioFlinger::createTrack(
379        pid_t pid,
380        audio_stream_type_t streamType,
381        uint32_t sampleRate,
382        audio_format_t format,
383        uint32_t channelMask,
384        int frameCount,
385        uint32_t flags,
386        const sp<IMemory>& sharedBuffer,
387        int output,
388        int *sessionId,
389        status_t *status)
390{
391    sp<PlaybackThread::Track> track;
392    sp<TrackHandle> trackHandle;
393    sp<Client> client;
394    wp<Client> wclient;
395    status_t lStatus;
396    int lSessionId;
397
398    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
399    // but if someone uses binder directly they could bypass that and cause us to crash
400    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
401        ALOGE("createTrack() invalid stream type %d", streamType);
402        lStatus = BAD_VALUE;
403        goto Exit;
404    }
405
406    {
407        Mutex::Autolock _l(mLock);
408        PlaybackThread *thread = checkPlaybackThread_l(output);
409        PlaybackThread *effectThread = NULL;
410        if (thread == NULL) {
411            ALOGE("unknown output thread");
412            lStatus = BAD_VALUE;
413            goto Exit;
414        }
415
416        wclient = mClients.valueFor(pid);
417
418        if (wclient != NULL) {
419            client = wclient.promote();
420        } else {
421            client = new Client(this, pid);
422            mClients.add(pid, client);
423        }
424
425        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
426        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
427            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
428                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
429                if (mPlaybackThreads.keyAt(i) != output) {
430                    // prevent same audio session on different output threads
431                    uint32_t sessions = t->hasAudioSession(*sessionId);
432                    if (sessions & PlaybackThread::TRACK_SESSION) {
433                        ALOGE("createTrack() session ID %d already in use", *sessionId);
434                        lStatus = BAD_VALUE;
435                        goto Exit;
436                    }
437                    // check if an effect with same session ID is waiting for a track to be created
438                    if (sessions & PlaybackThread::EFFECT_SESSION) {
439                        effectThread = t.get();
440                    }
441                }
442            }
443            lSessionId = *sessionId;
444        } else {
445            // if no audio session id is provided, create one here
446            lSessionId = nextUniqueId();
447            if (sessionId != NULL) {
448                *sessionId = lSessionId;
449            }
450        }
451        ALOGV("createTrack() lSessionId: %d", lSessionId);
452
453        track = thread->createTrack_l(client, streamType, sampleRate, format,
454                channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
455
456        // move effect chain to this output thread if an effect on same session was waiting
457        // for a track to be created
458        if (lStatus == NO_ERROR && effectThread != NULL) {
459            Mutex::Autolock _dl(thread->mLock);
460            Mutex::Autolock _sl(effectThread->mLock);
461            moveEffectChain_l(lSessionId, effectThread, thread, true);
462        }
463    }
464    if (lStatus == NO_ERROR) {
465        trackHandle = new TrackHandle(track);
466    } else {
467        // remove local strong reference to Client before deleting the Track so that the Client
468        // destructor is called by the TrackBase destructor with mLock held
469        client.clear();
470        track.clear();
471    }
472
473Exit:
474    if(status) {
475        *status = lStatus;
476    }
477    return trackHandle;
478}
479
480uint32_t AudioFlinger::sampleRate(int output) const
481{
482    Mutex::Autolock _l(mLock);
483    PlaybackThread *thread = checkPlaybackThread_l(output);
484    if (thread == NULL) {
485        ALOGW("sampleRate() unknown thread %d", output);
486        return 0;
487    }
488    return thread->sampleRate();
489}
490
491int AudioFlinger::channelCount(int output) const
492{
493    Mutex::Autolock _l(mLock);
494    PlaybackThread *thread = checkPlaybackThread_l(output);
495    if (thread == NULL) {
496        ALOGW("channelCount() unknown thread %d", output);
497        return 0;
498    }
499    return thread->channelCount();
500}
501
502audio_format_t AudioFlinger::format(int output) const
503{
504    Mutex::Autolock _l(mLock);
505    PlaybackThread *thread = checkPlaybackThread_l(output);
506    if (thread == NULL) {
507        ALOGW("format() unknown thread %d", output);
508        return AUDIO_FORMAT_INVALID;
509    }
510    return thread->format();
511}
512
513size_t AudioFlinger::frameCount(int output) const
514{
515    Mutex::Autolock _l(mLock);
516    PlaybackThread *thread = checkPlaybackThread_l(output);
517    if (thread == NULL) {
518        ALOGW("frameCount() unknown thread %d", output);
519        return 0;
520    }
521    return thread->frameCount();
522}
523
524uint32_t AudioFlinger::latency(int output) const
525{
526    Mutex::Autolock _l(mLock);
527    PlaybackThread *thread = checkPlaybackThread_l(output);
528    if (thread == NULL) {
529        ALOGW("latency() unknown thread %d", output);
530        return 0;
531    }
532    return thread->latency();
533}
534
535status_t AudioFlinger::setMasterVolume(float value)
536{
537    status_t ret = initCheck();
538    if (ret != NO_ERROR) {
539        return ret;
540    }
541
542    // check calling permissions
543    if (!settingsAllowed()) {
544        return PERMISSION_DENIED;
545    }
546
547    // when hw supports master volume, don't scale in sw mixer
548    { // scope for the lock
549        AutoMutex lock(mHardwareLock);
550        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
551        if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
552            value = 1.0f;
553        }
554        mHardwareStatus = AUDIO_HW_IDLE;
555    }
556
557    Mutex::Autolock _l(mLock);
558    mMasterVolume = value;
559    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
560       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
561
562    return NO_ERROR;
563}
564
565status_t AudioFlinger::setMode(audio_mode_t mode)
566{
567    status_t ret = initCheck();
568    if (ret != NO_ERROR) {
569        return ret;
570    }
571
572    // check calling permissions
573    if (!settingsAllowed()) {
574        return PERMISSION_DENIED;
575    }
576    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
577        ALOGW("Illegal value: setMode(%d)", mode);
578        return BAD_VALUE;
579    }
580
581    { // scope for the lock
582        AutoMutex lock(mHardwareLock);
583        mHardwareStatus = AUDIO_HW_SET_MODE;
584        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
585        mHardwareStatus = AUDIO_HW_IDLE;
586    }
587
588    if (NO_ERROR == ret) {
589        Mutex::Autolock _l(mLock);
590        mMode = mode;
591        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
592           mPlaybackThreads.valueAt(i)->setMode(mode);
593    }
594
595    return ret;
596}
597
598status_t AudioFlinger::setMicMute(bool state)
599{
600    status_t ret = initCheck();
601    if (ret != NO_ERROR) {
602        return ret;
603    }
604
605    // check calling permissions
606    if (!settingsAllowed()) {
607        return PERMISSION_DENIED;
608    }
609
610    AutoMutex lock(mHardwareLock);
611    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
612    ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
613    mHardwareStatus = AUDIO_HW_IDLE;
614    return ret;
615}
616
617bool AudioFlinger::getMicMute() const
618{
619    status_t ret = initCheck();
620    if (ret != NO_ERROR) {
621        return false;
622    }
623
624    bool state = AUDIO_MODE_INVALID;
625    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
626    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
627    mHardwareStatus = AUDIO_HW_IDLE;
628    return state;
629}
630
631status_t AudioFlinger::setMasterMute(bool muted)
632{
633    // check calling permissions
634    if (!settingsAllowed()) {
635        return PERMISSION_DENIED;
636    }
637
638    Mutex::Autolock _l(mLock);
639    mMasterMute = muted;
640    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
641       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
642
643    return NO_ERROR;
644}
645
646float AudioFlinger::masterVolume() const
647{
648    Mutex::Autolock _l(mLock);
649    return masterVolume_l();
650}
651
652bool AudioFlinger::masterMute() const
653{
654    Mutex::Autolock _l(mLock);
655    return masterMute_l();
656}
657
658status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output)
659{
660    // check calling permissions
661    if (!settingsAllowed()) {
662        return PERMISSION_DENIED;
663    }
664
665    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
666        ALOGE("setStreamVolume() invalid stream %d", stream);
667        return BAD_VALUE;
668    }
669
670    AutoMutex lock(mLock);
671    PlaybackThread *thread = NULL;
672    if (output) {
673        thread = checkPlaybackThread_l(output);
674        if (thread == NULL) {
675            return BAD_VALUE;
676        }
677    }
678
679    mStreamTypes[stream].volume = value;
680
681    if (thread == NULL) {
682        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
683           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
684        }
685    } else {
686        thread->setStreamVolume(stream, value);
687    }
688
689    return NO_ERROR;
690}
691
692status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
693{
694    // check calling permissions
695    if (!settingsAllowed()) {
696        return PERMISSION_DENIED;
697    }
698
699    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
700        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
701        ALOGE("setStreamMute() invalid stream %d", stream);
702        return BAD_VALUE;
703    }
704
705    AutoMutex lock(mLock);
706    mStreamTypes[stream].mute = muted;
707    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
708       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
709
710    return NO_ERROR;
711}
712
713float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const
714{
715    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
716        return 0.0f;
717    }
718
719    AutoMutex lock(mLock);
720    float volume;
721    if (output) {
722        PlaybackThread *thread = checkPlaybackThread_l(output);
723        if (thread == NULL) {
724            return 0.0f;
725        }
726        volume = thread->streamVolume(stream);
727    } else {
728        volume = mStreamTypes[stream].volume;
729    }
730
731    return volume;
732}
733
734bool AudioFlinger::streamMute(audio_stream_type_t stream) const
735{
736    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
737        return true;
738    }
739
740    return mStreamTypes[stream].mute;
741}
742
743status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
744{
745    status_t result;
746
747    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
748            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
749    // check calling permissions
750    if (!settingsAllowed()) {
751        return PERMISSION_DENIED;
752    }
753
754    // ioHandle == 0 means the parameters are global to the audio hardware interface
755    if (ioHandle == 0) {
756        AutoMutex lock(mHardwareLock);
757        mHardwareStatus = AUDIO_SET_PARAMETER;
758        status_t final_result = NO_ERROR;
759        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
760            audio_hw_device_t *dev = mAudioHwDevs[i];
761            result = dev->set_parameters(dev, keyValuePairs.string());
762            final_result = result ?: final_result;
763        }
764        mHardwareStatus = AUDIO_HW_IDLE;
765        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
766        AudioParameter param = AudioParameter(keyValuePairs);
767        String8 value;
768        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
769            Mutex::Autolock _l(mLock);
770            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
771            if (mBtNrecIsOff != btNrecIsOff) {
772                for (size_t i = 0; i < mRecordThreads.size(); i++) {
773                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
774                    RecordThread::RecordTrack *track = thread->track();
775                    if (track != NULL) {
776                        audio_devices_t device = (audio_devices_t)(
777                                thread->device() & AUDIO_DEVICE_IN_ALL);
778                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
779                        thread->setEffectSuspended(FX_IID_AEC,
780                                                   suspend,
781                                                   track->sessionId());
782                        thread->setEffectSuspended(FX_IID_NS,
783                                                   suspend,
784                                                   track->sessionId());
785                    }
786                }
787                mBtNrecIsOff = btNrecIsOff;
788            }
789        }
790        return final_result;
791    }
792
793    // hold a strong ref on thread in case closeOutput() or closeInput() is called
794    // and the thread is exited once the lock is released
795    sp<ThreadBase> thread;
796    {
797        Mutex::Autolock _l(mLock);
798        thread = checkPlaybackThread_l(ioHandle);
799        if (thread == NULL) {
800            thread = checkRecordThread_l(ioHandle);
801        } else if (thread == primaryPlaybackThread_l()) {
802            // indicate output device change to all input threads for pre processing
803            AudioParameter param = AudioParameter(keyValuePairs);
804            int value;
805            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
806                for (size_t i = 0; i < mRecordThreads.size(); i++) {
807                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
808                }
809            }
810        }
811    }
812    if (thread != 0) {
813        return thread->setParameters(keyValuePairs);
814    }
815    return BAD_VALUE;
816}
817
818String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) const
819{
820//    ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
821//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
822
823    if (ioHandle == 0) {
824        String8 out_s8;
825
826        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
827            audio_hw_device_t *dev = mAudioHwDevs[i];
828            char *s = dev->get_parameters(dev, keys.string());
829            out_s8 += String8(s);
830            free(s);
831        }
832        return out_s8;
833    }
834
835    Mutex::Autolock _l(mLock);
836
837    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
838    if (playbackThread != NULL) {
839        return playbackThread->getParameters(keys);
840    }
841    RecordThread *recordThread = checkRecordThread_l(ioHandle);
842    if (recordThread != NULL) {
843        return recordThread->getParameters(keys);
844    }
845    return String8("");
846}
847
848size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
849{
850    status_t ret = initCheck();
851    if (ret != NO_ERROR) {
852        return 0;
853    }
854
855    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
856}
857
858unsigned int AudioFlinger::getInputFramesLost(int ioHandle) const
859{
860    if (ioHandle == 0) {
861        return 0;
862    }
863
864    Mutex::Autolock _l(mLock);
865
866    RecordThread *recordThread = checkRecordThread_l(ioHandle);
867    if (recordThread != NULL) {
868        return recordThread->getInputFramesLost();
869    }
870    return 0;
871}
872
873status_t AudioFlinger::setVoiceVolume(float value)
874{
875    status_t ret = initCheck();
876    if (ret != NO_ERROR) {
877        return ret;
878    }
879
880    // check calling permissions
881    if (!settingsAllowed()) {
882        return PERMISSION_DENIED;
883    }
884
885    AutoMutex lock(mHardwareLock);
886    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
887    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
888    mHardwareStatus = AUDIO_HW_IDLE;
889
890    return ret;
891}
892
893status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) const
894{
895    status_t status;
896
897    Mutex::Autolock _l(mLock);
898
899    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
900    if (playbackThread != NULL) {
901        return playbackThread->getRenderPosition(halFrames, dspFrames);
902    }
903
904    return BAD_VALUE;
905}
906
907void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
908{
909
910    Mutex::Autolock _l(mLock);
911
912    int pid = IPCThreadState::self()->getCallingPid();
913    if (mNotificationClients.indexOfKey(pid) < 0) {
914        sp<NotificationClient> notificationClient = new NotificationClient(this,
915                                                                            client,
916                                                                            pid);
917        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
918
919        mNotificationClients.add(pid, notificationClient);
920
921        sp<IBinder> binder = client->asBinder();
922        binder->linkToDeath(notificationClient);
923
924        // the config change is always sent from playback or record threads to avoid deadlock
925        // with AudioSystem::gLock
926        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
927            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
928        }
929
930        for (size_t i = 0; i < mRecordThreads.size(); i++) {
931            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
932        }
933    }
934}
935
936void AudioFlinger::removeNotificationClient(pid_t pid)
937{
938    Mutex::Autolock _l(mLock);
939
940    int index = mNotificationClients.indexOfKey(pid);
941    if (index >= 0) {
942        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
943        ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
944        mNotificationClients.removeItem(pid);
945    }
946
947    ALOGV("%d died, releasing its sessions", pid);
948    int num = mAudioSessionRefs.size();
949    bool removed = false;
950    for (int i = 0; i< num; i++) {
951        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
952        ALOGV(" pid %d @ %d", ref->pid, i);
953        if (ref->pid == pid) {
954            ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
955            mAudioSessionRefs.removeAt(i);
956            delete ref;
957            removed = true;
958            i--;
959            num--;
960        }
961    }
962    if (removed) {
963        purgeStaleEffects_l();
964    }
965}
966
967// audioConfigChanged_l() must be called with AudioFlinger::mLock held
968void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
969{
970    size_t size = mNotificationClients.size();
971    for (size_t i = 0; i < size; i++) {
972        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
973                                                                               param2);
974    }
975}
976
977// removeClient_l() must be called with AudioFlinger::mLock held
978void AudioFlinger::removeClient_l(pid_t pid)
979{
980    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
981    mClients.removeItem(pid);
982}
983
984
985// ----------------------------------------------------------------------------
986
987AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device,
988        type_t type)
989    :   Thread(false),
990        mType(type),
991        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0),
992        // mChannelMask
993        mChannelCount(0),
994        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
995        mParamStatus(NO_ERROR),
996        mStandby(false), mId(id), mExiting(false),
997        mDevice(device),
998        mDeathRecipient(new PMDeathRecipient(this))
999{
1000}
1001
1002AudioFlinger::ThreadBase::~ThreadBase()
1003{
1004    mParamCond.broadcast();
1005    // do not lock the mutex in destructor
1006    releaseWakeLock_l();
1007    if (mPowerManager != 0) {
1008        sp<IBinder> binder = mPowerManager->asBinder();
1009        binder->unlinkToDeath(mDeathRecipient);
1010    }
1011}
1012
1013void AudioFlinger::ThreadBase::exit()
1014{
1015    // keep a strong ref on ourself so that we won't get
1016    // destroyed in the middle of requestExitAndWait()
1017    sp <ThreadBase> strongMe = this;
1018
1019    ALOGV("ThreadBase::exit");
1020    {
1021        AutoMutex lock(mLock);
1022        mExiting = true;
1023        requestExit();
1024        mWaitWorkCV.signal();
1025    }
1026    requestExitAndWait();
1027}
1028
1029uint32_t AudioFlinger::ThreadBase::sampleRate() const
1030{
1031    return mSampleRate;
1032}
1033
1034int AudioFlinger::ThreadBase::channelCount() const
1035{
1036    return (int)mChannelCount;
1037}
1038
1039audio_format_t AudioFlinger::ThreadBase::format() const
1040{
1041    return mFormat;
1042}
1043
1044size_t AudioFlinger::ThreadBase::frameCount() const
1045{
1046    return mFrameCount;
1047}
1048
1049status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1050{
1051    status_t status;
1052
1053    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1054    Mutex::Autolock _l(mLock);
1055
1056    mNewParameters.add(keyValuePairs);
1057    mWaitWorkCV.signal();
1058    // wait condition with timeout in case the thread loop has exited
1059    // before the request could be processed
1060    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
1061        status = mParamStatus;
1062        mWaitWorkCV.signal();
1063    } else {
1064        status = TIMED_OUT;
1065    }
1066    return status;
1067}
1068
1069void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1070{
1071    Mutex::Autolock _l(mLock);
1072    sendConfigEvent_l(event, param);
1073}
1074
1075// sendConfigEvent_l() must be called with ThreadBase::mLock held
1076void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1077{
1078    ConfigEvent configEvent;
1079    configEvent.mEvent = event;
1080    configEvent.mParam = param;
1081    mConfigEvents.add(configEvent);
1082    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1083    mWaitWorkCV.signal();
1084}
1085
1086void AudioFlinger::ThreadBase::processConfigEvents()
1087{
1088    mLock.lock();
1089    while(!mConfigEvents.isEmpty()) {
1090        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1091        ConfigEvent configEvent = mConfigEvents[0];
1092        mConfigEvents.removeAt(0);
1093        // release mLock before locking AudioFlinger mLock: lock order is always
1094        // AudioFlinger then ThreadBase to avoid cross deadlock
1095        mLock.unlock();
1096        mAudioFlinger->mLock.lock();
1097        audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
1098        mAudioFlinger->mLock.unlock();
1099        mLock.lock();
1100    }
1101    mLock.unlock();
1102}
1103
1104status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1105{
1106    const size_t SIZE = 256;
1107    char buffer[SIZE];
1108    String8 result;
1109
1110    bool locked = tryLock(mLock);
1111    if (!locked) {
1112        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1113        write(fd, buffer, strlen(buffer));
1114    }
1115
1116    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1117    result.append(buffer);
1118    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1119    result.append(buffer);
1120    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1121    result.append(buffer);
1122    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1123    result.append(buffer);
1124    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1125    result.append(buffer);
1126    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1127    result.append(buffer);
1128    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
1129    result.append(buffer);
1130
1131    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1132    result.append(buffer);
1133    result.append(" Index Command");
1134    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1135        snprintf(buffer, SIZE, "\n %02d    ", i);
1136        result.append(buffer);
1137        result.append(mNewParameters[i]);
1138    }
1139
1140    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1141    result.append(buffer);
1142    snprintf(buffer, SIZE, " Index event param\n");
1143    result.append(buffer);
1144    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1145        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
1146        result.append(buffer);
1147    }
1148    result.append("\n");
1149
1150    write(fd, result.string(), result.size());
1151
1152    if (locked) {
1153        mLock.unlock();
1154    }
1155    return NO_ERROR;
1156}
1157
1158status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1159{
1160    const size_t SIZE = 256;
1161    char buffer[SIZE];
1162    String8 result;
1163
1164    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1165    write(fd, buffer, strlen(buffer));
1166
1167    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1168        sp<EffectChain> chain = mEffectChains[i];
1169        if (chain != 0) {
1170            chain->dump(fd, args);
1171        }
1172    }
1173    return NO_ERROR;
1174}
1175
1176void AudioFlinger::ThreadBase::acquireWakeLock()
1177{
1178    Mutex::Autolock _l(mLock);
1179    acquireWakeLock_l();
1180}
1181
1182void AudioFlinger::ThreadBase::acquireWakeLock_l()
1183{
1184    if (mPowerManager == 0) {
1185        // use checkService() to avoid blocking if power service is not up yet
1186        sp<IBinder> binder =
1187            defaultServiceManager()->checkService(String16("power"));
1188        if (binder == 0) {
1189            ALOGW("Thread %s cannot connect to the power manager service", mName);
1190        } else {
1191            mPowerManager = interface_cast<IPowerManager>(binder);
1192            binder->linkToDeath(mDeathRecipient);
1193        }
1194    }
1195    if (mPowerManager != 0) {
1196        sp<IBinder> binder = new BBinder();
1197        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
1198                                                         binder,
1199                                                         String16(mName));
1200        if (status == NO_ERROR) {
1201            mWakeLockToken = binder;
1202        }
1203        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
1204    }
1205}
1206
1207void AudioFlinger::ThreadBase::releaseWakeLock()
1208{
1209    Mutex::Autolock _l(mLock);
1210    releaseWakeLock_l();
1211}
1212
1213void AudioFlinger::ThreadBase::releaseWakeLock_l()
1214{
1215    if (mWakeLockToken != 0) {
1216        ALOGV("releaseWakeLock_l() %s", mName);
1217        if (mPowerManager != 0) {
1218            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
1219        }
1220        mWakeLockToken.clear();
1221    }
1222}
1223
1224void AudioFlinger::ThreadBase::clearPowerManager()
1225{
1226    Mutex::Autolock _l(mLock);
1227    releaseWakeLock_l();
1228    mPowerManager.clear();
1229}
1230
1231void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
1232{
1233    sp<ThreadBase> thread = mThread.promote();
1234    if (thread != 0) {
1235        thread->clearPowerManager();
1236    }
1237    ALOGW("power manager service died !!!");
1238}
1239
1240void AudioFlinger::ThreadBase::setEffectSuspended(
1241        const effect_uuid_t *type, bool suspend, int sessionId)
1242{
1243    Mutex::Autolock _l(mLock);
1244    setEffectSuspended_l(type, suspend, sessionId);
1245}
1246
1247void AudioFlinger::ThreadBase::setEffectSuspended_l(
1248        const effect_uuid_t *type, bool suspend, int sessionId)
1249{
1250    sp<EffectChain> chain = getEffectChain_l(sessionId);
1251    if (chain != 0) {
1252        if (type != NULL) {
1253            chain->setEffectSuspended_l(type, suspend);
1254        } else {
1255            chain->setEffectSuspendedAll_l(suspend);
1256        }
1257    }
1258
1259    updateSuspendedSessions_l(type, suspend, sessionId);
1260}
1261
1262void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1263{
1264    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1265    if (index < 0) {
1266        return;
1267    }
1268
1269    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1270            mSuspendedSessions.editValueAt(index);
1271
1272    for (size_t i = 0; i < sessionEffects.size(); i++) {
1273        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1274        for (int j = 0; j < desc->mRefCount; j++) {
1275            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1276                chain->setEffectSuspendedAll_l(true);
1277            } else {
1278                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1279                     desc->mType.timeLow);
1280                chain->setEffectSuspended_l(&desc->mType, true);
1281            }
1282        }
1283    }
1284}
1285
1286void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1287                                                         bool suspend,
1288                                                         int sessionId)
1289{
1290    int index = mSuspendedSessions.indexOfKey(sessionId);
1291
1292    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1293
1294    if (suspend) {
1295        if (index >= 0) {
1296            sessionEffects = mSuspendedSessions.editValueAt(index);
1297        } else {
1298            mSuspendedSessions.add(sessionId, sessionEffects);
1299        }
1300    } else {
1301        if (index < 0) {
1302            return;
1303        }
1304        sessionEffects = mSuspendedSessions.editValueAt(index);
1305    }
1306
1307
1308    int key = EffectChain::kKeyForSuspendAll;
1309    if (type != NULL) {
1310        key = type->timeLow;
1311    }
1312    index = sessionEffects.indexOfKey(key);
1313
1314    sp <SuspendedSessionDesc> desc;
1315    if (suspend) {
1316        if (index >= 0) {
1317            desc = sessionEffects.valueAt(index);
1318        } else {
1319            desc = new SuspendedSessionDesc();
1320            if (type != NULL) {
1321                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1322            }
1323            sessionEffects.add(key, desc);
1324            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1325        }
1326        desc->mRefCount++;
1327    } else {
1328        if (index < 0) {
1329            return;
1330        }
1331        desc = sessionEffects.valueAt(index);
1332        if (--desc->mRefCount == 0) {
1333            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1334            sessionEffects.removeItemsAt(index);
1335            if (sessionEffects.isEmpty()) {
1336                ALOGV("updateSuspendedSessions_l() restore removing session %d",
1337                                 sessionId);
1338                mSuspendedSessions.removeItem(sessionId);
1339            }
1340        }
1341    }
1342    if (!sessionEffects.isEmpty()) {
1343        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1344    }
1345}
1346
1347void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1348                                                            bool enabled,
1349                                                            int sessionId)
1350{
1351    Mutex::Autolock _l(mLock);
1352    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1353}
1354
1355void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1356                                                            bool enabled,
1357                                                            int sessionId)
1358{
1359    if (mType != RECORD) {
1360        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1361        // another session. This gives the priority to well behaved effect control panels
1362        // and applications not using global effects.
1363        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1364            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1365        }
1366    }
1367
1368    sp<EffectChain> chain = getEffectChain_l(sessionId);
1369    if (chain != 0) {
1370        chain->checkSuspendOnEffectEnabled(effect, enabled);
1371    }
1372}
1373
1374// ----------------------------------------------------------------------------
1375
1376AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1377                                             AudioStreamOut* output,
1378                                             int id,
1379                                             uint32_t device,
1380                                             type_t type)
1381    :   ThreadBase(audioFlinger, id, device, type),
1382        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
1383        // Assumes constructor is called by AudioFlinger with it's mLock held,
1384        // but it would be safer to explicitly pass initial masterMute as parameter
1385        mMasterMute(audioFlinger->masterMute_l()),
1386        // mStreamTypes[] initialized in constructor body
1387        mOutput(output),
1388        // Assumes constructor is called by AudioFlinger with it's mLock held,
1389        // but it would be safer to explicitly pass initial masterVolume as parameter
1390        mMasterVolume(audioFlinger->masterVolume_l()),
1391        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1392{
1393    snprintf(mName, kNameLength, "AudioOut_%d", id);
1394
1395    readOutputParameters();
1396
1397    // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
1398    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
1399    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
1400            stream = (audio_stream_type_t) (stream + 1)) {
1401        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1402        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1403        // initialized by stream_type_t default constructor
1404        // mStreamTypes[stream].valid = true;
1405    }
1406}
1407
1408AudioFlinger::PlaybackThread::~PlaybackThread()
1409{
1410    delete [] mMixBuffer;
1411}
1412
1413status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1414{
1415    dumpInternals(fd, args);
1416    dumpTracks(fd, args);
1417    dumpEffectChains(fd, args);
1418    return NO_ERROR;
1419}
1420
1421status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1422{
1423    const size_t SIZE = 256;
1424    char buffer[SIZE];
1425    String8 result;
1426
1427    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1428    result.append(buffer);
1429    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1430    for (size_t i = 0; i < mTracks.size(); ++i) {
1431        sp<Track> track = mTracks[i];
1432        if (track != 0) {
1433            track->dump(buffer, SIZE);
1434            result.append(buffer);
1435        }
1436    }
1437
1438    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1439    result.append(buffer);
1440    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1441    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1442        sp<Track> track = mActiveTracks[i].promote();
1443        if (track != 0) {
1444            track->dump(buffer, SIZE);
1445            result.append(buffer);
1446        }
1447    }
1448    write(fd, result.string(), result.size());
1449    return NO_ERROR;
1450}
1451
1452status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1453{
1454    const size_t SIZE = 256;
1455    char buffer[SIZE];
1456    String8 result;
1457
1458    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1459    result.append(buffer);
1460    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1461    result.append(buffer);
1462    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1463    result.append(buffer);
1464    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1465    result.append(buffer);
1466    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1467    result.append(buffer);
1468    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1469    result.append(buffer);
1470    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1471    result.append(buffer);
1472    write(fd, result.string(), result.size());
1473
1474    dumpBase(fd, args);
1475
1476    return NO_ERROR;
1477}
1478
1479// Thread virtuals
1480status_t AudioFlinger::PlaybackThread::readyToRun()
1481{
1482    status_t status = initCheck();
1483    if (status == NO_ERROR) {
1484        ALOGI("AudioFlinger's thread %p ready to run", this);
1485    } else {
1486        ALOGE("No working audio driver found.");
1487    }
1488    return status;
1489}
1490
1491void AudioFlinger::PlaybackThread::onFirstRef()
1492{
1493    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1494}
1495
1496// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1497sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1498        const sp<AudioFlinger::Client>& client,
1499        audio_stream_type_t streamType,
1500        uint32_t sampleRate,
1501        audio_format_t format,
1502        uint32_t channelMask,
1503        int frameCount,
1504        const sp<IMemory>& sharedBuffer,
1505        int sessionId,
1506        status_t *status)
1507{
1508    sp<Track> track;
1509    status_t lStatus;
1510
1511    if (mType == DIRECT) {
1512        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1513            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1514                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1515                        "for output %p with format %d",
1516                        sampleRate, format, channelMask, mOutput, mFormat);
1517                lStatus = BAD_VALUE;
1518                goto Exit;
1519            }
1520        }
1521    } else {
1522        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1523        if (sampleRate > mSampleRate*2) {
1524            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1525            lStatus = BAD_VALUE;
1526            goto Exit;
1527        }
1528    }
1529
1530    lStatus = initCheck();
1531    if (lStatus != NO_ERROR) {
1532        ALOGE("Audio driver not initialized.");
1533        goto Exit;
1534    }
1535
1536    { // scope for mLock
1537        Mutex::Autolock _l(mLock);
1538
1539        // all tracks in same audio session must share the same routing strategy otherwise
1540        // conflicts will happen when tracks are moved from one output to another by audio policy
1541        // manager
1542        uint32_t strategy =
1543                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1544        for (size_t i = 0; i < mTracks.size(); ++i) {
1545            sp<Track> t = mTracks[i];
1546            if (t != 0) {
1547                uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
1548                if (sessionId == t->sessionId() && strategy != actual) {
1549                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1550                            strategy, actual);
1551                    lStatus = BAD_VALUE;
1552                    goto Exit;
1553                }
1554            }
1555        }
1556
1557        track = new Track(this, client, streamType, sampleRate, format,
1558                channelMask, frameCount, sharedBuffer, sessionId);
1559        if (track->getCblk() == NULL || track->name() < 0) {
1560            lStatus = NO_MEMORY;
1561            goto Exit;
1562        }
1563        mTracks.add(track);
1564
1565        sp<EffectChain> chain = getEffectChain_l(sessionId);
1566        if (chain != 0) {
1567            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1568            track->setMainBuffer(chain->inBuffer());
1569            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1570            chain->incTrackCnt();
1571        }
1572
1573        // invalidate track immediately if the stream type was moved to another thread since
1574        // createTrack() was called by the client process.
1575        if (!mStreamTypes[streamType].valid) {
1576            ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1577                 this, streamType);
1578            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1579        }
1580    }
1581    lStatus = NO_ERROR;
1582
1583Exit:
1584    if(status) {
1585        *status = lStatus;
1586    }
1587    return track;
1588}
1589
1590uint32_t AudioFlinger::PlaybackThread::latency() const
1591{
1592    Mutex::Autolock _l(mLock);
1593    if (initCheck() == NO_ERROR) {
1594        return mOutput->stream->get_latency(mOutput->stream);
1595    } else {
1596        return 0;
1597    }
1598}
1599
1600status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1601{
1602    mMasterVolume = value;
1603    return NO_ERROR;
1604}
1605
1606status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1607{
1608    mMasterMute = muted;
1609    return NO_ERROR;
1610}
1611
1612float AudioFlinger::PlaybackThread::masterVolume() const
1613{
1614    return mMasterVolume;
1615}
1616
1617bool AudioFlinger::PlaybackThread::masterMute() const
1618{
1619    return mMasterMute;
1620}
1621
1622status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
1623{
1624    mStreamTypes[stream].volume = value;
1625    return NO_ERROR;
1626}
1627
1628status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
1629{
1630    mStreamTypes[stream].mute = muted;
1631    return NO_ERROR;
1632}
1633
1634float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
1635{
1636    return mStreamTypes[stream].volume;
1637}
1638
1639bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const
1640{
1641    return mStreamTypes[stream].mute;
1642}
1643
1644// addTrack_l() must be called with ThreadBase::mLock held
1645status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1646{
1647    status_t status = ALREADY_EXISTS;
1648
1649    // set retry count for buffer fill
1650    track->mRetryCount = kMaxTrackStartupRetries;
1651    if (mActiveTracks.indexOf(track) < 0) {
1652        // the track is newly added, make sure it fills up all its
1653        // buffers before playing. This is to ensure the client will
1654        // effectively get the latency it requested.
1655        track->mFillingUpStatus = Track::FS_FILLING;
1656        track->mResetDone = false;
1657        mActiveTracks.add(track);
1658        if (track->mainBuffer() != mMixBuffer) {
1659            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1660            if (chain != 0) {
1661                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1662                chain->incActiveTrackCnt();
1663            }
1664        }
1665
1666        status = NO_ERROR;
1667    }
1668
1669    ALOGV("mWaitWorkCV.broadcast");
1670    mWaitWorkCV.broadcast();
1671
1672    return status;
1673}
1674
1675// destroyTrack_l() must be called with ThreadBase::mLock held
1676void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1677{
1678    track->mState = TrackBase::TERMINATED;
1679    if (mActiveTracks.indexOf(track) < 0) {
1680        removeTrack_l(track);
1681    }
1682}
1683
1684void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1685{
1686    mTracks.remove(track);
1687    deleteTrackName_l(track->name());
1688    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1689    if (chain != 0) {
1690        chain->decTrackCnt();
1691    }
1692}
1693
1694String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1695{
1696    String8 out_s8 = String8("");
1697    char *s;
1698
1699    Mutex::Autolock _l(mLock);
1700    if (initCheck() != NO_ERROR) {
1701        return out_s8;
1702    }
1703
1704    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1705    out_s8 = String8(s);
1706    free(s);
1707    return out_s8;
1708}
1709
1710// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1711void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1712    AudioSystem::OutputDescriptor desc;
1713    void *param2 = NULL;
1714
1715    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1716
1717    switch (event) {
1718    case AudioSystem::OUTPUT_OPENED:
1719    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1720        desc.channels = mChannelMask;
1721        desc.samplingRate = mSampleRate;
1722        desc.format = mFormat;
1723        desc.frameCount = mFrameCount;
1724        desc.latency = latency();
1725        param2 = &desc;
1726        break;
1727
1728    case AudioSystem::STREAM_CONFIG_CHANGED:
1729        param2 = &param;
1730    case AudioSystem::OUTPUT_CLOSED:
1731    default:
1732        break;
1733    }
1734    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1735}
1736
1737void AudioFlinger::PlaybackThread::readOutputParameters()
1738{
1739    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1740    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1741    mChannelCount = (uint16_t)popcount(mChannelMask);
1742    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1743    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
1744    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1745
1746    // FIXME - Current mixer implementation only supports stereo output: Always
1747    // Allocate a stereo buffer even if HW output is mono.
1748    delete[] mMixBuffer;
1749    mMixBuffer = new int16_t[mFrameCount * 2];
1750    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1751
1752    // force reconfiguration of effect chains and engines to take new buffer size and audio
1753    // parameters into account
1754    // Note that mLock is not held when readOutputParameters() is called from the constructor
1755    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1756    // matter.
1757    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1758    Vector< sp<EffectChain> > effectChains = mEffectChains;
1759    for (size_t i = 0; i < effectChains.size(); i ++) {
1760        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1761    }
1762}
1763
1764status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1765{
1766    if (halFrames == NULL || dspFrames == NULL) {
1767        return BAD_VALUE;
1768    }
1769    Mutex::Autolock _l(mLock);
1770    if (initCheck() != NO_ERROR) {
1771        return INVALID_OPERATION;
1772    }
1773    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1774
1775    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1776}
1777
1778uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1779{
1780    Mutex::Autolock _l(mLock);
1781    uint32_t result = 0;
1782    if (getEffectChain_l(sessionId) != 0) {
1783        result = EFFECT_SESSION;
1784    }
1785
1786    for (size_t i = 0; i < mTracks.size(); ++i) {
1787        sp<Track> track = mTracks[i];
1788        if (sessionId == track->sessionId() &&
1789                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1790            result |= TRACK_SESSION;
1791            break;
1792        }
1793    }
1794
1795    return result;
1796}
1797
1798uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1799{
1800    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1801    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1802    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1803        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1804    }
1805    for (size_t i = 0; i < mTracks.size(); i++) {
1806        sp<Track> track = mTracks[i];
1807        if (sessionId == track->sessionId() &&
1808                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1809            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1810        }
1811    }
1812    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1813}
1814
1815
1816AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
1817{
1818    Mutex::Autolock _l(mLock);
1819    return mOutput;
1820}
1821
1822AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1823{
1824    Mutex::Autolock _l(mLock);
1825    AudioStreamOut *output = mOutput;
1826    mOutput = NULL;
1827    return output;
1828}
1829
1830// this method must always be called either with ThreadBase mLock held or inside the thread loop
1831audio_stream_t* AudioFlinger::PlaybackThread::stream()
1832{
1833    if (mOutput == NULL) {
1834        return NULL;
1835    }
1836    return &mOutput->stream->common;
1837}
1838
1839uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
1840{
1841    // A2DP output latency is not due only to buffering capacity. It also reflects encoding,
1842    // decoding and transfer time. So sleeping for half of the latency would likely cause
1843    // underruns
1844    if (audio_is_a2dp_device((audio_devices_t)mDevice)) {
1845        return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000);
1846    } else {
1847        return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
1848    }
1849}
1850
1851// ----------------------------------------------------------------------------
1852
1853AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
1854        int id, uint32_t device, type_t type)
1855    :   PlaybackThread(audioFlinger, output, id, device, type),
1856        mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)),
1857        mPrevMixerStatus(MIXER_IDLE)
1858{
1859    // FIXME - Current mixer implementation only supports stereo output
1860    if (mChannelCount == 1) {
1861        ALOGE("Invalid audio hardware channel count");
1862    }
1863}
1864
1865AudioFlinger::MixerThread::~MixerThread()
1866{
1867    delete mAudioMixer;
1868}
1869
1870bool AudioFlinger::MixerThread::threadLoop()
1871{
1872    Vector< sp<Track> > tracksToRemove;
1873    mixer_state mixerStatus = MIXER_IDLE;
1874    nsecs_t standbyTime = systemTime();
1875    size_t mixBufferSize = mFrameCount * mFrameSize;
1876    // FIXME: Relaxed timing because of a certain device that can't meet latency
1877    // Should be reduced to 2x after the vendor fixes the driver issue
1878    // increase threshold again due to low power audio mode. The way this warning threshold is
1879    // calculated and its usefulness should be reconsidered anyway.
1880    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1881    nsecs_t lastWarning = 0;
1882    bool longStandbyExit = false;
1883    uint32_t activeSleepTime = activeSleepTimeUs();
1884    uint32_t idleSleepTime = idleSleepTimeUs();
1885    uint32_t sleepTime = idleSleepTime;
1886    uint32_t sleepTimeShift = 0;
1887    Vector< sp<EffectChain> > effectChains;
1888#ifdef DEBUG_CPU_USAGE
1889    ThreadCpuUsage cpu;
1890    const CentralTendencyStatistics& stats = cpu.statistics();
1891#endif
1892
1893    acquireWakeLock();
1894
1895    while (!exitPending())
1896    {
1897#ifdef DEBUG_CPU_USAGE
1898        cpu.sampleAndEnable();
1899        unsigned n = stats.n();
1900        // cpu.elapsed() is expensive, so don't call it every loop
1901        if ((n & 127) == 1) {
1902            long long elapsed = cpu.elapsed();
1903            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1904                double perLoop = elapsed / (double) n;
1905                double perLoop100 = perLoop * 0.01;
1906                double mean = stats.mean();
1907                double stddev = stats.stddev();
1908                double minimum = stats.minimum();
1909                double maximum = stats.maximum();
1910                cpu.resetStatistics();
1911                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",
1912                        elapsed * .000000001, n, perLoop * .000001,
1913                        mean * .001,
1914                        stddev * .001,
1915                        minimum * .001,
1916                        maximum * .001,
1917                        mean / perLoop100,
1918                        stddev / perLoop100,
1919                        minimum / perLoop100,
1920                        maximum / perLoop100);
1921            }
1922        }
1923#endif
1924        processConfigEvents();
1925
1926        mixerStatus = MIXER_IDLE;
1927        { // scope for mLock
1928
1929            Mutex::Autolock _l(mLock);
1930
1931            if (checkForNewParameters_l()) {
1932                mixBufferSize = mFrameCount * mFrameSize;
1933                // FIXME: Relaxed timing because of a certain device that can't meet latency
1934                // Should be reduced to 2x after the vendor fixes the driver issue
1935                // increase threshold again due to low power audio mode. The way this warning
1936                // threshold is calculated and its usefulness should be reconsidered anyway.
1937                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1938                activeSleepTime = activeSleepTimeUs();
1939                idleSleepTime = idleSleepTimeUs();
1940            }
1941
1942            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1943
1944            // put audio hardware into standby after short delay
1945            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1946                        mSuspended)) {
1947                if (!mStandby) {
1948                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1949                    mOutput->stream->common.standby(&mOutput->stream->common);
1950                    mStandby = true;
1951                    mBytesWritten = 0;
1952                }
1953
1954                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1955                    // we're about to wait, flush the binder command buffer
1956                    IPCThreadState::self()->flushCommands();
1957
1958                    if (exitPending()) break;
1959
1960                    releaseWakeLock_l();
1961                    // wait until we have something to do...
1962                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1963                    mWaitWorkCV.wait(mLock);
1964                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1965                    acquireWakeLock_l();
1966
1967                    mPrevMixerStatus = MIXER_IDLE;
1968                    if (!mMasterMute) {
1969                        char value[PROPERTY_VALUE_MAX];
1970                        property_get("ro.audio.silent", value, "0");
1971                        if (atoi(value)) {
1972                            ALOGD("Silence is golden");
1973                            setMasterMute(true);
1974                        }
1975                    }
1976
1977                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1978                    sleepTime = idleSleepTime;
1979                    sleepTimeShift = 0;
1980                    continue;
1981                }
1982            }
1983
1984            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1985
1986            // prevent any changes in effect chain list and in each effect chain
1987            // during mixing and effect process as the audio buffers could be deleted
1988            // or modified if an effect is created or deleted
1989            lockEffectChains_l(effectChains);
1990        }
1991
1992        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1993            // mix buffers...
1994            mAudioMixer->process();
1995            // increase sleep time progressively when application underrun condition clears.
1996            // Only increase sleep time if the mixer is ready for two consecutive times to avoid
1997            // that a steady state of alternating ready/not ready conditions keeps the sleep time
1998            // such that we would underrun the audio HAL.
1999            if ((sleepTime == 0) && (sleepTimeShift > 0)) {
2000                sleepTimeShift--;
2001            }
2002            sleepTime = 0;
2003            standbyTime = systemTime() + kStandbyTimeInNsecs;
2004            //TODO: delay standby when effects have a tail
2005        } else {
2006            // If no tracks are ready, sleep once for the duration of an output
2007            // buffer size, then write 0s to the output
2008            if (sleepTime == 0) {
2009                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2010                    sleepTime = activeSleepTime >> sleepTimeShift;
2011                    if (sleepTime < kMinThreadSleepTimeUs) {
2012                        sleepTime = kMinThreadSleepTimeUs;
2013                    }
2014                    // reduce sleep time in case of consecutive application underruns to avoid
2015                    // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
2016                    // duration we would end up writing less data than needed by the audio HAL if
2017                    // the condition persists.
2018                    if (sleepTimeShift < kMaxThreadSleepTimeShift) {
2019                        sleepTimeShift++;
2020                    }
2021                } else {
2022                    sleepTime = idleSleepTime;
2023                }
2024            } else if (mBytesWritten != 0 ||
2025                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
2026                memset (mMixBuffer, 0, mixBufferSize);
2027                sleepTime = 0;
2028                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
2029            }
2030            // TODO add standby time extension fct of effect tail
2031        }
2032
2033        if (mSuspended) {
2034            sleepTime = suspendSleepTimeUs();
2035        }
2036        // sleepTime == 0 means we must write to audio hardware
2037        if (sleepTime == 0) {
2038            for (size_t i = 0; i < effectChains.size(); i ++) {
2039                effectChains[i]->process_l();
2040            }
2041            // enable changes in effect chain
2042            unlockEffectChains(effectChains);
2043            mLastWriteTime = systemTime();
2044            mInWrite = true;
2045            mBytesWritten += mixBufferSize;
2046
2047            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2048            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2049            mNumWrites++;
2050            mInWrite = false;
2051            nsecs_t now = systemTime();
2052            nsecs_t delta = now - mLastWriteTime;
2053            if (!mStandby && delta > maxPeriod) {
2054                mNumDelayedWrites++;
2055                if ((now - lastWarning) > kWarningThrottleNs) {
2056                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2057                            ns2ms(delta), mNumDelayedWrites, this);
2058                    lastWarning = now;
2059                }
2060                if (mStandby) {
2061                    longStandbyExit = true;
2062                }
2063            }
2064            mStandby = false;
2065        } else {
2066            // enable changes in effect chain
2067            unlockEffectChains(effectChains);
2068            usleep(sleepTime);
2069        }
2070
2071        // finally let go of all our tracks, without the lock held
2072        // since we can't guarantee the destructors won't acquire that
2073        // same lock.
2074        tracksToRemove.clear();
2075
2076        // Effect chains will be actually deleted here if they were removed from
2077        // mEffectChains list during mixing or effects processing
2078        effectChains.clear();
2079    }
2080
2081    if (!mStandby) {
2082        mOutput->stream->common.standby(&mOutput->stream->common);
2083    }
2084
2085    releaseWakeLock();
2086
2087    ALOGV("MixerThread %p exiting", this);
2088    return false;
2089}
2090
2091// prepareTracks_l() must be called with ThreadBase::mLock held
2092AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
2093        const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2094{
2095
2096    mixer_state mixerStatus = MIXER_IDLE;
2097    // find out which tracks need to be processed
2098    size_t count = activeTracks.size();
2099    size_t mixedTracks = 0;
2100    size_t tracksWithEffect = 0;
2101
2102    float masterVolume = mMasterVolume;
2103    bool  masterMute = mMasterMute;
2104
2105    if (masterMute) {
2106        masterVolume = 0;
2107    }
2108    // Delegate master volume control to effect in output mix effect chain if needed
2109    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2110    if (chain != 0) {
2111        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2112        chain->setVolume_l(&v, &v);
2113        masterVolume = (float)((v + (1 << 23)) >> 24);
2114        chain.clear();
2115    }
2116
2117    for (size_t i=0 ; i<count ; i++) {
2118        sp<Track> t = activeTracks[i].promote();
2119        if (t == 0) continue;
2120
2121        // this const just means the local variable doesn't change
2122        Track* const track = t.get();
2123        audio_track_cblk_t* cblk = track->cblk();
2124
2125        // The first time a track is added we wait
2126        // for all its buffers to be filled before processing it
2127        int name = track->name();
2128        // make sure that we have enough frames to mix one full buffer.
2129        // enforce this condition only once to enable draining the buffer in case the client
2130        // app does not call stop() and relies on underrun to stop:
2131        // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
2132        // during last round
2133        uint32_t minFrames = 1;
2134        if (!track->isStopped() && !track->isPausing() &&
2135                (mPrevMixerStatus == MIXER_TRACKS_READY)) {
2136            if (t->sampleRate() == (int)mSampleRate) {
2137                minFrames = mFrameCount;
2138            } else {
2139                // +1 for rounding and +1 for additional sample needed for interpolation
2140                minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
2141                // add frames already consumed but not yet released by the resampler
2142                // because cblk->framesReady() will  include these frames
2143                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
2144                // the minimum track buffer size is normally twice the number of frames necessary
2145                // to fill one buffer and the resampler should not leave more than one buffer worth
2146                // of unreleased frames after each pass, but just in case...
2147                ALOG_ASSERT(minFrames <= cblk->frameCount);
2148            }
2149        }
2150        if ((cblk->framesReady() >= minFrames) && track->isReady() &&
2151                !track->isPaused() && !track->isTerminated())
2152        {
2153            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
2154
2155            mixedTracks++;
2156
2157            // track->mainBuffer() != mMixBuffer means there is an effect chain
2158            // connected to the track
2159            chain.clear();
2160            if (track->mainBuffer() != mMixBuffer) {
2161                chain = getEffectChain_l(track->sessionId());
2162                // Delegate volume control to effect in track effect chain if needed
2163                if (chain != 0) {
2164                    tracksWithEffect++;
2165                } else {
2166                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
2167                            name, track->sessionId());
2168                }
2169            }
2170
2171
2172            int param = AudioMixer::VOLUME;
2173            if (track->mFillingUpStatus == Track::FS_FILLED) {
2174                // no ramp for the first volume setting
2175                track->mFillingUpStatus = Track::FS_ACTIVE;
2176                if (track->mState == TrackBase::RESUMING) {
2177                    track->mState = TrackBase::ACTIVE;
2178                    param = AudioMixer::RAMP_VOLUME;
2179                }
2180                mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2181            } else if (cblk->server != 0) {
2182                // If the track is stopped before the first frame was mixed,
2183                // do not apply ramp
2184                param = AudioMixer::RAMP_VOLUME;
2185            }
2186
2187            // compute volume for this track
2188            uint32_t vl, vr, va;
2189            if (track->isMuted() || track->isPausing() ||
2190                mStreamTypes[track->type()].mute) {
2191                vl = vr = va = 0;
2192                if (track->isPausing()) {
2193                    track->setPaused();
2194                }
2195            } else {
2196
2197                // read original volumes with volume control
2198                float typeVolume = mStreamTypes[track->type()].volume;
2199                float v = masterVolume * typeVolume;
2200                uint32_t vlr = cblk->getVolumeLR();
2201                vl = vlr & 0xFFFF;
2202                vr = vlr >> 16;
2203                // track volumes come from shared memory, so can't be trusted and must be clamped
2204                if (vl > MAX_GAIN_INT) {
2205                    ALOGV("Track left volume out of range: %04X", vl);
2206                    vl = MAX_GAIN_INT;
2207                }
2208                if (vr > MAX_GAIN_INT) {
2209                    ALOGV("Track right volume out of range: %04X", vr);
2210                    vr = MAX_GAIN_INT;
2211                }
2212                // now apply the master volume and stream type volume
2213                vl = (uint32_t)(v * vl) << 12;
2214                vr = (uint32_t)(v * vr) << 12;
2215                // assuming master volume and stream type volume each go up to 1.0,
2216                // vl and vr are now in 8.24 format
2217
2218                uint16_t sendLevel = cblk->getSendLevel_U4_12();
2219                // send level comes from shared memory and so may be corrupt
2220                if (sendLevel >= MAX_GAIN_INT) {
2221                    ALOGV("Track send level out of range: %04X", sendLevel);
2222                    sendLevel = MAX_GAIN_INT;
2223                }
2224                va = (uint32_t)(v * sendLevel);
2225            }
2226            // Delegate volume control to effect in track effect chain if needed
2227            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2228                // Do not ramp volume if volume is controlled by effect
2229                param = AudioMixer::VOLUME;
2230                track->mHasVolumeController = true;
2231            } else {
2232                // force no volume ramp when volume controller was just disabled or removed
2233                // from effect chain to avoid volume spike
2234                if (track->mHasVolumeController) {
2235                    param = AudioMixer::VOLUME;
2236                }
2237                track->mHasVolumeController = false;
2238            }
2239
2240            // Convert volumes from 8.24 to 4.12 format
2241            int16_t left, right, aux;
2242            // This additional clamping is needed in case chain->setVolume_l() overshot
2243            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2244            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2245            left = int16_t(v_clamped);
2246            v_clamped = (vr + (1 << 11)) >> 12;
2247            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2248            right = int16_t(v_clamped);
2249
2250            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2251            aux = int16_t(va);
2252
2253            // XXX: these things DON'T need to be done each time
2254            mAudioMixer->setBufferProvider(name, track);
2255            mAudioMixer->enable(name);
2256
2257            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left);
2258            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right);
2259            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux);
2260            mAudioMixer->setParameter(
2261                name,
2262                AudioMixer::TRACK,
2263                AudioMixer::FORMAT, (void *)track->format());
2264            mAudioMixer->setParameter(
2265                name,
2266                AudioMixer::TRACK,
2267                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2268            mAudioMixer->setParameter(
2269                name,
2270                AudioMixer::RESAMPLE,
2271                AudioMixer::SAMPLE_RATE,
2272                (void *)(cblk->sampleRate));
2273            mAudioMixer->setParameter(
2274                name,
2275                AudioMixer::TRACK,
2276                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2277            mAudioMixer->setParameter(
2278                name,
2279                AudioMixer::TRACK,
2280                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2281
2282            // reset retry count
2283            track->mRetryCount = kMaxTrackRetries;
2284            // If one track is ready, set the mixer ready if:
2285            //  - the mixer was not ready during previous round OR
2286            //  - no other track is not ready
2287            if (mPrevMixerStatus != MIXER_TRACKS_READY ||
2288                    mixerStatus != MIXER_TRACKS_ENABLED) {
2289                mixerStatus = MIXER_TRACKS_READY;
2290            }
2291        } else {
2292            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
2293            if (track->isStopped()) {
2294                track->reset();
2295            }
2296            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2297                // We have consumed all the buffers of this track.
2298                // Remove it from the list of active tracks.
2299                tracksToRemove->add(track);
2300            } else {
2301                // No buffers for this track. Give it a few chances to
2302                // fill a buffer, then remove it from active list.
2303                if (--(track->mRetryCount) <= 0) {
2304                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
2305                    tracksToRemove->add(track);
2306                    // indicate to client process that the track was disabled because of underrun
2307                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2308                // If one track is not ready, mark the mixer also not ready if:
2309                //  - the mixer was ready during previous round OR
2310                //  - no other track is ready
2311                } else if (mPrevMixerStatus == MIXER_TRACKS_READY ||
2312                                mixerStatus != MIXER_TRACKS_READY) {
2313                    mixerStatus = MIXER_TRACKS_ENABLED;
2314                }
2315            }
2316            mAudioMixer->disable(name);
2317        }
2318    }
2319
2320    // remove all the tracks that need to be...
2321    count = tracksToRemove->size();
2322    if (CC_UNLIKELY(count)) {
2323        for (size_t i=0 ; i<count ; i++) {
2324            const sp<Track>& track = tracksToRemove->itemAt(i);
2325            mActiveTracks.remove(track);
2326            if (track->mainBuffer() != mMixBuffer) {
2327                chain = getEffectChain_l(track->sessionId());
2328                if (chain != 0) {
2329                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2330                    chain->decActiveTrackCnt();
2331                }
2332            }
2333            if (track->isTerminated()) {
2334                removeTrack_l(track);
2335            }
2336        }
2337    }
2338
2339    // mix buffer must be cleared if all tracks are connected to an
2340    // effect chain as in this case the mixer will not write to
2341    // mix buffer and track effects will accumulate into it
2342    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2343        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2344    }
2345
2346    mPrevMixerStatus = mixerStatus;
2347    return mixerStatus;
2348}
2349
2350void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType)
2351{
2352    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2353            this,  streamType, mTracks.size());
2354    Mutex::Autolock _l(mLock);
2355
2356    size_t size = mTracks.size();
2357    for (size_t i = 0; i < size; i++) {
2358        sp<Track> t = mTracks[i];
2359        if (t->type() == streamType) {
2360            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2361            t->mCblk->cv.signal();
2362        }
2363    }
2364}
2365
2366void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid)
2367{
2368    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2369            this,  streamType, valid);
2370    Mutex::Autolock _l(mLock);
2371
2372    mStreamTypes[streamType].valid = valid;
2373}
2374
2375// getTrackName_l() must be called with ThreadBase::mLock held
2376int AudioFlinger::MixerThread::getTrackName_l()
2377{
2378    return mAudioMixer->getTrackName();
2379}
2380
2381// deleteTrackName_l() must be called with ThreadBase::mLock held
2382void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2383{
2384    ALOGV("remove track (%d) and delete from mixer", name);
2385    mAudioMixer->deleteTrackName(name);
2386}
2387
2388// checkForNewParameters_l() must be called with ThreadBase::mLock held
2389bool AudioFlinger::MixerThread::checkForNewParameters_l()
2390{
2391    bool reconfig = false;
2392
2393    while (!mNewParameters.isEmpty()) {
2394        status_t status = NO_ERROR;
2395        String8 keyValuePair = mNewParameters[0];
2396        AudioParameter param = AudioParameter(keyValuePair);
2397        int value;
2398
2399        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2400            reconfig = true;
2401        }
2402        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2403            if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
2404                status = BAD_VALUE;
2405            } else {
2406                reconfig = true;
2407            }
2408        }
2409        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2410            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2411                status = BAD_VALUE;
2412            } else {
2413                reconfig = true;
2414            }
2415        }
2416        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2417            // do not accept frame count changes if tracks are open as the track buffer
2418            // size depends on frame count and correct behavior would not be guaranteed
2419            // if frame count is changed after track creation
2420            if (!mTracks.isEmpty()) {
2421                status = INVALID_OPERATION;
2422            } else {
2423                reconfig = true;
2424            }
2425        }
2426        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2427            // when changing the audio output device, call addBatteryData to notify
2428            // the change
2429            if ((int)mDevice != value) {
2430                uint32_t params = 0;
2431                // check whether speaker is on
2432                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2433                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2434                }
2435
2436                int deviceWithoutSpeaker
2437                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2438                // check if any other device (except speaker) is on
2439                if (value & deviceWithoutSpeaker ) {
2440                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2441                }
2442
2443                if (params != 0) {
2444                    addBatteryData(params);
2445                }
2446            }
2447
2448            // forward device change to effects that have requested to be
2449            // aware of attached audio device.
2450            mDevice = (uint32_t)value;
2451            for (size_t i = 0; i < mEffectChains.size(); i++) {
2452                mEffectChains[i]->setDevice_l(mDevice);
2453            }
2454        }
2455
2456        if (status == NO_ERROR) {
2457            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2458                                                    keyValuePair.string());
2459            if (!mStandby && status == INVALID_OPERATION) {
2460               mOutput->stream->common.standby(&mOutput->stream->common);
2461               mStandby = true;
2462               mBytesWritten = 0;
2463               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2464                                                       keyValuePair.string());
2465            }
2466            if (status == NO_ERROR && reconfig) {
2467                delete mAudioMixer;
2468                // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
2469                mAudioMixer = NULL;
2470                readOutputParameters();
2471                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2472                for (size_t i = 0; i < mTracks.size() ; i++) {
2473                    int name = getTrackName_l();
2474                    if (name < 0) break;
2475                    mTracks[i]->mName = name;
2476                    // limit track sample rate to 2 x new output sample rate
2477                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2478                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2479                    }
2480                }
2481                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2482            }
2483        }
2484
2485        mNewParameters.removeAt(0);
2486
2487        mParamStatus = status;
2488        mParamCond.signal();
2489        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2490        // already timed out waiting for the status and will never signal the condition.
2491        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2492    }
2493    return reconfig;
2494}
2495
2496status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2497{
2498    const size_t SIZE = 256;
2499    char buffer[SIZE];
2500    String8 result;
2501
2502    PlaybackThread::dumpInternals(fd, args);
2503
2504    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2505    result.append(buffer);
2506    write(fd, result.string(), result.size());
2507    return NO_ERROR;
2508}
2509
2510uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2511{
2512    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2513}
2514
2515uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2516{
2517    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2518}
2519
2520// ----------------------------------------------------------------------------
2521AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2522    :   PlaybackThread(audioFlinger, output, id, device, DIRECT)
2523        // mLeftVolFloat, mRightVolFloat
2524        // mLeftVolShort, mRightVolShort
2525{
2526}
2527
2528AudioFlinger::DirectOutputThread::~DirectOutputThread()
2529{
2530}
2531
2532static inline
2533int32_t mul(int16_t in, int16_t v)
2534{
2535#if defined(__arm__) && !defined(__thumb__)
2536    int32_t out;
2537    asm( "smulbb %[out], %[in], %[v] \n"
2538         : [out]"=r"(out)
2539         : [in]"%r"(in), [v]"r"(v)
2540         : );
2541    return out;
2542#else
2543    return in * int32_t(v);
2544#endif
2545}
2546
2547void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2548{
2549    // Do not apply volume on compressed audio
2550    if (!audio_is_linear_pcm(mFormat)) {
2551        return;
2552    }
2553
2554    // convert to signed 16 bit before volume calculation
2555    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2556        size_t count = mFrameCount * mChannelCount;
2557        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2558        int16_t *dst = mMixBuffer + count-1;
2559        while(count--) {
2560            *dst-- = (int16_t)(*src--^0x80) << 8;
2561        }
2562    }
2563
2564    size_t frameCount = mFrameCount;
2565    int16_t *out = mMixBuffer;
2566    if (ramp) {
2567        if (mChannelCount == 1) {
2568            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2569            int32_t vlInc = d / (int32_t)frameCount;
2570            int32_t vl = ((int32_t)mLeftVolShort << 16);
2571            do {
2572                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2573                out++;
2574                vl += vlInc;
2575            } while (--frameCount);
2576
2577        } else {
2578            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2579            int32_t vlInc = d / (int32_t)frameCount;
2580            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2581            int32_t vrInc = d / (int32_t)frameCount;
2582            int32_t vl = ((int32_t)mLeftVolShort << 16);
2583            int32_t vr = ((int32_t)mRightVolShort << 16);
2584            do {
2585                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2586                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2587                out += 2;
2588                vl += vlInc;
2589                vr += vrInc;
2590            } while (--frameCount);
2591        }
2592    } else {
2593        if (mChannelCount == 1) {
2594            do {
2595                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2596                out++;
2597            } while (--frameCount);
2598        } else {
2599            do {
2600                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2601                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2602                out += 2;
2603            } while (--frameCount);
2604        }
2605    }
2606
2607    // convert back to unsigned 8 bit after volume calculation
2608    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2609        size_t count = mFrameCount * mChannelCount;
2610        int16_t *src = mMixBuffer;
2611        uint8_t *dst = (uint8_t *)mMixBuffer;
2612        while(count--) {
2613            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2614        }
2615    }
2616
2617    mLeftVolShort = leftVol;
2618    mRightVolShort = rightVol;
2619}
2620
2621bool AudioFlinger::DirectOutputThread::threadLoop()
2622{
2623    mixer_state mixerStatus = MIXER_IDLE;
2624    sp<Track> trackToRemove;
2625    sp<Track> activeTrack;
2626    nsecs_t standbyTime = systemTime();
2627    int8_t *curBuf;
2628    size_t mixBufferSize = mFrameCount*mFrameSize;
2629    uint32_t activeSleepTime = activeSleepTimeUs();
2630    uint32_t idleSleepTime = idleSleepTimeUs();
2631    uint32_t sleepTime = idleSleepTime;
2632    // use shorter standby delay as on normal output to release
2633    // hardware resources as soon as possible
2634    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2635
2636    acquireWakeLock();
2637
2638    while (!exitPending())
2639    {
2640        bool rampVolume;
2641        uint16_t leftVol;
2642        uint16_t rightVol;
2643        Vector< sp<EffectChain> > effectChains;
2644
2645        processConfigEvents();
2646
2647        mixerStatus = MIXER_IDLE;
2648
2649        { // scope for the mLock
2650
2651            Mutex::Autolock _l(mLock);
2652
2653            if (checkForNewParameters_l()) {
2654                mixBufferSize = mFrameCount*mFrameSize;
2655                activeSleepTime = activeSleepTimeUs();
2656                idleSleepTime = idleSleepTimeUs();
2657                standbyDelay = microseconds(activeSleepTime*2);
2658            }
2659
2660            // put audio hardware into standby after short delay
2661            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2662                        mSuspended)) {
2663                // wait until we have something to do...
2664                if (!mStandby) {
2665                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2666                    mOutput->stream->common.standby(&mOutput->stream->common);
2667                    mStandby = true;
2668                    mBytesWritten = 0;
2669                }
2670
2671                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2672                    // we're about to wait, flush the binder command buffer
2673                    IPCThreadState::self()->flushCommands();
2674
2675                    if (exitPending()) break;
2676
2677                    releaseWakeLock_l();
2678                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2679                    mWaitWorkCV.wait(mLock);
2680                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2681                    acquireWakeLock_l();
2682
2683                    if (!mMasterMute) {
2684                        char value[PROPERTY_VALUE_MAX];
2685                        property_get("ro.audio.silent", value, "0");
2686                        if (atoi(value)) {
2687                            ALOGD("Silence is golden");
2688                            setMasterMute(true);
2689                        }
2690                    }
2691
2692                    standbyTime = systemTime() + standbyDelay;
2693                    sleepTime = idleSleepTime;
2694                    continue;
2695                }
2696            }
2697
2698            effectChains = mEffectChains;
2699
2700            // find out which tracks need to be processed
2701            if (mActiveTracks.size() != 0) {
2702                sp<Track> t = mActiveTracks[0].promote();
2703                if (t == 0) continue;
2704
2705                Track* const track = t.get();
2706                audio_track_cblk_t* cblk = track->cblk();
2707
2708                // The first time a track is added we wait
2709                // for all its buffers to be filled before processing it
2710                if (cblk->framesReady() && track->isReady() &&
2711                        !track->isPaused() && !track->isTerminated())
2712                {
2713                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2714
2715                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2716                        track->mFillingUpStatus = Track::FS_ACTIVE;
2717                        mLeftVolFloat = mRightVolFloat = 0;
2718                        mLeftVolShort = mRightVolShort = 0;
2719                        if (track->mState == TrackBase::RESUMING) {
2720                            track->mState = TrackBase::ACTIVE;
2721                            rampVolume = true;
2722                        }
2723                    } else if (cblk->server != 0) {
2724                        // If the track is stopped before the first frame was mixed,
2725                        // do not apply ramp
2726                        rampVolume = true;
2727                    }
2728                    // compute volume for this track
2729                    float left, right;
2730                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2731                        mStreamTypes[track->type()].mute) {
2732                        left = right = 0;
2733                        if (track->isPausing()) {
2734                            track->setPaused();
2735                        }
2736                    } else {
2737                        float typeVolume = mStreamTypes[track->type()].volume;
2738                        float v = mMasterVolume * typeVolume;
2739                        uint32_t vlr = cblk->getVolumeLR();
2740                        float v_clamped = v * (vlr & 0xFFFF);
2741                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2742                        left = v_clamped/MAX_GAIN;
2743                        v_clamped = v * (vlr >> 16);
2744                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2745                        right = v_clamped/MAX_GAIN;
2746                    }
2747
2748                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2749                        mLeftVolFloat = left;
2750                        mRightVolFloat = right;
2751
2752                        // If audio HAL implements volume control,
2753                        // force software volume to nominal value
2754                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2755                            left = 1.0f;
2756                            right = 1.0f;
2757                        }
2758
2759                        // Convert volumes from float to 8.24
2760                        uint32_t vl = (uint32_t)(left * (1 << 24));
2761                        uint32_t vr = (uint32_t)(right * (1 << 24));
2762
2763                        // Delegate volume control to effect in track effect chain if needed
2764                        // only one effect chain can be present on DirectOutputThread, so if
2765                        // there is one, the track is connected to it
2766                        if (!effectChains.isEmpty()) {
2767                            // Do not ramp volume if volume is controlled by effect
2768                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2769                                rampVolume = false;
2770                            }
2771                        }
2772
2773                        // Convert volumes from 8.24 to 4.12 format
2774                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2775                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2776                        leftVol = (uint16_t)v_clamped;
2777                        v_clamped = (vr + (1 << 11)) >> 12;
2778                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2779                        rightVol = (uint16_t)v_clamped;
2780                    } else {
2781                        leftVol = mLeftVolShort;
2782                        rightVol = mRightVolShort;
2783                        rampVolume = false;
2784                    }
2785
2786                    // reset retry count
2787                    track->mRetryCount = kMaxTrackRetriesDirect;
2788                    activeTrack = t;
2789                    mixerStatus = MIXER_TRACKS_READY;
2790                } else {
2791                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2792                    if (track->isStopped()) {
2793                        track->reset();
2794                    }
2795                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2796                        // We have consumed all the buffers of this track.
2797                        // Remove it from the list of active tracks.
2798                        trackToRemove = track;
2799                    } else {
2800                        // No buffers for this track. Give it a few chances to
2801                        // fill a buffer, then remove it from active list.
2802                        if (--(track->mRetryCount) <= 0) {
2803                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2804                            trackToRemove = track;
2805                        } else {
2806                            mixerStatus = MIXER_TRACKS_ENABLED;
2807                        }
2808                    }
2809                }
2810            }
2811
2812            // remove all the tracks that need to be...
2813            if (CC_UNLIKELY(trackToRemove != 0)) {
2814                mActiveTracks.remove(trackToRemove);
2815                if (!effectChains.isEmpty()) {
2816                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2817                            trackToRemove->sessionId());
2818                    effectChains[0]->decActiveTrackCnt();
2819                }
2820                if (trackToRemove->isTerminated()) {
2821                    removeTrack_l(trackToRemove);
2822                }
2823            }
2824
2825            lockEffectChains_l(effectChains);
2826       }
2827
2828        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2829            AudioBufferProvider::Buffer buffer;
2830            size_t frameCount = mFrameCount;
2831            curBuf = (int8_t *)mMixBuffer;
2832            // output audio to hardware
2833            while (frameCount) {
2834                buffer.frameCount = frameCount;
2835                activeTrack->getNextBuffer(&buffer);
2836                if (CC_UNLIKELY(buffer.raw == NULL)) {
2837                    memset(curBuf, 0, frameCount * mFrameSize);
2838                    break;
2839                }
2840                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2841                frameCount -= buffer.frameCount;
2842                curBuf += buffer.frameCount * mFrameSize;
2843                activeTrack->releaseBuffer(&buffer);
2844            }
2845            sleepTime = 0;
2846            standbyTime = systemTime() + standbyDelay;
2847        } else {
2848            if (sleepTime == 0) {
2849                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2850                    sleepTime = activeSleepTime;
2851                } else {
2852                    sleepTime = idleSleepTime;
2853                }
2854            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2855                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2856                sleepTime = 0;
2857            }
2858        }
2859
2860        if (mSuspended) {
2861            sleepTime = suspendSleepTimeUs();
2862        }
2863        // sleepTime == 0 means we must write to audio hardware
2864        if (sleepTime == 0) {
2865            if (mixerStatus == MIXER_TRACKS_READY) {
2866                applyVolume(leftVol, rightVol, rampVolume);
2867            }
2868            for (size_t i = 0; i < effectChains.size(); i ++) {
2869                effectChains[i]->process_l();
2870            }
2871            unlockEffectChains(effectChains);
2872
2873            mLastWriteTime = systemTime();
2874            mInWrite = true;
2875            mBytesWritten += mixBufferSize;
2876            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2877            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2878            mNumWrites++;
2879            mInWrite = false;
2880            mStandby = false;
2881        } else {
2882            unlockEffectChains(effectChains);
2883            usleep(sleepTime);
2884        }
2885
2886        // finally let go of removed track, without the lock held
2887        // since we can't guarantee the destructors won't acquire that
2888        // same lock.
2889        trackToRemove.clear();
2890        activeTrack.clear();
2891
2892        // Effect chains will be actually deleted here if they were removed from
2893        // mEffectChains list during mixing or effects processing
2894        effectChains.clear();
2895    }
2896
2897    if (!mStandby) {
2898        mOutput->stream->common.standby(&mOutput->stream->common);
2899    }
2900
2901    releaseWakeLock();
2902
2903    ALOGV("DirectOutputThread %p exiting", this);
2904    return false;
2905}
2906
2907// getTrackName_l() must be called with ThreadBase::mLock held
2908int AudioFlinger::DirectOutputThread::getTrackName_l()
2909{
2910    return 0;
2911}
2912
2913// deleteTrackName_l() must be called with ThreadBase::mLock held
2914void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2915{
2916}
2917
2918// checkForNewParameters_l() must be called with ThreadBase::mLock held
2919bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2920{
2921    bool reconfig = false;
2922
2923    while (!mNewParameters.isEmpty()) {
2924        status_t status = NO_ERROR;
2925        String8 keyValuePair = mNewParameters[0];
2926        AudioParameter param = AudioParameter(keyValuePair);
2927        int value;
2928
2929        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2930            // do not accept frame count changes if tracks are open as the track buffer
2931            // size depends on frame count and correct behavior would not be garantied
2932            // if frame count is changed after track creation
2933            if (!mTracks.isEmpty()) {
2934                status = INVALID_OPERATION;
2935            } else {
2936                reconfig = true;
2937            }
2938        }
2939        if (status == NO_ERROR) {
2940            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2941                                                    keyValuePair.string());
2942            if (!mStandby && status == INVALID_OPERATION) {
2943               mOutput->stream->common.standby(&mOutput->stream->common);
2944               mStandby = true;
2945               mBytesWritten = 0;
2946               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2947                                                       keyValuePair.string());
2948            }
2949            if (status == NO_ERROR && reconfig) {
2950                readOutputParameters();
2951                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2952            }
2953        }
2954
2955        mNewParameters.removeAt(0);
2956
2957        mParamStatus = status;
2958        mParamCond.signal();
2959        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2960        // already timed out waiting for the status and will never signal the condition.
2961        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2962    }
2963    return reconfig;
2964}
2965
2966uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2967{
2968    uint32_t time;
2969    if (audio_is_linear_pcm(mFormat)) {
2970        time = PlaybackThread::activeSleepTimeUs();
2971    } else {
2972        time = 10000;
2973    }
2974    return time;
2975}
2976
2977uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2978{
2979    uint32_t time;
2980    if (audio_is_linear_pcm(mFormat)) {
2981        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2982    } else {
2983        time = 10000;
2984    }
2985    return time;
2986}
2987
2988uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2989{
2990    uint32_t time;
2991    if (audio_is_linear_pcm(mFormat)) {
2992        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2993    } else {
2994        time = 10000;
2995    }
2996    return time;
2997}
2998
2999
3000// ----------------------------------------------------------------------------
3001
3002AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
3003        AudioFlinger::MixerThread* mainThread, int id)
3004    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING),
3005        mWaitTimeMs(UINT_MAX)
3006{
3007    addOutputTrack(mainThread);
3008}
3009
3010AudioFlinger::DuplicatingThread::~DuplicatingThread()
3011{
3012    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3013        mOutputTracks[i]->destroy();
3014    }
3015    mOutputTracks.clear();
3016}
3017
3018bool AudioFlinger::DuplicatingThread::threadLoop()
3019{
3020    Vector< sp<Track> > tracksToRemove;
3021    mixer_state mixerStatus = MIXER_IDLE;
3022    nsecs_t standbyTime = systemTime();
3023    size_t mixBufferSize = mFrameCount*mFrameSize;
3024    SortedVector< sp<OutputTrack> > outputTracks;
3025    uint32_t writeFrames = 0;
3026    uint32_t activeSleepTime = activeSleepTimeUs();
3027    uint32_t idleSleepTime = idleSleepTimeUs();
3028    uint32_t sleepTime = idleSleepTime;
3029    Vector< sp<EffectChain> > effectChains;
3030
3031    acquireWakeLock();
3032
3033    while (!exitPending())
3034    {
3035        processConfigEvents();
3036
3037        mixerStatus = MIXER_IDLE;
3038        { // scope for the mLock
3039
3040            Mutex::Autolock _l(mLock);
3041
3042            if (checkForNewParameters_l()) {
3043                mixBufferSize = mFrameCount*mFrameSize;
3044                updateWaitTime();
3045                activeSleepTime = activeSleepTimeUs();
3046                idleSleepTime = idleSleepTimeUs();
3047            }
3048
3049            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
3050
3051            for (size_t i = 0; i < mOutputTracks.size(); i++) {
3052                outputTracks.add(mOutputTracks[i]);
3053            }
3054
3055            // put audio hardware into standby after short delay
3056            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
3057                         mSuspended)) {
3058                if (!mStandby) {
3059                    for (size_t i = 0; i < outputTracks.size(); i++) {
3060                        outputTracks[i]->stop();
3061                    }
3062                    mStandby = true;
3063                    mBytesWritten = 0;
3064                }
3065
3066                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
3067                    // we're about to wait, flush the binder command buffer
3068                    IPCThreadState::self()->flushCommands();
3069                    outputTracks.clear();
3070
3071                    if (exitPending()) break;
3072
3073                    releaseWakeLock_l();
3074                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3075                    mWaitWorkCV.wait(mLock);
3076                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3077                    acquireWakeLock_l();
3078
3079                    mPrevMixerStatus = MIXER_IDLE;
3080                    if (!mMasterMute) {
3081                        char value[PROPERTY_VALUE_MAX];
3082                        property_get("ro.audio.silent", value, "0");
3083                        if (atoi(value)) {
3084                            ALOGD("Silence is golden");
3085                            setMasterMute(true);
3086                        }
3087                    }
3088
3089                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3090                    sleepTime = idleSleepTime;
3091                    continue;
3092                }
3093            }
3094
3095            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3096
3097            // prevent any changes in effect chain list and in each effect chain
3098            // during mixing and effect process as the audio buffers could be deleted
3099            // or modified if an effect is created or deleted
3100            lockEffectChains_l(effectChains);
3101        }
3102
3103        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3104            // mix buffers...
3105            if (outputsReady(outputTracks)) {
3106                mAudioMixer->process();
3107            } else {
3108                memset(mMixBuffer, 0, mixBufferSize);
3109            }
3110            sleepTime = 0;
3111            writeFrames = mFrameCount;
3112        } else {
3113            if (sleepTime == 0) {
3114                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3115                    sleepTime = activeSleepTime;
3116                } else {
3117                    sleepTime = idleSleepTime;
3118                }
3119            } else if (mBytesWritten != 0) {
3120                // flush remaining overflow buffers in output tracks
3121                for (size_t i = 0; i < outputTracks.size(); i++) {
3122                    if (outputTracks[i]->isActive()) {
3123                        sleepTime = 0;
3124                        writeFrames = 0;
3125                        memset(mMixBuffer, 0, mixBufferSize);
3126                        break;
3127                    }
3128                }
3129            }
3130        }
3131
3132        if (mSuspended) {
3133            sleepTime = suspendSleepTimeUs();
3134        }
3135        // sleepTime == 0 means we must write to audio hardware
3136        if (sleepTime == 0) {
3137            for (size_t i = 0; i < effectChains.size(); i ++) {
3138                effectChains[i]->process_l();
3139            }
3140            // enable changes in effect chain
3141            unlockEffectChains(effectChains);
3142
3143            standbyTime = systemTime() + kStandbyTimeInNsecs;
3144            for (size_t i = 0; i < outputTracks.size(); i++) {
3145                outputTracks[i]->write(mMixBuffer, writeFrames);
3146            }
3147            mStandby = false;
3148            mBytesWritten += mixBufferSize;
3149        } else {
3150            // enable changes in effect chain
3151            unlockEffectChains(effectChains);
3152            usleep(sleepTime);
3153        }
3154
3155        // finally let go of all our tracks, without the lock held
3156        // since we can't guarantee the destructors won't acquire that
3157        // same lock.
3158        tracksToRemove.clear();
3159        outputTracks.clear();
3160
3161        // Effect chains will be actually deleted here if they were removed from
3162        // mEffectChains list during mixing or effects processing
3163        effectChains.clear();
3164    }
3165
3166    releaseWakeLock();
3167
3168    return false;
3169}
3170
3171void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3172{
3173    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3174    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3175                                            this,
3176                                            mSampleRate,
3177                                            mFormat,
3178                                            mChannelMask,
3179                                            frameCount);
3180    if (outputTrack->cblk() != NULL) {
3181        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3182        mOutputTracks.add(outputTrack);
3183        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3184        updateWaitTime();
3185    }
3186}
3187
3188void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3189{
3190    Mutex::Autolock _l(mLock);
3191    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3192        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3193            mOutputTracks[i]->destroy();
3194            mOutputTracks.removeAt(i);
3195            updateWaitTime();
3196            return;
3197        }
3198    }
3199    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3200}
3201
3202void AudioFlinger::DuplicatingThread::updateWaitTime()
3203{
3204    mWaitTimeMs = UINT_MAX;
3205    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3206        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3207        if (strong != 0) {
3208            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3209            if (waitTimeMs < mWaitTimeMs) {
3210                mWaitTimeMs = waitTimeMs;
3211            }
3212        }
3213    }
3214}
3215
3216
3217bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3218{
3219    for (size_t i = 0; i < outputTracks.size(); i++) {
3220        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3221        if (thread == 0) {
3222            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3223            return false;
3224        }
3225        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3226        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3227            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3228            return false;
3229        }
3230    }
3231    return true;
3232}
3233
3234uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3235{
3236    return (mWaitTimeMs * 1000) / 2;
3237}
3238
3239// ----------------------------------------------------------------------------
3240
3241// TrackBase constructor must be called with AudioFlinger::mLock held
3242AudioFlinger::ThreadBase::TrackBase::TrackBase(
3243            const wp<ThreadBase>& thread,
3244            const sp<Client>& client,
3245            uint32_t sampleRate,
3246            audio_format_t format,
3247            uint32_t channelMask,
3248            int frameCount,
3249            uint32_t flags,
3250            const sp<IMemory>& sharedBuffer,
3251            int sessionId)
3252    :   RefBase(),
3253        mThread(thread),
3254        mClient(client),
3255        mCblk(NULL),
3256        // mBuffer
3257        // mBufferEnd
3258        mFrameCount(0),
3259        mState(IDLE),
3260        mFormat(format),
3261        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3262        mSessionId(sessionId)
3263        // mChannelCount
3264        // mChannelMask
3265{
3266    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3267
3268    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3269   size_t size = sizeof(audio_track_cblk_t);
3270   uint8_t channelCount = popcount(channelMask);
3271   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3272   if (sharedBuffer == 0) {
3273       size += bufferSize;
3274   }
3275
3276   if (client != NULL) {
3277        mCblkMemory = client->heap()->allocate(size);
3278        if (mCblkMemory != 0) {
3279            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3280            if (mCblk != NULL) { // construct the shared structure in-place.
3281                new(mCblk) audio_track_cblk_t();
3282                // clear all buffers
3283                mCblk->frameCount = frameCount;
3284                mCblk->sampleRate = sampleRate;
3285                mChannelCount = channelCount;
3286                mChannelMask = channelMask;
3287                if (sharedBuffer == 0) {
3288                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3289                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3290                    // Force underrun condition to avoid false underrun callback until first data is
3291                    // written to buffer (other flags are cleared)
3292                    mCblk->flags = CBLK_UNDERRUN_ON;
3293                } else {
3294                    mBuffer = sharedBuffer->pointer();
3295                }
3296                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3297            }
3298        } else {
3299            ALOGE("not enough memory for AudioTrack size=%u", size);
3300            client->heap()->dump("AudioTrack");
3301            return;
3302        }
3303   } else {
3304       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3305           // construct the shared structure in-place.
3306           new(mCblk) audio_track_cblk_t();
3307           // clear all buffers
3308           mCblk->frameCount = frameCount;
3309           mCblk->sampleRate = sampleRate;
3310           mChannelCount = channelCount;
3311           mChannelMask = channelMask;
3312           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3313           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3314           // Force underrun condition to avoid false underrun callback until first data is
3315           // written to buffer (other flags are cleared)
3316           mCblk->flags = CBLK_UNDERRUN_ON;
3317           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3318   }
3319}
3320
3321AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3322{
3323    if (mCblk != NULL) {
3324        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3325        if (mClient == NULL) {
3326            delete mCblk;
3327        }
3328    }
3329    mCblkMemory.clear();            // and free the shared memory
3330    if (mClient != 0) {
3331        // Client destructor must run with AudioFlinger mutex locked
3332        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3333        // If the client's reference count drops to zero, the associated destructor
3334        // must run with AudioFlinger lock held. Thus the explicit clear() rather than
3335        // relying on the automatic clear() at end of scope.
3336        mClient.clear();
3337    }
3338}
3339
3340void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3341{
3342    buffer->raw = NULL;
3343    mFrameCount = buffer->frameCount;
3344    step();
3345    buffer->frameCount = 0;
3346}
3347
3348bool AudioFlinger::ThreadBase::TrackBase::step() {
3349    bool result;
3350    audio_track_cblk_t* cblk = this->cblk();
3351
3352    result = cblk->stepServer(mFrameCount);
3353    if (!result) {
3354        ALOGV("stepServer failed acquiring cblk mutex");
3355        mFlags |= STEPSERVER_FAILED;
3356    }
3357    return result;
3358}
3359
3360void AudioFlinger::ThreadBase::TrackBase::reset() {
3361    audio_track_cblk_t* cblk = this->cblk();
3362
3363    cblk->user = 0;
3364    cblk->server = 0;
3365    cblk->userBase = 0;
3366    cblk->serverBase = 0;
3367    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3368    ALOGV("TrackBase::reset");
3369}
3370
3371sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3372{
3373    return mCblkMemory;
3374}
3375
3376int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3377    return (int)mCblk->sampleRate;
3378}
3379
3380int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3381    return (const int)mChannelCount;
3382}
3383
3384uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3385    return mChannelMask;
3386}
3387
3388void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3389    audio_track_cblk_t* cblk = this->cblk();
3390    size_t frameSize = cblk->frameSize;
3391    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
3392    int8_t *bufferEnd = bufferStart + frames * frameSize;
3393
3394    // Check validity of returned pointer in case the track control block would have been corrupted.
3395    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3396        ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) {
3397        ALOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3398                server %d, serverBase %d, user %d, userBase %d",
3399                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3400                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3401        return NULL;
3402    }
3403
3404    return bufferStart;
3405}
3406
3407// ----------------------------------------------------------------------------
3408
3409// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3410AudioFlinger::PlaybackThread::Track::Track(
3411            const wp<ThreadBase>& thread,
3412            const sp<Client>& client,
3413            audio_stream_type_t streamType,
3414            uint32_t sampleRate,
3415            audio_format_t format,
3416            uint32_t channelMask,
3417            int frameCount,
3418            const sp<IMemory>& sharedBuffer,
3419            int sessionId)
3420    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3421    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3422    mAuxEffectId(0), mHasVolumeController(false)
3423{
3424    if (mCblk != NULL) {
3425        sp<ThreadBase> baseThread = thread.promote();
3426        if (baseThread != 0) {
3427            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3428            mName = playbackThread->getTrackName_l();
3429            mMainBuffer = playbackThread->mixBuffer();
3430        }
3431        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3432        if (mName < 0) {
3433            ALOGE("no more track names available");
3434        }
3435        mStreamType = streamType;
3436        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3437        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3438        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3439    }
3440}
3441
3442AudioFlinger::PlaybackThread::Track::~Track()
3443{
3444    ALOGV("PlaybackThread::Track destructor");
3445    sp<ThreadBase> thread = mThread.promote();
3446    if (thread != 0) {
3447        Mutex::Autolock _l(thread->mLock);
3448        mState = TERMINATED;
3449    }
3450}
3451
3452void AudioFlinger::PlaybackThread::Track::destroy()
3453{
3454    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3455    // by removing it from mTracks vector, so there is a risk that this Tracks's
3456    // desctructor is called. As the destructor needs to lock mLock,
3457    // we must acquire a strong reference on this Track before locking mLock
3458    // here so that the destructor is called only when exiting this function.
3459    // On the other hand, as long as Track::destroy() is only called by
3460    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3461    // this Track with its member mTrack.
3462    sp<Track> keep(this);
3463    { // scope for mLock
3464        sp<ThreadBase> thread = mThread.promote();
3465        if (thread != 0) {
3466            if (!isOutputTrack()) {
3467                if (mState == ACTIVE || mState == RESUMING) {
3468                    AudioSystem::stopOutput(thread->id(),
3469                                            (audio_stream_type_t)mStreamType,
3470                                            mSessionId);
3471
3472                    // to track the speaker usage
3473                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3474                }
3475                AudioSystem::releaseOutput(thread->id());
3476            }
3477            Mutex::Autolock _l(thread->mLock);
3478            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3479            playbackThread->destroyTrack_l(this);
3480        }
3481    }
3482}
3483
3484void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3485{
3486    uint32_t vlr = mCblk->getVolumeLR();
3487    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",
3488            mName - AudioMixer::TRACK0,
3489            (mClient == 0) ? getpid() : mClient->pid(),
3490            mStreamType,
3491            mFormat,
3492            mChannelMask,
3493            mSessionId,
3494            mFrameCount,
3495            mState,
3496            mMute,
3497            mFillingUpStatus,
3498            mCblk->sampleRate,
3499            vlr & 0xFFFF,
3500            vlr >> 16,
3501            mCblk->server,
3502            mCblk->user,
3503            (int)mMainBuffer,
3504            (int)mAuxBuffer);
3505}
3506
3507status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3508{
3509     audio_track_cblk_t* cblk = this->cblk();
3510     uint32_t framesReady;
3511     uint32_t framesReq = buffer->frameCount;
3512
3513     // Check if last stepServer failed, try to step now
3514     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3515         if (!step())  goto getNextBuffer_exit;
3516         ALOGV("stepServer recovered");
3517         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3518     }
3519
3520     framesReady = cblk->framesReady();
3521
3522     if (CC_LIKELY(framesReady)) {
3523        uint32_t s = cblk->server;
3524        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3525
3526        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3527        if (framesReq > framesReady) {
3528            framesReq = framesReady;
3529        }
3530        if (s + framesReq > bufferEnd) {
3531            framesReq = bufferEnd - s;
3532        }
3533
3534         buffer->raw = getBuffer(s, framesReq);
3535         if (buffer->raw == NULL) goto getNextBuffer_exit;
3536
3537         buffer->frameCount = framesReq;
3538        return NO_ERROR;
3539     }
3540
3541getNextBuffer_exit:
3542     buffer->raw = NULL;
3543     buffer->frameCount = 0;
3544     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3545     return NOT_ENOUGH_DATA;
3546}
3547
3548bool AudioFlinger::PlaybackThread::Track::isReady() const {
3549    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3550
3551    if (mCblk->framesReady() >= mCblk->frameCount ||
3552            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3553        mFillingUpStatus = FS_FILLED;
3554        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3555        return true;
3556    }
3557    return false;
3558}
3559
3560status_t AudioFlinger::PlaybackThread::Track::start()
3561{
3562    status_t status = NO_ERROR;
3563    ALOGV("start(%d), calling thread %d session %d",
3564            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3565    sp<ThreadBase> thread = mThread.promote();
3566    if (thread != 0) {
3567        Mutex::Autolock _l(thread->mLock);
3568        track_state state = mState;
3569        // here the track could be either new, or restarted
3570        // in both cases "unstop" the track
3571        if (mState == PAUSED) {
3572            mState = TrackBase::RESUMING;
3573            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3574        } else {
3575            mState = TrackBase::ACTIVE;
3576            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3577        }
3578
3579        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3580            thread->mLock.unlock();
3581            status = AudioSystem::startOutput(thread->id(),
3582                                              (audio_stream_type_t)mStreamType,
3583                                              mSessionId);
3584            thread->mLock.lock();
3585
3586            // to track the speaker usage
3587            if (status == NO_ERROR) {
3588                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3589            }
3590        }
3591        if (status == NO_ERROR) {
3592            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3593            playbackThread->addTrack_l(this);
3594        } else {
3595            mState = state;
3596        }
3597    } else {
3598        status = BAD_VALUE;
3599    }
3600    return status;
3601}
3602
3603void AudioFlinger::PlaybackThread::Track::stop()
3604{
3605    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3606    sp<ThreadBase> thread = mThread.promote();
3607    if (thread != 0) {
3608        Mutex::Autolock _l(thread->mLock);
3609        track_state state = mState;
3610        if (mState > STOPPED) {
3611            mState = STOPPED;
3612            // If the track is not active (PAUSED and buffers full), flush buffers
3613            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3614            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3615                reset();
3616            }
3617            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3618        }
3619        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3620            thread->mLock.unlock();
3621            AudioSystem::stopOutput(thread->id(),
3622                                    (audio_stream_type_t)mStreamType,
3623                                    mSessionId);
3624            thread->mLock.lock();
3625
3626            // to track the speaker usage
3627            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3628        }
3629    }
3630}
3631
3632void AudioFlinger::PlaybackThread::Track::pause()
3633{
3634    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3635    sp<ThreadBase> thread = mThread.promote();
3636    if (thread != 0) {
3637        Mutex::Autolock _l(thread->mLock);
3638        if (mState == ACTIVE || mState == RESUMING) {
3639            mState = PAUSING;
3640            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3641            if (!isOutputTrack()) {
3642                thread->mLock.unlock();
3643                AudioSystem::stopOutput(thread->id(),
3644                                        (audio_stream_type_t)mStreamType,
3645                                        mSessionId);
3646                thread->mLock.lock();
3647
3648                // to track the speaker usage
3649                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3650            }
3651        }
3652    }
3653}
3654
3655void AudioFlinger::PlaybackThread::Track::flush()
3656{
3657    ALOGV("flush(%d)", mName);
3658    sp<ThreadBase> thread = mThread.promote();
3659    if (thread != 0) {
3660        Mutex::Autolock _l(thread->mLock);
3661        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3662            return;
3663        }
3664        // No point remaining in PAUSED state after a flush => go to
3665        // STOPPED state
3666        mState = STOPPED;
3667
3668        // do not reset the track if it is still in the process of being stopped or paused.
3669        // this will be done by prepareTracks_l() when the track is stopped.
3670        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3671        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3672            reset();
3673        }
3674    }
3675}
3676
3677void AudioFlinger::PlaybackThread::Track::reset()
3678{
3679    // Do not reset twice to avoid discarding data written just after a flush and before
3680    // the audioflinger thread detects the track is stopped.
3681    if (!mResetDone) {
3682        TrackBase::reset();
3683        // Force underrun condition to avoid false underrun callback until first data is
3684        // written to buffer
3685        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3686        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3687        mFillingUpStatus = FS_FILLING;
3688        mResetDone = true;
3689    }
3690}
3691
3692void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3693{
3694    mMute = muted;
3695}
3696
3697status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3698{
3699    status_t status = DEAD_OBJECT;
3700    sp<ThreadBase> thread = mThread.promote();
3701    if (thread != 0) {
3702       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3703       status = playbackThread->attachAuxEffect(this, EffectId);
3704    }
3705    return status;
3706}
3707
3708void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3709{
3710    mAuxEffectId = EffectId;
3711    mAuxBuffer = buffer;
3712}
3713
3714// ----------------------------------------------------------------------------
3715
3716// RecordTrack constructor must be called with AudioFlinger::mLock held
3717AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3718            const wp<ThreadBase>& thread,
3719            const sp<Client>& client,
3720            uint32_t sampleRate,
3721            audio_format_t format,
3722            uint32_t channelMask,
3723            int frameCount,
3724            uint32_t flags,
3725            int sessionId)
3726    :   TrackBase(thread, client, sampleRate, format,
3727                  channelMask, frameCount, flags, 0, sessionId),
3728        mOverflow(false)
3729{
3730    if (mCblk != NULL) {
3731       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3732       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3733           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3734       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3735           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3736       } else {
3737           mCblk->frameSize = sizeof(int8_t);
3738       }
3739    }
3740}
3741
3742AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3743{
3744    sp<ThreadBase> thread = mThread.promote();
3745    if (thread != 0) {
3746        AudioSystem::releaseInput(thread->id());
3747    }
3748}
3749
3750status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3751{
3752    audio_track_cblk_t* cblk = this->cblk();
3753    uint32_t framesAvail;
3754    uint32_t framesReq = buffer->frameCount;
3755
3756     // Check if last stepServer failed, try to step now
3757    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3758        if (!step()) goto getNextBuffer_exit;
3759        ALOGV("stepServer recovered");
3760        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3761    }
3762
3763    framesAvail = cblk->framesAvailable_l();
3764
3765    if (CC_LIKELY(framesAvail)) {
3766        uint32_t s = cblk->server;
3767        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3768
3769        if (framesReq > framesAvail) {
3770            framesReq = framesAvail;
3771        }
3772        if (s + framesReq > bufferEnd) {
3773            framesReq = bufferEnd - s;
3774        }
3775
3776        buffer->raw = getBuffer(s, framesReq);
3777        if (buffer->raw == NULL) goto getNextBuffer_exit;
3778
3779        buffer->frameCount = framesReq;
3780        return NO_ERROR;
3781    }
3782
3783getNextBuffer_exit:
3784    buffer->raw = NULL;
3785    buffer->frameCount = 0;
3786    return NOT_ENOUGH_DATA;
3787}
3788
3789status_t AudioFlinger::RecordThread::RecordTrack::start()
3790{
3791    sp<ThreadBase> thread = mThread.promote();
3792    if (thread != 0) {
3793        RecordThread *recordThread = (RecordThread *)thread.get();
3794        return recordThread->start(this);
3795    } else {
3796        return BAD_VALUE;
3797    }
3798}
3799
3800void AudioFlinger::RecordThread::RecordTrack::stop()
3801{
3802    sp<ThreadBase> thread = mThread.promote();
3803    if (thread != 0) {
3804        RecordThread *recordThread = (RecordThread *)thread.get();
3805        recordThread->stop(this);
3806        TrackBase::reset();
3807        // Force overerrun condition to avoid false overrun callback until first data is
3808        // read from buffer
3809        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3810    }
3811}
3812
3813void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3814{
3815    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3816            (mClient == 0) ? getpid() : mClient->pid(),
3817            mFormat,
3818            mChannelMask,
3819            mSessionId,
3820            mFrameCount,
3821            mState,
3822            mCblk->sampleRate,
3823            mCblk->server,
3824            mCblk->user);
3825}
3826
3827
3828// ----------------------------------------------------------------------------
3829
3830AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3831            const wp<ThreadBase>& thread,
3832            DuplicatingThread *sourceThread,
3833            uint32_t sampleRate,
3834            audio_format_t format,
3835            uint32_t channelMask,
3836            int frameCount)
3837    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3838    mActive(false), mSourceThread(sourceThread)
3839{
3840
3841    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3842    if (mCblk != NULL) {
3843        mCblk->flags |= CBLK_DIRECTION_OUT;
3844        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3845        mOutBuffer.frameCount = 0;
3846        playbackThread->mTracks.add(this);
3847        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3848                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3849                mCblk, mBuffer, mCblk->buffers,
3850                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3851    } else {
3852        ALOGW("Error creating output track on thread %p", playbackThread);
3853    }
3854}
3855
3856AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3857{
3858    clearBufferQueue();
3859}
3860
3861status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3862{
3863    status_t status = Track::start();
3864    if (status != NO_ERROR) {
3865        return status;
3866    }
3867
3868    mActive = true;
3869    mRetryCount = 127;
3870    return status;
3871}
3872
3873void AudioFlinger::PlaybackThread::OutputTrack::stop()
3874{
3875    Track::stop();
3876    clearBufferQueue();
3877    mOutBuffer.frameCount = 0;
3878    mActive = false;
3879}
3880
3881bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3882{
3883    Buffer *pInBuffer;
3884    Buffer inBuffer;
3885    uint32_t channelCount = mChannelCount;
3886    bool outputBufferFull = false;
3887    inBuffer.frameCount = frames;
3888    inBuffer.i16 = data;
3889
3890    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3891
3892    if (!mActive && frames != 0) {
3893        start();
3894        sp<ThreadBase> thread = mThread.promote();
3895        if (thread != 0) {
3896            MixerThread *mixerThread = (MixerThread *)thread.get();
3897            if (mCblk->frameCount > frames){
3898                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3899                    uint32_t startFrames = (mCblk->frameCount - frames);
3900                    pInBuffer = new Buffer;
3901                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3902                    pInBuffer->frameCount = startFrames;
3903                    pInBuffer->i16 = pInBuffer->mBuffer;
3904                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3905                    mBufferQueue.add(pInBuffer);
3906                } else {
3907                    ALOGW ("OutputTrack::write() %p no more buffers in queue", this);
3908                }
3909            }
3910        }
3911    }
3912
3913    while (waitTimeLeftMs) {
3914        // First write pending buffers, then new data
3915        if (mBufferQueue.size()) {
3916            pInBuffer = mBufferQueue.itemAt(0);
3917        } else {
3918            pInBuffer = &inBuffer;
3919        }
3920
3921        if (pInBuffer->frameCount == 0) {
3922            break;
3923        }
3924
3925        if (mOutBuffer.frameCount == 0) {
3926            mOutBuffer.frameCount = pInBuffer->frameCount;
3927            nsecs_t startTime = systemTime();
3928            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
3929                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3930                outputBufferFull = true;
3931                break;
3932            }
3933            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3934            if (waitTimeLeftMs >= waitTimeMs) {
3935                waitTimeLeftMs -= waitTimeMs;
3936            } else {
3937                waitTimeLeftMs = 0;
3938            }
3939        }
3940
3941        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3942        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3943        mCblk->stepUser(outFrames);
3944        pInBuffer->frameCount -= outFrames;
3945        pInBuffer->i16 += outFrames * channelCount;
3946        mOutBuffer.frameCount -= outFrames;
3947        mOutBuffer.i16 += outFrames * channelCount;
3948
3949        if (pInBuffer->frameCount == 0) {
3950            if (mBufferQueue.size()) {
3951                mBufferQueue.removeAt(0);
3952                delete [] pInBuffer->mBuffer;
3953                delete pInBuffer;
3954                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3955            } else {
3956                break;
3957            }
3958        }
3959    }
3960
3961    // If we could not write all frames, allocate a buffer and queue it for next time.
3962    if (inBuffer.frameCount) {
3963        sp<ThreadBase> thread = mThread.promote();
3964        if (thread != 0 && !thread->standby()) {
3965            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3966                pInBuffer = new Buffer;
3967                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3968                pInBuffer->frameCount = inBuffer.frameCount;
3969                pInBuffer->i16 = pInBuffer->mBuffer;
3970                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3971                mBufferQueue.add(pInBuffer);
3972                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3973            } else {
3974                ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3975            }
3976        }
3977    }
3978
3979    // Calling write() with a 0 length buffer, means that no more data will be written:
3980    // If no more buffers are pending, fill output track buffer to make sure it is started
3981    // by output mixer.
3982    if (frames == 0 && mBufferQueue.size() == 0) {
3983        if (mCblk->user < mCblk->frameCount) {
3984            frames = mCblk->frameCount - mCblk->user;
3985            pInBuffer = new Buffer;
3986            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3987            pInBuffer->frameCount = frames;
3988            pInBuffer->i16 = pInBuffer->mBuffer;
3989            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3990            mBufferQueue.add(pInBuffer);
3991        } else if (mActive) {
3992            stop();
3993        }
3994    }
3995
3996    return outputBufferFull;
3997}
3998
3999status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
4000{
4001    int active;
4002    status_t result;
4003    audio_track_cblk_t* cblk = mCblk;
4004    uint32_t framesReq = buffer->frameCount;
4005
4006//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
4007    buffer->frameCount  = 0;
4008
4009    uint32_t framesAvail = cblk->framesAvailable();
4010
4011
4012    if (framesAvail == 0) {
4013        Mutex::Autolock _l(cblk->lock);
4014        goto start_loop_here;
4015        while (framesAvail == 0) {
4016            active = mActive;
4017            if (CC_UNLIKELY(!active)) {
4018                ALOGV("Not active and NO_MORE_BUFFERS");
4019                return NO_MORE_BUFFERS;
4020            }
4021            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
4022            if (result != NO_ERROR) {
4023                return NO_MORE_BUFFERS;
4024            }
4025            // read the server count again
4026        start_loop_here:
4027            framesAvail = cblk->framesAvailable_l();
4028        }
4029    }
4030
4031//    if (framesAvail < framesReq) {
4032//        return NO_MORE_BUFFERS;
4033//    }
4034
4035    if (framesReq > framesAvail) {
4036        framesReq = framesAvail;
4037    }
4038
4039    uint32_t u = cblk->user;
4040    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
4041
4042    if (u + framesReq > bufferEnd) {
4043        framesReq = bufferEnd - u;
4044    }
4045
4046    buffer->frameCount  = framesReq;
4047    buffer->raw         = (void *)cblk->buffer(u);
4048    return NO_ERROR;
4049}
4050
4051
4052void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
4053{
4054    size_t size = mBufferQueue.size();
4055    Buffer *pBuffer;
4056
4057    for (size_t i = 0; i < size; i++) {
4058        pBuffer = mBufferQueue.itemAt(i);
4059        delete [] pBuffer->mBuffer;
4060        delete pBuffer;
4061    }
4062    mBufferQueue.clear();
4063}
4064
4065// ----------------------------------------------------------------------------
4066
4067AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
4068    :   RefBase(),
4069        mAudioFlinger(audioFlinger),
4070        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
4071        mPid(pid)
4072{
4073    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
4074}
4075
4076// Client destructor must be called with AudioFlinger::mLock held
4077AudioFlinger::Client::~Client()
4078{
4079    mAudioFlinger->removeClient_l(mPid);
4080}
4081
4082sp<MemoryDealer> AudioFlinger::Client::heap() const
4083{
4084    return mMemoryDealer;
4085}
4086
4087// ----------------------------------------------------------------------------
4088
4089AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
4090                                                     const sp<IAudioFlingerClient>& client,
4091                                                     pid_t pid)
4092    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
4093{
4094}
4095
4096AudioFlinger::NotificationClient::~NotificationClient()
4097{
4098}
4099
4100void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
4101{
4102    sp<NotificationClient> keep(this);
4103    {
4104        mAudioFlinger->removeNotificationClient(mPid);
4105    }
4106}
4107
4108// ----------------------------------------------------------------------------
4109
4110AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4111    : BnAudioTrack(),
4112      mTrack(track)
4113{
4114}
4115
4116AudioFlinger::TrackHandle::~TrackHandle() {
4117    // just stop the track on deletion, associated resources
4118    // will be freed from the main thread once all pending buffers have
4119    // been played. Unless it's not in the active track list, in which
4120    // case we free everything now...
4121    mTrack->destroy();
4122}
4123
4124sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4125    return mTrack->getCblk();
4126}
4127
4128status_t AudioFlinger::TrackHandle::start() {
4129    return mTrack->start();
4130}
4131
4132void AudioFlinger::TrackHandle::stop() {
4133    mTrack->stop();
4134}
4135
4136void AudioFlinger::TrackHandle::flush() {
4137    mTrack->flush();
4138}
4139
4140void AudioFlinger::TrackHandle::mute(bool e) {
4141    mTrack->mute(e);
4142}
4143
4144void AudioFlinger::TrackHandle::pause() {
4145    mTrack->pause();
4146}
4147
4148status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4149{
4150    return mTrack->attachAuxEffect(EffectId);
4151}
4152
4153status_t AudioFlinger::TrackHandle::onTransact(
4154    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4155{
4156    return BnAudioTrack::onTransact(code, data, reply, flags);
4157}
4158
4159// ----------------------------------------------------------------------------
4160
4161sp<IAudioRecord> AudioFlinger::openRecord(
4162        pid_t pid,
4163        int input,
4164        uint32_t sampleRate,
4165        audio_format_t format,
4166        uint32_t channelMask,
4167        int frameCount,
4168        uint32_t flags,
4169        int *sessionId,
4170        status_t *status)
4171{
4172    sp<RecordThread::RecordTrack> recordTrack;
4173    sp<RecordHandle> recordHandle;
4174    sp<Client> client;
4175    wp<Client> wclient;
4176    status_t lStatus;
4177    RecordThread *thread;
4178    size_t inFrameCount;
4179    int lSessionId;
4180
4181    // check calling permissions
4182    if (!recordingAllowed()) {
4183        lStatus = PERMISSION_DENIED;
4184        goto Exit;
4185    }
4186
4187    // add client to list
4188    { // scope for mLock
4189        Mutex::Autolock _l(mLock);
4190        thread = checkRecordThread_l(input);
4191        if (thread == NULL) {
4192            lStatus = BAD_VALUE;
4193            goto Exit;
4194        }
4195
4196        wclient = mClients.valueFor(pid);
4197        if (wclient != NULL) {
4198            client = wclient.promote();
4199        } else {
4200            client = new Client(this, pid);
4201            mClients.add(pid, client);
4202        }
4203
4204        // If no audio session id is provided, create one here
4205        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4206            lSessionId = *sessionId;
4207        } else {
4208            lSessionId = nextUniqueId();
4209            if (sessionId != NULL) {
4210                *sessionId = lSessionId;
4211            }
4212        }
4213        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4214        recordTrack = thread->createRecordTrack_l(client,
4215                                                sampleRate,
4216                                                format,
4217                                                channelMask,
4218                                                frameCount,
4219                                                flags,
4220                                                lSessionId,
4221                                                &lStatus);
4222    }
4223    if (lStatus != NO_ERROR) {
4224        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4225        // destructor is called by the TrackBase destructor with mLock held
4226        client.clear();
4227        recordTrack.clear();
4228        goto Exit;
4229    }
4230
4231    // return to handle to client
4232    recordHandle = new RecordHandle(recordTrack);
4233    lStatus = NO_ERROR;
4234
4235Exit:
4236    if (status) {
4237        *status = lStatus;
4238    }
4239    return recordHandle;
4240}
4241
4242// ----------------------------------------------------------------------------
4243
4244AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4245    : BnAudioRecord(),
4246    mRecordTrack(recordTrack)
4247{
4248}
4249
4250AudioFlinger::RecordHandle::~RecordHandle() {
4251    stop();
4252}
4253
4254sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4255    return mRecordTrack->getCblk();
4256}
4257
4258status_t AudioFlinger::RecordHandle::start() {
4259    ALOGV("RecordHandle::start()");
4260    return mRecordTrack->start();
4261}
4262
4263void AudioFlinger::RecordHandle::stop() {
4264    ALOGV("RecordHandle::stop()");
4265    mRecordTrack->stop();
4266}
4267
4268status_t AudioFlinger::RecordHandle::onTransact(
4269    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4270{
4271    return BnAudioRecord::onTransact(code, data, reply, flags);
4272}
4273
4274// ----------------------------------------------------------------------------
4275
4276AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4277                                         AudioStreamIn *input,
4278                                         uint32_t sampleRate,
4279                                         uint32_t channels,
4280                                         int id,
4281                                         uint32_t device) :
4282    ThreadBase(audioFlinger, id, device, RECORD),
4283    mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
4284    // mRsmpInIndex and mInputBytes set by readInputParameters()
4285    mReqChannelCount(popcount(channels)),
4286    mReqSampleRate(sampleRate)
4287    // mBytesRead is only meaningful while active, and so is cleared in start()
4288    // (but might be better to also clear here for dump?)
4289{
4290    snprintf(mName, kNameLength, "AudioIn_%d", id);
4291
4292    readInputParameters();
4293}
4294
4295
4296AudioFlinger::RecordThread::~RecordThread()
4297{
4298    delete[] mRsmpInBuffer;
4299    delete mResampler;
4300    delete[] mRsmpOutBuffer;
4301}
4302
4303void AudioFlinger::RecordThread::onFirstRef()
4304{
4305    run(mName, PRIORITY_URGENT_AUDIO);
4306}
4307
4308status_t AudioFlinger::RecordThread::readyToRun()
4309{
4310    status_t status = initCheck();
4311    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4312    return status;
4313}
4314
4315bool AudioFlinger::RecordThread::threadLoop()
4316{
4317    AudioBufferProvider::Buffer buffer;
4318    sp<RecordTrack> activeTrack;
4319    Vector< sp<EffectChain> > effectChains;
4320
4321    nsecs_t lastWarning = 0;
4322
4323    acquireWakeLock();
4324
4325    // start recording
4326    while (!exitPending()) {
4327
4328        processConfigEvents();
4329
4330        { // scope for mLock
4331            Mutex::Autolock _l(mLock);
4332            checkForNewParameters_l();
4333            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4334                if (!mStandby) {
4335                    mInput->stream->common.standby(&mInput->stream->common);
4336                    mStandby = true;
4337                }
4338
4339                if (exitPending()) break;
4340
4341                releaseWakeLock_l();
4342                ALOGV("RecordThread: loop stopping");
4343                // go to sleep
4344                mWaitWorkCV.wait(mLock);
4345                ALOGV("RecordThread: loop starting");
4346                acquireWakeLock_l();
4347                continue;
4348            }
4349            if (mActiveTrack != 0) {
4350                if (mActiveTrack->mState == TrackBase::PAUSING) {
4351                    if (!mStandby) {
4352                        mInput->stream->common.standby(&mInput->stream->common);
4353                        mStandby = true;
4354                    }
4355                    mActiveTrack.clear();
4356                    mStartStopCond.broadcast();
4357                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4358                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4359                        mActiveTrack.clear();
4360                        mStartStopCond.broadcast();
4361                    } else if (mBytesRead != 0) {
4362                        // record start succeeds only if first read from audio input
4363                        // succeeds
4364                        if (mBytesRead > 0) {
4365                            mActiveTrack->mState = TrackBase::ACTIVE;
4366                        } else {
4367                            mActiveTrack.clear();
4368                        }
4369                        mStartStopCond.broadcast();
4370                    }
4371                    mStandby = false;
4372                }
4373            }
4374            lockEffectChains_l(effectChains);
4375        }
4376
4377        if (mActiveTrack != 0) {
4378            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4379                mActiveTrack->mState != TrackBase::RESUMING) {
4380                unlockEffectChains(effectChains);
4381                usleep(kRecordThreadSleepUs);
4382                continue;
4383            }
4384            for (size_t i = 0; i < effectChains.size(); i ++) {
4385                effectChains[i]->process_l();
4386            }
4387
4388            buffer.frameCount = mFrameCount;
4389            if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4390                size_t framesOut = buffer.frameCount;
4391                if (mResampler == NULL) {
4392                    // no resampling
4393                    while (framesOut) {
4394                        size_t framesIn = mFrameCount - mRsmpInIndex;
4395                        if (framesIn) {
4396                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4397                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4398                            if (framesIn > framesOut)
4399                                framesIn = framesOut;
4400                            mRsmpInIndex += framesIn;
4401                            framesOut -= framesIn;
4402                            if ((int)mChannelCount == mReqChannelCount ||
4403                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4404                                memcpy(dst, src, framesIn * mFrameSize);
4405                            } else {
4406                                int16_t *src16 = (int16_t *)src;
4407                                int16_t *dst16 = (int16_t *)dst;
4408                                if (mChannelCount == 1) {
4409                                    while (framesIn--) {
4410                                        *dst16++ = *src16;
4411                                        *dst16++ = *src16++;
4412                                    }
4413                                } else {
4414                                    while (framesIn--) {
4415                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4416                                        src16 += 2;
4417                                    }
4418                                }
4419                            }
4420                        }
4421                        if (framesOut && mFrameCount == mRsmpInIndex) {
4422                            if (framesOut == mFrameCount &&
4423                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4424                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4425                                framesOut = 0;
4426                            } else {
4427                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4428                                mRsmpInIndex = 0;
4429                            }
4430                            if (mBytesRead < 0) {
4431                                ALOGE("Error reading audio input");
4432                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4433                                    // Force input into standby so that it tries to
4434                                    // recover at next read attempt
4435                                    mInput->stream->common.standby(&mInput->stream->common);
4436                                    usleep(kRecordThreadSleepUs);
4437                                }
4438                                mRsmpInIndex = mFrameCount;
4439                                framesOut = 0;
4440                                buffer.frameCount = 0;
4441                            }
4442                        }
4443                    }
4444                } else {
4445                    // resampling
4446
4447                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4448                    // alter output frame count as if we were expecting stereo samples
4449                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4450                        framesOut >>= 1;
4451                    }
4452                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4453                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4454                    // are 32 bit aligned which should be always true.
4455                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4456                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4457                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4458                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4459                        int16_t *dst = buffer.i16;
4460                        while (framesOut--) {
4461                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4462                            src += 2;
4463                        }
4464                    } else {
4465                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4466                    }
4467
4468                }
4469                mActiveTrack->releaseBuffer(&buffer);
4470                mActiveTrack->overflow();
4471            }
4472            // client isn't retrieving buffers fast enough
4473            else {
4474                if (!mActiveTrack->setOverflow()) {
4475                    nsecs_t now = systemTime();
4476                    if ((now - lastWarning) > kWarningThrottleNs) {
4477                        ALOGW("RecordThread: buffer overflow");
4478                        lastWarning = now;
4479                    }
4480                }
4481                // Release the processor for a while before asking for a new buffer.
4482                // This will give the application more chance to read from the buffer and
4483                // clear the overflow.
4484                usleep(kRecordThreadSleepUs);
4485            }
4486        }
4487        // enable changes in effect chain
4488        unlockEffectChains(effectChains);
4489        effectChains.clear();
4490    }
4491
4492    if (!mStandby) {
4493        mInput->stream->common.standby(&mInput->stream->common);
4494    }
4495    mActiveTrack.clear();
4496
4497    mStartStopCond.broadcast();
4498
4499    releaseWakeLock();
4500
4501    ALOGV("RecordThread %p exiting", this);
4502    return false;
4503}
4504
4505
4506sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4507        const sp<AudioFlinger::Client>& client,
4508        uint32_t sampleRate,
4509        audio_format_t format,
4510        int channelMask,
4511        int frameCount,
4512        uint32_t flags,
4513        int sessionId,
4514        status_t *status)
4515{
4516    sp<RecordTrack> track;
4517    status_t lStatus;
4518
4519    lStatus = initCheck();
4520    if (lStatus != NO_ERROR) {
4521        ALOGE("Audio driver not initialized.");
4522        goto Exit;
4523    }
4524
4525    { // scope for mLock
4526        Mutex::Autolock _l(mLock);
4527
4528        track = new RecordTrack(this, client, sampleRate,
4529                      format, channelMask, frameCount, flags, sessionId);
4530
4531        if (track->getCblk() == 0) {
4532            lStatus = NO_MEMORY;
4533            goto Exit;
4534        }
4535
4536        mTrack = track.get();
4537        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4538        bool suspend = audio_is_bluetooth_sco_device(
4539                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4540        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4541        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4542    }
4543    lStatus = NO_ERROR;
4544
4545Exit:
4546    if (status) {
4547        *status = lStatus;
4548    }
4549    return track;
4550}
4551
4552status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4553{
4554    ALOGV("RecordThread::start");
4555    sp <ThreadBase> strongMe = this;
4556    status_t status = NO_ERROR;
4557    {
4558        AutoMutex lock(mLock);
4559        if (mActiveTrack != 0) {
4560            if (recordTrack != mActiveTrack.get()) {
4561                status = -EBUSY;
4562            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4563                mActiveTrack->mState = TrackBase::ACTIVE;
4564            }
4565            return status;
4566        }
4567
4568        recordTrack->mState = TrackBase::IDLE;
4569        mActiveTrack = recordTrack;
4570        mLock.unlock();
4571        status_t status = AudioSystem::startInput(mId);
4572        mLock.lock();
4573        if (status != NO_ERROR) {
4574            mActiveTrack.clear();
4575            return status;
4576        }
4577        mRsmpInIndex = mFrameCount;
4578        mBytesRead = 0;
4579        if (mResampler != NULL) {
4580            mResampler->reset();
4581        }
4582        mActiveTrack->mState = TrackBase::RESUMING;
4583        // signal thread to start
4584        ALOGV("Signal record thread");
4585        mWaitWorkCV.signal();
4586        // do not wait for mStartStopCond if exiting
4587        if (mExiting) {
4588            mActiveTrack.clear();
4589            status = INVALID_OPERATION;
4590            goto startError;
4591        }
4592        mStartStopCond.wait(mLock);
4593        if (mActiveTrack == 0) {
4594            ALOGV("Record failed to start");
4595            status = BAD_VALUE;
4596            goto startError;
4597        }
4598        ALOGV("Record started OK");
4599        return status;
4600    }
4601startError:
4602    AudioSystem::stopInput(mId);
4603    return status;
4604}
4605
4606void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4607    ALOGV("RecordThread::stop");
4608    sp <ThreadBase> strongMe = this;
4609    {
4610        AutoMutex lock(mLock);
4611        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4612            mActiveTrack->mState = TrackBase::PAUSING;
4613            // do not wait for mStartStopCond if exiting
4614            if (mExiting) {
4615                return;
4616            }
4617            mStartStopCond.wait(mLock);
4618            // if we have been restarted, recordTrack == mActiveTrack.get() here
4619            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4620                mLock.unlock();
4621                AudioSystem::stopInput(mId);
4622                mLock.lock();
4623                ALOGV("Record stopped OK");
4624            }
4625        }
4626    }
4627}
4628
4629status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4630{
4631    const size_t SIZE = 256;
4632    char buffer[SIZE];
4633    String8 result;
4634
4635    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4636    result.append(buffer);
4637
4638    if (mActiveTrack != 0) {
4639        result.append("Active Track:\n");
4640        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4641        mActiveTrack->dump(buffer, SIZE);
4642        result.append(buffer);
4643
4644        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4645        result.append(buffer);
4646        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4647        result.append(buffer);
4648        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4649        result.append(buffer);
4650        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4651        result.append(buffer);
4652        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4653        result.append(buffer);
4654
4655
4656    } else {
4657        result.append("No record client\n");
4658    }
4659    write(fd, result.string(), result.size());
4660
4661    dumpBase(fd, args);
4662    dumpEffectChains(fd, args);
4663
4664    return NO_ERROR;
4665}
4666
4667status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4668{
4669    size_t framesReq = buffer->frameCount;
4670    size_t framesReady = mFrameCount - mRsmpInIndex;
4671    int channelCount;
4672
4673    if (framesReady == 0) {
4674        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4675        if (mBytesRead < 0) {
4676            ALOGE("RecordThread::getNextBuffer() Error reading audio input");
4677            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4678                // Force input into standby so that it tries to
4679                // recover at next read attempt
4680                mInput->stream->common.standby(&mInput->stream->common);
4681                usleep(kRecordThreadSleepUs);
4682            }
4683            buffer->raw = NULL;
4684            buffer->frameCount = 0;
4685            return NOT_ENOUGH_DATA;
4686        }
4687        mRsmpInIndex = 0;
4688        framesReady = mFrameCount;
4689    }
4690
4691    if (framesReq > framesReady) {
4692        framesReq = framesReady;
4693    }
4694
4695    if (mChannelCount == 1 && mReqChannelCount == 2) {
4696        channelCount = 1;
4697    } else {
4698        channelCount = 2;
4699    }
4700    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4701    buffer->frameCount = framesReq;
4702    return NO_ERROR;
4703}
4704
4705void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4706{
4707    mRsmpInIndex += buffer->frameCount;
4708    buffer->frameCount = 0;
4709}
4710
4711bool AudioFlinger::RecordThread::checkForNewParameters_l()
4712{
4713    bool reconfig = false;
4714
4715    while (!mNewParameters.isEmpty()) {
4716        status_t status = NO_ERROR;
4717        String8 keyValuePair = mNewParameters[0];
4718        AudioParameter param = AudioParameter(keyValuePair);
4719        int value;
4720        audio_format_t reqFormat = mFormat;
4721        int reqSamplingRate = mReqSampleRate;
4722        int reqChannelCount = mReqChannelCount;
4723
4724        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4725            reqSamplingRate = value;
4726            reconfig = true;
4727        }
4728        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4729            reqFormat = (audio_format_t) value;
4730            reconfig = true;
4731        }
4732        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4733            reqChannelCount = popcount(value);
4734            reconfig = true;
4735        }
4736        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4737            // do not accept frame count changes if tracks are open as the track buffer
4738            // size depends on frame count and correct behavior would not be garantied
4739            // if frame count is changed after track creation
4740            if (mActiveTrack != 0) {
4741                status = INVALID_OPERATION;
4742            } else {
4743                reconfig = true;
4744            }
4745        }
4746        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4747            // forward device change to effects that have requested to be
4748            // aware of attached audio device.
4749            for (size_t i = 0; i < mEffectChains.size(); i++) {
4750                mEffectChains[i]->setDevice_l(value);
4751            }
4752            // store input device and output device but do not forward output device to audio HAL.
4753            // Note that status is ignored by the caller for output device
4754            // (see AudioFlinger::setParameters()
4755            if (value & AUDIO_DEVICE_OUT_ALL) {
4756                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4757                status = BAD_VALUE;
4758            } else {
4759                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4760                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4761                if (mTrack != NULL) {
4762                    bool suspend = audio_is_bluetooth_sco_device(
4763                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4764                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4765                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4766                }
4767            }
4768            mDevice |= (uint32_t)value;
4769        }
4770        if (status == NO_ERROR) {
4771            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4772            if (status == INVALID_OPERATION) {
4773               mInput->stream->common.standby(&mInput->stream->common);
4774               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4775            }
4776            if (reconfig) {
4777                if (status == BAD_VALUE &&
4778                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4779                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4780                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4781                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4782                    (reqChannelCount < 3)) {
4783                    status = NO_ERROR;
4784                }
4785                if (status == NO_ERROR) {
4786                    readInputParameters();
4787                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4788                }
4789            }
4790        }
4791
4792        mNewParameters.removeAt(0);
4793
4794        mParamStatus = status;
4795        mParamCond.signal();
4796        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4797        // already timed out waiting for the status and will never signal the condition.
4798        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4799    }
4800    return reconfig;
4801}
4802
4803String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4804{
4805    char *s;
4806    String8 out_s8 = String8();
4807
4808    Mutex::Autolock _l(mLock);
4809    if (initCheck() != NO_ERROR) {
4810        return out_s8;
4811    }
4812
4813    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4814    out_s8 = String8(s);
4815    free(s);
4816    return out_s8;
4817}
4818
4819void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4820    AudioSystem::OutputDescriptor desc;
4821    void *param2 = NULL;
4822
4823    switch (event) {
4824    case AudioSystem::INPUT_OPENED:
4825    case AudioSystem::INPUT_CONFIG_CHANGED:
4826        desc.channels = mChannelMask;
4827        desc.samplingRate = mSampleRate;
4828        desc.format = mFormat;
4829        desc.frameCount = mFrameCount;
4830        desc.latency = 0;
4831        param2 = &desc;
4832        break;
4833
4834    case AudioSystem::INPUT_CLOSED:
4835    default:
4836        break;
4837    }
4838    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4839}
4840
4841void AudioFlinger::RecordThread::readInputParameters()
4842{
4843    delete mRsmpInBuffer;
4844    // mRsmpInBuffer is always assigned a new[] below
4845    delete mRsmpOutBuffer;
4846    mRsmpOutBuffer = NULL;
4847    delete mResampler;
4848    mResampler = NULL;
4849
4850    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4851    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4852    mChannelCount = (uint16_t)popcount(mChannelMask);
4853    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4854    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
4855    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4856    mFrameCount = mInputBytes / mFrameSize;
4857    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4858
4859    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4860    {
4861        int channelCount;
4862         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4863         // stereo to mono post process as the resampler always outputs stereo.
4864        if (mChannelCount == 1 && mReqChannelCount == 2) {
4865            channelCount = 1;
4866        } else {
4867            channelCount = 2;
4868        }
4869        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4870        mResampler->setSampleRate(mSampleRate);
4871        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4872        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4873
4874        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4875        if (mChannelCount == 1 && mReqChannelCount == 1) {
4876            mFrameCount >>= 1;
4877        }
4878
4879    }
4880    mRsmpInIndex = mFrameCount;
4881}
4882
4883unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4884{
4885    Mutex::Autolock _l(mLock);
4886    if (initCheck() != NO_ERROR) {
4887        return 0;
4888    }
4889
4890    return mInput->stream->get_input_frames_lost(mInput->stream);
4891}
4892
4893uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4894{
4895    Mutex::Autolock _l(mLock);
4896    uint32_t result = 0;
4897    if (getEffectChain_l(sessionId) != 0) {
4898        result = EFFECT_SESSION;
4899    }
4900
4901    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4902        result |= TRACK_SESSION;
4903    }
4904
4905    return result;
4906}
4907
4908AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4909{
4910    Mutex::Autolock _l(mLock);
4911    return mTrack;
4912}
4913
4914AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const
4915{
4916    Mutex::Autolock _l(mLock);
4917    return mInput;
4918}
4919
4920AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4921{
4922    Mutex::Autolock _l(mLock);
4923    AudioStreamIn *input = mInput;
4924    mInput = NULL;
4925    return input;
4926}
4927
4928// this method must always be called either with ThreadBase mLock held or inside the thread loop
4929audio_stream_t* AudioFlinger::RecordThread::stream()
4930{
4931    if (mInput == NULL) {
4932        return NULL;
4933    }
4934    return &mInput->stream->common;
4935}
4936
4937
4938// ----------------------------------------------------------------------------
4939
4940int AudioFlinger::openOutput(uint32_t *pDevices,
4941                                uint32_t *pSamplingRate,
4942                                audio_format_t *pFormat,
4943                                uint32_t *pChannels,
4944                                uint32_t *pLatencyMs,
4945                                uint32_t flags)
4946{
4947    status_t status;
4948    PlaybackThread *thread = NULL;
4949    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4950    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4951    audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
4952    uint32_t channels = pChannels ? *pChannels : 0;
4953    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4954    audio_stream_out_t *outStream;
4955    audio_hw_device_t *outHwDev;
4956
4957    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4958            pDevices ? *pDevices : 0,
4959            samplingRate,
4960            format,
4961            channels,
4962            flags);
4963
4964    if (pDevices == NULL || *pDevices == 0) {
4965        return 0;
4966    }
4967
4968    Mutex::Autolock _l(mLock);
4969
4970    outHwDev = findSuitableHwDev_l(*pDevices);
4971    if (outHwDev == NULL)
4972        return 0;
4973
4974    status = outHwDev->open_output_stream(outHwDev, *pDevices, &format,
4975                                          &channels, &samplingRate, &outStream);
4976    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4977            outStream,
4978            samplingRate,
4979            format,
4980            channels,
4981            status);
4982
4983    mHardwareStatus = AUDIO_HW_IDLE;
4984    if (outStream != NULL) {
4985        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4986        int id = nextUniqueId();
4987
4988        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4989            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4990            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4991            thread = new DirectOutputThread(this, output, id, *pDevices);
4992            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4993        } else {
4994            thread = new MixerThread(this, output, id, *pDevices);
4995            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4996        }
4997        mPlaybackThreads.add(id, thread);
4998
4999        if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
5000        if (pFormat != NULL) *pFormat = format;
5001        if (pChannels != NULL) *pChannels = channels;
5002        if (pLatencyMs != NULL) *pLatencyMs = thread->latency();
5003
5004        // notify client processes of the new output creation
5005        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
5006        return id;
5007    }
5008
5009    return 0;
5010}
5011
5012int AudioFlinger::openDuplicateOutput(int output1, int output2)
5013{
5014    Mutex::Autolock _l(mLock);
5015    MixerThread *thread1 = checkMixerThread_l(output1);
5016    MixerThread *thread2 = checkMixerThread_l(output2);
5017
5018    if (thread1 == NULL || thread2 == NULL) {
5019        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
5020        return 0;
5021    }
5022
5023    int id = nextUniqueId();
5024    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
5025    thread->addOutputTrack(thread2);
5026    mPlaybackThreads.add(id, thread);
5027    // notify client processes of the new output creation
5028    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
5029    return id;
5030}
5031
5032status_t AudioFlinger::closeOutput(int output)
5033{
5034    // keep strong reference on the playback thread so that
5035    // it is not destroyed while exit() is executed
5036    sp <PlaybackThread> thread;
5037    {
5038        Mutex::Autolock _l(mLock);
5039        thread = checkPlaybackThread_l(output);
5040        if (thread == NULL) {
5041            return BAD_VALUE;
5042        }
5043
5044        ALOGV("closeOutput() %d", output);
5045
5046        if (thread->type() == ThreadBase::MIXER) {
5047            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5048                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
5049                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
5050                    dupThread->removeOutputTrack((MixerThread *)thread.get());
5051                }
5052            }
5053        }
5054        void *param2 = NULL;
5055        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
5056        mPlaybackThreads.removeItem(output);
5057    }
5058    thread->exit();
5059
5060    if (thread->type() != ThreadBase::DUPLICATING) {
5061        AudioStreamOut *out = thread->clearOutput();
5062        assert(out != NULL);
5063        // from now on thread->mOutput is NULL
5064        out->hwDev->close_output_stream(out->hwDev, out->stream);
5065        delete out;
5066    }
5067    return NO_ERROR;
5068}
5069
5070status_t AudioFlinger::suspendOutput(int output)
5071{
5072    Mutex::Autolock _l(mLock);
5073    PlaybackThread *thread = checkPlaybackThread_l(output);
5074
5075    if (thread == NULL) {
5076        return BAD_VALUE;
5077    }
5078
5079    ALOGV("suspendOutput() %d", output);
5080    thread->suspend();
5081
5082    return NO_ERROR;
5083}
5084
5085status_t AudioFlinger::restoreOutput(int output)
5086{
5087    Mutex::Autolock _l(mLock);
5088    PlaybackThread *thread = checkPlaybackThread_l(output);
5089
5090    if (thread == NULL) {
5091        return BAD_VALUE;
5092    }
5093
5094    ALOGV("restoreOutput() %d", output);
5095
5096    thread->restore();
5097
5098    return NO_ERROR;
5099}
5100
5101int AudioFlinger::openInput(uint32_t *pDevices,
5102                                uint32_t *pSamplingRate,
5103                                audio_format_t *pFormat,
5104                                uint32_t *pChannels,
5105                                audio_in_acoustics_t acoustics)
5106{
5107    status_t status;
5108    RecordThread *thread = NULL;
5109    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5110    audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
5111    uint32_t channels = pChannels ? *pChannels : 0;
5112    uint32_t reqSamplingRate = samplingRate;
5113    audio_format_t reqFormat = format;
5114    uint32_t reqChannels = channels;
5115    audio_stream_in_t *inStream;
5116    audio_hw_device_t *inHwDev;
5117
5118    if (pDevices == NULL || *pDevices == 0) {
5119        return 0;
5120    }
5121
5122    Mutex::Autolock _l(mLock);
5123
5124    inHwDev = findSuitableHwDev_l(*pDevices);
5125    if (inHwDev == NULL)
5126        return 0;
5127
5128    status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
5129                                        &channels, &samplingRate,
5130                                        acoustics,
5131                                        &inStream);
5132    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5133            inStream,
5134            samplingRate,
5135            format,
5136            channels,
5137            acoustics,
5138            status);
5139
5140    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5141    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5142    // or stereo to mono conversions on 16 bit PCM inputs.
5143    if (inStream == NULL && status == BAD_VALUE &&
5144        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5145        (samplingRate <= 2 * reqSamplingRate) &&
5146        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5147        ALOGV("openInput() reopening with proposed sampling rate and channels");
5148        status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
5149                                            &channels, &samplingRate,
5150                                            acoustics,
5151                                            &inStream);
5152    }
5153
5154    if (inStream != NULL) {
5155        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5156
5157        int id = nextUniqueId();
5158        // Start record thread
5159        // RecorThread require both input and output device indication to forward to audio
5160        // pre processing modules
5161        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5162        thread = new RecordThread(this,
5163                                  input,
5164                                  reqSamplingRate,
5165                                  reqChannels,
5166                                  id,
5167                                  device);
5168        mRecordThreads.add(id, thread);
5169        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5170        if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
5171        if (pFormat != NULL) *pFormat = format;
5172        if (pChannels != NULL) *pChannels = reqChannels;
5173
5174        input->stream->common.standby(&input->stream->common);
5175
5176        // notify client processes of the new input creation
5177        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5178        return id;
5179    }
5180
5181    return 0;
5182}
5183
5184status_t AudioFlinger::closeInput(int input)
5185{
5186    // keep strong reference on the record thread so that
5187    // it is not destroyed while exit() is executed
5188    sp <RecordThread> thread;
5189    {
5190        Mutex::Autolock _l(mLock);
5191        thread = checkRecordThread_l(input);
5192        if (thread == NULL) {
5193            return BAD_VALUE;
5194        }
5195
5196        ALOGV("closeInput() %d", input);
5197        void *param2 = NULL;
5198        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5199        mRecordThreads.removeItem(input);
5200    }
5201    thread->exit();
5202
5203    AudioStreamIn *in = thread->clearInput();
5204    assert(in != NULL);
5205    // from now on thread->mInput is NULL
5206    in->hwDev->close_input_stream(in->hwDev, in->stream);
5207    delete in;
5208
5209    return NO_ERROR;
5210}
5211
5212status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output)
5213{
5214    Mutex::Autolock _l(mLock);
5215    MixerThread *dstThread = checkMixerThread_l(output);
5216    if (dstThread == NULL) {
5217        ALOGW("setStreamOutput() bad output id %d", output);
5218        return BAD_VALUE;
5219    }
5220
5221    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5222    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5223
5224    dstThread->setStreamValid(stream, true);
5225
5226    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5227        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5228        if (thread != dstThread &&
5229            thread->type() != ThreadBase::DIRECT) {
5230            MixerThread *srcThread = (MixerThread *)thread;
5231            srcThread->setStreamValid(stream, false);
5232            srcThread->invalidateTracks(stream);
5233        }
5234    }
5235
5236    return NO_ERROR;
5237}
5238
5239
5240int AudioFlinger::newAudioSessionId()
5241{
5242    return nextUniqueId();
5243}
5244
5245void AudioFlinger::acquireAudioSessionId(int audioSession)
5246{
5247    Mutex::Autolock _l(mLock);
5248    int caller = IPCThreadState::self()->getCallingPid();
5249    ALOGV("acquiring %d from %d", audioSession, caller);
5250    int num = mAudioSessionRefs.size();
5251    for (int i = 0; i< num; i++) {
5252        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5253        if (ref->sessionid == audioSession && ref->pid == caller) {
5254            ref->cnt++;
5255            ALOGV(" incremented refcount to %d", ref->cnt);
5256            return;
5257        }
5258    }
5259    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
5260    ALOGV(" added new entry for %d", audioSession);
5261}
5262
5263void AudioFlinger::releaseAudioSessionId(int audioSession)
5264{
5265    Mutex::Autolock _l(mLock);
5266    int caller = IPCThreadState::self()->getCallingPid();
5267    ALOGV("releasing %d from %d", audioSession, caller);
5268    int num = mAudioSessionRefs.size();
5269    for (int i = 0; i< num; i++) {
5270        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5271        if (ref->sessionid == audioSession && ref->pid == caller) {
5272            ref->cnt--;
5273            ALOGV(" decremented refcount to %d", ref->cnt);
5274            if (ref->cnt == 0) {
5275                mAudioSessionRefs.removeAt(i);
5276                delete ref;
5277                purgeStaleEffects_l();
5278            }
5279            return;
5280        }
5281    }
5282    ALOGW("session id %d not found for pid %d", audioSession, caller);
5283}
5284
5285void AudioFlinger::purgeStaleEffects_l() {
5286
5287    ALOGV("purging stale effects");
5288
5289    Vector< sp<EffectChain> > chains;
5290
5291    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5292        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5293        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5294            sp<EffectChain> ec = t->mEffectChains[j];
5295            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5296                chains.push(ec);
5297            }
5298        }
5299    }
5300    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5301        sp<RecordThread> t = mRecordThreads.valueAt(i);
5302        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5303            sp<EffectChain> ec = t->mEffectChains[j];
5304            chains.push(ec);
5305        }
5306    }
5307
5308    for (size_t i = 0; i < chains.size(); i++) {
5309        sp<EffectChain> ec = chains[i];
5310        int sessionid = ec->sessionId();
5311        sp<ThreadBase> t = ec->mThread.promote();
5312        if (t == 0) {
5313            continue;
5314        }
5315        size_t numsessionrefs = mAudioSessionRefs.size();
5316        bool found = false;
5317        for (size_t k = 0; k < numsessionrefs; k++) {
5318            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5319            if (ref->sessionid == sessionid) {
5320                ALOGV(" session %d still exists for %d with %d refs",
5321                     sessionid, ref->pid, ref->cnt);
5322                found = true;
5323                break;
5324            }
5325        }
5326        if (!found) {
5327            // remove all effects from the chain
5328            while (ec->mEffects.size()) {
5329                sp<EffectModule> effect = ec->mEffects[0];
5330                effect->unPin();
5331                Mutex::Autolock _l (t->mLock);
5332                t->removeEffect_l(effect);
5333                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5334                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5335                    if (handle != 0) {
5336                        handle->mEffect.clear();
5337                        if (handle->mHasControl && handle->mEnabled) {
5338                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5339                        }
5340                    }
5341                }
5342                AudioSystem::unregisterEffect(effect->id());
5343            }
5344        }
5345    }
5346    return;
5347}
5348
5349// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5350AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5351{
5352    PlaybackThread *thread = NULL;
5353    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5354        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5355    }
5356    return thread;
5357}
5358
5359// checkMixerThread_l() must be called with AudioFlinger::mLock held
5360AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5361{
5362    PlaybackThread *thread = checkPlaybackThread_l(output);
5363    if (thread != NULL) {
5364        if (thread->type() == ThreadBase::DIRECT) {
5365            thread = NULL;
5366        }
5367    }
5368    return (MixerThread *)thread;
5369}
5370
5371// checkRecordThread_l() must be called with AudioFlinger::mLock held
5372AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5373{
5374    RecordThread *thread = NULL;
5375    if (mRecordThreads.indexOfKey(input) >= 0) {
5376        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5377    }
5378    return thread;
5379}
5380
5381uint32_t AudioFlinger::nextUniqueId()
5382{
5383    return android_atomic_inc(&mNextUniqueId);
5384}
5385
5386AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5387{
5388    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5389        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5390        AudioStreamOut *output = thread->getOutput();
5391        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5392            return thread;
5393        }
5394    }
5395    return NULL;
5396}
5397
5398uint32_t AudioFlinger::primaryOutputDevice_l()
5399{
5400    PlaybackThread *thread = primaryPlaybackThread_l();
5401
5402    if (thread == NULL) {
5403        return 0;
5404    }
5405
5406    return thread->device();
5407}
5408
5409
5410// ----------------------------------------------------------------------------
5411//  Effect management
5412// ----------------------------------------------------------------------------
5413
5414
5415status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
5416{
5417    Mutex::Autolock _l(mLock);
5418    return EffectQueryNumberEffects(numEffects);
5419}
5420
5421status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
5422{
5423    Mutex::Autolock _l(mLock);
5424    return EffectQueryEffect(index, descriptor);
5425}
5426
5427status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid,
5428        effect_descriptor_t *descriptor) const
5429{
5430    Mutex::Autolock _l(mLock);
5431    return EffectGetDescriptor(pUuid, descriptor);
5432}
5433
5434
5435sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5436        effect_descriptor_t *pDesc,
5437        const sp<IEffectClient>& effectClient,
5438        int32_t priority,
5439        int io,
5440        int sessionId,
5441        status_t *status,
5442        int *id,
5443        int *enabled)
5444{
5445    status_t lStatus = NO_ERROR;
5446    sp<EffectHandle> handle;
5447    effect_descriptor_t desc;
5448    sp<Client> client;
5449    wp<Client> wclient;
5450
5451    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5452            pid, effectClient.get(), priority, sessionId, io);
5453
5454    if (pDesc == NULL) {
5455        lStatus = BAD_VALUE;
5456        goto Exit;
5457    }
5458
5459    // check audio settings permission for global effects
5460    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5461        lStatus = PERMISSION_DENIED;
5462        goto Exit;
5463    }
5464
5465    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5466    // that can only be created by audio policy manager (running in same process)
5467    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5468        lStatus = PERMISSION_DENIED;
5469        goto Exit;
5470    }
5471
5472    if (io == 0) {
5473        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5474            // output must be specified by AudioPolicyManager when using session
5475            // AUDIO_SESSION_OUTPUT_STAGE
5476            lStatus = BAD_VALUE;
5477            goto Exit;
5478        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5479            // if the output returned by getOutputForEffect() is removed before we lock the
5480            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5481            // and we will exit safely
5482            io = AudioSystem::getOutputForEffect(&desc);
5483        }
5484    }
5485
5486    {
5487        Mutex::Autolock _l(mLock);
5488
5489
5490        if (!EffectIsNullUuid(&pDesc->uuid)) {
5491            // if uuid is specified, request effect descriptor
5492            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5493            if (lStatus < 0) {
5494                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5495                goto Exit;
5496            }
5497        } else {
5498            // if uuid is not specified, look for an available implementation
5499            // of the required type in effect factory
5500            if (EffectIsNullUuid(&pDesc->type)) {
5501                ALOGW("createEffect() no effect type");
5502                lStatus = BAD_VALUE;
5503                goto Exit;
5504            }
5505            uint32_t numEffects = 0;
5506            effect_descriptor_t d;
5507            d.flags = 0; // prevent compiler warning
5508            bool found = false;
5509
5510            lStatus = EffectQueryNumberEffects(&numEffects);
5511            if (lStatus < 0) {
5512                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5513                goto Exit;
5514            }
5515            for (uint32_t i = 0; i < numEffects; i++) {
5516                lStatus = EffectQueryEffect(i, &desc);
5517                if (lStatus < 0) {
5518                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5519                    continue;
5520                }
5521                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5522                    // If matching type found save effect descriptor. If the session is
5523                    // 0 and the effect is not auxiliary, continue enumeration in case
5524                    // an auxiliary version of this effect type is available
5525                    found = true;
5526                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5527                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5528                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5529                        break;
5530                    }
5531                }
5532            }
5533            if (!found) {
5534                lStatus = BAD_VALUE;
5535                ALOGW("createEffect() effect not found");
5536                goto Exit;
5537            }
5538            // For same effect type, chose auxiliary version over insert version if
5539            // connect to output mix (Compliance to OpenSL ES)
5540            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5541                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5542                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5543            }
5544        }
5545
5546        // Do not allow auxiliary effects on a session different from 0 (output mix)
5547        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5548             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5549            lStatus = INVALID_OPERATION;
5550            goto Exit;
5551        }
5552
5553        // check recording permission for visualizer
5554        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5555            !recordingAllowed()) {
5556            lStatus = PERMISSION_DENIED;
5557            goto Exit;
5558        }
5559
5560        // return effect descriptor
5561        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5562
5563        // If output is not specified try to find a matching audio session ID in one of the
5564        // output threads.
5565        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5566        // because of code checking output when entering the function.
5567        // Note: io is never 0 when creating an effect on an input
5568        if (io == 0) {
5569             // look for the thread where the specified audio session is present
5570            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5571                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5572                    io = mPlaybackThreads.keyAt(i);
5573                    break;
5574                }
5575            }
5576            if (io == 0) {
5577               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5578                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5579                       io = mRecordThreads.keyAt(i);
5580                       break;
5581                   }
5582               }
5583            }
5584            // If no output thread contains the requested session ID, default to
5585            // first output. The effect chain will be moved to the correct output
5586            // thread when a track with the same session ID is created
5587            if (io == 0 && mPlaybackThreads.size()) {
5588                io = mPlaybackThreads.keyAt(0);
5589            }
5590            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5591        }
5592        ThreadBase *thread = checkRecordThread_l(io);
5593        if (thread == NULL) {
5594            thread = checkPlaybackThread_l(io);
5595            if (thread == NULL) {
5596                ALOGE("createEffect() unknown output thread");
5597                lStatus = BAD_VALUE;
5598                goto Exit;
5599            }
5600        }
5601
5602        wclient = mClients.valueFor(pid);
5603
5604        if (wclient != NULL) {
5605            client = wclient.promote();
5606        } else {
5607            client = new Client(this, pid);
5608            mClients.add(pid, client);
5609        }
5610
5611        // create effect on selected output thread
5612        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5613                &desc, enabled, &lStatus);
5614        if (handle != 0 && id != NULL) {
5615            *id = handle->id();
5616        }
5617    }
5618
5619Exit:
5620    if(status) {
5621        *status = lStatus;
5622    }
5623    return handle;
5624}
5625
5626status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5627{
5628    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5629            sessionId, srcOutput, dstOutput);
5630    Mutex::Autolock _l(mLock);
5631    if (srcOutput == dstOutput) {
5632        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
5633        return NO_ERROR;
5634    }
5635    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5636    if (srcThread == NULL) {
5637        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
5638        return BAD_VALUE;
5639    }
5640    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5641    if (dstThread == NULL) {
5642        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
5643        return BAD_VALUE;
5644    }
5645
5646    Mutex::Autolock _dl(dstThread->mLock);
5647    Mutex::Autolock _sl(srcThread->mLock);
5648    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5649
5650    return NO_ERROR;
5651}
5652
5653// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5654status_t AudioFlinger::moveEffectChain_l(int sessionId,
5655                                   AudioFlinger::PlaybackThread *srcThread,
5656                                   AudioFlinger::PlaybackThread *dstThread,
5657                                   bool reRegister)
5658{
5659    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5660            sessionId, srcThread, dstThread);
5661
5662    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5663    if (chain == 0) {
5664        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5665                sessionId, srcThread);
5666        return INVALID_OPERATION;
5667    }
5668
5669    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5670    // so that a new chain is created with correct parameters when first effect is added. This is
5671    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5672    // removed.
5673    srcThread->removeEffectChain_l(chain);
5674
5675    // transfer all effects one by one so that new effect chain is created on new thread with
5676    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5677    int dstOutput = dstThread->id();
5678    sp<EffectChain> dstChain;
5679    uint32_t strategy = 0; // prevent compiler warning
5680    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5681    while (effect != 0) {
5682        srcThread->removeEffect_l(effect);
5683        dstThread->addEffect_l(effect);
5684        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5685        if (effect->state() == EffectModule::ACTIVE ||
5686                effect->state() == EffectModule::STOPPING) {
5687            effect->start();
5688        }
5689        // if the move request is not received from audio policy manager, the effect must be
5690        // re-registered with the new strategy and output
5691        if (dstChain == 0) {
5692            dstChain = effect->chain().promote();
5693            if (dstChain == 0) {
5694                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5695                srcThread->addEffect_l(effect);
5696                return NO_INIT;
5697            }
5698            strategy = dstChain->strategy();
5699        }
5700        if (reRegister) {
5701            AudioSystem::unregisterEffect(effect->id());
5702            AudioSystem::registerEffect(&effect->desc(),
5703                                        dstOutput,
5704                                        strategy,
5705                                        sessionId,
5706                                        effect->id());
5707        }
5708        effect = chain->getEffectFromId_l(0);
5709    }
5710
5711    return NO_ERROR;
5712}
5713
5714
5715// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5716sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5717        const sp<AudioFlinger::Client>& client,
5718        const sp<IEffectClient>& effectClient,
5719        int32_t priority,
5720        int sessionId,
5721        effect_descriptor_t *desc,
5722        int *enabled,
5723        status_t *status
5724        )
5725{
5726    sp<EffectModule> effect;
5727    sp<EffectHandle> handle;
5728    status_t lStatus;
5729    sp<EffectChain> chain;
5730    bool chainCreated = false;
5731    bool effectCreated = false;
5732    bool effectRegistered = false;
5733
5734    lStatus = initCheck();
5735    if (lStatus != NO_ERROR) {
5736        ALOGW("createEffect_l() Audio driver not initialized.");
5737        goto Exit;
5738    }
5739
5740    // Do not allow effects with session ID 0 on direct output or duplicating threads
5741    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5742    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5743        ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5744                desc->name, sessionId);
5745        lStatus = BAD_VALUE;
5746        goto Exit;
5747    }
5748    // Only Pre processor effects are allowed on input threads and only on input threads
5749    if ((mType == RECORD &&
5750            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5751            (mType != RECORD &&
5752                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5753        ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5754                desc->name, desc->flags, mType);
5755        lStatus = BAD_VALUE;
5756        goto Exit;
5757    }
5758
5759    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5760
5761    { // scope for mLock
5762        Mutex::Autolock _l(mLock);
5763
5764        // check for existing effect chain with the requested audio session
5765        chain = getEffectChain_l(sessionId);
5766        if (chain == 0) {
5767            // create a new chain for this session
5768            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5769            chain = new EffectChain(this, sessionId);
5770            addEffectChain_l(chain);
5771            chain->setStrategy(getStrategyForSession_l(sessionId));
5772            chainCreated = true;
5773        } else {
5774            effect = chain->getEffectFromDesc_l(desc);
5775        }
5776
5777        ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
5778
5779        if (effect == 0) {
5780            int id = mAudioFlinger->nextUniqueId();
5781            // Check CPU and memory usage
5782            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5783            if (lStatus != NO_ERROR) {
5784                goto Exit;
5785            }
5786            effectRegistered = true;
5787            // create a new effect module if none present in the chain
5788            effect = new EffectModule(this, chain, desc, id, sessionId);
5789            lStatus = effect->status();
5790            if (lStatus != NO_ERROR) {
5791                goto Exit;
5792            }
5793            lStatus = chain->addEffect_l(effect);
5794            if (lStatus != NO_ERROR) {
5795                goto Exit;
5796            }
5797            effectCreated = true;
5798
5799            effect->setDevice(mDevice);
5800            effect->setMode(mAudioFlinger->getMode());
5801        }
5802        // create effect handle and connect it to effect module
5803        handle = new EffectHandle(effect, client, effectClient, priority);
5804        lStatus = effect->addHandle(handle);
5805        if (enabled != NULL) {
5806            *enabled = (int)effect->isEnabled();
5807        }
5808    }
5809
5810Exit:
5811    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5812        Mutex::Autolock _l(mLock);
5813        if (effectCreated) {
5814            chain->removeEffect_l(effect);
5815        }
5816        if (effectRegistered) {
5817            AudioSystem::unregisterEffect(effect->id());
5818        }
5819        if (chainCreated) {
5820            removeEffectChain_l(chain);
5821        }
5822        handle.clear();
5823    }
5824
5825    if(status) {
5826        *status = lStatus;
5827    }
5828    return handle;
5829}
5830
5831sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5832{
5833    sp<EffectChain> chain = getEffectChain_l(sessionId);
5834    return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
5835}
5836
5837// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5838// PlaybackThread::mLock held
5839status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5840{
5841    // check for existing effect chain with the requested audio session
5842    int sessionId = effect->sessionId();
5843    sp<EffectChain> chain = getEffectChain_l(sessionId);
5844    bool chainCreated = false;
5845
5846    if (chain == 0) {
5847        // create a new chain for this session
5848        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5849        chain = new EffectChain(this, sessionId);
5850        addEffectChain_l(chain);
5851        chain->setStrategy(getStrategyForSession_l(sessionId));
5852        chainCreated = true;
5853    }
5854    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5855
5856    if (chain->getEffectFromId_l(effect->id()) != 0) {
5857        ALOGW("addEffect_l() %p effect %s already present in chain %p",
5858                this, effect->desc().name, chain.get());
5859        return BAD_VALUE;
5860    }
5861
5862    status_t status = chain->addEffect_l(effect);
5863    if (status != NO_ERROR) {
5864        if (chainCreated) {
5865            removeEffectChain_l(chain);
5866        }
5867        return status;
5868    }
5869
5870    effect->setDevice(mDevice);
5871    effect->setMode(mAudioFlinger->getMode());
5872    return NO_ERROR;
5873}
5874
5875void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5876
5877    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5878    effect_descriptor_t desc = effect->desc();
5879    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5880        detachAuxEffect_l(effect->id());
5881    }
5882
5883    sp<EffectChain> chain = effect->chain().promote();
5884    if (chain != 0) {
5885        // remove effect chain if removing last effect
5886        if (chain->removeEffect_l(effect) == 0) {
5887            removeEffectChain_l(chain);
5888        }
5889    } else {
5890        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5891    }
5892}
5893
5894void AudioFlinger::ThreadBase::lockEffectChains_l(
5895        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5896{
5897    effectChains = mEffectChains;
5898    for (size_t i = 0; i < mEffectChains.size(); i++) {
5899        mEffectChains[i]->lock();
5900    }
5901}
5902
5903void AudioFlinger::ThreadBase::unlockEffectChains(
5904        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5905{
5906    for (size_t i = 0; i < effectChains.size(); i++) {
5907        effectChains[i]->unlock();
5908    }
5909}
5910
5911sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5912{
5913    Mutex::Autolock _l(mLock);
5914    return getEffectChain_l(sessionId);
5915}
5916
5917sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5918{
5919    size_t size = mEffectChains.size();
5920    for (size_t i = 0; i < size; i++) {
5921        if (mEffectChains[i]->sessionId() == sessionId) {
5922            return mEffectChains[i];
5923        }
5924    }
5925    return 0;
5926}
5927
5928void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
5929{
5930    Mutex::Autolock _l(mLock);
5931    size_t size = mEffectChains.size();
5932    for (size_t i = 0; i < size; i++) {
5933        mEffectChains[i]->setMode_l(mode);
5934    }
5935}
5936
5937void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5938                                                    const wp<EffectHandle>& handle,
5939                                                    bool unpiniflast) {
5940
5941    Mutex::Autolock _l(mLock);
5942    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5943    // delete the effect module if removing last handle on it
5944    if (effect->removeHandle(handle) == 0) {
5945        if (!effect->isPinned() || unpiniflast) {
5946            removeEffect_l(effect);
5947            AudioSystem::unregisterEffect(effect->id());
5948        }
5949    }
5950}
5951
5952status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5953{
5954    int session = chain->sessionId();
5955    int16_t *buffer = mMixBuffer;
5956    bool ownsBuffer = false;
5957
5958    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5959    if (session > 0) {
5960        // Only one effect chain can be present in direct output thread and it uses
5961        // the mix buffer as input
5962        if (mType != DIRECT) {
5963            size_t numSamples = mFrameCount * mChannelCount;
5964            buffer = new int16_t[numSamples];
5965            memset(buffer, 0, numSamples * sizeof(int16_t));
5966            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5967            ownsBuffer = true;
5968        }
5969
5970        // Attach all tracks with same session ID to this chain.
5971        for (size_t i = 0; i < mTracks.size(); ++i) {
5972            sp<Track> track = mTracks[i];
5973            if (session == track->sessionId()) {
5974                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5975                track->setMainBuffer(buffer);
5976                chain->incTrackCnt();
5977            }
5978        }
5979
5980        // indicate all active tracks in the chain
5981        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5982            sp<Track> track = mActiveTracks[i].promote();
5983            if (track == 0) continue;
5984            if (session == track->sessionId()) {
5985                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5986                chain->incActiveTrackCnt();
5987            }
5988        }
5989    }
5990
5991    chain->setInBuffer(buffer, ownsBuffer);
5992    chain->setOutBuffer(mMixBuffer);
5993    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5994    // chains list in order to be processed last as it contains output stage effects
5995    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5996    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5997    // after track specific effects and before output stage
5998    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5999    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
6000    // Effect chain for other sessions are inserted at beginning of effect
6001    // chains list to be processed before output mix effects. Relative order between other
6002    // sessions is not important
6003    size_t size = mEffectChains.size();
6004    size_t i = 0;
6005    for (i = 0; i < size; i++) {
6006        if (mEffectChains[i]->sessionId() < session) break;
6007    }
6008    mEffectChains.insertAt(chain, i);
6009    checkSuspendOnAddEffectChain_l(chain);
6010
6011    return NO_ERROR;
6012}
6013
6014size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
6015{
6016    int session = chain->sessionId();
6017
6018    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
6019
6020    for (size_t i = 0; i < mEffectChains.size(); i++) {
6021        if (chain == mEffectChains[i]) {
6022            mEffectChains.removeAt(i);
6023            // detach all active tracks from the chain
6024            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
6025                sp<Track> track = mActiveTracks[i].promote();
6026                if (track == 0) continue;
6027                if (session == track->sessionId()) {
6028                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
6029                            chain.get(), session);
6030                    chain->decActiveTrackCnt();
6031                }
6032            }
6033
6034            // detach all tracks with same session ID from this chain
6035            for (size_t i = 0; i < mTracks.size(); ++i) {
6036                sp<Track> track = mTracks[i];
6037                if (session == track->sessionId()) {
6038                    track->setMainBuffer(mMixBuffer);
6039                    chain->decTrackCnt();
6040                }
6041            }
6042            break;
6043        }
6044    }
6045    return mEffectChains.size();
6046}
6047
6048status_t AudioFlinger::PlaybackThread::attachAuxEffect(
6049        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6050{
6051    Mutex::Autolock _l(mLock);
6052    return attachAuxEffect_l(track, EffectId);
6053}
6054
6055status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6056        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6057{
6058    status_t status = NO_ERROR;
6059
6060    if (EffectId == 0) {
6061        track->setAuxBuffer(0, NULL);
6062    } else {
6063        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6064        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6065        if (effect != 0) {
6066            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6067                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6068            } else {
6069                status = INVALID_OPERATION;
6070            }
6071        } else {
6072            status = BAD_VALUE;
6073        }
6074    }
6075    return status;
6076}
6077
6078void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6079{
6080     for (size_t i = 0; i < mTracks.size(); ++i) {
6081        sp<Track> track = mTracks[i];
6082        if (track->auxEffectId() == effectId) {
6083            attachAuxEffect_l(track, 0);
6084        }
6085    }
6086}
6087
6088status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6089{
6090    // only one chain per input thread
6091    if (mEffectChains.size() != 0) {
6092        return INVALID_OPERATION;
6093    }
6094    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6095
6096    chain->setInBuffer(NULL);
6097    chain->setOutBuffer(NULL);
6098
6099    checkSuspendOnAddEffectChain_l(chain);
6100
6101    mEffectChains.add(chain);
6102
6103    return NO_ERROR;
6104}
6105
6106size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6107{
6108    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6109    ALOGW_IF(mEffectChains.size() != 1,
6110            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6111            chain.get(), mEffectChains.size(), this);
6112    if (mEffectChains.size() == 1) {
6113        mEffectChains.removeAt(0);
6114    }
6115    return 0;
6116}
6117
6118// ----------------------------------------------------------------------------
6119//  EffectModule implementation
6120// ----------------------------------------------------------------------------
6121
6122#undef LOG_TAG
6123#define LOG_TAG "AudioFlinger::EffectModule"
6124
6125AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6126                                        const wp<AudioFlinger::EffectChain>& chain,
6127                                        effect_descriptor_t *desc,
6128                                        int id,
6129                                        int sessionId)
6130    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6131      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6132{
6133    ALOGV("Constructor %p", this);
6134    int lStatus;
6135    sp<ThreadBase> thread = mThread.promote();
6136    if (thread == 0) {
6137        return;
6138    }
6139
6140    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6141
6142    // create effect engine from effect factory
6143    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6144
6145    if (mStatus != NO_ERROR) {
6146        return;
6147    }
6148    lStatus = init();
6149    if (lStatus < 0) {
6150        mStatus = lStatus;
6151        goto Error;
6152    }
6153
6154    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6155        mPinned = true;
6156    }
6157    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6158    return;
6159Error:
6160    EffectRelease(mEffectInterface);
6161    mEffectInterface = NULL;
6162    ALOGV("Constructor Error %d", mStatus);
6163}
6164
6165AudioFlinger::EffectModule::~EffectModule()
6166{
6167    ALOGV("Destructor %p", this);
6168    if (mEffectInterface != NULL) {
6169        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6170                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6171            sp<ThreadBase> thread = mThread.promote();
6172            if (thread != 0) {
6173                audio_stream_t *stream = thread->stream();
6174                if (stream != NULL) {
6175                    stream->remove_audio_effect(stream, mEffectInterface);
6176                }
6177            }
6178        }
6179        // release effect engine
6180        EffectRelease(mEffectInterface);
6181    }
6182}
6183
6184status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle)
6185{
6186    status_t status;
6187
6188    Mutex::Autolock _l(mLock);
6189    // First handle in mHandles has highest priority and controls the effect module
6190    int priority = handle->priority();
6191    size_t size = mHandles.size();
6192    sp<EffectHandle> h;
6193    size_t i;
6194    for (i = 0; i < size; i++) {
6195        h = mHandles[i].promote();
6196        if (h == 0) continue;
6197        if (h->priority() <= priority) break;
6198    }
6199    // if inserted in first place, move effect control from previous owner to this handle
6200    if (i == 0) {
6201        bool enabled = false;
6202        if (h != 0) {
6203            enabled = h->enabled();
6204            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6205        }
6206        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6207        status = NO_ERROR;
6208    } else {
6209        status = ALREADY_EXISTS;
6210    }
6211    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6212    mHandles.insertAt(handle, i);
6213    return status;
6214}
6215
6216size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6217{
6218    Mutex::Autolock _l(mLock);
6219    size_t size = mHandles.size();
6220    size_t i;
6221    for (i = 0; i < size; i++) {
6222        if (mHandles[i] == handle) break;
6223    }
6224    if (i == size) {
6225        return size;
6226    }
6227    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6228
6229    bool enabled = false;
6230    EffectHandle *hdl = handle.unsafe_get();
6231    if (hdl != NULL) {
6232        ALOGV("removeHandle() unsafe_get OK");
6233        enabled = hdl->enabled();
6234    }
6235    mHandles.removeAt(i);
6236    size = mHandles.size();
6237    // if removed from first place, move effect control from this handle to next in line
6238    if (i == 0 && size != 0) {
6239        sp<EffectHandle> h = mHandles[0].promote();
6240        if (h != 0) {
6241            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6242        }
6243    }
6244
6245    // Prevent calls to process() and other functions on effect interface from now on.
6246    // The effect engine will be released by the destructor when the last strong reference on
6247    // this object is released which can happen after next process is called.
6248    if (size == 0 && !mPinned) {
6249        mState = DESTROYED;
6250    }
6251
6252    return size;
6253}
6254
6255sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6256{
6257    Mutex::Autolock _l(mLock);
6258    return mHandles.size() != 0 ? mHandles[0].promote() : 0;
6259}
6260
6261void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6262{
6263    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6264    // keep a strong reference on this EffectModule to avoid calling the
6265    // destructor before we exit
6266    sp<EffectModule> keep(this);
6267    {
6268        sp<ThreadBase> thread = mThread.promote();
6269        if (thread != 0) {
6270            thread->disconnectEffect(keep, handle, unpiniflast);
6271        }
6272    }
6273}
6274
6275void AudioFlinger::EffectModule::updateState() {
6276    Mutex::Autolock _l(mLock);
6277
6278    switch (mState) {
6279    case RESTART:
6280        reset_l();
6281        // FALL THROUGH
6282
6283    case STARTING:
6284        // clear auxiliary effect input buffer for next accumulation
6285        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6286            memset(mConfig.inputCfg.buffer.raw,
6287                   0,
6288                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6289        }
6290        start_l();
6291        mState = ACTIVE;
6292        break;
6293    case STOPPING:
6294        stop_l();
6295        mDisableWaitCnt = mMaxDisableWaitCnt;
6296        mState = STOPPED;
6297        break;
6298    case STOPPED:
6299        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6300        // turn off sequence.
6301        if (--mDisableWaitCnt == 0) {
6302            reset_l();
6303            mState = IDLE;
6304        }
6305        break;
6306    default: //IDLE , ACTIVE, DESTROYED
6307        break;
6308    }
6309}
6310
6311void AudioFlinger::EffectModule::process()
6312{
6313    Mutex::Autolock _l(mLock);
6314
6315    if (mState == DESTROYED || mEffectInterface == NULL ||
6316            mConfig.inputCfg.buffer.raw == NULL ||
6317            mConfig.outputCfg.buffer.raw == NULL) {
6318        return;
6319    }
6320
6321    if (isProcessEnabled()) {
6322        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6323        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6324            ditherAndClamp(mConfig.inputCfg.buffer.s32,
6325                                        mConfig.inputCfg.buffer.s32,
6326                                        mConfig.inputCfg.buffer.frameCount/2);
6327        }
6328
6329        // do the actual processing in the effect engine
6330        int ret = (*mEffectInterface)->process(mEffectInterface,
6331                                               &mConfig.inputCfg.buffer,
6332                                               &mConfig.outputCfg.buffer);
6333
6334        // force transition to IDLE state when engine is ready
6335        if (mState == STOPPED && ret == -ENODATA) {
6336            mDisableWaitCnt = 1;
6337        }
6338
6339        // clear auxiliary effect input buffer for next accumulation
6340        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6341            memset(mConfig.inputCfg.buffer.raw, 0,
6342                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6343        }
6344    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6345                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6346        // If an insert effect is idle and input buffer is different from output buffer,
6347        // accumulate input onto output
6348        sp<EffectChain> chain = mChain.promote();
6349        if (chain != 0 && chain->activeTrackCnt() != 0) {
6350            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6351            int16_t *in = mConfig.inputCfg.buffer.s16;
6352            int16_t *out = mConfig.outputCfg.buffer.s16;
6353            for (size_t i = 0; i < frameCnt; i++) {
6354                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6355            }
6356        }
6357    }
6358}
6359
6360void AudioFlinger::EffectModule::reset_l()
6361{
6362    if (mEffectInterface == NULL) {
6363        return;
6364    }
6365    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6366}
6367
6368status_t AudioFlinger::EffectModule::configure()
6369{
6370    uint32_t channels;
6371    if (mEffectInterface == NULL) {
6372        return NO_INIT;
6373    }
6374
6375    sp<ThreadBase> thread = mThread.promote();
6376    if (thread == 0) {
6377        return DEAD_OBJECT;
6378    }
6379
6380    // TODO: handle configuration of effects replacing track process
6381    if (thread->channelCount() == 1) {
6382        channels = AUDIO_CHANNEL_OUT_MONO;
6383    } else {
6384        channels = AUDIO_CHANNEL_OUT_STEREO;
6385    }
6386
6387    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6388        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6389    } else {
6390        mConfig.inputCfg.channels = channels;
6391    }
6392    mConfig.outputCfg.channels = channels;
6393    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6394    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6395    mConfig.inputCfg.samplingRate = thread->sampleRate();
6396    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6397    mConfig.inputCfg.bufferProvider.cookie = NULL;
6398    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6399    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6400    mConfig.outputCfg.bufferProvider.cookie = NULL;
6401    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6402    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6403    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6404    // Insert effect:
6405    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6406    // always overwrites output buffer: input buffer == output buffer
6407    // - in other sessions:
6408    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6409    //      other effect: overwrites output buffer: input buffer == output buffer
6410    // Auxiliary effect:
6411    //      accumulates in output buffer: input buffer != output buffer
6412    // Therefore: accumulate <=> input buffer != output buffer
6413    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6414        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6415    } else {
6416        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6417    }
6418    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6419    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6420    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6421    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6422
6423    ALOGV("configure() %p thread %p buffer %p framecount %d",
6424            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6425
6426    status_t cmdStatus;
6427    uint32_t size = sizeof(int);
6428    status_t status = (*mEffectInterface)->command(mEffectInterface,
6429                                                   EFFECT_CMD_SET_CONFIG,
6430                                                   sizeof(effect_config_t),
6431                                                   &mConfig,
6432                                                   &size,
6433                                                   &cmdStatus);
6434    if (status == 0) {
6435        status = cmdStatus;
6436    }
6437
6438    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6439            (1000 * mConfig.outputCfg.buffer.frameCount);
6440
6441    return status;
6442}
6443
6444status_t AudioFlinger::EffectModule::init()
6445{
6446    Mutex::Autolock _l(mLock);
6447    if (mEffectInterface == NULL) {
6448        return NO_INIT;
6449    }
6450    status_t cmdStatus;
6451    uint32_t size = sizeof(status_t);
6452    status_t status = (*mEffectInterface)->command(mEffectInterface,
6453                                                   EFFECT_CMD_INIT,
6454                                                   0,
6455                                                   NULL,
6456                                                   &size,
6457                                                   &cmdStatus);
6458    if (status == 0) {
6459        status = cmdStatus;
6460    }
6461    return status;
6462}
6463
6464status_t AudioFlinger::EffectModule::start()
6465{
6466    Mutex::Autolock _l(mLock);
6467    return start_l();
6468}
6469
6470status_t AudioFlinger::EffectModule::start_l()
6471{
6472    if (mEffectInterface == NULL) {
6473        return NO_INIT;
6474    }
6475    status_t cmdStatus;
6476    uint32_t size = sizeof(status_t);
6477    status_t status = (*mEffectInterface)->command(mEffectInterface,
6478                                                   EFFECT_CMD_ENABLE,
6479                                                   0,
6480                                                   NULL,
6481                                                   &size,
6482                                                   &cmdStatus);
6483    if (status == 0) {
6484        status = cmdStatus;
6485    }
6486    if (status == 0 &&
6487            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6488             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6489        sp<ThreadBase> thread = mThread.promote();
6490        if (thread != 0) {
6491            audio_stream_t *stream = thread->stream();
6492            if (stream != NULL) {
6493                stream->add_audio_effect(stream, mEffectInterface);
6494            }
6495        }
6496    }
6497    return status;
6498}
6499
6500status_t AudioFlinger::EffectModule::stop()
6501{
6502    Mutex::Autolock _l(mLock);
6503    return stop_l();
6504}
6505
6506status_t AudioFlinger::EffectModule::stop_l()
6507{
6508    if (mEffectInterface == NULL) {
6509        return NO_INIT;
6510    }
6511    status_t cmdStatus;
6512    uint32_t size = sizeof(status_t);
6513    status_t status = (*mEffectInterface)->command(mEffectInterface,
6514                                                   EFFECT_CMD_DISABLE,
6515                                                   0,
6516                                                   NULL,
6517                                                   &size,
6518                                                   &cmdStatus);
6519    if (status == 0) {
6520        status = cmdStatus;
6521    }
6522    if (status == 0 &&
6523            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6524             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6525        sp<ThreadBase> thread = mThread.promote();
6526        if (thread != 0) {
6527            audio_stream_t *stream = thread->stream();
6528            if (stream != NULL) {
6529                stream->remove_audio_effect(stream, mEffectInterface);
6530            }
6531        }
6532    }
6533    return status;
6534}
6535
6536status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6537                                             uint32_t cmdSize,
6538                                             void *pCmdData,
6539                                             uint32_t *replySize,
6540                                             void *pReplyData)
6541{
6542    Mutex::Autolock _l(mLock);
6543//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6544
6545    if (mState == DESTROYED || mEffectInterface == NULL) {
6546        return NO_INIT;
6547    }
6548    status_t status = (*mEffectInterface)->command(mEffectInterface,
6549                                                   cmdCode,
6550                                                   cmdSize,
6551                                                   pCmdData,
6552                                                   replySize,
6553                                                   pReplyData);
6554    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6555        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6556        for (size_t i = 1; i < mHandles.size(); i++) {
6557            sp<EffectHandle> h = mHandles[i].promote();
6558            if (h != 0) {
6559                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6560            }
6561        }
6562    }
6563    return status;
6564}
6565
6566status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6567{
6568
6569    Mutex::Autolock _l(mLock);
6570    ALOGV("setEnabled %p enabled %d", this, enabled);
6571
6572    if (enabled != isEnabled()) {
6573        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6574        if (enabled && status != NO_ERROR) {
6575            return status;
6576        }
6577
6578        switch (mState) {
6579        // going from disabled to enabled
6580        case IDLE:
6581            mState = STARTING;
6582            break;
6583        case STOPPED:
6584            mState = RESTART;
6585            break;
6586        case STOPPING:
6587            mState = ACTIVE;
6588            break;
6589
6590        // going from enabled to disabled
6591        case RESTART:
6592            mState = STOPPED;
6593            break;
6594        case STARTING:
6595            mState = IDLE;
6596            break;
6597        case ACTIVE:
6598            mState = STOPPING;
6599            break;
6600        case DESTROYED:
6601            return NO_ERROR; // simply ignore as we are being destroyed
6602        }
6603        for (size_t i = 1; i < mHandles.size(); i++) {
6604            sp<EffectHandle> h = mHandles[i].promote();
6605            if (h != 0) {
6606                h->setEnabled(enabled);
6607            }
6608        }
6609    }
6610    return NO_ERROR;
6611}
6612
6613bool AudioFlinger::EffectModule::isEnabled()
6614{
6615    switch (mState) {
6616    case RESTART:
6617    case STARTING:
6618    case ACTIVE:
6619        return true;
6620    case IDLE:
6621    case STOPPING:
6622    case STOPPED:
6623    case DESTROYED:
6624    default:
6625        return false;
6626    }
6627}
6628
6629bool AudioFlinger::EffectModule::isProcessEnabled()
6630{
6631    switch (mState) {
6632    case RESTART:
6633    case ACTIVE:
6634    case STOPPING:
6635    case STOPPED:
6636        return true;
6637    case IDLE:
6638    case STARTING:
6639    case DESTROYED:
6640    default:
6641        return false;
6642    }
6643}
6644
6645status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6646{
6647    Mutex::Autolock _l(mLock);
6648    status_t status = NO_ERROR;
6649
6650    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6651    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6652    if (isProcessEnabled() &&
6653            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6654            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6655        status_t cmdStatus;
6656        uint32_t volume[2];
6657        uint32_t *pVolume = NULL;
6658        uint32_t size = sizeof(volume);
6659        volume[0] = *left;
6660        volume[1] = *right;
6661        if (controller) {
6662            pVolume = volume;
6663        }
6664        status = (*mEffectInterface)->command(mEffectInterface,
6665                                              EFFECT_CMD_SET_VOLUME,
6666                                              size,
6667                                              volume,
6668                                              &size,
6669                                              pVolume);
6670        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6671            *left = volume[0];
6672            *right = volume[1];
6673        }
6674    }
6675    return status;
6676}
6677
6678status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6679{
6680    Mutex::Autolock _l(mLock);
6681    status_t status = NO_ERROR;
6682    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6683        // audio pre processing modules on RecordThread can receive both output and
6684        // input device indication in the same call
6685        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6686        if (dev) {
6687            status_t cmdStatus;
6688            uint32_t size = sizeof(status_t);
6689
6690            status = (*mEffectInterface)->command(mEffectInterface,
6691                                                  EFFECT_CMD_SET_DEVICE,
6692                                                  sizeof(uint32_t),
6693                                                  &dev,
6694                                                  &size,
6695                                                  &cmdStatus);
6696            if (status == NO_ERROR) {
6697                status = cmdStatus;
6698            }
6699        }
6700        dev = device & AUDIO_DEVICE_IN_ALL;
6701        if (dev) {
6702            status_t cmdStatus;
6703            uint32_t size = sizeof(status_t);
6704
6705            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6706                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6707                                                  sizeof(uint32_t),
6708                                                  &dev,
6709                                                  &size,
6710                                                  &cmdStatus);
6711            if (status2 == NO_ERROR) {
6712                status2 = cmdStatus;
6713            }
6714            if (status == NO_ERROR) {
6715                status = status2;
6716            }
6717        }
6718    }
6719    return status;
6720}
6721
6722status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
6723{
6724    Mutex::Autolock _l(mLock);
6725    status_t status = NO_ERROR;
6726    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6727        status_t cmdStatus;
6728        uint32_t size = sizeof(status_t);
6729        status = (*mEffectInterface)->command(mEffectInterface,
6730                                              EFFECT_CMD_SET_AUDIO_MODE,
6731                                              sizeof(audio_mode_t),
6732                                              &mode,
6733                                              &size,
6734                                              &cmdStatus);
6735        if (status == NO_ERROR) {
6736            status = cmdStatus;
6737        }
6738    }
6739    return status;
6740}
6741
6742void AudioFlinger::EffectModule::setSuspended(bool suspended)
6743{
6744    Mutex::Autolock _l(mLock);
6745    mSuspended = suspended;
6746}
6747
6748bool AudioFlinger::EffectModule::suspended() const
6749{
6750    Mutex::Autolock _l(mLock);
6751    return mSuspended;
6752}
6753
6754status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6755{
6756    const size_t SIZE = 256;
6757    char buffer[SIZE];
6758    String8 result;
6759
6760    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6761    result.append(buffer);
6762
6763    bool locked = tryLock(mLock);
6764    // failed to lock - AudioFlinger is probably deadlocked
6765    if (!locked) {
6766        result.append("\t\tCould not lock Fx mutex:\n");
6767    }
6768
6769    result.append("\t\tSession Status State Engine:\n");
6770    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6771            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6772    result.append(buffer);
6773
6774    result.append("\t\tDescriptor:\n");
6775    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6776            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6777            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6778            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6779    result.append(buffer);
6780    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6781                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6782                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6783                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6784    result.append(buffer);
6785    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6786            mDescriptor.apiVersion,
6787            mDescriptor.flags);
6788    result.append(buffer);
6789    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6790            mDescriptor.name);
6791    result.append(buffer);
6792    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6793            mDescriptor.implementor);
6794    result.append(buffer);
6795
6796    result.append("\t\t- Input configuration:\n");
6797    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6798    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6799            (uint32_t)mConfig.inputCfg.buffer.raw,
6800            mConfig.inputCfg.buffer.frameCount,
6801            mConfig.inputCfg.samplingRate,
6802            mConfig.inputCfg.channels,
6803            mConfig.inputCfg.format);
6804    result.append(buffer);
6805
6806    result.append("\t\t- Output configuration:\n");
6807    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6808    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6809            (uint32_t)mConfig.outputCfg.buffer.raw,
6810            mConfig.outputCfg.buffer.frameCount,
6811            mConfig.outputCfg.samplingRate,
6812            mConfig.outputCfg.channels,
6813            mConfig.outputCfg.format);
6814    result.append(buffer);
6815
6816    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6817    result.append(buffer);
6818    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6819    for (size_t i = 0; i < mHandles.size(); ++i) {
6820        sp<EffectHandle> handle = mHandles[i].promote();
6821        if (handle != 0) {
6822            handle->dump(buffer, SIZE);
6823            result.append(buffer);
6824        }
6825    }
6826
6827    result.append("\n");
6828
6829    write(fd, result.string(), result.length());
6830
6831    if (locked) {
6832        mLock.unlock();
6833    }
6834
6835    return NO_ERROR;
6836}
6837
6838// ----------------------------------------------------------------------------
6839//  EffectHandle implementation
6840// ----------------------------------------------------------------------------
6841
6842#undef LOG_TAG
6843#define LOG_TAG "AudioFlinger::EffectHandle"
6844
6845AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6846                                        const sp<AudioFlinger::Client>& client,
6847                                        const sp<IEffectClient>& effectClient,
6848                                        int32_t priority)
6849    : BnEffect(),
6850    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6851    mPriority(priority), mHasControl(false), mEnabled(false)
6852{
6853    ALOGV("constructor %p", this);
6854
6855    if (client == 0) {
6856        return;
6857    }
6858    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6859    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6860    if (mCblkMemory != 0) {
6861        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6862
6863        if (mCblk != NULL) {
6864            new(mCblk) effect_param_cblk_t();
6865            mBuffer = (uint8_t *)mCblk + bufOffset;
6866         }
6867    } else {
6868        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6869        return;
6870    }
6871}
6872
6873AudioFlinger::EffectHandle::~EffectHandle()
6874{
6875    ALOGV("Destructor %p", this);
6876    disconnect(false);
6877    ALOGV("Destructor DONE %p", this);
6878}
6879
6880status_t AudioFlinger::EffectHandle::enable()
6881{
6882    ALOGV("enable %p", this);
6883    if (!mHasControl) return INVALID_OPERATION;
6884    if (mEffect == 0) return DEAD_OBJECT;
6885
6886    if (mEnabled) {
6887        return NO_ERROR;
6888    }
6889
6890    mEnabled = true;
6891
6892    sp<ThreadBase> thread = mEffect->thread().promote();
6893    if (thread != 0) {
6894        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6895    }
6896
6897    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6898    if (mEffect->suspended()) {
6899        return NO_ERROR;
6900    }
6901
6902    status_t status = mEffect->setEnabled(true);
6903    if (status != NO_ERROR) {
6904        if (thread != 0) {
6905            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6906        }
6907        mEnabled = false;
6908    }
6909    return status;
6910}
6911
6912status_t AudioFlinger::EffectHandle::disable()
6913{
6914    ALOGV("disable %p", this);
6915    if (!mHasControl) return INVALID_OPERATION;
6916    if (mEffect == 0) return DEAD_OBJECT;
6917
6918    if (!mEnabled) {
6919        return NO_ERROR;
6920    }
6921    mEnabled = false;
6922
6923    if (mEffect->suspended()) {
6924        return NO_ERROR;
6925    }
6926
6927    status_t status = mEffect->setEnabled(false);
6928
6929    sp<ThreadBase> thread = mEffect->thread().promote();
6930    if (thread != 0) {
6931        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6932    }
6933
6934    return status;
6935}
6936
6937void AudioFlinger::EffectHandle::disconnect()
6938{
6939    disconnect(true);
6940}
6941
6942void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6943{
6944    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6945    if (mEffect == 0) {
6946        return;
6947    }
6948    mEffect->disconnect(this, unpiniflast);
6949
6950    if (mHasControl && mEnabled) {
6951        sp<ThreadBase> thread = mEffect->thread().promote();
6952        if (thread != 0) {
6953            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6954        }
6955    }
6956
6957    // release sp on module => module destructor can be called now
6958    mEffect.clear();
6959    if (mClient != 0) {
6960        if (mCblk != NULL) {
6961            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6962        }
6963        mCblkMemory.clear();            // and free the shared memory
6964        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6965        mClient.clear();
6966    }
6967}
6968
6969status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6970                                             uint32_t cmdSize,
6971                                             void *pCmdData,
6972                                             uint32_t *replySize,
6973                                             void *pReplyData)
6974{
6975//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6976//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6977
6978    // only get parameter command is permitted for applications not controlling the effect
6979    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6980        return INVALID_OPERATION;
6981    }
6982    if (mEffect == 0) return DEAD_OBJECT;
6983    if (mClient == 0) return INVALID_OPERATION;
6984
6985    // handle commands that are not forwarded transparently to effect engine
6986    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6987        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6988        // no risk to block the whole media server process or mixer threads is we are stuck here
6989        Mutex::Autolock _l(mCblk->lock);
6990        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6991            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6992            mCblk->serverIndex = 0;
6993            mCblk->clientIndex = 0;
6994            return BAD_VALUE;
6995        }
6996        status_t status = NO_ERROR;
6997        while (mCblk->serverIndex < mCblk->clientIndex) {
6998            int reply;
6999            uint32_t rsize = sizeof(int);
7000            int *p = (int *)(mBuffer + mCblk->serverIndex);
7001            int size = *p++;
7002            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
7003                ALOGW("command(): invalid parameter block size");
7004                break;
7005            }
7006            effect_param_t *param = (effect_param_t *)p;
7007            if (param->psize == 0 || param->vsize == 0) {
7008                ALOGW("command(): null parameter or value size");
7009                mCblk->serverIndex += size;
7010                continue;
7011            }
7012            uint32_t psize = sizeof(effect_param_t) +
7013                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
7014                             param->vsize;
7015            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
7016                                            psize,
7017                                            p,
7018                                            &rsize,
7019                                            &reply);
7020            // stop at first error encountered
7021            if (ret != NO_ERROR) {
7022                status = ret;
7023                *(int *)pReplyData = reply;
7024                break;
7025            } else if (reply != NO_ERROR) {
7026                *(int *)pReplyData = reply;
7027                break;
7028            }
7029            mCblk->serverIndex += size;
7030        }
7031        mCblk->serverIndex = 0;
7032        mCblk->clientIndex = 0;
7033        return status;
7034    } else if (cmdCode == EFFECT_CMD_ENABLE) {
7035        *(int *)pReplyData = NO_ERROR;
7036        return enable();
7037    } else if (cmdCode == EFFECT_CMD_DISABLE) {
7038        *(int *)pReplyData = NO_ERROR;
7039        return disable();
7040    }
7041
7042    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7043}
7044
7045sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
7046    return mCblkMemory;
7047}
7048
7049void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
7050{
7051    ALOGV("setControl %p control %d", this, hasControl);
7052
7053    mHasControl = hasControl;
7054    mEnabled = enabled;
7055
7056    if (signal && mEffectClient != 0) {
7057        mEffectClient->controlStatusChanged(hasControl);
7058    }
7059}
7060
7061void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7062                                                 uint32_t cmdSize,
7063                                                 void *pCmdData,
7064                                                 uint32_t replySize,
7065                                                 void *pReplyData)
7066{
7067    if (mEffectClient != 0) {
7068        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7069    }
7070}
7071
7072
7073
7074void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7075{
7076    if (mEffectClient != 0) {
7077        mEffectClient->enableStatusChanged(enabled);
7078    }
7079}
7080
7081status_t AudioFlinger::EffectHandle::onTransact(
7082    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7083{
7084    return BnEffect::onTransact(code, data, reply, flags);
7085}
7086
7087
7088void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7089{
7090    bool locked = mCblk != NULL && tryLock(mCblk->lock);
7091
7092    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7093            (mClient == 0) ? getpid() : mClient->pid(),
7094            mPriority,
7095            mHasControl,
7096            !locked,
7097            mCblk ? mCblk->clientIndex : 0,
7098            mCblk ? mCblk->serverIndex : 0
7099            );
7100
7101    if (locked) {
7102        mCblk->lock.unlock();
7103    }
7104}
7105
7106#undef LOG_TAG
7107#define LOG_TAG "AudioFlinger::EffectChain"
7108
7109AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7110                                        int sessionId)
7111    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7112      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7113      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7114{
7115    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7116    sp<ThreadBase> thread = mThread.promote();
7117    if (thread == 0) {
7118        return;
7119    }
7120    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7121                                    thread->frameCount();
7122}
7123
7124AudioFlinger::EffectChain::~EffectChain()
7125{
7126    if (mOwnInBuffer) {
7127        delete mInBuffer;
7128    }
7129
7130}
7131
7132// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7133sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7134{
7135    size_t size = mEffects.size();
7136
7137    for (size_t i = 0; i < size; i++) {
7138        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7139            return mEffects[i];
7140        }
7141    }
7142    return 0;
7143}
7144
7145// getEffectFromId_l() must be called with ThreadBase::mLock held
7146sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7147{
7148    size_t size = mEffects.size();
7149
7150    for (size_t i = 0; i < size; i++) {
7151        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7152        if (id == 0 || mEffects[i]->id() == id) {
7153            return mEffects[i];
7154        }
7155    }
7156    return 0;
7157}
7158
7159// getEffectFromType_l() must be called with ThreadBase::mLock held
7160sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7161        const effect_uuid_t *type)
7162{
7163    size_t size = mEffects.size();
7164
7165    for (size_t i = 0; i < size; i++) {
7166        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7167            return mEffects[i];
7168        }
7169    }
7170    return 0;
7171}
7172
7173// Must be called with EffectChain::mLock locked
7174void AudioFlinger::EffectChain::process_l()
7175{
7176    sp<ThreadBase> thread = mThread.promote();
7177    if (thread == 0) {
7178        ALOGW("process_l(): cannot promote mixer thread");
7179        return;
7180    }
7181    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7182            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7183    // always process effects unless no more tracks are on the session and the effect tail
7184    // has been rendered
7185    bool doProcess = true;
7186    if (!isGlobalSession) {
7187        bool tracksOnSession = (trackCnt() != 0);
7188
7189        if (!tracksOnSession && mTailBufferCount == 0) {
7190            doProcess = false;
7191        }
7192
7193        if (activeTrackCnt() == 0) {
7194            // if no track is active and the effect tail has not been rendered,
7195            // the input buffer must be cleared here as the mixer process will not do it
7196            if (tracksOnSession || mTailBufferCount > 0) {
7197                size_t numSamples = thread->frameCount() * thread->channelCount();
7198                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7199                if (mTailBufferCount > 0) {
7200                    mTailBufferCount--;
7201                }
7202            }
7203        }
7204    }
7205
7206    size_t size = mEffects.size();
7207    if (doProcess) {
7208        for (size_t i = 0; i < size; i++) {
7209            mEffects[i]->process();
7210        }
7211    }
7212    for (size_t i = 0; i < size; i++) {
7213        mEffects[i]->updateState();
7214    }
7215}
7216
7217// addEffect_l() must be called with PlaybackThread::mLock held
7218status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7219{
7220    effect_descriptor_t desc = effect->desc();
7221    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7222
7223    Mutex::Autolock _l(mLock);
7224    effect->setChain(this);
7225    sp<ThreadBase> thread = mThread.promote();
7226    if (thread == 0) {
7227        return NO_INIT;
7228    }
7229    effect->setThread(thread);
7230
7231    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7232        // Auxiliary effects are inserted at the beginning of mEffects vector as
7233        // they are processed first and accumulated in chain input buffer
7234        mEffects.insertAt(effect, 0);
7235
7236        // the input buffer for auxiliary effect contains mono samples in
7237        // 32 bit format. This is to avoid saturation in AudoMixer
7238        // accumulation stage. Saturation is done in EffectModule::process() before
7239        // calling the process in effect engine
7240        size_t numSamples = thread->frameCount();
7241        int32_t *buffer = new int32_t[numSamples];
7242        memset(buffer, 0, numSamples * sizeof(int32_t));
7243        effect->setInBuffer((int16_t *)buffer);
7244        // auxiliary effects output samples to chain input buffer for further processing
7245        // by insert effects
7246        effect->setOutBuffer(mInBuffer);
7247    } else {
7248        // Insert effects are inserted at the end of mEffects vector as they are processed
7249        //  after track and auxiliary effects.
7250        // Insert effect order as a function of indicated preference:
7251        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7252        //  another effect is present
7253        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7254        //  last effect claiming first position
7255        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7256        //  first effect claiming last position
7257        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7258        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7259        // already present
7260
7261        int size = (int)mEffects.size();
7262        int idx_insert = size;
7263        int idx_insert_first = -1;
7264        int idx_insert_last = -1;
7265
7266        for (int i = 0; i < size; i++) {
7267            effect_descriptor_t d = mEffects[i]->desc();
7268            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7269            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7270            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7271                // check invalid effect chaining combinations
7272                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7273                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7274                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7275                    return INVALID_OPERATION;
7276                }
7277                // remember position of first insert effect and by default
7278                // select this as insert position for new effect
7279                if (idx_insert == size) {
7280                    idx_insert = i;
7281                }
7282                // remember position of last insert effect claiming
7283                // first position
7284                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7285                    idx_insert_first = i;
7286                }
7287                // remember position of first insert effect claiming
7288                // last position
7289                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7290                    idx_insert_last == -1) {
7291                    idx_insert_last = i;
7292                }
7293            }
7294        }
7295
7296        // modify idx_insert from first position if needed
7297        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7298            if (idx_insert_last != -1) {
7299                idx_insert = idx_insert_last;
7300            } else {
7301                idx_insert = size;
7302            }
7303        } else {
7304            if (idx_insert_first != -1) {
7305                idx_insert = idx_insert_first + 1;
7306            }
7307        }
7308
7309        // always read samples from chain input buffer
7310        effect->setInBuffer(mInBuffer);
7311
7312        // if last effect in the chain, output samples to chain
7313        // output buffer, otherwise to chain input buffer
7314        if (idx_insert == size) {
7315            if (idx_insert != 0) {
7316                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7317                mEffects[idx_insert-1]->configure();
7318            }
7319            effect->setOutBuffer(mOutBuffer);
7320        } else {
7321            effect->setOutBuffer(mInBuffer);
7322        }
7323        mEffects.insertAt(effect, idx_insert);
7324
7325        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7326    }
7327    effect->configure();
7328    return NO_ERROR;
7329}
7330
7331// removeEffect_l() must be called with PlaybackThread::mLock held
7332size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7333{
7334    Mutex::Autolock _l(mLock);
7335    int size = (int)mEffects.size();
7336    int i;
7337    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7338
7339    for (i = 0; i < size; i++) {
7340        if (effect == mEffects[i]) {
7341            // calling stop here will remove pre-processing effect from the audio HAL.
7342            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7343            // the middle of a read from audio HAL
7344            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7345                    mEffects[i]->state() == EffectModule::STOPPING) {
7346                mEffects[i]->stop();
7347            }
7348            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7349                delete[] effect->inBuffer();
7350            } else {
7351                if (i == size - 1 && i != 0) {
7352                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7353                    mEffects[i - 1]->configure();
7354                }
7355            }
7356            mEffects.removeAt(i);
7357            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7358            break;
7359        }
7360    }
7361
7362    return mEffects.size();
7363}
7364
7365// setDevice_l() must be called with PlaybackThread::mLock held
7366void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7367{
7368    size_t size = mEffects.size();
7369    for (size_t i = 0; i < size; i++) {
7370        mEffects[i]->setDevice(device);
7371    }
7372}
7373
7374// setMode_l() must be called with PlaybackThread::mLock held
7375void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode)
7376{
7377    size_t size = mEffects.size();
7378    for (size_t i = 0; i < size; i++) {
7379        mEffects[i]->setMode(mode);
7380    }
7381}
7382
7383// setVolume_l() must be called with PlaybackThread::mLock held
7384bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7385{
7386    uint32_t newLeft = *left;
7387    uint32_t newRight = *right;
7388    bool hasControl = false;
7389    int ctrlIdx = -1;
7390    size_t size = mEffects.size();
7391
7392    // first update volume controller
7393    for (size_t i = size; i > 0; i--) {
7394        if (mEffects[i - 1]->isProcessEnabled() &&
7395            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7396            ctrlIdx = i - 1;
7397            hasControl = true;
7398            break;
7399        }
7400    }
7401
7402    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7403        if (hasControl) {
7404            *left = mNewLeftVolume;
7405            *right = mNewRightVolume;
7406        }
7407        return hasControl;
7408    }
7409
7410    mVolumeCtrlIdx = ctrlIdx;
7411    mLeftVolume = newLeft;
7412    mRightVolume = newRight;
7413
7414    // second get volume update from volume controller
7415    if (ctrlIdx >= 0) {
7416        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7417        mNewLeftVolume = newLeft;
7418        mNewRightVolume = newRight;
7419    }
7420    // then indicate volume to all other effects in chain.
7421    // Pass altered volume to effects before volume controller
7422    // and requested volume to effects after controller
7423    uint32_t lVol = newLeft;
7424    uint32_t rVol = newRight;
7425
7426    for (size_t i = 0; i < size; i++) {
7427        if ((int)i == ctrlIdx) continue;
7428        // this also works for ctrlIdx == -1 when there is no volume controller
7429        if ((int)i > ctrlIdx) {
7430            lVol = *left;
7431            rVol = *right;
7432        }
7433        mEffects[i]->setVolume(&lVol, &rVol, false);
7434    }
7435    *left = newLeft;
7436    *right = newRight;
7437
7438    return hasControl;
7439}
7440
7441status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7442{
7443    const size_t SIZE = 256;
7444    char buffer[SIZE];
7445    String8 result;
7446
7447    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7448    result.append(buffer);
7449
7450    bool locked = tryLock(mLock);
7451    // failed to lock - AudioFlinger is probably deadlocked
7452    if (!locked) {
7453        result.append("\tCould not lock mutex:\n");
7454    }
7455
7456    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7457    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7458            mEffects.size(),
7459            (uint32_t)mInBuffer,
7460            (uint32_t)mOutBuffer,
7461            mActiveTrackCnt);
7462    result.append(buffer);
7463    write(fd, result.string(), result.size());
7464
7465    for (size_t i = 0; i < mEffects.size(); ++i) {
7466        sp<EffectModule> effect = mEffects[i];
7467        if (effect != 0) {
7468            effect->dump(fd, args);
7469        }
7470    }
7471
7472    if (locked) {
7473        mLock.unlock();
7474    }
7475
7476    return NO_ERROR;
7477}
7478
7479// must be called with ThreadBase::mLock held
7480void AudioFlinger::EffectChain::setEffectSuspended_l(
7481        const effect_uuid_t *type, bool suspend)
7482{
7483    sp<SuspendedEffectDesc> desc;
7484    // use effect type UUID timelow as key as there is no real risk of identical
7485    // timeLow fields among effect type UUIDs.
7486    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7487    if (suspend) {
7488        if (index >= 0) {
7489            desc = mSuspendedEffects.valueAt(index);
7490        } else {
7491            desc = new SuspendedEffectDesc();
7492            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7493            mSuspendedEffects.add(type->timeLow, desc);
7494            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7495        }
7496        if (desc->mRefCount++ == 0) {
7497            sp<EffectModule> effect = getEffectIfEnabled(type);
7498            if (effect != 0) {
7499                desc->mEffect = effect;
7500                effect->setSuspended(true);
7501                effect->setEnabled(false);
7502            }
7503        }
7504    } else {
7505        if (index < 0) {
7506            return;
7507        }
7508        desc = mSuspendedEffects.valueAt(index);
7509        if (desc->mRefCount <= 0) {
7510            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7511            desc->mRefCount = 1;
7512        }
7513        if (--desc->mRefCount == 0) {
7514            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7515            if (desc->mEffect != 0) {
7516                sp<EffectModule> effect = desc->mEffect.promote();
7517                if (effect != 0) {
7518                    effect->setSuspended(false);
7519                    sp<EffectHandle> handle = effect->controlHandle();
7520                    if (handle != 0) {
7521                        effect->setEnabled(handle->enabled());
7522                    }
7523                }
7524                desc->mEffect.clear();
7525            }
7526            mSuspendedEffects.removeItemsAt(index);
7527        }
7528    }
7529}
7530
7531// must be called with ThreadBase::mLock held
7532void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7533{
7534    sp<SuspendedEffectDesc> desc;
7535
7536    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7537    if (suspend) {
7538        if (index >= 0) {
7539            desc = mSuspendedEffects.valueAt(index);
7540        } else {
7541            desc = new SuspendedEffectDesc();
7542            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7543            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7544        }
7545        if (desc->mRefCount++ == 0) {
7546            Vector< sp<EffectModule> > effects;
7547            getSuspendEligibleEffects(effects);
7548            for (size_t i = 0; i < effects.size(); i++) {
7549                setEffectSuspended_l(&effects[i]->desc().type, true);
7550            }
7551        }
7552    } else {
7553        if (index < 0) {
7554            return;
7555        }
7556        desc = mSuspendedEffects.valueAt(index);
7557        if (desc->mRefCount <= 0) {
7558            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7559            desc->mRefCount = 1;
7560        }
7561        if (--desc->mRefCount == 0) {
7562            Vector<const effect_uuid_t *> types;
7563            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7564                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7565                    continue;
7566                }
7567                types.add(&mSuspendedEffects.valueAt(i)->mType);
7568            }
7569            for (size_t i = 0; i < types.size(); i++) {
7570                setEffectSuspended_l(types[i], false);
7571            }
7572            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7573            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7574        }
7575    }
7576}
7577
7578
7579// The volume effect is used for automated tests only
7580#ifndef OPENSL_ES_H_
7581static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7582                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7583const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7584#endif //OPENSL_ES_H_
7585
7586bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7587{
7588    // auxiliary effects and visualizer are never suspended on output mix
7589    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7590        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7591         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7592         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7593        return false;
7594    }
7595    return true;
7596}
7597
7598void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects)
7599{
7600    effects.clear();
7601    for (size_t i = 0; i < mEffects.size(); i++) {
7602        if (isEffectEligibleForSuspend(mEffects[i]->desc())) {
7603            effects.add(mEffects[i]);
7604        }
7605    }
7606}
7607
7608sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7609                                                            const effect_uuid_t *type)
7610{
7611    sp<EffectModule> effect = getEffectFromType_l(type);
7612    return effect != 0 && effect->isEnabled() ? effect : 0;
7613}
7614
7615void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7616                                                            bool enabled)
7617{
7618    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7619    if (enabled) {
7620        if (index < 0) {
7621            // if the effect is not suspend check if all effects are suspended
7622            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7623            if (index < 0) {
7624                return;
7625            }
7626            if (!isEffectEligibleForSuspend(effect->desc())) {
7627                return;
7628            }
7629            setEffectSuspended_l(&effect->desc().type, enabled);
7630            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7631            if (index < 0) {
7632                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7633                return;
7634            }
7635        }
7636        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7637             effect->desc().type.timeLow);
7638        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7639        // if effect is requested to suspended but was not yet enabled, supend it now.
7640        if (desc->mEffect == 0) {
7641            desc->mEffect = effect;
7642            effect->setEnabled(false);
7643            effect->setSuspended(true);
7644        }
7645    } else {
7646        if (index < 0) {
7647            return;
7648        }
7649        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7650             effect->desc().type.timeLow);
7651        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7652        desc->mEffect.clear();
7653        effect->setSuspended(false);
7654    }
7655}
7656
7657#undef LOG_TAG
7658#define LOG_TAG "AudioFlinger"
7659
7660// ----------------------------------------------------------------------------
7661
7662status_t AudioFlinger::onTransact(
7663        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7664{
7665    return BnAudioFlinger::onTransact(code, data, reply, flags);
7666}
7667
7668}; // namespace android
7669