AudioFlinger.cpp revision bb001926447d0b7dc71ca8bb3c9856f3136d8f4c 
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 != NULL) {
813        result = thread->setParameters(keyValuePairs);
814        return result;
815    }
816    return BAD_VALUE;
817}
818
819String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
820{
821//    ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
822//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
823
824    if (ioHandle == 0) {
825        String8 out_s8;
826
827        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
828            audio_hw_device_t *dev = mAudioHwDevs[i];
829            char *s = dev->get_parameters(dev, keys.string());
830            out_s8 += String8(s);
831            free(s);
832        }
833        return out_s8;
834    }
835
836    Mutex::Autolock _l(mLock);
837
838    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
839    if (playbackThread != NULL) {
840        return playbackThread->getParameters(keys);
841    }
842    RecordThread *recordThread = checkRecordThread_l(ioHandle);
843    if (recordThread != NULL) {
844        return recordThread->getParameters(keys);
845    }
846    return String8("");
847}
848
849size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount)
850{
851    status_t ret = initCheck();
852    if (ret != NO_ERROR) {
853        return 0;
854    }
855
856    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
857}
858
859unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
860{
861    if (ioHandle == 0) {
862        return 0;
863    }
864
865    Mutex::Autolock _l(mLock);
866
867    RecordThread *recordThread = checkRecordThread_l(ioHandle);
868    if (recordThread != NULL) {
869        return recordThread->getInputFramesLost();
870    }
871    return 0;
872}
873
874status_t AudioFlinger::setVoiceVolume(float value)
875{
876    status_t ret = initCheck();
877    if (ret != NO_ERROR) {
878        return ret;
879    }
880
881    // check calling permissions
882    if (!settingsAllowed()) {
883        return PERMISSION_DENIED;
884    }
885
886    AutoMutex lock(mHardwareLock);
887    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
888    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
889    mHardwareStatus = AUDIO_HW_IDLE;
890
891    return ret;
892}
893
894status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
895{
896    status_t status;
897
898    Mutex::Autolock _l(mLock);
899
900    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
901    if (playbackThread != NULL) {
902        return playbackThread->getRenderPosition(halFrames, dspFrames);
903    }
904
905    return BAD_VALUE;
906}
907
908void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
909{
910
911    Mutex::Autolock _l(mLock);
912
913    pid_t pid = IPCThreadState::self()->getCallingPid();
914    if (mNotificationClients.indexOfKey(pid) < 0) {
915        sp<NotificationClient> notificationClient = new NotificationClient(this,
916                                                                            client,
917                                                                            pid);
918        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
919
920        mNotificationClients.add(pid, notificationClient);
921
922        sp<IBinder> binder = client->asBinder();
923        binder->linkToDeath(notificationClient);
924
925        // the config change is always sent from playback or record threads to avoid deadlock
926        // with AudioSystem::gLock
927        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
928            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
929        }
930
931        for (size_t i = 0; i < mRecordThreads.size(); i++) {
932            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
933        }
934    }
935}
936
937void AudioFlinger::removeNotificationClient(pid_t pid)
938{
939    Mutex::Autolock _l(mLock);
940
941    int index = mNotificationClients.indexOfKey(pid);
942    if (index >= 0) {
943        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
944        ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
945        mNotificationClients.removeItem(pid);
946    }
947
948    ALOGV("%d died, releasing its sessions", pid);
949    int num = mAudioSessionRefs.size();
950    bool removed = false;
951    for (int i = 0; i< num; i++) {
952        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
953        ALOGV(" pid %d @ %d", ref->pid, i);
954        if (ref->pid == pid) {
955            ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
956            mAudioSessionRefs.removeAt(i);
957            delete ref;
958            removed = true;
959            i--;
960            num--;
961        }
962    }
963    if (removed) {
964        purgeStaleEffects_l();
965    }
966}
967
968// audioConfigChanged_l() must be called with AudioFlinger::mLock held
969void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
970{
971    size_t size = mNotificationClients.size();
972    for (size_t i = 0; i < size; i++) {
973        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
974                                                                               param2);
975    }
976}
977
978// removeClient_l() must be called with AudioFlinger::mLock held
979void AudioFlinger::removeClient_l(pid_t pid)
980{
981    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
982    mClients.removeItem(pid);
983}
984
985
986// ----------------------------------------------------------------------------
987
988AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device,
989        type_t type)
990    :   Thread(false),
991        mType(type),
992        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0),
993        // mChannelMask
994        mChannelCount(0),
995        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
996        mParamStatus(NO_ERROR),
997        mStandby(false), mId(id), mExiting(false),
998        mDevice(device),
999        mDeathRecipient(new PMDeathRecipient(this))
1000{
1001}
1002
1003AudioFlinger::ThreadBase::~ThreadBase()
1004{
1005    mParamCond.broadcast();
1006    // do not lock the mutex in destructor
1007    releaseWakeLock_l();
1008    if (mPowerManager != 0) {
1009        sp<IBinder> binder = mPowerManager->asBinder();
1010        binder->unlinkToDeath(mDeathRecipient);
1011    }
1012}
1013
1014void AudioFlinger::ThreadBase::exit()
1015{
1016    // keep a strong ref on ourself so that we won't get
1017    // destroyed in the middle of requestExitAndWait()
1018    sp <ThreadBase> strongMe = this;
1019
1020    ALOGV("ThreadBase::exit");
1021    {
1022        AutoMutex lock(mLock);
1023        mExiting = true;
1024        requestExit();
1025        mWaitWorkCV.signal();
1026    }
1027    requestExitAndWait();
1028}
1029
1030uint32_t AudioFlinger::ThreadBase::sampleRate() const
1031{
1032    return mSampleRate;
1033}
1034
1035int AudioFlinger::ThreadBase::channelCount() const
1036{
1037    return (int)mChannelCount;
1038}
1039
1040audio_format_t AudioFlinger::ThreadBase::format() const
1041{
1042    return mFormat;
1043}
1044
1045size_t AudioFlinger::ThreadBase::frameCount() const
1046{
1047    return mFrameCount;
1048}
1049
1050status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1051{
1052    status_t status;
1053
1054    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1055    Mutex::Autolock _l(mLock);
1056
1057    mNewParameters.add(keyValuePairs);
1058    mWaitWorkCV.signal();
1059    // wait condition with timeout in case the thread loop has exited
1060    // before the request could be processed
1061    if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
1062        status = mParamStatus;
1063        mWaitWorkCV.signal();
1064    } else {
1065        status = TIMED_OUT;
1066    }
1067    return status;
1068}
1069
1070void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1071{
1072    Mutex::Autolock _l(mLock);
1073    sendConfigEvent_l(event, param);
1074}
1075
1076// sendConfigEvent_l() must be called with ThreadBase::mLock held
1077void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1078{
1079    ConfigEvent configEvent;
1080    configEvent.mEvent = event;
1081    configEvent.mParam = param;
1082    mConfigEvents.add(configEvent);
1083    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1084    mWaitWorkCV.signal();
1085}
1086
1087void AudioFlinger::ThreadBase::processConfigEvents()
1088{
1089    mLock.lock();
1090    while(!mConfigEvents.isEmpty()) {
1091        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1092        ConfigEvent configEvent = mConfigEvents[0];
1093        mConfigEvents.removeAt(0);
1094        // release mLock before locking AudioFlinger mLock: lock order is always
1095        // AudioFlinger then ThreadBase to avoid cross deadlock
1096        mLock.unlock();
1097        mAudioFlinger->mLock.lock();
1098        audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
1099        mAudioFlinger->mLock.unlock();
1100        mLock.lock();
1101    }
1102    mLock.unlock();
1103}
1104
1105status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1106{
1107    const size_t SIZE = 256;
1108    char buffer[SIZE];
1109    String8 result;
1110
1111    bool locked = tryLock(mLock);
1112    if (!locked) {
1113        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1114        write(fd, buffer, strlen(buffer));
1115    }
1116
1117    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1118    result.append(buffer);
1119    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1120    result.append(buffer);
1121    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1122    result.append(buffer);
1123    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1124    result.append(buffer);
1125    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1126    result.append(buffer);
1127    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1128    result.append(buffer);
1129    snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
1130    result.append(buffer);
1131
1132    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1133    result.append(buffer);
1134    result.append(" Index Command");
1135    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1136        snprintf(buffer, SIZE, "\n %02d    ", i);
1137        result.append(buffer);
1138        result.append(mNewParameters[i]);
1139    }
1140
1141    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1142    result.append(buffer);
1143    snprintf(buffer, SIZE, " Index event param\n");
1144    result.append(buffer);
1145    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1146        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
1147        result.append(buffer);
1148    }
1149    result.append("\n");
1150
1151    write(fd, result.string(), result.size());
1152
1153    if (locked) {
1154        mLock.unlock();
1155    }
1156    return NO_ERROR;
1157}
1158
1159status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1160{
1161    const size_t SIZE = 256;
1162    char buffer[SIZE];
1163    String8 result;
1164
1165    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1166    write(fd, buffer, strlen(buffer));
1167
1168    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1169        sp<EffectChain> chain = mEffectChains[i];
1170        if (chain != 0) {
1171            chain->dump(fd, args);
1172        }
1173    }
1174    return NO_ERROR;
1175}
1176
1177void AudioFlinger::ThreadBase::acquireWakeLock()
1178{
1179    Mutex::Autolock _l(mLock);
1180    acquireWakeLock_l();
1181}
1182
1183void AudioFlinger::ThreadBase::acquireWakeLock_l()
1184{
1185    if (mPowerManager == 0) {
1186        // use checkService() to avoid blocking if power service is not up yet
1187        sp<IBinder> binder =
1188            defaultServiceManager()->checkService(String16("power"));
1189        if (binder == 0) {
1190            ALOGW("Thread %s cannot connect to the power manager service", mName);
1191        } else {
1192            mPowerManager = interface_cast<IPowerManager>(binder);
1193            binder->linkToDeath(mDeathRecipient);
1194        }
1195    }
1196    if (mPowerManager != 0) {
1197        sp<IBinder> binder = new BBinder();
1198        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
1199                                                         binder,
1200                                                         String16(mName));
1201        if (status == NO_ERROR) {
1202            mWakeLockToken = binder;
1203        }
1204        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
1205    }
1206}
1207
1208void AudioFlinger::ThreadBase::releaseWakeLock()
1209{
1210    Mutex::Autolock _l(mLock);
1211    releaseWakeLock_l();
1212}
1213
1214void AudioFlinger::ThreadBase::releaseWakeLock_l()
1215{
1216    if (mWakeLockToken != 0) {
1217        ALOGV("releaseWakeLock_l() %s", mName);
1218        if (mPowerManager != 0) {
1219            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
1220        }
1221        mWakeLockToken.clear();
1222    }
1223}
1224
1225void AudioFlinger::ThreadBase::clearPowerManager()
1226{
1227    Mutex::Autolock _l(mLock);
1228    releaseWakeLock_l();
1229    mPowerManager.clear();
1230}
1231
1232void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
1233{
1234    sp<ThreadBase> thread = mThread.promote();
1235    if (thread != 0) {
1236        thread->clearPowerManager();
1237    }
1238    ALOGW("power manager service died !!!");
1239}
1240
1241void AudioFlinger::ThreadBase::setEffectSuspended(
1242        const effect_uuid_t *type, bool suspend, int sessionId)
1243{
1244    Mutex::Autolock _l(mLock);
1245    setEffectSuspended_l(type, suspend, sessionId);
1246}
1247
1248void AudioFlinger::ThreadBase::setEffectSuspended_l(
1249        const effect_uuid_t *type, bool suspend, int sessionId)
1250{
1251    sp<EffectChain> chain;
1252    chain = getEffectChain_l(sessionId);
1253    if (chain != 0) {
1254        if (type != NULL) {
1255            chain->setEffectSuspended_l(type, suspend);
1256        } else {
1257            chain->setEffectSuspendedAll_l(suspend);
1258        }
1259    }
1260
1261    updateSuspendedSessions_l(type, suspend, sessionId);
1262}
1263
1264void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1265{
1266    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1267    if (index < 0) {
1268        return;
1269    }
1270
1271    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1272            mSuspendedSessions.editValueAt(index);
1273
1274    for (size_t i = 0; i < sessionEffects.size(); i++) {
1275        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1276        for (int j = 0; j < desc->mRefCount; j++) {
1277            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1278                chain->setEffectSuspendedAll_l(true);
1279            } else {
1280                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1281                     desc->mType.timeLow);
1282                chain->setEffectSuspended_l(&desc->mType, true);
1283            }
1284        }
1285    }
1286}
1287
1288void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1289                                                         bool suspend,
1290                                                         int sessionId)
1291{
1292    int index = mSuspendedSessions.indexOfKey(sessionId);
1293
1294    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1295
1296    if (suspend) {
1297        if (index >= 0) {
1298            sessionEffects = mSuspendedSessions.editValueAt(index);
1299        } else {
1300            mSuspendedSessions.add(sessionId, sessionEffects);
1301        }
1302    } else {
1303        if (index < 0) {
1304            return;
1305        }
1306        sessionEffects = mSuspendedSessions.editValueAt(index);
1307    }
1308
1309
1310    int key = EffectChain::kKeyForSuspendAll;
1311    if (type != NULL) {
1312        key = type->timeLow;
1313    }
1314    index = sessionEffects.indexOfKey(key);
1315
1316    sp <SuspendedSessionDesc> desc;
1317    if (suspend) {
1318        if (index >= 0) {
1319            desc = sessionEffects.valueAt(index);
1320        } else {
1321            desc = new SuspendedSessionDesc();
1322            if (type != NULL) {
1323                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1324            }
1325            sessionEffects.add(key, desc);
1326            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1327        }
1328        desc->mRefCount++;
1329    } else {
1330        if (index < 0) {
1331            return;
1332        }
1333        desc = sessionEffects.valueAt(index);
1334        if (--desc->mRefCount == 0) {
1335            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1336            sessionEffects.removeItemsAt(index);
1337            if (sessionEffects.isEmpty()) {
1338                ALOGV("updateSuspendedSessions_l() restore removing session %d",
1339                                 sessionId);
1340                mSuspendedSessions.removeItem(sessionId);
1341            }
1342        }
1343    }
1344    if (!sessionEffects.isEmpty()) {
1345        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1346    }
1347}
1348
1349void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1350                                                            bool enabled,
1351                                                            int sessionId)
1352{
1353    Mutex::Autolock _l(mLock);
1354    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1355}
1356
1357void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1358                                                            bool enabled,
1359                                                            int sessionId)
1360{
1361    if (mType != RECORD) {
1362        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1363        // another session. This gives the priority to well behaved effect control panels
1364        // and applications not using global effects.
1365        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1366            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1367        }
1368    }
1369
1370    sp<EffectChain> chain = getEffectChain_l(sessionId);
1371    if (chain != 0) {
1372        chain->checkSuspendOnEffectEnabled(effect, enabled);
1373    }
1374}
1375
1376// ----------------------------------------------------------------------------
1377
1378AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1379                                             AudioStreamOut* output,
1380                                             int id,
1381                                             uint32_t device,
1382                                             type_t type)
1383    :   ThreadBase(audioFlinger, id, device, type),
1384        mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
1385        // Assumes constructor is called by AudioFlinger with it's mLock held,
1386        // but it would be safer to explicitly pass initial masterMute as parameter
1387        mMasterMute(audioFlinger->masterMute_l()),
1388        // mStreamTypes[] initialized in constructor body
1389        mOutput(output),
1390        // Assumes constructor is called by AudioFlinger with it's mLock held,
1391        // but it would be safer to explicitly pass initial masterVolume as parameter
1392        mMasterVolume(audioFlinger->masterVolume_l()),
1393        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1394{
1395    snprintf(mName, kNameLength, "AudioOut_%d", id);
1396
1397    readOutputParameters();
1398
1399    // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
1400    // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
1401    for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
1402            stream = (audio_stream_type_t) (stream + 1)) {
1403        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1404        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1405        // initialized by stream_type_t default constructor
1406        // mStreamTypes[stream].valid = true;
1407    }
1408}
1409
1410AudioFlinger::PlaybackThread::~PlaybackThread()
1411{
1412    delete [] mMixBuffer;
1413}
1414
1415status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1416{
1417    dumpInternals(fd, args);
1418    dumpTracks(fd, args);
1419    dumpEffectChains(fd, args);
1420    return NO_ERROR;
1421}
1422
1423status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1424{
1425    const size_t SIZE = 256;
1426    char buffer[SIZE];
1427    String8 result;
1428
1429    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1430    result.append(buffer);
1431    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1432    for (size_t i = 0; i < mTracks.size(); ++i) {
1433        sp<Track> track = mTracks[i];
1434        if (track != 0) {
1435            track->dump(buffer, SIZE);
1436            result.append(buffer);
1437        }
1438    }
1439
1440    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1441    result.append(buffer);
1442    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1443    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1444        sp<Track> track = mActiveTracks[i].promote();
1445        if (track != 0) {
1446            track->dump(buffer, SIZE);
1447            result.append(buffer);
1448        }
1449    }
1450    write(fd, result.string(), result.size());
1451    return NO_ERROR;
1452}
1453
1454status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1455{
1456    const size_t SIZE = 256;
1457    char buffer[SIZE];
1458    String8 result;
1459
1460    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1461    result.append(buffer);
1462    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1463    result.append(buffer);
1464    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1465    result.append(buffer);
1466    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1467    result.append(buffer);
1468    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1469    result.append(buffer);
1470    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1471    result.append(buffer);
1472    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1473    result.append(buffer);
1474    write(fd, result.string(), result.size());
1475
1476    dumpBase(fd, args);
1477
1478    return NO_ERROR;
1479}
1480
1481// Thread virtuals
1482status_t AudioFlinger::PlaybackThread::readyToRun()
1483{
1484    status_t status = initCheck();
1485    if (status == NO_ERROR) {
1486        ALOGI("AudioFlinger's thread %p ready to run", this);
1487    } else {
1488        ALOGE("No working audio driver found.");
1489    }
1490    return status;
1491}
1492
1493void AudioFlinger::PlaybackThread::onFirstRef()
1494{
1495    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1496}
1497
1498// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1499sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1500        const sp<AudioFlinger::Client>& client,
1501        audio_stream_type_t streamType,
1502        uint32_t sampleRate,
1503        audio_format_t format,
1504        uint32_t channelMask,
1505        int frameCount,
1506        const sp<IMemory>& sharedBuffer,
1507        int sessionId,
1508        status_t *status)
1509{
1510    sp<Track> track;
1511    status_t lStatus;
1512
1513    if (mType == DIRECT) {
1514        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1515            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1516                ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1517                        "for output %p with format %d",
1518                        sampleRate, format, channelMask, mOutput, mFormat);
1519                lStatus = BAD_VALUE;
1520                goto Exit;
1521            }
1522        }
1523    } else {
1524        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1525        if (sampleRate > mSampleRate*2) {
1526            ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1527            lStatus = BAD_VALUE;
1528            goto Exit;
1529        }
1530    }
1531
1532    lStatus = initCheck();
1533    if (lStatus != NO_ERROR) {
1534        ALOGE("Audio driver not initialized.");
1535        goto Exit;
1536    }
1537
1538    { // scope for mLock
1539        Mutex::Autolock _l(mLock);
1540
1541        // all tracks in same audio session must share the same routing strategy otherwise
1542        // conflicts will happen when tracks are moved from one output to another by audio policy
1543        // manager
1544        uint32_t strategy =
1545                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1546        for (size_t i = 0; i < mTracks.size(); ++i) {
1547            sp<Track> t = mTracks[i];
1548            if (t != 0) {
1549                uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
1550                if (sessionId == t->sessionId() && strategy != actual) {
1551                    ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1552                            strategy, actual);
1553                    lStatus = BAD_VALUE;
1554                    goto Exit;
1555                }
1556            }
1557        }
1558
1559        track = new Track(this, client, streamType, sampleRate, format,
1560                channelMask, frameCount, sharedBuffer, sessionId);
1561        if (track->getCblk() == NULL || track->name() < 0) {
1562            lStatus = NO_MEMORY;
1563            goto Exit;
1564        }
1565        mTracks.add(track);
1566
1567        sp<EffectChain> chain = getEffectChain_l(sessionId);
1568        if (chain != 0) {
1569            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1570            track->setMainBuffer(chain->inBuffer());
1571            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1572            chain->incTrackCnt();
1573        }
1574
1575        // invalidate track immediately if the stream type was moved to another thread since
1576        // createTrack() was called by the client process.
1577        if (!mStreamTypes[streamType].valid) {
1578            ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1579                 this, streamType);
1580            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1581        }
1582    }
1583    lStatus = NO_ERROR;
1584
1585Exit:
1586    if(status) {
1587        *status = lStatus;
1588    }
1589    return track;
1590}
1591
1592uint32_t AudioFlinger::PlaybackThread::latency() const
1593{
1594    Mutex::Autolock _l(mLock);
1595    if (initCheck() == NO_ERROR) {
1596        return mOutput->stream->get_latency(mOutput->stream);
1597    } else {
1598        return 0;
1599    }
1600}
1601
1602status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1603{
1604    mMasterVolume = value;
1605    return NO_ERROR;
1606}
1607
1608status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1609{
1610    mMasterMute = muted;
1611    return NO_ERROR;
1612}
1613
1614float AudioFlinger::PlaybackThread::masterVolume() const
1615{
1616    return mMasterVolume;
1617}
1618
1619bool AudioFlinger::PlaybackThread::masterMute() const
1620{
1621    return mMasterMute;
1622}
1623
1624status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
1625{
1626    mStreamTypes[stream].volume = value;
1627    return NO_ERROR;
1628}
1629
1630status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
1631{
1632    mStreamTypes[stream].mute = muted;
1633    return NO_ERROR;
1634}
1635
1636float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
1637{
1638    return mStreamTypes[stream].volume;
1639}
1640
1641bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const
1642{
1643    return mStreamTypes[stream].mute;
1644}
1645
1646// addTrack_l() must be called with ThreadBase::mLock held
1647status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1648{
1649    status_t status = ALREADY_EXISTS;
1650
1651    // set retry count for buffer fill
1652    track->mRetryCount = kMaxTrackStartupRetries;
1653    if (mActiveTracks.indexOf(track) < 0) {
1654        // the track is newly added, make sure it fills up all its
1655        // buffers before playing. This is to ensure the client will
1656        // effectively get the latency it requested.
1657        track->mFillingUpStatus = Track::FS_FILLING;
1658        track->mResetDone = false;
1659        mActiveTracks.add(track);
1660        if (track->mainBuffer() != mMixBuffer) {
1661            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1662            if (chain != 0) {
1663                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1664                chain->incActiveTrackCnt();
1665            }
1666        }
1667
1668        status = NO_ERROR;
1669    }
1670
1671    ALOGV("mWaitWorkCV.broadcast");
1672    mWaitWorkCV.broadcast();
1673
1674    return status;
1675}
1676
1677// destroyTrack_l() must be called with ThreadBase::mLock held
1678void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1679{
1680    track->mState = TrackBase::TERMINATED;
1681    if (mActiveTracks.indexOf(track) < 0) {
1682        removeTrack_l(track);
1683    }
1684}
1685
1686void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1687{
1688    mTracks.remove(track);
1689    deleteTrackName_l(track->name());
1690    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1691    if (chain != 0) {
1692        chain->decTrackCnt();
1693    }
1694}
1695
1696String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1697{
1698    String8 out_s8 = String8("");
1699    char *s;
1700
1701    Mutex::Autolock _l(mLock);
1702    if (initCheck() != NO_ERROR) {
1703        return out_s8;
1704    }
1705
1706    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1707    out_s8 = String8(s);
1708    free(s);
1709    return out_s8;
1710}
1711
1712// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1713void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1714    AudioSystem::OutputDescriptor desc;
1715    void *param2 = NULL;
1716
1717    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1718
1719    switch (event) {
1720    case AudioSystem::OUTPUT_OPENED:
1721    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1722        desc.channels = mChannelMask;
1723        desc.samplingRate = mSampleRate;
1724        desc.format = mFormat;
1725        desc.frameCount = mFrameCount;
1726        desc.latency = latency();
1727        param2 = &desc;
1728        break;
1729
1730    case AudioSystem::STREAM_CONFIG_CHANGED:
1731        param2 = &param;
1732    case AudioSystem::OUTPUT_CLOSED:
1733    default:
1734        break;
1735    }
1736    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1737}
1738
1739void AudioFlinger::PlaybackThread::readOutputParameters()
1740{
1741    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1742    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1743    mChannelCount = (uint16_t)popcount(mChannelMask);
1744    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1745    mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
1746    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1747
1748    // FIXME - Current mixer implementation only supports stereo output: Always
1749    // Allocate a stereo buffer even if HW output is mono.
1750    delete[] mMixBuffer;
1751    mMixBuffer = new int16_t[mFrameCount * 2];
1752    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1753
1754    // force reconfiguration of effect chains and engines to take new buffer size and audio
1755    // parameters into account
1756    // Note that mLock is not held when readOutputParameters() is called from the constructor
1757    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1758    // matter.
1759    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1760    Vector< sp<EffectChain> > effectChains = mEffectChains;
1761    for (size_t i = 0; i < effectChains.size(); i ++) {
1762        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1763    }
1764}
1765
1766status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1767{
1768    if (halFrames == NULL || dspFrames == NULL) {
1769        return BAD_VALUE;
1770    }
1771    Mutex::Autolock _l(mLock);
1772    if (initCheck() != NO_ERROR) {
1773        return INVALID_OPERATION;
1774    }
1775    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1776
1777    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1778}
1779
1780uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1781{
1782    Mutex::Autolock _l(mLock);
1783    uint32_t result = 0;
1784    if (getEffectChain_l(sessionId) != 0) {
1785        result = EFFECT_SESSION;
1786    }
1787
1788    for (size_t i = 0; i < mTracks.size(); ++i) {
1789        sp<Track> track = mTracks[i];
1790        if (sessionId == track->sessionId() &&
1791                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1792            result |= TRACK_SESSION;
1793            break;
1794        }
1795    }
1796
1797    return result;
1798}
1799
1800uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1801{
1802    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1803    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1804    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1805        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1806    }
1807    for (size_t i = 0; i < mTracks.size(); i++) {
1808        sp<Track> track = mTracks[i];
1809        if (sessionId == track->sessionId() &&
1810                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1811            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1812        }
1813    }
1814    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1815}
1816
1817
1818AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
1819{
1820    Mutex::Autolock _l(mLock);
1821    return mOutput;
1822}
1823
1824AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1825{
1826    Mutex::Autolock _l(mLock);
1827    AudioStreamOut *output = mOutput;
1828    mOutput = NULL;
1829    return output;
1830}
1831
1832// this method must always be called either with ThreadBase mLock held or inside the thread loop
1833audio_stream_t* AudioFlinger::PlaybackThread::stream()
1834{
1835    if (mOutput == NULL) {
1836        return NULL;
1837    }
1838    return &mOutput->stream->common;
1839}
1840
1841uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
1842{
1843    // A2DP output latency is not due only to buffering capacity. It also reflects encoding,
1844    // decoding and transfer time. So sleeping for half of the latency would likely cause
1845    // underruns
1846    if (audio_is_a2dp_device((audio_devices_t)mDevice)) {
1847        return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000);
1848    } else {
1849        return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
1850    }
1851}
1852
1853// ----------------------------------------------------------------------------
1854
1855AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
1856        int id, uint32_t device, type_t type)
1857    :   PlaybackThread(audioFlinger, output, id, device, type),
1858        mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)),
1859        mPrevMixerStatus(MIXER_IDLE)
1860{
1861    // FIXME - Current mixer implementation only supports stereo output
1862    if (mChannelCount == 1) {
1863        ALOGE("Invalid audio hardware channel count");
1864    }
1865}
1866
1867AudioFlinger::MixerThread::~MixerThread()
1868{
1869    delete mAudioMixer;
1870}
1871
1872bool AudioFlinger::MixerThread::threadLoop()
1873{
1874    Vector< sp<Track> > tracksToRemove;
1875    mixer_state mixerStatus = MIXER_IDLE;
1876    nsecs_t standbyTime = systemTime();
1877    size_t mixBufferSize = mFrameCount * mFrameSize;
1878    // FIXME: Relaxed timing because of a certain device that can't meet latency
1879    // Should be reduced to 2x after the vendor fixes the driver issue
1880    // increase threshold again due to low power audio mode. The way this warning threshold is
1881    // calculated and its usefulness should be reconsidered anyway.
1882    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1883    nsecs_t lastWarning = 0;
1884    bool longStandbyExit = false;
1885    uint32_t activeSleepTime = activeSleepTimeUs();
1886    uint32_t idleSleepTime = idleSleepTimeUs();
1887    uint32_t sleepTime = idleSleepTime;
1888    uint32_t sleepTimeShift = 0;
1889    Vector< sp<EffectChain> > effectChains;
1890#ifdef DEBUG_CPU_USAGE
1891    ThreadCpuUsage cpu;
1892    const CentralTendencyStatistics& stats = cpu.statistics();
1893#endif
1894
1895    acquireWakeLock();
1896
1897    while (!exitPending())
1898    {
1899#ifdef DEBUG_CPU_USAGE
1900        cpu.sampleAndEnable();
1901        unsigned n = stats.n();
1902        // cpu.elapsed() is expensive, so don't call it every loop
1903        if ((n & 127) == 1) {
1904            long long elapsed = cpu.elapsed();
1905            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1906                double perLoop = elapsed / (double) n;
1907                double perLoop100 = perLoop * 0.01;
1908                double mean = stats.mean();
1909                double stddev = stats.stddev();
1910                double minimum = stats.minimum();
1911                double maximum = stats.maximum();
1912                cpu.resetStatistics();
1913                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",
1914                        elapsed * .000000001, n, perLoop * .000001,
1915                        mean * .001,
1916                        stddev * .001,
1917                        minimum * .001,
1918                        maximum * .001,
1919                        mean / perLoop100,
1920                        stddev / perLoop100,
1921                        minimum / perLoop100,
1922                        maximum / perLoop100);
1923            }
1924        }
1925#endif
1926        processConfigEvents();
1927
1928        mixerStatus = MIXER_IDLE;
1929        { // scope for mLock
1930
1931            Mutex::Autolock _l(mLock);
1932
1933            if (checkForNewParameters_l()) {
1934                mixBufferSize = mFrameCount * mFrameSize;
1935                // FIXME: Relaxed timing because of a certain device that can't meet latency
1936                // Should be reduced to 2x after the vendor fixes the driver issue
1937                // increase threshold again due to low power audio mode. The way this warning
1938                // threshold is calculated and its usefulness should be reconsidered anyway.
1939                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1940                activeSleepTime = activeSleepTimeUs();
1941                idleSleepTime = idleSleepTimeUs();
1942            }
1943
1944            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1945
1946            // put audio hardware into standby after short delay
1947            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1948                        mSuspended)) {
1949                if (!mStandby) {
1950                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1951                    mOutput->stream->common.standby(&mOutput->stream->common);
1952                    mStandby = true;
1953                    mBytesWritten = 0;
1954                }
1955
1956                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1957                    // we're about to wait, flush the binder command buffer
1958                    IPCThreadState::self()->flushCommands();
1959
1960                    if (exitPending()) break;
1961
1962                    releaseWakeLock_l();
1963                    // wait until we have something to do...
1964                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1965                    mWaitWorkCV.wait(mLock);
1966                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1967                    acquireWakeLock_l();
1968
1969                    mPrevMixerStatus = MIXER_IDLE;
1970                    if (!mMasterMute) {
1971                        char value[PROPERTY_VALUE_MAX];
1972                        property_get("ro.audio.silent", value, "0");
1973                        if (atoi(value)) {
1974                            ALOGD("Silence is golden");
1975                            setMasterMute(true);
1976                        }
1977                    }
1978
1979                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1980                    sleepTime = idleSleepTime;
1981                    sleepTimeShift = 0;
1982                    continue;
1983                }
1984            }
1985
1986            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1987
1988            // prevent any changes in effect chain list and in each effect chain
1989            // during mixing and effect process as the audio buffers could be deleted
1990            // or modified if an effect is created or deleted
1991            lockEffectChains_l(effectChains);
1992        }
1993
1994        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1995            // mix buffers...
1996            mAudioMixer->process();
1997            // increase sleep time progressively when application underrun condition clears.
1998            // Only increase sleep time if the mixer is ready for two consecutive times to avoid
1999            // that a steady state of alternating ready/not ready conditions keeps the sleep time
2000            // such that we would underrun the audio HAL.
2001            if ((sleepTime == 0) && (sleepTimeShift > 0)) {
2002                sleepTimeShift--;
2003            }
2004            sleepTime = 0;
2005            standbyTime = systemTime() + kStandbyTimeInNsecs;
2006            //TODO: delay standby when effects have a tail
2007        } else {
2008            // If no tracks are ready, sleep once for the duration of an output
2009            // buffer size, then write 0s to the output
2010            if (sleepTime == 0) {
2011                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2012                    sleepTime = activeSleepTime >> sleepTimeShift;
2013                    if (sleepTime < kMinThreadSleepTimeUs) {
2014                        sleepTime = kMinThreadSleepTimeUs;
2015                    }
2016                    // reduce sleep time in case of consecutive application underruns to avoid
2017                    // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
2018                    // duration we would end up writing less data than needed by the audio HAL if
2019                    // the condition persists.
2020                    if (sleepTimeShift < kMaxThreadSleepTimeShift) {
2021                        sleepTimeShift++;
2022                    }
2023                } else {
2024                    sleepTime = idleSleepTime;
2025                }
2026            } else if (mBytesWritten != 0 ||
2027                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
2028                memset (mMixBuffer, 0, mixBufferSize);
2029                sleepTime = 0;
2030                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
2031            }
2032            // TODO add standby time extension fct of effect tail
2033        }
2034
2035        if (mSuspended) {
2036            sleepTime = suspendSleepTimeUs();
2037        }
2038        // sleepTime == 0 means we must write to audio hardware
2039        if (sleepTime == 0) {
2040            for (size_t i = 0; i < effectChains.size(); i ++) {
2041                effectChains[i]->process_l();
2042            }
2043            // enable changes in effect chain
2044            unlockEffectChains(effectChains);
2045            mLastWriteTime = systemTime();
2046            mInWrite = true;
2047            mBytesWritten += mixBufferSize;
2048
2049            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2050            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2051            mNumWrites++;
2052            mInWrite = false;
2053            nsecs_t now = systemTime();
2054            nsecs_t delta = now - mLastWriteTime;
2055            if (!mStandby && delta > maxPeriod) {
2056                mNumDelayedWrites++;
2057                if ((now - lastWarning) > kWarningThrottleNs) {
2058                    ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2059                            ns2ms(delta), mNumDelayedWrites, this);
2060                    lastWarning = now;
2061                }
2062                if (mStandby) {
2063                    longStandbyExit = true;
2064                }
2065            }
2066            mStandby = false;
2067        } else {
2068            // enable changes in effect chain
2069            unlockEffectChains(effectChains);
2070            usleep(sleepTime);
2071        }
2072
2073        // finally let go of all our tracks, without the lock held
2074        // since we can't guarantee the destructors won't acquire that
2075        // same lock.
2076        tracksToRemove.clear();
2077
2078        // Effect chains will be actually deleted here if they were removed from
2079        // mEffectChains list during mixing or effects processing
2080        effectChains.clear();
2081    }
2082
2083    if (!mStandby) {
2084        mOutput->stream->common.standby(&mOutput->stream->common);
2085    }
2086
2087    releaseWakeLock();
2088
2089    ALOGV("MixerThread %p exiting", this);
2090    return false;
2091}
2092
2093// prepareTracks_l() must be called with ThreadBase::mLock held
2094AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
2095        const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2096{
2097
2098    mixer_state mixerStatus = MIXER_IDLE;
2099    // find out which tracks need to be processed
2100    size_t count = activeTracks.size();
2101    size_t mixedTracks = 0;
2102    size_t tracksWithEffect = 0;
2103
2104    float masterVolume = mMasterVolume;
2105    bool  masterMute = mMasterMute;
2106
2107    if (masterMute) {
2108        masterVolume = 0;
2109    }
2110    // Delegate master volume control to effect in output mix effect chain if needed
2111    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2112    if (chain != 0) {
2113        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2114        chain->setVolume_l(&v, &v);
2115        masterVolume = (float)((v + (1 << 23)) >> 24);
2116        chain.clear();
2117    }
2118
2119    for (size_t i=0 ; i<count ; i++) {
2120        sp<Track> t = activeTracks[i].promote();
2121        if (t == 0) continue;
2122
2123        // this const just means the local variable doesn't change
2124        Track* const track = t.get();
2125        audio_track_cblk_t* cblk = track->cblk();
2126
2127        // The first time a track is added we wait
2128        // for all its buffers to be filled before processing it
2129        int name = track->name();
2130        // make sure that we have enough frames to mix one full buffer.
2131        // enforce this condition only once to enable draining the buffer in case the client
2132        // app does not call stop() and relies on underrun to stop:
2133        // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
2134        // during last round
2135        uint32_t minFrames = 1;
2136        if (!track->isStopped() && !track->isPausing() &&
2137                (mPrevMixerStatus == MIXER_TRACKS_READY)) {
2138            if (t->sampleRate() == (int)mSampleRate) {
2139                minFrames = mFrameCount;
2140            } else {
2141                // +1 for rounding and +1 for additional sample needed for interpolation
2142                minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
2143                // add frames already consumed but not yet released by the resampler
2144                // because cblk->framesReady() will  include these frames
2145                minFrames += mAudioMixer->getUnreleasedFrames(track->name());
2146                // the minimum track buffer size is normally twice the number of frames necessary
2147                // to fill one buffer and the resampler should not leave more than one buffer worth
2148                // of unreleased frames after each pass, but just in case...
2149                ALOG_ASSERT(minFrames <= cblk->frameCount);
2150            }
2151        }
2152        if ((cblk->framesReady() >= minFrames) && track->isReady() &&
2153                !track->isPaused() && !track->isTerminated())
2154        {
2155            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
2156
2157            mixedTracks++;
2158
2159            // track->mainBuffer() != mMixBuffer means there is an effect chain
2160            // connected to the track
2161            chain.clear();
2162            if (track->mainBuffer() != mMixBuffer) {
2163                chain = getEffectChain_l(track->sessionId());
2164                // Delegate volume control to effect in track effect chain if needed
2165                if (chain != 0) {
2166                    tracksWithEffect++;
2167                } else {
2168                    ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
2169                            name, track->sessionId());
2170                }
2171            }
2172
2173
2174            int param = AudioMixer::VOLUME;
2175            if (track->mFillingUpStatus == Track::FS_FILLED) {
2176                // no ramp for the first volume setting
2177                track->mFillingUpStatus = Track::FS_ACTIVE;
2178                if (track->mState == TrackBase::RESUMING) {
2179                    track->mState = TrackBase::ACTIVE;
2180                    param = AudioMixer::RAMP_VOLUME;
2181                }
2182                mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2183            } else if (cblk->server != 0) {
2184                // If the track is stopped before the first frame was mixed,
2185                // do not apply ramp
2186                param = AudioMixer::RAMP_VOLUME;
2187            }
2188
2189            // compute volume for this track
2190            uint32_t vl, vr, va;
2191            if (track->isMuted() || track->isPausing() ||
2192                mStreamTypes[track->type()].mute) {
2193                vl = vr = va = 0;
2194                if (track->isPausing()) {
2195                    track->setPaused();
2196                }
2197            } else {
2198
2199                // read original volumes with volume control
2200                float typeVolume = mStreamTypes[track->type()].volume;
2201                float v = masterVolume * typeVolume;
2202                uint32_t vlr = cblk->getVolumeLR();
2203                vl = vlr & 0xFFFF;
2204                vr = vlr >> 16;
2205                // track volumes come from shared memory, so can't be trusted and must be clamped
2206                if (vl > MAX_GAIN_INT) {
2207                    ALOGV("Track left volume out of range: %04X", vl);
2208                    vl = MAX_GAIN_INT;
2209                }
2210                if (vr > MAX_GAIN_INT) {
2211                    ALOGV("Track right volume out of range: %04X", vr);
2212                    vr = MAX_GAIN_INT;
2213                }
2214                // now apply the master volume and stream type volume
2215                vl = (uint32_t)(v * vl) << 12;
2216                vr = (uint32_t)(v * vr) << 12;
2217                // assuming master volume and stream type volume each go up to 1.0,
2218                // vl and vr are now in 8.24 format
2219
2220                uint16_t sendLevel = cblk->getSendLevel_U4_12();
2221                // send level comes from shared memory and so may be corrupt
2222                if (sendLevel >= MAX_GAIN_INT) {
2223                    ALOGV("Track send level out of range: %04X", sendLevel);
2224                    sendLevel = MAX_GAIN_INT;
2225                }
2226                va = (uint32_t)(v * sendLevel);
2227            }
2228            // Delegate volume control to effect in track effect chain if needed
2229            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2230                // Do not ramp volume if volume is controlled by effect
2231                param = AudioMixer::VOLUME;
2232                track->mHasVolumeController = true;
2233            } else {
2234                // force no volume ramp when volume controller was just disabled or removed
2235                // from effect chain to avoid volume spike
2236                if (track->mHasVolumeController) {
2237                    param = AudioMixer::VOLUME;
2238                }
2239                track->mHasVolumeController = false;
2240            }
2241
2242            // Convert volumes from 8.24 to 4.12 format
2243            int16_t left, right, aux;
2244            // This additional clamping is needed in case chain->setVolume_l() overshot
2245            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2246            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2247            left = int16_t(v_clamped);
2248            v_clamped = (vr + (1 << 11)) >> 12;
2249            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2250            right = int16_t(v_clamped);
2251
2252            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2253            aux = int16_t(va);
2254
2255            // XXX: these things DON'T need to be done each time
2256            mAudioMixer->setBufferProvider(name, track);
2257            mAudioMixer->enable(name);
2258
2259            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left);
2260            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right);
2261            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux);
2262            mAudioMixer->setParameter(
2263                name,
2264                AudioMixer::TRACK,
2265                AudioMixer::FORMAT, (void *)track->format());
2266            mAudioMixer->setParameter(
2267                name,
2268                AudioMixer::TRACK,
2269                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2270            mAudioMixer->setParameter(
2271                name,
2272                AudioMixer::RESAMPLE,
2273                AudioMixer::SAMPLE_RATE,
2274                (void *)(cblk->sampleRate));
2275            mAudioMixer->setParameter(
2276                name,
2277                AudioMixer::TRACK,
2278                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2279            mAudioMixer->setParameter(
2280                name,
2281                AudioMixer::TRACK,
2282                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2283
2284            // reset retry count
2285            track->mRetryCount = kMaxTrackRetries;
2286            // If one track is ready, set the mixer ready if:
2287            //  - the mixer was not ready during previous round OR
2288            //  - no other track is not ready
2289            if (mPrevMixerStatus != MIXER_TRACKS_READY ||
2290                    mixerStatus != MIXER_TRACKS_ENABLED) {
2291                mixerStatus = MIXER_TRACKS_READY;
2292            }
2293        } else {
2294            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
2295            if (track->isStopped()) {
2296                track->reset();
2297            }
2298            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2299                // We have consumed all the buffers of this track.
2300                // Remove it from the list of active tracks.
2301                tracksToRemove->add(track);
2302            } else {
2303                // No buffers for this track. Give it a few chances to
2304                // fill a buffer, then remove it from active list.
2305                if (--(track->mRetryCount) <= 0) {
2306                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
2307                    tracksToRemove->add(track);
2308                    // indicate to client process that the track was disabled because of underrun
2309                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2310                // If one track is not ready, mark the mixer also not ready if:
2311                //  - the mixer was ready during previous round OR
2312                //  - no other track is ready
2313                } else if (mPrevMixerStatus == MIXER_TRACKS_READY ||
2314                                mixerStatus != MIXER_TRACKS_READY) {
2315                    mixerStatus = MIXER_TRACKS_ENABLED;
2316                }
2317            }
2318            mAudioMixer->disable(name);
2319        }
2320    }
2321
2322    // remove all the tracks that need to be...
2323    count = tracksToRemove->size();
2324    if (CC_UNLIKELY(count)) {
2325        for (size_t i=0 ; i<count ; i++) {
2326            const sp<Track>& track = tracksToRemove->itemAt(i);
2327            mActiveTracks.remove(track);
2328            if (track->mainBuffer() != mMixBuffer) {
2329                chain = getEffectChain_l(track->sessionId());
2330                if (chain != 0) {
2331                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2332                    chain->decActiveTrackCnt();
2333                }
2334            }
2335            if (track->isTerminated()) {
2336                removeTrack_l(track);
2337            }
2338        }
2339    }
2340
2341    // mix buffer must be cleared if all tracks are connected to an
2342    // effect chain as in this case the mixer will not write to
2343    // mix buffer and track effects will accumulate into it
2344    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2345        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2346    }
2347
2348    mPrevMixerStatus = mixerStatus;
2349    return mixerStatus;
2350}
2351
2352void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType)
2353{
2354    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2355            this,  streamType, mTracks.size());
2356    Mutex::Autolock _l(mLock);
2357
2358    size_t size = mTracks.size();
2359    for (size_t i = 0; i < size; i++) {
2360        sp<Track> t = mTracks[i];
2361        if (t->type() == streamType) {
2362            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2363            t->mCblk->cv.signal();
2364        }
2365    }
2366}
2367
2368void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid)
2369{
2370    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2371            this,  streamType, valid);
2372    Mutex::Autolock _l(mLock);
2373
2374    mStreamTypes[streamType].valid = valid;
2375}
2376
2377// getTrackName_l() must be called with ThreadBase::mLock held
2378int AudioFlinger::MixerThread::getTrackName_l()
2379{
2380    return mAudioMixer->getTrackName();
2381}
2382
2383// deleteTrackName_l() must be called with ThreadBase::mLock held
2384void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2385{
2386    ALOGV("remove track (%d) and delete from mixer", name);
2387    mAudioMixer->deleteTrackName(name);
2388}
2389
2390// checkForNewParameters_l() must be called with ThreadBase::mLock held
2391bool AudioFlinger::MixerThread::checkForNewParameters_l()
2392{
2393    bool reconfig = false;
2394
2395    while (!mNewParameters.isEmpty()) {
2396        status_t status = NO_ERROR;
2397        String8 keyValuePair = mNewParameters[0];
2398        AudioParameter param = AudioParameter(keyValuePair);
2399        int value;
2400
2401        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2402            reconfig = true;
2403        }
2404        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2405            if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
2406                status = BAD_VALUE;
2407            } else {
2408                reconfig = true;
2409            }
2410        }
2411        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2412            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2413                status = BAD_VALUE;
2414            } else {
2415                reconfig = true;
2416            }
2417        }
2418        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2419            // do not accept frame count changes if tracks are open as the track buffer
2420            // size depends on frame count and correct behavior would not be guaranteed
2421            // if frame count is changed after track creation
2422            if (!mTracks.isEmpty()) {
2423                status = INVALID_OPERATION;
2424            } else {
2425                reconfig = true;
2426            }
2427        }
2428        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2429            // when changing the audio output device, call addBatteryData to notify
2430            // the change
2431            if ((int)mDevice != value) {
2432                uint32_t params = 0;
2433                // check whether speaker is on
2434                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2435                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2436                }
2437
2438                int deviceWithoutSpeaker
2439                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2440                // check if any other device (except speaker) is on
2441                if (value & deviceWithoutSpeaker ) {
2442                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2443                }
2444
2445                if (params != 0) {
2446                    addBatteryData(params);
2447                }
2448            }
2449
2450            // forward device change to effects that have requested to be
2451            // aware of attached audio device.
2452            mDevice = (uint32_t)value;
2453            for (size_t i = 0; i < mEffectChains.size(); i++) {
2454                mEffectChains[i]->setDevice_l(mDevice);
2455            }
2456        }
2457
2458        if (status == NO_ERROR) {
2459            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2460                                                    keyValuePair.string());
2461            if (!mStandby && status == INVALID_OPERATION) {
2462               mOutput->stream->common.standby(&mOutput->stream->common);
2463               mStandby = true;
2464               mBytesWritten = 0;
2465               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2466                                                       keyValuePair.string());
2467            }
2468            if (status == NO_ERROR && reconfig) {
2469                delete mAudioMixer;
2470                // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
2471                mAudioMixer = NULL;
2472                readOutputParameters();
2473                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2474                for (size_t i = 0; i < mTracks.size() ; i++) {
2475                    int name = getTrackName_l();
2476                    if (name < 0) break;
2477                    mTracks[i]->mName = name;
2478                    // limit track sample rate to 2 x new output sample rate
2479                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2480                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2481                    }
2482                }
2483                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2484            }
2485        }
2486
2487        mNewParameters.removeAt(0);
2488
2489        mParamStatus = status;
2490        mParamCond.signal();
2491        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2492        // already timed out waiting for the status and will never signal the condition.
2493        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2494    }
2495    return reconfig;
2496}
2497
2498status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2499{
2500    const size_t SIZE = 256;
2501    char buffer[SIZE];
2502    String8 result;
2503
2504    PlaybackThread::dumpInternals(fd, args);
2505
2506    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2507    result.append(buffer);
2508    write(fd, result.string(), result.size());
2509    return NO_ERROR;
2510}
2511
2512uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2513{
2514    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2515}
2516
2517uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2518{
2519    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2520}
2521
2522// ----------------------------------------------------------------------------
2523AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2524    :   PlaybackThread(audioFlinger, output, id, device, DIRECT)
2525        // mLeftVolFloat, mRightVolFloat
2526        // mLeftVolShort, mRightVolShort
2527{
2528}
2529
2530AudioFlinger::DirectOutputThread::~DirectOutputThread()
2531{
2532}
2533
2534static inline
2535int32_t mul(int16_t in, int16_t v)
2536{
2537#if defined(__arm__) && !defined(__thumb__)
2538    int32_t out;
2539    asm( "smulbb %[out], %[in], %[v] \n"
2540         : [out]"=r"(out)
2541         : [in]"%r"(in), [v]"r"(v)
2542         : );
2543    return out;
2544#else
2545    return in * int32_t(v);
2546#endif
2547}
2548
2549void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2550{
2551    // Do not apply volume on compressed audio
2552    if (!audio_is_linear_pcm(mFormat)) {
2553        return;
2554    }
2555
2556    // convert to signed 16 bit before volume calculation
2557    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2558        size_t count = mFrameCount * mChannelCount;
2559        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2560        int16_t *dst = mMixBuffer + count-1;
2561        while(count--) {
2562            *dst-- = (int16_t)(*src--^0x80) << 8;
2563        }
2564    }
2565
2566    size_t frameCount = mFrameCount;
2567    int16_t *out = mMixBuffer;
2568    if (ramp) {
2569        if (mChannelCount == 1) {
2570            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2571            int32_t vlInc = d / (int32_t)frameCount;
2572            int32_t vl = ((int32_t)mLeftVolShort << 16);
2573            do {
2574                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2575                out++;
2576                vl += vlInc;
2577            } while (--frameCount);
2578
2579        } else {
2580            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2581            int32_t vlInc = d / (int32_t)frameCount;
2582            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2583            int32_t vrInc = d / (int32_t)frameCount;
2584            int32_t vl = ((int32_t)mLeftVolShort << 16);
2585            int32_t vr = ((int32_t)mRightVolShort << 16);
2586            do {
2587                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2588                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2589                out += 2;
2590                vl += vlInc;
2591                vr += vrInc;
2592            } while (--frameCount);
2593        }
2594    } else {
2595        if (mChannelCount == 1) {
2596            do {
2597                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2598                out++;
2599            } while (--frameCount);
2600        } else {
2601            do {
2602                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2603                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2604                out += 2;
2605            } while (--frameCount);
2606        }
2607    }
2608
2609    // convert back to unsigned 8 bit after volume calculation
2610    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2611        size_t count = mFrameCount * mChannelCount;
2612        int16_t *src = mMixBuffer;
2613        uint8_t *dst = (uint8_t *)mMixBuffer;
2614        while(count--) {
2615            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2616        }
2617    }
2618
2619    mLeftVolShort = leftVol;
2620    mRightVolShort = rightVol;
2621}
2622
2623bool AudioFlinger::DirectOutputThread::threadLoop()
2624{
2625    mixer_state mixerStatus = MIXER_IDLE;
2626    sp<Track> trackToRemove;
2627    sp<Track> activeTrack;
2628    nsecs_t standbyTime = systemTime();
2629    int8_t *curBuf;
2630    size_t mixBufferSize = mFrameCount*mFrameSize;
2631    uint32_t activeSleepTime = activeSleepTimeUs();
2632    uint32_t idleSleepTime = idleSleepTimeUs();
2633    uint32_t sleepTime = idleSleepTime;
2634    // use shorter standby delay as on normal output to release
2635    // hardware resources as soon as possible
2636    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2637
2638    acquireWakeLock();
2639
2640    while (!exitPending())
2641    {
2642        bool rampVolume;
2643        uint16_t leftVol;
2644        uint16_t rightVol;
2645        Vector< sp<EffectChain> > effectChains;
2646
2647        processConfigEvents();
2648
2649        mixerStatus = MIXER_IDLE;
2650
2651        { // scope for the mLock
2652
2653            Mutex::Autolock _l(mLock);
2654
2655            if (checkForNewParameters_l()) {
2656                mixBufferSize = mFrameCount*mFrameSize;
2657                activeSleepTime = activeSleepTimeUs();
2658                idleSleepTime = idleSleepTimeUs();
2659                standbyDelay = microseconds(activeSleepTime*2);
2660            }
2661
2662            // put audio hardware into standby after short delay
2663            if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2664                        mSuspended)) {
2665                // wait until we have something to do...
2666                if (!mStandby) {
2667                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2668                    mOutput->stream->common.standby(&mOutput->stream->common);
2669                    mStandby = true;
2670                    mBytesWritten = 0;
2671                }
2672
2673                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2674                    // we're about to wait, flush the binder command buffer
2675                    IPCThreadState::self()->flushCommands();
2676
2677                    if (exitPending()) break;
2678
2679                    releaseWakeLock_l();
2680                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2681                    mWaitWorkCV.wait(mLock);
2682                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2683                    acquireWakeLock_l();
2684
2685                    if (!mMasterMute) {
2686                        char value[PROPERTY_VALUE_MAX];
2687                        property_get("ro.audio.silent", value, "0");
2688                        if (atoi(value)) {
2689                            ALOGD("Silence is golden");
2690                            setMasterMute(true);
2691                        }
2692                    }
2693
2694                    standbyTime = systemTime() + standbyDelay;
2695                    sleepTime = idleSleepTime;
2696                    continue;
2697                }
2698            }
2699
2700            effectChains = mEffectChains;
2701
2702            // find out which tracks need to be processed
2703            if (mActiveTracks.size() != 0) {
2704                sp<Track> t = mActiveTracks[0].promote();
2705                if (t == 0) continue;
2706
2707                Track* const track = t.get();
2708                audio_track_cblk_t* cblk = track->cblk();
2709
2710                // The first time a track is added we wait
2711                // for all its buffers to be filled before processing it
2712                if (cblk->framesReady() && track->isReady() &&
2713                        !track->isPaused() && !track->isTerminated())
2714                {
2715                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2716
2717                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2718                        track->mFillingUpStatus = Track::FS_ACTIVE;
2719                        mLeftVolFloat = mRightVolFloat = 0;
2720                        mLeftVolShort = mRightVolShort = 0;
2721                        if (track->mState == TrackBase::RESUMING) {
2722                            track->mState = TrackBase::ACTIVE;
2723                            rampVolume = true;
2724                        }
2725                    } else if (cblk->server != 0) {
2726                        // If the track is stopped before the first frame was mixed,
2727                        // do not apply ramp
2728                        rampVolume = true;
2729                    }
2730                    // compute volume for this track
2731                    float left, right;
2732                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2733                        mStreamTypes[track->type()].mute) {
2734                        left = right = 0;
2735                        if (track->isPausing()) {
2736                            track->setPaused();
2737                        }
2738                    } else {
2739                        float typeVolume = mStreamTypes[track->type()].volume;
2740                        float v = mMasterVolume * typeVolume;
2741                        uint32_t vlr = cblk->getVolumeLR();
2742                        float v_clamped = v * (vlr & 0xFFFF);
2743                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2744                        left = v_clamped/MAX_GAIN;
2745                        v_clamped = v * (vlr >> 16);
2746                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2747                        right = v_clamped/MAX_GAIN;
2748                    }
2749
2750                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2751                        mLeftVolFloat = left;
2752                        mRightVolFloat = right;
2753
2754                        // If audio HAL implements volume control,
2755                        // force software volume to nominal value
2756                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2757                            left = 1.0f;
2758                            right = 1.0f;
2759                        }
2760
2761                        // Convert volumes from float to 8.24
2762                        uint32_t vl = (uint32_t)(left * (1 << 24));
2763                        uint32_t vr = (uint32_t)(right * (1 << 24));
2764
2765                        // Delegate volume control to effect in track effect chain if needed
2766                        // only one effect chain can be present on DirectOutputThread, so if
2767                        // there is one, the track is connected to it
2768                        if (!effectChains.isEmpty()) {
2769                            // Do not ramp volume if volume is controlled by effect
2770                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2771                                rampVolume = false;
2772                            }
2773                        }
2774
2775                        // Convert volumes from 8.24 to 4.12 format
2776                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2777                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2778                        leftVol = (uint16_t)v_clamped;
2779                        v_clamped = (vr + (1 << 11)) >> 12;
2780                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2781                        rightVol = (uint16_t)v_clamped;
2782                    } else {
2783                        leftVol = mLeftVolShort;
2784                        rightVol = mRightVolShort;
2785                        rampVolume = false;
2786                    }
2787
2788                    // reset retry count
2789                    track->mRetryCount = kMaxTrackRetriesDirect;
2790                    activeTrack = t;
2791                    mixerStatus = MIXER_TRACKS_READY;
2792                } else {
2793                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2794                    if (track->isStopped()) {
2795                        track->reset();
2796                    }
2797                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2798                        // We have consumed all the buffers of this track.
2799                        // Remove it from the list of active tracks.
2800                        trackToRemove = track;
2801                    } else {
2802                        // No buffers for this track. Give it a few chances to
2803                        // fill a buffer, then remove it from active list.
2804                        if (--(track->mRetryCount) <= 0) {
2805                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2806                            trackToRemove = track;
2807                        } else {
2808                            mixerStatus = MIXER_TRACKS_ENABLED;
2809                        }
2810                    }
2811                }
2812            }
2813
2814            // remove all the tracks that need to be...
2815            if (CC_UNLIKELY(trackToRemove != 0)) {
2816                mActiveTracks.remove(trackToRemove);
2817                if (!effectChains.isEmpty()) {
2818                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2819                            trackToRemove->sessionId());
2820                    effectChains[0]->decActiveTrackCnt();
2821                }
2822                if (trackToRemove->isTerminated()) {
2823                    removeTrack_l(trackToRemove);
2824                }
2825            }
2826
2827            lockEffectChains_l(effectChains);
2828       }
2829
2830        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2831            AudioBufferProvider::Buffer buffer;
2832            size_t frameCount = mFrameCount;
2833            curBuf = (int8_t *)mMixBuffer;
2834            // output audio to hardware
2835            while (frameCount) {
2836                buffer.frameCount = frameCount;
2837                activeTrack->getNextBuffer(&buffer);
2838                if (CC_UNLIKELY(buffer.raw == NULL)) {
2839                    memset(curBuf, 0, frameCount * mFrameSize);
2840                    break;
2841                }
2842                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2843                frameCount -= buffer.frameCount;
2844                curBuf += buffer.frameCount * mFrameSize;
2845                activeTrack->releaseBuffer(&buffer);
2846            }
2847            sleepTime = 0;
2848            standbyTime = systemTime() + standbyDelay;
2849        } else {
2850            if (sleepTime == 0) {
2851                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2852                    sleepTime = activeSleepTime;
2853                } else {
2854                    sleepTime = idleSleepTime;
2855                }
2856            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2857                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2858                sleepTime = 0;
2859            }
2860        }
2861
2862        if (mSuspended) {
2863            sleepTime = suspendSleepTimeUs();
2864        }
2865        // sleepTime == 0 means we must write to audio hardware
2866        if (sleepTime == 0) {
2867            if (mixerStatus == MIXER_TRACKS_READY) {
2868                applyVolume(leftVol, rightVol, rampVolume);
2869            }
2870            for (size_t i = 0; i < effectChains.size(); i ++) {
2871                effectChains[i]->process_l();
2872            }
2873            unlockEffectChains(effectChains);
2874
2875            mLastWriteTime = systemTime();
2876            mInWrite = true;
2877            mBytesWritten += mixBufferSize;
2878            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2879            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2880            mNumWrites++;
2881            mInWrite = false;
2882            mStandby = false;
2883        } else {
2884            unlockEffectChains(effectChains);
2885            usleep(sleepTime);
2886        }
2887
2888        // finally let go of removed track, without the lock held
2889        // since we can't guarantee the destructors won't acquire that
2890        // same lock.
2891        trackToRemove.clear();
2892        activeTrack.clear();
2893
2894        // Effect chains will be actually deleted here if they were removed from
2895        // mEffectChains list during mixing or effects processing
2896        effectChains.clear();
2897    }
2898
2899    if (!mStandby) {
2900        mOutput->stream->common.standby(&mOutput->stream->common);
2901    }
2902
2903    releaseWakeLock();
2904
2905    ALOGV("DirectOutputThread %p exiting", this);
2906    return false;
2907}
2908
2909// getTrackName_l() must be called with ThreadBase::mLock held
2910int AudioFlinger::DirectOutputThread::getTrackName_l()
2911{
2912    return 0;
2913}
2914
2915// deleteTrackName_l() must be called with ThreadBase::mLock held
2916void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2917{
2918}
2919
2920// checkForNewParameters_l() must be called with ThreadBase::mLock held
2921bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2922{
2923    bool reconfig = false;
2924
2925    while (!mNewParameters.isEmpty()) {
2926        status_t status = NO_ERROR;
2927        String8 keyValuePair = mNewParameters[0];
2928        AudioParameter param = AudioParameter(keyValuePair);
2929        int value;
2930
2931        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2932            // do not accept frame count changes if tracks are open as the track buffer
2933            // size depends on frame count and correct behavior would not be garantied
2934            // if frame count is changed after track creation
2935            if (!mTracks.isEmpty()) {
2936                status = INVALID_OPERATION;
2937            } else {
2938                reconfig = true;
2939            }
2940        }
2941        if (status == NO_ERROR) {
2942            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2943                                                    keyValuePair.string());
2944            if (!mStandby && status == INVALID_OPERATION) {
2945               mOutput->stream->common.standby(&mOutput->stream->common);
2946               mStandby = true;
2947               mBytesWritten = 0;
2948               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2949                                                       keyValuePair.string());
2950            }
2951            if (status == NO_ERROR && reconfig) {
2952                readOutputParameters();
2953                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2954            }
2955        }
2956
2957        mNewParameters.removeAt(0);
2958
2959        mParamStatus = status;
2960        mParamCond.signal();
2961        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2962        // already timed out waiting for the status and will never signal the condition.
2963        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
2964    }
2965    return reconfig;
2966}
2967
2968uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2969{
2970    uint32_t time;
2971    if (audio_is_linear_pcm(mFormat)) {
2972        time = PlaybackThread::activeSleepTimeUs();
2973    } else {
2974        time = 10000;
2975    }
2976    return time;
2977}
2978
2979uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2980{
2981    uint32_t time;
2982    if (audio_is_linear_pcm(mFormat)) {
2983        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2984    } else {
2985        time = 10000;
2986    }
2987    return time;
2988}
2989
2990uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2991{
2992    uint32_t time;
2993    if (audio_is_linear_pcm(mFormat)) {
2994        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2995    } else {
2996        time = 10000;
2997    }
2998    return time;
2999}
3000
3001
3002// ----------------------------------------------------------------------------
3003
3004AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
3005        AudioFlinger::MixerThread* mainThread, int id)
3006    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING),
3007        mWaitTimeMs(UINT_MAX)
3008{
3009    addOutputTrack(mainThread);
3010}
3011
3012AudioFlinger::DuplicatingThread::~DuplicatingThread()
3013{
3014    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3015        mOutputTracks[i]->destroy();
3016    }
3017    mOutputTracks.clear();
3018}
3019
3020bool AudioFlinger::DuplicatingThread::threadLoop()
3021{
3022    Vector< sp<Track> > tracksToRemove;
3023    mixer_state mixerStatus = MIXER_IDLE;
3024    nsecs_t standbyTime = systemTime();
3025    size_t mixBufferSize = mFrameCount*mFrameSize;
3026    SortedVector< sp<OutputTrack> > outputTracks;
3027    uint32_t writeFrames = 0;
3028    uint32_t activeSleepTime = activeSleepTimeUs();
3029    uint32_t idleSleepTime = idleSleepTimeUs();
3030    uint32_t sleepTime = idleSleepTime;
3031    Vector< sp<EffectChain> > effectChains;
3032
3033    acquireWakeLock();
3034
3035    while (!exitPending())
3036    {
3037        processConfigEvents();
3038
3039        mixerStatus = MIXER_IDLE;
3040        { // scope for the mLock
3041
3042            Mutex::Autolock _l(mLock);
3043
3044            if (checkForNewParameters_l()) {
3045                mixBufferSize = mFrameCount*mFrameSize;
3046                updateWaitTime();
3047                activeSleepTime = activeSleepTimeUs();
3048                idleSleepTime = idleSleepTimeUs();
3049            }
3050
3051            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
3052
3053            for (size_t i = 0; i < mOutputTracks.size(); i++) {
3054                outputTracks.add(mOutputTracks[i]);
3055            }
3056
3057            // put audio hardware into standby after short delay
3058            if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
3059                         mSuspended)) {
3060                if (!mStandby) {
3061                    for (size_t i = 0; i < outputTracks.size(); i++) {
3062                        outputTracks[i]->stop();
3063                    }
3064                    mStandby = true;
3065                    mBytesWritten = 0;
3066                }
3067
3068                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
3069                    // we're about to wait, flush the binder command buffer
3070                    IPCThreadState::self()->flushCommands();
3071                    outputTracks.clear();
3072
3073                    if (exitPending()) break;
3074
3075                    releaseWakeLock_l();
3076                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
3077                    mWaitWorkCV.wait(mLock);
3078                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
3079                    acquireWakeLock_l();
3080
3081                    mPrevMixerStatus = MIXER_IDLE;
3082                    if (!mMasterMute) {
3083                        char value[PROPERTY_VALUE_MAX];
3084                        property_get("ro.audio.silent", value, "0");
3085                        if (atoi(value)) {
3086                            ALOGD("Silence is golden");
3087                            setMasterMute(true);
3088                        }
3089                    }
3090
3091                    standbyTime = systemTime() + kStandbyTimeInNsecs;
3092                    sleepTime = idleSleepTime;
3093                    continue;
3094                }
3095            }
3096
3097            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
3098
3099            // prevent any changes in effect chain list and in each effect chain
3100            // during mixing and effect process as the audio buffers could be deleted
3101            // or modified if an effect is created or deleted
3102            lockEffectChains_l(effectChains);
3103        }
3104
3105        if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
3106            // mix buffers...
3107            if (outputsReady(outputTracks)) {
3108                mAudioMixer->process();
3109            } else {
3110                memset(mMixBuffer, 0, mixBufferSize);
3111            }
3112            sleepTime = 0;
3113            writeFrames = mFrameCount;
3114        } else {
3115            if (sleepTime == 0) {
3116                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3117                    sleepTime = activeSleepTime;
3118                } else {
3119                    sleepTime = idleSleepTime;
3120                }
3121            } else if (mBytesWritten != 0) {
3122                // flush remaining overflow buffers in output tracks
3123                for (size_t i = 0; i < outputTracks.size(); i++) {
3124                    if (outputTracks[i]->isActive()) {
3125                        sleepTime = 0;
3126                        writeFrames = 0;
3127                        memset(mMixBuffer, 0, mixBufferSize);
3128                        break;
3129                    }
3130                }
3131            }
3132        }
3133
3134        if (mSuspended) {
3135            sleepTime = suspendSleepTimeUs();
3136        }
3137        // sleepTime == 0 means we must write to audio hardware
3138        if (sleepTime == 0) {
3139            for (size_t i = 0; i < effectChains.size(); i ++) {
3140                effectChains[i]->process_l();
3141            }
3142            // enable changes in effect chain
3143            unlockEffectChains(effectChains);
3144
3145            standbyTime = systemTime() + kStandbyTimeInNsecs;
3146            for (size_t i = 0; i < outputTracks.size(); i++) {
3147                outputTracks[i]->write(mMixBuffer, writeFrames);
3148            }
3149            mStandby = false;
3150            mBytesWritten += mixBufferSize;
3151        } else {
3152            // enable changes in effect chain
3153            unlockEffectChains(effectChains);
3154            usleep(sleepTime);
3155        }
3156
3157        // finally let go of all our tracks, without the lock held
3158        // since we can't guarantee the destructors won't acquire that
3159        // same lock.
3160        tracksToRemove.clear();
3161        outputTracks.clear();
3162
3163        // Effect chains will be actually deleted here if they were removed from
3164        // mEffectChains list during mixing or effects processing
3165        effectChains.clear();
3166    }
3167
3168    releaseWakeLock();
3169
3170    return false;
3171}
3172
3173void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3174{
3175    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3176    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3177                                            this,
3178                                            mSampleRate,
3179                                            mFormat,
3180                                            mChannelMask,
3181                                            frameCount);
3182    if (outputTrack->cblk() != NULL) {
3183        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3184        mOutputTracks.add(outputTrack);
3185        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3186        updateWaitTime();
3187    }
3188}
3189
3190void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3191{
3192    Mutex::Autolock _l(mLock);
3193    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3194        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3195            mOutputTracks[i]->destroy();
3196            mOutputTracks.removeAt(i);
3197            updateWaitTime();
3198            return;
3199        }
3200    }
3201    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3202}
3203
3204void AudioFlinger::DuplicatingThread::updateWaitTime()
3205{
3206    mWaitTimeMs = UINT_MAX;
3207    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3208        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3209        if (strong != NULL) {
3210            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3211            if (waitTimeMs < mWaitTimeMs) {
3212                mWaitTimeMs = waitTimeMs;
3213            }
3214        }
3215    }
3216}
3217
3218
3219bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3220{
3221    for (size_t i = 0; i < outputTracks.size(); i++) {
3222        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3223        if (thread == 0) {
3224            ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3225            return false;
3226        }
3227        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3228        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3229            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3230            return false;
3231        }
3232    }
3233    return true;
3234}
3235
3236uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3237{
3238    return (mWaitTimeMs * 1000) / 2;
3239}
3240
3241// ----------------------------------------------------------------------------
3242
3243// TrackBase constructor must be called with AudioFlinger::mLock held
3244AudioFlinger::ThreadBase::TrackBase::TrackBase(
3245            const wp<ThreadBase>& thread,
3246            const sp<Client>& client,
3247            uint32_t sampleRate,
3248            audio_format_t format,
3249            uint32_t channelMask,
3250            int frameCount,
3251            uint32_t flags,
3252            const sp<IMemory>& sharedBuffer,
3253            int sessionId)
3254    :   RefBase(),
3255        mThread(thread),
3256        mClient(client),
3257        mCblk(NULL),
3258        // mBuffer
3259        // mBufferEnd
3260        mFrameCount(0),
3261        mState(IDLE),
3262        mClientTid(-1),
3263        mFormat(format),
3264        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3265        mSessionId(sessionId)
3266        // mChannelCount
3267        // mChannelMask
3268{
3269    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3270
3271    // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3272   size_t size = sizeof(audio_track_cblk_t);
3273   uint8_t channelCount = popcount(channelMask);
3274   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3275   if (sharedBuffer == 0) {
3276       size += bufferSize;
3277   }
3278
3279   if (client != NULL) {
3280        mCblkMemory = client->heap()->allocate(size);
3281        if (mCblkMemory != 0) {
3282            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3283            if (mCblk != NULL) { // construct the shared structure in-place.
3284                new(mCblk) audio_track_cblk_t();
3285                // clear all buffers
3286                mCblk->frameCount = frameCount;
3287                mCblk->sampleRate = sampleRate;
3288                mChannelCount = channelCount;
3289                mChannelMask = channelMask;
3290                if (sharedBuffer == 0) {
3291                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3292                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3293                    // Force underrun condition to avoid false underrun callback until first data is
3294                    // written to buffer (other flags are cleared)
3295                    mCblk->flags = CBLK_UNDERRUN_ON;
3296                } else {
3297                    mBuffer = sharedBuffer->pointer();
3298                }
3299                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3300            }
3301        } else {
3302            ALOGE("not enough memory for AudioTrack size=%u", size);
3303            client->heap()->dump("AudioTrack");
3304            return;
3305        }
3306   } else {
3307       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3308           // construct the shared structure in-place.
3309           new(mCblk) audio_track_cblk_t();
3310           // clear all buffers
3311           mCblk->frameCount = frameCount;
3312           mCblk->sampleRate = sampleRate;
3313           mChannelCount = channelCount;
3314           mChannelMask = channelMask;
3315           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3316           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3317           // Force underrun condition to avoid false underrun callback until first data is
3318           // written to buffer (other flags are cleared)
3319           mCblk->flags = CBLK_UNDERRUN_ON;
3320           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3321   }
3322}
3323
3324AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3325{
3326    if (mCblk != NULL) {
3327        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3328        if (mClient == NULL) {
3329            delete mCblk;
3330        }
3331    }
3332    mCblkMemory.clear();            // and free the shared memory
3333    if (mClient != NULL) {
3334        // Client destructor must run with AudioFlinger mutex locked
3335        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
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 == NULL) ? 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 == NULL) ? 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() == NULL) {
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    pid_t pid = 0;
4635
4636    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4637    result.append(buffer);
4638
4639    if (mActiveTrack != 0) {
4640        result.append("Active Track:\n");
4641        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4642        mActiveTrack->dump(buffer, SIZE);
4643        result.append(buffer);
4644
4645        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4646        result.append(buffer);
4647        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4648        result.append(buffer);
4649        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4650        result.append(buffer);
4651        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4652        result.append(buffer);
4653        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4654        result.append(buffer);
4655
4656
4657    } else {
4658        result.append("No record client\n");
4659    }
4660    write(fd, result.string(), result.size());
4661
4662    dumpBase(fd, args);
4663    dumpEffectChains(fd, args);
4664
4665    return NO_ERROR;
4666}
4667
4668status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4669{
4670    size_t framesReq = buffer->frameCount;
4671    size_t framesReady = mFrameCount - mRsmpInIndex;
4672    int channelCount;
4673
4674    if (framesReady == 0) {
4675        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4676        if (mBytesRead < 0) {
4677            ALOGE("RecordThread::getNextBuffer() Error reading audio input");
4678            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4679                // Force input into standby so that it tries to
4680                // recover at next read attempt
4681                mInput->stream->common.standby(&mInput->stream->common);
4682                usleep(kRecordThreadSleepUs);
4683            }
4684            buffer->raw = NULL;
4685            buffer->frameCount = 0;
4686            return NOT_ENOUGH_DATA;
4687        }
4688        mRsmpInIndex = 0;
4689        framesReady = mFrameCount;
4690    }
4691
4692    if (framesReq > framesReady) {
4693        framesReq = framesReady;
4694    }
4695
4696    if (mChannelCount == 1 && mReqChannelCount == 2) {
4697        channelCount = 1;
4698    } else {
4699        channelCount = 2;
4700    }
4701    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4702    buffer->frameCount = framesReq;
4703    return NO_ERROR;
4704}
4705
4706void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4707{
4708    mRsmpInIndex += buffer->frameCount;
4709    buffer->frameCount = 0;
4710}
4711
4712bool AudioFlinger::RecordThread::checkForNewParameters_l()
4713{
4714    bool reconfig = false;
4715
4716    while (!mNewParameters.isEmpty()) {
4717        status_t status = NO_ERROR;
4718        String8 keyValuePair = mNewParameters[0];
4719        AudioParameter param = AudioParameter(keyValuePair);
4720        int value;
4721        audio_format_t reqFormat = mFormat;
4722        int reqSamplingRate = mReqSampleRate;
4723        int reqChannelCount = mReqChannelCount;
4724
4725        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4726            reqSamplingRate = value;
4727            reconfig = true;
4728        }
4729        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4730            reqFormat = (audio_format_t) value;
4731            reconfig = true;
4732        }
4733        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4734            reqChannelCount = popcount(value);
4735            reconfig = true;
4736        }
4737        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4738            // do not accept frame count changes if tracks are open as the track buffer
4739            // size depends on frame count and correct behavior would not be garantied
4740            // if frame count is changed after track creation
4741            if (mActiveTrack != 0) {
4742                status = INVALID_OPERATION;
4743            } else {
4744                reconfig = true;
4745            }
4746        }
4747        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4748            // forward device change to effects that have requested to be
4749            // aware of attached audio device.
4750            for (size_t i = 0; i < mEffectChains.size(); i++) {
4751                mEffectChains[i]->setDevice_l(value);
4752            }
4753            // store input device and output device but do not forward output device to audio HAL.
4754            // Note that status is ignored by the caller for output device
4755            // (see AudioFlinger::setParameters()
4756            if (value & AUDIO_DEVICE_OUT_ALL) {
4757                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4758                status = BAD_VALUE;
4759            } else {
4760                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4761                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4762                if (mTrack != NULL) {
4763                    bool suspend = audio_is_bluetooth_sco_device(
4764                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4765                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4766                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4767                }
4768            }
4769            mDevice |= (uint32_t)value;
4770        }
4771        if (status == NO_ERROR) {
4772            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4773            if (status == INVALID_OPERATION) {
4774               mInput->stream->common.standby(&mInput->stream->common);
4775               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4776            }
4777            if (reconfig) {
4778                if (status == BAD_VALUE &&
4779                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4780                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4781                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4782                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4783                    (reqChannelCount < 3)) {
4784                    status = NO_ERROR;
4785                }
4786                if (status == NO_ERROR) {
4787                    readInputParameters();
4788                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4789                }
4790            }
4791        }
4792
4793        mNewParameters.removeAt(0);
4794
4795        mParamStatus = status;
4796        mParamCond.signal();
4797        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4798        // already timed out waiting for the status and will never signal the condition.
4799        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4800    }
4801    return reconfig;
4802}
4803
4804String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4805{
4806    char *s;
4807    String8 out_s8 = String8();
4808
4809    Mutex::Autolock _l(mLock);
4810    if (initCheck() != NO_ERROR) {
4811        return out_s8;
4812    }
4813
4814    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4815    out_s8 = String8(s);
4816    free(s);
4817    return out_s8;
4818}
4819
4820void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4821    AudioSystem::OutputDescriptor desc;
4822    void *param2 = NULL;
4823
4824    switch (event) {
4825    case AudioSystem::INPUT_OPENED:
4826    case AudioSystem::INPUT_CONFIG_CHANGED:
4827        desc.channels = mChannelMask;
4828        desc.samplingRate = mSampleRate;
4829        desc.format = mFormat;
4830        desc.frameCount = mFrameCount;
4831        desc.latency = 0;
4832        param2 = &desc;
4833        break;
4834
4835    case AudioSystem::INPUT_CLOSED:
4836    default:
4837        break;
4838    }
4839    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4840}
4841
4842void AudioFlinger::RecordThread::readInputParameters()
4843{
4844    delete mRsmpInBuffer;
4845    // mRsmpInBuffer is always assigned a new[] below
4846    delete mRsmpOutBuffer;
4847    mRsmpOutBuffer = NULL;
4848    delete mResampler;
4849    mResampler = NULL;
4850
4851    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4852    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4853    mChannelCount = (uint16_t)popcount(mChannelMask);
4854    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4855    mFrameSize = audio_stream_frame_size(&mInput->stream->common);
4856    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4857    mFrameCount = mInputBytes / mFrameSize;
4858    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4859
4860    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4861    {
4862        int channelCount;
4863         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4864         // stereo to mono post process as the resampler always outputs stereo.
4865        if (mChannelCount == 1 && mReqChannelCount == 2) {
4866            channelCount = 1;
4867        } else {
4868            channelCount = 2;
4869        }
4870        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4871        mResampler->setSampleRate(mSampleRate);
4872        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4873        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4874
4875        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4876        if (mChannelCount == 1 && mReqChannelCount == 1) {
4877            mFrameCount >>= 1;
4878        }
4879
4880    }
4881    mRsmpInIndex = mFrameCount;
4882}
4883
4884unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4885{
4886    Mutex::Autolock _l(mLock);
4887    if (initCheck() != NO_ERROR) {
4888        return 0;
4889    }
4890
4891    return mInput->stream->get_input_frames_lost(mInput->stream);
4892}
4893
4894uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4895{
4896    Mutex::Autolock _l(mLock);
4897    uint32_t result = 0;
4898    if (getEffectChain_l(sessionId) != 0) {
4899        result = EFFECT_SESSION;
4900    }
4901
4902    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4903        result |= TRACK_SESSION;
4904    }
4905
4906    return result;
4907}
4908
4909AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4910{
4911    Mutex::Autolock _l(mLock);
4912    return mTrack;
4913}
4914
4915AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const
4916{
4917    Mutex::Autolock _l(mLock);
4918    return mInput;
4919}
4920
4921AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4922{
4923    Mutex::Autolock _l(mLock);
4924    AudioStreamIn *input = mInput;
4925    mInput = NULL;
4926    return input;
4927}
4928
4929// this method must always be called either with ThreadBase mLock held or inside the thread loop
4930audio_stream_t* AudioFlinger::RecordThread::stream()
4931{
4932    if (mInput == NULL) {
4933        return NULL;
4934    }
4935    return &mInput->stream->common;
4936}
4937
4938
4939// ----------------------------------------------------------------------------
4940
4941int AudioFlinger::openOutput(uint32_t *pDevices,
4942                                uint32_t *pSamplingRate,
4943                                audio_format_t *pFormat,
4944                                uint32_t *pChannels,
4945                                uint32_t *pLatencyMs,
4946                                uint32_t flags)
4947{
4948    status_t status;
4949    PlaybackThread *thread = NULL;
4950    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4951    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4952    audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
4953    uint32_t channels = pChannels ? *pChannels : 0;
4954    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4955    audio_stream_out_t *outStream;
4956    audio_hw_device_t *outHwDev;
4957
4958    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4959            pDevices ? *pDevices : 0,
4960            samplingRate,
4961            format,
4962            channels,
4963            flags);
4964
4965    if (pDevices == NULL || *pDevices == 0) {
4966        return 0;
4967    }
4968
4969    Mutex::Autolock _l(mLock);
4970
4971    outHwDev = findSuitableHwDev_l(*pDevices);
4972    if (outHwDev == NULL)
4973        return 0;
4974
4975    status = outHwDev->open_output_stream(outHwDev, *pDevices, &format,
4976                                          &channels, &samplingRate, &outStream);
4977    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4978            outStream,
4979            samplingRate,
4980            format,
4981            channels,
4982            status);
4983
4984    mHardwareStatus = AUDIO_HW_IDLE;
4985    if (outStream != NULL) {
4986        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4987        int id = nextUniqueId();
4988
4989        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4990            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4991            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4992            thread = new DirectOutputThread(this, output, id, *pDevices);
4993            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4994        } else {
4995            thread = new MixerThread(this, output, id, *pDevices);
4996            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4997        }
4998        mPlaybackThreads.add(id, thread);
4999
5000        if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
5001        if (pFormat != NULL) *pFormat = format;
5002        if (pChannels != NULL) *pChannels = channels;
5003        if (pLatencyMs != NULL) *pLatencyMs = thread->latency();
5004
5005        // notify client processes of the new output creation
5006        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
5007        return id;
5008    }
5009
5010    return 0;
5011}
5012
5013int AudioFlinger::openDuplicateOutput(int output1, int output2)
5014{
5015    Mutex::Autolock _l(mLock);
5016    MixerThread *thread1 = checkMixerThread_l(output1);
5017    MixerThread *thread2 = checkMixerThread_l(output2);
5018
5019    if (thread1 == NULL || thread2 == NULL) {
5020        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
5021        return 0;
5022    }
5023
5024    int id = nextUniqueId();
5025    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
5026    thread->addOutputTrack(thread2);
5027    mPlaybackThreads.add(id, thread);
5028    // notify client processes of the new output creation
5029    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
5030    return id;
5031}
5032
5033status_t AudioFlinger::closeOutput(int output)
5034{
5035    // keep strong reference on the playback thread so that
5036    // it is not destroyed while exit() is executed
5037    sp <PlaybackThread> thread;
5038    {
5039        Mutex::Autolock _l(mLock);
5040        thread = checkPlaybackThread_l(output);
5041        if (thread == NULL) {
5042            return BAD_VALUE;
5043        }
5044
5045        ALOGV("closeOutput() %d", output);
5046
5047        if (thread->type() == ThreadBase::MIXER) {
5048            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5049                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
5050                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
5051                    dupThread->removeOutputTrack((MixerThread *)thread.get());
5052                }
5053            }
5054        }
5055        void *param2 = NULL;
5056        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
5057        mPlaybackThreads.removeItem(output);
5058    }
5059    thread->exit();
5060
5061    if (thread->type() != ThreadBase::DUPLICATING) {
5062        AudioStreamOut *out = thread->clearOutput();
5063        assert(out != NULL);
5064        // from now on thread->mOutput is NULL
5065        out->hwDev->close_output_stream(out->hwDev, out->stream);
5066        delete out;
5067    }
5068    return NO_ERROR;
5069}
5070
5071status_t AudioFlinger::suspendOutput(int output)
5072{
5073    Mutex::Autolock _l(mLock);
5074    PlaybackThread *thread = checkPlaybackThread_l(output);
5075
5076    if (thread == NULL) {
5077        return BAD_VALUE;
5078    }
5079
5080    ALOGV("suspendOutput() %d", output);
5081    thread->suspend();
5082
5083    return NO_ERROR;
5084}
5085
5086status_t AudioFlinger::restoreOutput(int output)
5087{
5088    Mutex::Autolock _l(mLock);
5089    PlaybackThread *thread = checkPlaybackThread_l(output);
5090
5091    if (thread == NULL) {
5092        return BAD_VALUE;
5093    }
5094
5095    ALOGV("restoreOutput() %d", output);
5096
5097    thread->restore();
5098
5099    return NO_ERROR;
5100}
5101
5102int AudioFlinger::openInput(uint32_t *pDevices,
5103                                uint32_t *pSamplingRate,
5104                                audio_format_t *pFormat,
5105                                uint32_t *pChannels,
5106                                audio_in_acoustics_t acoustics)
5107{
5108    status_t status;
5109    RecordThread *thread = NULL;
5110    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5111    audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
5112    uint32_t channels = pChannels ? *pChannels : 0;
5113    uint32_t reqSamplingRate = samplingRate;
5114    audio_format_t reqFormat = format;
5115    uint32_t reqChannels = channels;
5116    audio_stream_in_t *inStream;
5117    audio_hw_device_t *inHwDev;
5118
5119    if (pDevices == NULL || *pDevices == 0) {
5120        return 0;
5121    }
5122
5123    Mutex::Autolock _l(mLock);
5124
5125    inHwDev = findSuitableHwDev_l(*pDevices);
5126    if (inHwDev == NULL)
5127        return 0;
5128
5129    status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
5130                                        &channels, &samplingRate,
5131                                        acoustics,
5132                                        &inStream);
5133    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5134            inStream,
5135            samplingRate,
5136            format,
5137            channels,
5138            acoustics,
5139            status);
5140
5141    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5142    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5143    // or stereo to mono conversions on 16 bit PCM inputs.
5144    if (inStream == NULL && status == BAD_VALUE &&
5145        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5146        (samplingRate <= 2 * reqSamplingRate) &&
5147        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5148        ALOGV("openInput() reopening with proposed sampling rate and channels");
5149        status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
5150                                            &channels, &samplingRate,
5151                                            acoustics,
5152                                            &inStream);
5153    }
5154
5155    if (inStream != NULL) {
5156        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5157
5158        int id = nextUniqueId();
5159        // Start record thread
5160        // RecorThread require both input and output device indication to forward to audio
5161        // pre processing modules
5162        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5163        thread = new RecordThread(this,
5164                                  input,
5165                                  reqSamplingRate,
5166                                  reqChannels,
5167                                  id,
5168                                  device);
5169        mRecordThreads.add(id, thread);
5170        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5171        if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
5172        if (pFormat != NULL) *pFormat = format;
5173        if (pChannels != NULL) *pChannels = reqChannels;
5174
5175        input->stream->common.standby(&input->stream->common);
5176
5177        // notify client processes of the new input creation
5178        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5179        return id;
5180    }
5181
5182    return 0;
5183}
5184
5185status_t AudioFlinger::closeInput(int input)
5186{
5187    // keep strong reference on the record thread so that
5188    // it is not destroyed while exit() is executed
5189    sp <RecordThread> thread;
5190    {
5191        Mutex::Autolock _l(mLock);
5192        thread = checkRecordThread_l(input);
5193        if (thread == NULL) {
5194            return BAD_VALUE;
5195        }
5196
5197        ALOGV("closeInput() %d", input);
5198        void *param2 = NULL;
5199        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5200        mRecordThreads.removeItem(input);
5201    }
5202    thread->exit();
5203
5204    AudioStreamIn *in = thread->clearInput();
5205    assert(in != NULL);
5206    // from now on thread->mInput is NULL
5207    in->hwDev->close_input_stream(in->hwDev, in->stream);
5208    delete in;
5209
5210    return NO_ERROR;
5211}
5212
5213status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output)
5214{
5215    Mutex::Autolock _l(mLock);
5216    MixerThread *dstThread = checkMixerThread_l(output);
5217    if (dstThread == NULL) {
5218        ALOGW("setStreamOutput() bad output id %d", output);
5219        return BAD_VALUE;
5220    }
5221
5222    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5223    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5224
5225    dstThread->setStreamValid(stream, true);
5226
5227    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5228        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5229        if (thread != dstThread &&
5230            thread->type() != ThreadBase::DIRECT) {
5231            MixerThread *srcThread = (MixerThread *)thread;
5232            srcThread->setStreamValid(stream, false);
5233            srcThread->invalidateTracks(stream);
5234        }
5235    }
5236
5237    return NO_ERROR;
5238}
5239
5240
5241int AudioFlinger::newAudioSessionId()
5242{
5243    return nextUniqueId();
5244}
5245
5246void AudioFlinger::acquireAudioSessionId(int audioSession)
5247{
5248    Mutex::Autolock _l(mLock);
5249    pid_t caller = IPCThreadState::self()->getCallingPid();
5250    ALOGV("acquiring %d from %d", audioSession, caller);
5251    int num = mAudioSessionRefs.size();
5252    for (int i = 0; i< num; i++) {
5253        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5254        if (ref->sessionid == audioSession && ref->pid == caller) {
5255            ref->cnt++;
5256            ALOGV(" incremented refcount to %d", ref->cnt);
5257            return;
5258        }
5259    }
5260    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
5261    ALOGV(" added new entry for %d", audioSession);
5262}
5263
5264void AudioFlinger::releaseAudioSessionId(int audioSession)
5265{
5266    Mutex::Autolock _l(mLock);
5267    pid_t caller = IPCThreadState::self()->getCallingPid();
5268    ALOGV("releasing %d from %d", audioSession, caller);
5269    int num = mAudioSessionRefs.size();
5270    for (int i = 0; i< num; i++) {
5271        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5272        if (ref->sessionid == audioSession && ref->pid == caller) {
5273            ref->cnt--;
5274            ALOGV(" decremented refcount to %d", ref->cnt);
5275            if (ref->cnt == 0) {
5276                mAudioSessionRefs.removeAt(i);
5277                delete ref;
5278                purgeStaleEffects_l();
5279            }
5280            return;
5281        }
5282    }
5283    ALOGW("session id %d not found for pid %d", audioSession, caller);
5284}
5285
5286void AudioFlinger::purgeStaleEffects_l() {
5287
5288    ALOGV("purging stale effects");
5289
5290    Vector< sp<EffectChain> > chains;
5291
5292    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5293        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5294        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5295            sp<EffectChain> ec = t->mEffectChains[j];
5296            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5297                chains.push(ec);
5298            }
5299        }
5300    }
5301    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5302        sp<RecordThread> t = mRecordThreads.valueAt(i);
5303        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5304            sp<EffectChain> ec = t->mEffectChains[j];
5305            chains.push(ec);
5306        }
5307    }
5308
5309    for (size_t i = 0; i < chains.size(); i++) {
5310        sp<EffectChain> ec = chains[i];
5311        int sessionid = ec->sessionId();
5312        sp<ThreadBase> t = ec->mThread.promote();
5313        if (t == 0) {
5314            continue;
5315        }
5316        size_t numsessionrefs = mAudioSessionRefs.size();
5317        bool found = false;
5318        for (size_t k = 0; k < numsessionrefs; k++) {
5319            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5320            if (ref->sessionid == sessionid) {
5321                ALOGV(" session %d still exists for %d with %d refs",
5322                     sessionid, ref->pid, ref->cnt);
5323                found = true;
5324                break;
5325            }
5326        }
5327        if (!found) {
5328            // remove all effects from the chain
5329            while (ec->mEffects.size()) {
5330                sp<EffectModule> effect = ec->mEffects[0];
5331                effect->unPin();
5332                Mutex::Autolock _l (t->mLock);
5333                t->removeEffect_l(effect);
5334                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5335                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5336                    if (handle != 0) {
5337                        handle->mEffect.clear();
5338                        if (handle->mHasControl && handle->mEnabled) {
5339                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5340                        }
5341                    }
5342                }
5343                AudioSystem::unregisterEffect(effect->id());
5344            }
5345        }
5346    }
5347    return;
5348}
5349
5350// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5351AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5352{
5353    PlaybackThread *thread = NULL;
5354    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5355        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5356    }
5357    return thread;
5358}
5359
5360// checkMixerThread_l() must be called with AudioFlinger::mLock held
5361AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5362{
5363    PlaybackThread *thread = checkPlaybackThread_l(output);
5364    if (thread != NULL) {
5365        if (thread->type() == ThreadBase::DIRECT) {
5366            thread = NULL;
5367        }
5368    }
5369    return (MixerThread *)thread;
5370}
5371
5372// checkRecordThread_l() must be called with AudioFlinger::mLock held
5373AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5374{
5375    RecordThread *thread = NULL;
5376    if (mRecordThreads.indexOfKey(input) >= 0) {
5377        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5378    }
5379    return thread;
5380}
5381
5382uint32_t AudioFlinger::nextUniqueId()
5383{
5384    return android_atomic_inc(&mNextUniqueId);
5385}
5386
5387AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5388{
5389    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5390        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5391        AudioStreamOut *output = thread->getOutput();
5392        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5393            return thread;
5394        }
5395    }
5396    return NULL;
5397}
5398
5399uint32_t AudioFlinger::primaryOutputDevice_l()
5400{
5401    PlaybackThread *thread = primaryPlaybackThread_l();
5402
5403    if (thread == NULL) {
5404        return 0;
5405    }
5406
5407    return thread->device();
5408}
5409
5410
5411// ----------------------------------------------------------------------------
5412//  Effect management
5413// ----------------------------------------------------------------------------
5414
5415
5416status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5417{
5418    Mutex::Autolock _l(mLock);
5419    return EffectQueryNumberEffects(numEffects);
5420}
5421
5422status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5423{
5424    Mutex::Autolock _l(mLock);
5425    return EffectQueryEffect(index, descriptor);
5426}
5427
5428status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
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<EffectModule> effect;
5834
5835    sp<EffectChain> chain = getEffectChain_l(sessionId);
5836    if (chain != 0) {
5837        effect = chain->getEffectFromId_l(effectId);
5838    }
5839    return effect;
5840}
5841
5842// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5843// PlaybackThread::mLock held
5844status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5845{
5846    // check for existing effect chain with the requested audio session
5847    int sessionId = effect->sessionId();
5848    sp<EffectChain> chain = getEffectChain_l(sessionId);
5849    bool chainCreated = false;
5850
5851    if (chain == 0) {
5852        // create a new chain for this session
5853        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5854        chain = new EffectChain(this, sessionId);
5855        addEffectChain_l(chain);
5856        chain->setStrategy(getStrategyForSession_l(sessionId));
5857        chainCreated = true;
5858    }
5859    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5860
5861    if (chain->getEffectFromId_l(effect->id()) != 0) {
5862        ALOGW("addEffect_l() %p effect %s already present in chain %p",
5863                this, effect->desc().name, chain.get());
5864        return BAD_VALUE;
5865    }
5866
5867    status_t status = chain->addEffect_l(effect);
5868    if (status != NO_ERROR) {
5869        if (chainCreated) {
5870            removeEffectChain_l(chain);
5871        }
5872        return status;
5873    }
5874
5875    effect->setDevice(mDevice);
5876    effect->setMode(mAudioFlinger->getMode());
5877    return NO_ERROR;
5878}
5879
5880void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5881
5882    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5883    effect_descriptor_t desc = effect->desc();
5884    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5885        detachAuxEffect_l(effect->id());
5886    }
5887
5888    sp<EffectChain> chain = effect->chain().promote();
5889    if (chain != 0) {
5890        // remove effect chain if removing last effect
5891        if (chain->removeEffect_l(effect) == 0) {
5892            removeEffectChain_l(chain);
5893        }
5894    } else {
5895        ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5896    }
5897}
5898
5899void AudioFlinger::ThreadBase::lockEffectChains_l(
5900        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5901{
5902    effectChains = mEffectChains;
5903    for (size_t i = 0; i < mEffectChains.size(); i++) {
5904        mEffectChains[i]->lock();
5905    }
5906}
5907
5908void AudioFlinger::ThreadBase::unlockEffectChains(
5909        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5910{
5911    for (size_t i = 0; i < effectChains.size(); i++) {
5912        effectChains[i]->unlock();
5913    }
5914}
5915
5916sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5917{
5918    Mutex::Autolock _l(mLock);
5919    return getEffectChain_l(sessionId);
5920}
5921
5922sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5923{
5924    sp<EffectChain> chain;
5925
5926    size_t size = mEffectChains.size();
5927    for (size_t i = 0; i < size; i++) {
5928        if (mEffectChains[i]->sessionId() == sessionId) {
5929            chain = mEffectChains[i];
5930            break;
5931        }
5932    }
5933    return chain;
5934}
5935
5936void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
5937{
5938    Mutex::Autolock _l(mLock);
5939    size_t size = mEffectChains.size();
5940    for (size_t i = 0; i < size; i++) {
5941        mEffectChains[i]->setMode_l(mode);
5942    }
5943}
5944
5945void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5946                                                    const wp<EffectHandle>& handle,
5947                                                    bool unpiniflast) {
5948
5949    Mutex::Autolock _l(mLock);
5950    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5951    // delete the effect module if removing last handle on it
5952    if (effect->removeHandle(handle) == 0) {
5953        if (!effect->isPinned() || unpiniflast) {
5954            removeEffect_l(effect);
5955            AudioSystem::unregisterEffect(effect->id());
5956        }
5957    }
5958}
5959
5960status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5961{
5962    int session = chain->sessionId();
5963    int16_t *buffer = mMixBuffer;
5964    bool ownsBuffer = false;
5965
5966    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5967    if (session > 0) {
5968        // Only one effect chain can be present in direct output thread and it uses
5969        // the mix buffer as input
5970        if (mType != DIRECT) {
5971            size_t numSamples = mFrameCount * mChannelCount;
5972            buffer = new int16_t[numSamples];
5973            memset(buffer, 0, numSamples * sizeof(int16_t));
5974            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5975            ownsBuffer = true;
5976        }
5977
5978        // Attach all tracks with same session ID to this chain.
5979        for (size_t i = 0; i < mTracks.size(); ++i) {
5980            sp<Track> track = mTracks[i];
5981            if (session == track->sessionId()) {
5982                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5983                track->setMainBuffer(buffer);
5984                chain->incTrackCnt();
5985            }
5986        }
5987
5988        // indicate all active tracks in the chain
5989        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5990            sp<Track> track = mActiveTracks[i].promote();
5991            if (track == 0) continue;
5992            if (session == track->sessionId()) {
5993                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5994                chain->incActiveTrackCnt();
5995            }
5996        }
5997    }
5998
5999    chain->setInBuffer(buffer, ownsBuffer);
6000    chain->setOutBuffer(mMixBuffer);
6001    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
6002    // chains list in order to be processed last as it contains output stage effects
6003    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
6004    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
6005    // after track specific effects and before output stage
6006    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
6007    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
6008    // Effect chain for other sessions are inserted at beginning of effect
6009    // chains list to be processed before output mix effects. Relative order between other
6010    // sessions is not important
6011    size_t size = mEffectChains.size();
6012    size_t i = 0;
6013    for (i = 0; i < size; i++) {
6014        if (mEffectChains[i]->sessionId() < session) break;
6015    }
6016    mEffectChains.insertAt(chain, i);
6017    checkSuspendOnAddEffectChain_l(chain);
6018
6019    return NO_ERROR;
6020}
6021
6022size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
6023{
6024    int session = chain->sessionId();
6025
6026    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
6027
6028    for (size_t i = 0; i < mEffectChains.size(); i++) {
6029        if (chain == mEffectChains[i]) {
6030            mEffectChains.removeAt(i);
6031            // detach all active tracks from the chain
6032            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
6033                sp<Track> track = mActiveTracks[i].promote();
6034                if (track == 0) continue;
6035                if (session == track->sessionId()) {
6036                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
6037                            chain.get(), session);
6038                    chain->decActiveTrackCnt();
6039                }
6040            }
6041
6042            // detach all tracks with same session ID from this chain
6043            for (size_t i = 0; i < mTracks.size(); ++i) {
6044                sp<Track> track = mTracks[i];
6045                if (session == track->sessionId()) {
6046                    track->setMainBuffer(mMixBuffer);
6047                    chain->decTrackCnt();
6048                }
6049            }
6050            break;
6051        }
6052    }
6053    return mEffectChains.size();
6054}
6055
6056status_t AudioFlinger::PlaybackThread::attachAuxEffect(
6057        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6058{
6059    Mutex::Autolock _l(mLock);
6060    return attachAuxEffect_l(track, EffectId);
6061}
6062
6063status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
6064        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
6065{
6066    status_t status = NO_ERROR;
6067
6068    if (EffectId == 0) {
6069        track->setAuxBuffer(0, NULL);
6070    } else {
6071        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
6072        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
6073        if (effect != 0) {
6074            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6075                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
6076            } else {
6077                status = INVALID_OPERATION;
6078            }
6079        } else {
6080            status = BAD_VALUE;
6081        }
6082    }
6083    return status;
6084}
6085
6086void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
6087{
6088     for (size_t i = 0; i < mTracks.size(); ++i) {
6089        sp<Track> track = mTracks[i];
6090        if (track->auxEffectId() == effectId) {
6091            attachAuxEffect_l(track, 0);
6092        }
6093    }
6094}
6095
6096status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
6097{
6098    // only one chain per input thread
6099    if (mEffectChains.size() != 0) {
6100        return INVALID_OPERATION;
6101    }
6102    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
6103
6104    chain->setInBuffer(NULL);
6105    chain->setOutBuffer(NULL);
6106
6107    checkSuspendOnAddEffectChain_l(chain);
6108
6109    mEffectChains.add(chain);
6110
6111    return NO_ERROR;
6112}
6113
6114size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6115{
6116    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6117    ALOGW_IF(mEffectChains.size() != 1,
6118            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6119            chain.get(), mEffectChains.size(), this);
6120    if (mEffectChains.size() == 1) {
6121        mEffectChains.removeAt(0);
6122    }
6123    return 0;
6124}
6125
6126// ----------------------------------------------------------------------------
6127//  EffectModule implementation
6128// ----------------------------------------------------------------------------
6129
6130#undef LOG_TAG
6131#define LOG_TAG "AudioFlinger::EffectModule"
6132
6133AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6134                                        const wp<AudioFlinger::EffectChain>& chain,
6135                                        effect_descriptor_t *desc,
6136                                        int id,
6137                                        int sessionId)
6138    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6139      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6140{
6141    ALOGV("Constructor %p", this);
6142    int lStatus;
6143    sp<ThreadBase> thread = mThread.promote();
6144    if (thread == 0) {
6145        return;
6146    }
6147
6148    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6149
6150    // create effect engine from effect factory
6151    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6152
6153    if (mStatus != NO_ERROR) {
6154        return;
6155    }
6156    lStatus = init();
6157    if (lStatus < 0) {
6158        mStatus = lStatus;
6159        goto Error;
6160    }
6161
6162    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6163        mPinned = true;
6164    }
6165    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6166    return;
6167Error:
6168    EffectRelease(mEffectInterface);
6169    mEffectInterface = NULL;
6170    ALOGV("Constructor Error %d", mStatus);
6171}
6172
6173AudioFlinger::EffectModule::~EffectModule()
6174{
6175    ALOGV("Destructor %p", this);
6176    if (mEffectInterface != NULL) {
6177        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6178                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6179            sp<ThreadBase> thread = mThread.promote();
6180            if (thread != 0) {
6181                audio_stream_t *stream = thread->stream();
6182                if (stream != NULL) {
6183                    stream->remove_audio_effect(stream, mEffectInterface);
6184                }
6185            }
6186        }
6187        // release effect engine
6188        EffectRelease(mEffectInterface);
6189    }
6190}
6191
6192status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle)
6193{
6194    status_t status;
6195
6196    Mutex::Autolock _l(mLock);
6197    // First handle in mHandles has highest priority and controls the effect module
6198    int priority = handle->priority();
6199    size_t size = mHandles.size();
6200    sp<EffectHandle> h;
6201    size_t i;
6202    for (i = 0; i < size; i++) {
6203        h = mHandles[i].promote();
6204        if (h == 0) continue;
6205        if (h->priority() <= priority) break;
6206    }
6207    // if inserted in first place, move effect control from previous owner to this handle
6208    if (i == 0) {
6209        bool enabled = false;
6210        if (h != 0) {
6211            enabled = h->enabled();
6212            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6213        }
6214        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6215        status = NO_ERROR;
6216    } else {
6217        status = ALREADY_EXISTS;
6218    }
6219    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6220    mHandles.insertAt(handle, i);
6221    return status;
6222}
6223
6224size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6225{
6226    Mutex::Autolock _l(mLock);
6227    size_t size = mHandles.size();
6228    size_t i;
6229    for (i = 0; i < size; i++) {
6230        if (mHandles[i] == handle) break;
6231    }
6232    if (i == size) {
6233        return size;
6234    }
6235    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6236
6237    bool enabled = false;
6238    EffectHandle *hdl = handle.unsafe_get();
6239    if (hdl != NULL) {
6240        ALOGV("removeHandle() unsafe_get OK");
6241        enabled = hdl->enabled();
6242    }
6243    mHandles.removeAt(i);
6244    size = mHandles.size();
6245    // if removed from first place, move effect control from this handle to next in line
6246    if (i == 0 && size != 0) {
6247        sp<EffectHandle> h = mHandles[0].promote();
6248        if (h != 0) {
6249            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6250        }
6251    }
6252
6253    // Prevent calls to process() and other functions on effect interface from now on.
6254    // The effect engine will be released by the destructor when the last strong reference on
6255    // this object is released which can happen after next process is called.
6256    if (size == 0 && !mPinned) {
6257        mState = DESTROYED;
6258    }
6259
6260    return size;
6261}
6262
6263sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6264{
6265    Mutex::Autolock _l(mLock);
6266    sp<EffectHandle> handle;
6267    if (mHandles.size() != 0) {
6268        handle = mHandles[0].promote();
6269    }
6270    return handle;
6271}
6272
6273void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6274{
6275    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6276    // keep a strong reference on this EffectModule to avoid calling the
6277    // destructor before we exit
6278    sp<EffectModule> keep(this);
6279    {
6280        sp<ThreadBase> thread = mThread.promote();
6281        if (thread != 0) {
6282            thread->disconnectEffect(keep, handle, unpiniflast);
6283        }
6284    }
6285}
6286
6287void AudioFlinger::EffectModule::updateState() {
6288    Mutex::Autolock _l(mLock);
6289
6290    switch (mState) {
6291    case RESTART:
6292        reset_l();
6293        // FALL THROUGH
6294
6295    case STARTING:
6296        // clear auxiliary effect input buffer for next accumulation
6297        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6298            memset(mConfig.inputCfg.buffer.raw,
6299                   0,
6300                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6301        }
6302        start_l();
6303        mState = ACTIVE;
6304        break;
6305    case STOPPING:
6306        stop_l();
6307        mDisableWaitCnt = mMaxDisableWaitCnt;
6308        mState = STOPPED;
6309        break;
6310    case STOPPED:
6311        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6312        // turn off sequence.
6313        if (--mDisableWaitCnt == 0) {
6314            reset_l();
6315            mState = IDLE;
6316        }
6317        break;
6318    default: //IDLE , ACTIVE, DESTROYED
6319        break;
6320    }
6321}
6322
6323void AudioFlinger::EffectModule::process()
6324{
6325    Mutex::Autolock _l(mLock);
6326
6327    if (mState == DESTROYED || mEffectInterface == NULL ||
6328            mConfig.inputCfg.buffer.raw == NULL ||
6329            mConfig.outputCfg.buffer.raw == NULL) {
6330        return;
6331    }
6332
6333    if (isProcessEnabled()) {
6334        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6335        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6336            ditherAndClamp(mConfig.inputCfg.buffer.s32,
6337                                        mConfig.inputCfg.buffer.s32,
6338                                        mConfig.inputCfg.buffer.frameCount/2);
6339        }
6340
6341        // do the actual processing in the effect engine
6342        int ret = (*mEffectInterface)->process(mEffectInterface,
6343                                               &mConfig.inputCfg.buffer,
6344                                               &mConfig.outputCfg.buffer);
6345
6346        // force transition to IDLE state when engine is ready
6347        if (mState == STOPPED && ret == -ENODATA) {
6348            mDisableWaitCnt = 1;
6349        }
6350
6351        // clear auxiliary effect input buffer for next accumulation
6352        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6353            memset(mConfig.inputCfg.buffer.raw, 0,
6354                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6355        }
6356    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6357                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6358        // If an insert effect is idle and input buffer is different from output buffer,
6359        // accumulate input onto output
6360        sp<EffectChain> chain = mChain.promote();
6361        if (chain != 0 && chain->activeTrackCnt() != 0) {
6362            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6363            int16_t *in = mConfig.inputCfg.buffer.s16;
6364            int16_t *out = mConfig.outputCfg.buffer.s16;
6365            for (size_t i = 0; i < frameCnt; i++) {
6366                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6367            }
6368        }
6369    }
6370}
6371
6372void AudioFlinger::EffectModule::reset_l()
6373{
6374    if (mEffectInterface == NULL) {
6375        return;
6376    }
6377    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6378}
6379
6380status_t AudioFlinger::EffectModule::configure()
6381{
6382    uint32_t channels;
6383    if (mEffectInterface == NULL) {
6384        return NO_INIT;
6385    }
6386
6387    sp<ThreadBase> thread = mThread.promote();
6388    if (thread == 0) {
6389        return DEAD_OBJECT;
6390    }
6391
6392    // TODO: handle configuration of effects replacing track process
6393    if (thread->channelCount() == 1) {
6394        channels = AUDIO_CHANNEL_OUT_MONO;
6395    } else {
6396        channels = AUDIO_CHANNEL_OUT_STEREO;
6397    }
6398
6399    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6400        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6401    } else {
6402        mConfig.inputCfg.channels = channels;
6403    }
6404    mConfig.outputCfg.channels = channels;
6405    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6406    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6407    mConfig.inputCfg.samplingRate = thread->sampleRate();
6408    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6409    mConfig.inputCfg.bufferProvider.cookie = NULL;
6410    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6411    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6412    mConfig.outputCfg.bufferProvider.cookie = NULL;
6413    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6414    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6415    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6416    // Insert effect:
6417    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6418    // always overwrites output buffer: input buffer == output buffer
6419    // - in other sessions:
6420    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6421    //      other effect: overwrites output buffer: input buffer == output buffer
6422    // Auxiliary effect:
6423    //      accumulates in output buffer: input buffer != output buffer
6424    // Therefore: accumulate <=> input buffer != output buffer
6425    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6426        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6427    } else {
6428        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6429    }
6430    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6431    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6432    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6433    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6434
6435    ALOGV("configure() %p thread %p buffer %p framecount %d",
6436            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6437
6438    status_t cmdStatus;
6439    uint32_t size = sizeof(int);
6440    status_t status = (*mEffectInterface)->command(mEffectInterface,
6441                                                   EFFECT_CMD_SET_CONFIG,
6442                                                   sizeof(effect_config_t),
6443                                                   &mConfig,
6444                                                   &size,
6445                                                   &cmdStatus);
6446    if (status == 0) {
6447        status = cmdStatus;
6448    }
6449
6450    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6451            (1000 * mConfig.outputCfg.buffer.frameCount);
6452
6453    return status;
6454}
6455
6456status_t AudioFlinger::EffectModule::init()
6457{
6458    Mutex::Autolock _l(mLock);
6459    if (mEffectInterface == NULL) {
6460        return NO_INIT;
6461    }
6462    status_t cmdStatus;
6463    uint32_t size = sizeof(status_t);
6464    status_t status = (*mEffectInterface)->command(mEffectInterface,
6465                                                   EFFECT_CMD_INIT,
6466                                                   0,
6467                                                   NULL,
6468                                                   &size,
6469                                                   &cmdStatus);
6470    if (status == 0) {
6471        status = cmdStatus;
6472    }
6473    return status;
6474}
6475
6476status_t AudioFlinger::EffectModule::start()
6477{
6478    Mutex::Autolock _l(mLock);
6479    return start_l();
6480}
6481
6482status_t AudioFlinger::EffectModule::start_l()
6483{
6484    if (mEffectInterface == NULL) {
6485        return NO_INIT;
6486    }
6487    status_t cmdStatus;
6488    uint32_t size = sizeof(status_t);
6489    status_t status = (*mEffectInterface)->command(mEffectInterface,
6490                                                   EFFECT_CMD_ENABLE,
6491                                                   0,
6492                                                   NULL,
6493                                                   &size,
6494                                                   &cmdStatus);
6495    if (status == 0) {
6496        status = cmdStatus;
6497    }
6498    if (status == 0 &&
6499            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6500             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6501        sp<ThreadBase> thread = mThread.promote();
6502        if (thread != 0) {
6503            audio_stream_t *stream = thread->stream();
6504            if (stream != NULL) {
6505                stream->add_audio_effect(stream, mEffectInterface);
6506            }
6507        }
6508    }
6509    return status;
6510}
6511
6512status_t AudioFlinger::EffectModule::stop()
6513{
6514    Mutex::Autolock _l(mLock);
6515    return stop_l();
6516}
6517
6518status_t AudioFlinger::EffectModule::stop_l()
6519{
6520    if (mEffectInterface == NULL) {
6521        return NO_INIT;
6522    }
6523    status_t cmdStatus;
6524    uint32_t size = sizeof(status_t);
6525    status_t status = (*mEffectInterface)->command(mEffectInterface,
6526                                                   EFFECT_CMD_DISABLE,
6527                                                   0,
6528                                                   NULL,
6529                                                   &size,
6530                                                   &cmdStatus);
6531    if (status == 0) {
6532        status = cmdStatus;
6533    }
6534    if (status == 0 &&
6535            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6536             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6537        sp<ThreadBase> thread = mThread.promote();
6538        if (thread != 0) {
6539            audio_stream_t *stream = thread->stream();
6540            if (stream != NULL) {
6541                stream->remove_audio_effect(stream, mEffectInterface);
6542            }
6543        }
6544    }
6545    return status;
6546}
6547
6548status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6549                                             uint32_t cmdSize,
6550                                             void *pCmdData,
6551                                             uint32_t *replySize,
6552                                             void *pReplyData)
6553{
6554    Mutex::Autolock _l(mLock);
6555//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6556
6557    if (mState == DESTROYED || mEffectInterface == NULL) {
6558        return NO_INIT;
6559    }
6560    status_t status = (*mEffectInterface)->command(mEffectInterface,
6561                                                   cmdCode,
6562                                                   cmdSize,
6563                                                   pCmdData,
6564                                                   replySize,
6565                                                   pReplyData);
6566    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6567        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6568        for (size_t i = 1; i < mHandles.size(); i++) {
6569            sp<EffectHandle> h = mHandles[i].promote();
6570            if (h != 0) {
6571                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6572            }
6573        }
6574    }
6575    return status;
6576}
6577
6578status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6579{
6580
6581    Mutex::Autolock _l(mLock);
6582    ALOGV("setEnabled %p enabled %d", this, enabled);
6583
6584    if (enabled != isEnabled()) {
6585        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6586        if (enabled && status != NO_ERROR) {
6587            return status;
6588        }
6589
6590        switch (mState) {
6591        // going from disabled to enabled
6592        case IDLE:
6593            mState = STARTING;
6594            break;
6595        case STOPPED:
6596            mState = RESTART;
6597            break;
6598        case STOPPING:
6599            mState = ACTIVE;
6600            break;
6601
6602        // going from enabled to disabled
6603        case RESTART:
6604            mState = STOPPED;
6605            break;
6606        case STARTING:
6607            mState = IDLE;
6608            break;
6609        case ACTIVE:
6610            mState = STOPPING;
6611            break;
6612        case DESTROYED:
6613            return NO_ERROR; // simply ignore as we are being destroyed
6614        }
6615        for (size_t i = 1; i < mHandles.size(); i++) {
6616            sp<EffectHandle> h = mHandles[i].promote();
6617            if (h != 0) {
6618                h->setEnabled(enabled);
6619            }
6620        }
6621    }
6622    return NO_ERROR;
6623}
6624
6625bool AudioFlinger::EffectModule::isEnabled()
6626{
6627    switch (mState) {
6628    case RESTART:
6629    case STARTING:
6630    case ACTIVE:
6631        return true;
6632    case IDLE:
6633    case STOPPING:
6634    case STOPPED:
6635    case DESTROYED:
6636    default:
6637        return false;
6638    }
6639}
6640
6641bool AudioFlinger::EffectModule::isProcessEnabled()
6642{
6643    switch (mState) {
6644    case RESTART:
6645    case ACTIVE:
6646    case STOPPING:
6647    case STOPPED:
6648        return true;
6649    case IDLE:
6650    case STARTING:
6651    case DESTROYED:
6652    default:
6653        return false;
6654    }
6655}
6656
6657status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6658{
6659    Mutex::Autolock _l(mLock);
6660    status_t status = NO_ERROR;
6661
6662    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6663    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6664    if (isProcessEnabled() &&
6665            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6666            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6667        status_t cmdStatus;
6668        uint32_t volume[2];
6669        uint32_t *pVolume = NULL;
6670        uint32_t size = sizeof(volume);
6671        volume[0] = *left;
6672        volume[1] = *right;
6673        if (controller) {
6674            pVolume = volume;
6675        }
6676        status = (*mEffectInterface)->command(mEffectInterface,
6677                                              EFFECT_CMD_SET_VOLUME,
6678                                              size,
6679                                              volume,
6680                                              &size,
6681                                              pVolume);
6682        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6683            *left = volume[0];
6684            *right = volume[1];
6685        }
6686    }
6687    return status;
6688}
6689
6690status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6691{
6692    Mutex::Autolock _l(mLock);
6693    status_t status = NO_ERROR;
6694    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6695        // audio pre processing modules on RecordThread can receive both output and
6696        // input device indication in the same call
6697        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6698        if (dev) {
6699            status_t cmdStatus;
6700            uint32_t size = sizeof(status_t);
6701
6702            status = (*mEffectInterface)->command(mEffectInterface,
6703                                                  EFFECT_CMD_SET_DEVICE,
6704                                                  sizeof(uint32_t),
6705                                                  &dev,
6706                                                  &size,
6707                                                  &cmdStatus);
6708            if (status == NO_ERROR) {
6709                status = cmdStatus;
6710            }
6711        }
6712        dev = device & AUDIO_DEVICE_IN_ALL;
6713        if (dev) {
6714            status_t cmdStatus;
6715            uint32_t size = sizeof(status_t);
6716
6717            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6718                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6719                                                  sizeof(uint32_t),
6720                                                  &dev,
6721                                                  &size,
6722                                                  &cmdStatus);
6723            if (status2 == NO_ERROR) {
6724                status2 = cmdStatus;
6725            }
6726            if (status == NO_ERROR) {
6727                status = status2;
6728            }
6729        }
6730    }
6731    return status;
6732}
6733
6734status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
6735{
6736    Mutex::Autolock _l(mLock);
6737    status_t status = NO_ERROR;
6738    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6739        status_t cmdStatus;
6740        uint32_t size = sizeof(status_t);
6741        status = (*mEffectInterface)->command(mEffectInterface,
6742                                              EFFECT_CMD_SET_AUDIO_MODE,
6743                                              sizeof(audio_mode_t),
6744                                              &mode,
6745                                              &size,
6746                                              &cmdStatus);
6747        if (status == NO_ERROR) {
6748            status = cmdStatus;
6749        }
6750    }
6751    return status;
6752}
6753
6754void AudioFlinger::EffectModule::setSuspended(bool suspended)
6755{
6756    Mutex::Autolock _l(mLock);
6757    mSuspended = suspended;
6758}
6759
6760bool AudioFlinger::EffectModule::suspended() const
6761{
6762    Mutex::Autolock _l(mLock);
6763    return mSuspended;
6764}
6765
6766status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6767{
6768    const size_t SIZE = 256;
6769    char buffer[SIZE];
6770    String8 result;
6771
6772    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6773    result.append(buffer);
6774
6775    bool locked = tryLock(mLock);
6776    // failed to lock - AudioFlinger is probably deadlocked
6777    if (!locked) {
6778        result.append("\t\tCould not lock Fx mutex:\n");
6779    }
6780
6781    result.append("\t\tSession Status State Engine:\n");
6782    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6783            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6784    result.append(buffer);
6785
6786    result.append("\t\tDescriptor:\n");
6787    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6788            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6789            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6790            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6791    result.append(buffer);
6792    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6793                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6794                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6795                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6796    result.append(buffer);
6797    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6798            mDescriptor.apiVersion,
6799            mDescriptor.flags);
6800    result.append(buffer);
6801    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6802            mDescriptor.name);
6803    result.append(buffer);
6804    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6805            mDescriptor.implementor);
6806    result.append(buffer);
6807
6808    result.append("\t\t- Input configuration:\n");
6809    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6810    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6811            (uint32_t)mConfig.inputCfg.buffer.raw,
6812            mConfig.inputCfg.buffer.frameCount,
6813            mConfig.inputCfg.samplingRate,
6814            mConfig.inputCfg.channels,
6815            mConfig.inputCfg.format);
6816    result.append(buffer);
6817
6818    result.append("\t\t- Output configuration:\n");
6819    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6820    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6821            (uint32_t)mConfig.outputCfg.buffer.raw,
6822            mConfig.outputCfg.buffer.frameCount,
6823            mConfig.outputCfg.samplingRate,
6824            mConfig.outputCfg.channels,
6825            mConfig.outputCfg.format);
6826    result.append(buffer);
6827
6828    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6829    result.append(buffer);
6830    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6831    for (size_t i = 0; i < mHandles.size(); ++i) {
6832        sp<EffectHandle> handle = mHandles[i].promote();
6833        if (handle != 0) {
6834            handle->dump(buffer, SIZE);
6835            result.append(buffer);
6836        }
6837    }
6838
6839    result.append("\n");
6840
6841    write(fd, result.string(), result.length());
6842
6843    if (locked) {
6844        mLock.unlock();
6845    }
6846
6847    return NO_ERROR;
6848}
6849
6850// ----------------------------------------------------------------------------
6851//  EffectHandle implementation
6852// ----------------------------------------------------------------------------
6853
6854#undef LOG_TAG
6855#define LOG_TAG "AudioFlinger::EffectHandle"
6856
6857AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6858                                        const sp<AudioFlinger::Client>& client,
6859                                        const sp<IEffectClient>& effectClient,
6860                                        int32_t priority)
6861    : BnEffect(),
6862    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6863    mPriority(priority), mHasControl(false), mEnabled(false)
6864{
6865    ALOGV("constructor %p", this);
6866
6867    if (client == 0) {
6868        return;
6869    }
6870    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6871    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6872    if (mCblkMemory != 0) {
6873        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6874
6875        if (mCblk != NULL) {
6876            new(mCblk) effect_param_cblk_t();
6877            mBuffer = (uint8_t *)mCblk + bufOffset;
6878         }
6879    } else {
6880        ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6881        return;
6882    }
6883}
6884
6885AudioFlinger::EffectHandle::~EffectHandle()
6886{
6887    ALOGV("Destructor %p", this);
6888    disconnect(false);
6889    ALOGV("Destructor DONE %p", this);
6890}
6891
6892status_t AudioFlinger::EffectHandle::enable()
6893{
6894    ALOGV("enable %p", this);
6895    if (!mHasControl) return INVALID_OPERATION;
6896    if (mEffect == 0) return DEAD_OBJECT;
6897
6898    if (mEnabled) {
6899        return NO_ERROR;
6900    }
6901
6902    mEnabled = true;
6903
6904    sp<ThreadBase> thread = mEffect->thread().promote();
6905    if (thread != 0) {
6906        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6907    }
6908
6909    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6910    if (mEffect->suspended()) {
6911        return NO_ERROR;
6912    }
6913
6914    status_t status = mEffect->setEnabled(true);
6915    if (status != NO_ERROR) {
6916        if (thread != 0) {
6917            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6918        }
6919        mEnabled = false;
6920    }
6921    return status;
6922}
6923
6924status_t AudioFlinger::EffectHandle::disable()
6925{
6926    ALOGV("disable %p", this);
6927    if (!mHasControl) return INVALID_OPERATION;
6928    if (mEffect == 0) return DEAD_OBJECT;
6929
6930    if (!mEnabled) {
6931        return NO_ERROR;
6932    }
6933    mEnabled = false;
6934
6935    if (mEffect->suspended()) {
6936        return NO_ERROR;
6937    }
6938
6939    status_t status = mEffect->setEnabled(false);
6940
6941    sp<ThreadBase> thread = mEffect->thread().promote();
6942    if (thread != 0) {
6943        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6944    }
6945
6946    return status;
6947}
6948
6949void AudioFlinger::EffectHandle::disconnect()
6950{
6951    disconnect(true);
6952}
6953
6954void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6955{
6956    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6957    if (mEffect == 0) {
6958        return;
6959    }
6960    mEffect->disconnect(this, unpiniflast);
6961
6962    if (mHasControl && mEnabled) {
6963        sp<ThreadBase> thread = mEffect->thread().promote();
6964        if (thread != 0) {
6965            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6966        }
6967    }
6968
6969    // release sp on module => module destructor can be called now
6970    mEffect.clear();
6971    if (mClient != 0) {
6972        if (mCblk != NULL) {
6973            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6974        }
6975        mCblkMemory.clear();            // and free the shared memory
6976        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6977        mClient.clear();
6978    }
6979}
6980
6981status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6982                                             uint32_t cmdSize,
6983                                             void *pCmdData,
6984                                             uint32_t *replySize,
6985                                             void *pReplyData)
6986{
6987//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6988//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6989
6990    // only get parameter command is permitted for applications not controlling the effect
6991    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6992        return INVALID_OPERATION;
6993    }
6994    if (mEffect == 0) return DEAD_OBJECT;
6995    if (mClient == 0) return INVALID_OPERATION;
6996
6997    // handle commands that are not forwarded transparently to effect engine
6998    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6999        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
7000        // no risk to block the whole media server process or mixer threads is we are stuck here
7001        Mutex::Autolock _l(mCblk->lock);
7002        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
7003            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
7004            mCblk->serverIndex = 0;
7005            mCblk->clientIndex = 0;
7006            return BAD_VALUE;
7007        }
7008        status_t status = NO_ERROR;
7009        while (mCblk->serverIndex < mCblk->clientIndex) {
7010            int reply;
7011            uint32_t rsize = sizeof(int);
7012            int *p = (int *)(mBuffer + mCblk->serverIndex);
7013            int size = *p++;
7014            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
7015                ALOGW("command(): invalid parameter block size");
7016                break;
7017            }
7018            effect_param_t *param = (effect_param_t *)p;
7019            if (param->psize == 0 || param->vsize == 0) {
7020                ALOGW("command(): null parameter or value size");
7021                mCblk->serverIndex += size;
7022                continue;
7023            }
7024            uint32_t psize = sizeof(effect_param_t) +
7025                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
7026                             param->vsize;
7027            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
7028                                            psize,
7029                                            p,
7030                                            &rsize,
7031                                            &reply);
7032            // stop at first error encountered
7033            if (ret != NO_ERROR) {
7034                status = ret;
7035                *(int *)pReplyData = reply;
7036                break;
7037            } else if (reply != NO_ERROR) {
7038                *(int *)pReplyData = reply;
7039                break;
7040            }
7041            mCblk->serverIndex += size;
7042        }
7043        mCblk->serverIndex = 0;
7044        mCblk->clientIndex = 0;
7045        return status;
7046    } else if (cmdCode == EFFECT_CMD_ENABLE) {
7047        *(int *)pReplyData = NO_ERROR;
7048        return enable();
7049    } else if (cmdCode == EFFECT_CMD_DISABLE) {
7050        *(int *)pReplyData = NO_ERROR;
7051        return disable();
7052    }
7053
7054    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7055}
7056
7057sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
7058    return mCblkMemory;
7059}
7060
7061void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
7062{
7063    ALOGV("setControl %p control %d", this, hasControl);
7064
7065    mHasControl = hasControl;
7066    mEnabled = enabled;
7067
7068    if (signal && mEffectClient != 0) {
7069        mEffectClient->controlStatusChanged(hasControl);
7070    }
7071}
7072
7073void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
7074                                                 uint32_t cmdSize,
7075                                                 void *pCmdData,
7076                                                 uint32_t replySize,
7077                                                 void *pReplyData)
7078{
7079    if (mEffectClient != 0) {
7080        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
7081    }
7082}
7083
7084
7085
7086void AudioFlinger::EffectHandle::setEnabled(bool enabled)
7087{
7088    if (mEffectClient != 0) {
7089        mEffectClient->enableStatusChanged(enabled);
7090    }
7091}
7092
7093status_t AudioFlinger::EffectHandle::onTransact(
7094    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7095{
7096    return BnEffect::onTransact(code, data, reply, flags);
7097}
7098
7099
7100void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
7101{
7102    bool locked = mCblk != NULL && tryLock(mCblk->lock);
7103
7104    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
7105            (mClient == NULL) ? getpid() : mClient->pid(),
7106            mPriority,
7107            mHasControl,
7108            !locked,
7109            mCblk ? mCblk->clientIndex : 0,
7110            mCblk ? mCblk->serverIndex : 0
7111            );
7112
7113    if (locked) {
7114        mCblk->lock.unlock();
7115    }
7116}
7117
7118#undef LOG_TAG
7119#define LOG_TAG "AudioFlinger::EffectChain"
7120
7121AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7122                                        int sessionId)
7123    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
7124      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7125      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7126{
7127    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7128    sp<ThreadBase> thread = mThread.promote();
7129    if (thread == 0) {
7130        return;
7131    }
7132    mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
7133                                    thread->frameCount();
7134}
7135
7136AudioFlinger::EffectChain::~EffectChain()
7137{
7138    if (mOwnInBuffer) {
7139        delete mInBuffer;
7140    }
7141
7142}
7143
7144// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7145sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7146{
7147    sp<EffectModule> effect;
7148    size_t size = mEffects.size();
7149
7150    for (size_t i = 0; i < size; i++) {
7151        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7152            effect = mEffects[i];
7153            break;
7154        }
7155    }
7156    return effect;
7157}
7158
7159// getEffectFromId_l() must be called with ThreadBase::mLock held
7160sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7161{
7162    sp<EffectModule> effect;
7163    size_t size = mEffects.size();
7164
7165    for (size_t i = 0; i < size; i++) {
7166        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7167        if (id == 0 || mEffects[i]->id() == id) {
7168            effect = mEffects[i];
7169            break;
7170        }
7171    }
7172    return effect;
7173}
7174
7175// getEffectFromType_l() must be called with ThreadBase::mLock held
7176sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7177        const effect_uuid_t *type)
7178{
7179    sp<EffectModule> effect;
7180    size_t size = mEffects.size();
7181
7182    for (size_t i = 0; i < size; i++) {
7183        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7184            effect = mEffects[i];
7185            break;
7186        }
7187    }
7188    return effect;
7189}
7190
7191// Must be called with EffectChain::mLock locked
7192void AudioFlinger::EffectChain::process_l()
7193{
7194    sp<ThreadBase> thread = mThread.promote();
7195    if (thread == 0) {
7196        ALOGW("process_l(): cannot promote mixer thread");
7197        return;
7198    }
7199    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7200            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7201    // always process effects unless no more tracks are on the session and the effect tail
7202    // has been rendered
7203    bool doProcess = true;
7204    if (!isGlobalSession) {
7205        bool tracksOnSession = (trackCnt() != 0);
7206
7207        if (!tracksOnSession && mTailBufferCount == 0) {
7208            doProcess = false;
7209        }
7210
7211        if (activeTrackCnt() == 0) {
7212            // if no track is active and the effect tail has not been rendered,
7213            // the input buffer must be cleared here as the mixer process will not do it
7214            if (tracksOnSession || mTailBufferCount > 0) {
7215                size_t numSamples = thread->frameCount() * thread->channelCount();
7216                memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7217                if (mTailBufferCount > 0) {
7218                    mTailBufferCount--;
7219                }
7220            }
7221        }
7222    }
7223
7224    size_t size = mEffects.size();
7225    if (doProcess) {
7226        for (size_t i = 0; i < size; i++) {
7227            mEffects[i]->process();
7228        }
7229    }
7230    for (size_t i = 0; i < size; i++) {
7231        mEffects[i]->updateState();
7232    }
7233}
7234
7235// addEffect_l() must be called with PlaybackThread::mLock held
7236status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7237{
7238    effect_descriptor_t desc = effect->desc();
7239    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7240
7241    Mutex::Autolock _l(mLock);
7242    effect->setChain(this);
7243    sp<ThreadBase> thread = mThread.promote();
7244    if (thread == 0) {
7245        return NO_INIT;
7246    }
7247    effect->setThread(thread);
7248
7249    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7250        // Auxiliary effects are inserted at the beginning of mEffects vector as
7251        // they are processed first and accumulated in chain input buffer
7252        mEffects.insertAt(effect, 0);
7253
7254        // the input buffer for auxiliary effect contains mono samples in
7255        // 32 bit format. This is to avoid saturation in AudoMixer
7256        // accumulation stage. Saturation is done in EffectModule::process() before
7257        // calling the process in effect engine
7258        size_t numSamples = thread->frameCount();
7259        int32_t *buffer = new int32_t[numSamples];
7260        memset(buffer, 0, numSamples * sizeof(int32_t));
7261        effect->setInBuffer((int16_t *)buffer);
7262        // auxiliary effects output samples to chain input buffer for further processing
7263        // by insert effects
7264        effect->setOutBuffer(mInBuffer);
7265    } else {
7266        // Insert effects are inserted at the end of mEffects vector as they are processed
7267        //  after track and auxiliary effects.
7268        // Insert effect order as a function of indicated preference:
7269        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7270        //  another effect is present
7271        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7272        //  last effect claiming first position
7273        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7274        //  first effect claiming last position
7275        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7276        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7277        // already present
7278
7279        int size = (int)mEffects.size();
7280        int idx_insert = size;
7281        int idx_insert_first = -1;
7282        int idx_insert_last = -1;
7283
7284        for (int i = 0; i < size; i++) {
7285            effect_descriptor_t d = mEffects[i]->desc();
7286            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7287            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7288            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7289                // check invalid effect chaining combinations
7290                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7291                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7292                    ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7293                    return INVALID_OPERATION;
7294                }
7295                // remember position of first insert effect and by default
7296                // select this as insert position for new effect
7297                if (idx_insert == size) {
7298                    idx_insert = i;
7299                }
7300                // remember position of last insert effect claiming
7301                // first position
7302                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7303                    idx_insert_first = i;
7304                }
7305                // remember position of first insert effect claiming
7306                // last position
7307                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7308                    idx_insert_last == -1) {
7309                    idx_insert_last = i;
7310                }
7311            }
7312        }
7313
7314        // modify idx_insert from first position if needed
7315        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7316            if (idx_insert_last != -1) {
7317                idx_insert = idx_insert_last;
7318            } else {
7319                idx_insert = size;
7320            }
7321        } else {
7322            if (idx_insert_first != -1) {
7323                idx_insert = idx_insert_first + 1;
7324            }
7325        }
7326
7327        // always read samples from chain input buffer
7328        effect->setInBuffer(mInBuffer);
7329
7330        // if last effect in the chain, output samples to chain
7331        // output buffer, otherwise to chain input buffer
7332        if (idx_insert == size) {
7333            if (idx_insert != 0) {
7334                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7335                mEffects[idx_insert-1]->configure();
7336            }
7337            effect->setOutBuffer(mOutBuffer);
7338        } else {
7339            effect->setOutBuffer(mInBuffer);
7340        }
7341        mEffects.insertAt(effect, idx_insert);
7342
7343        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7344    }
7345    effect->configure();
7346    return NO_ERROR;
7347}
7348
7349// removeEffect_l() must be called with PlaybackThread::mLock held
7350size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7351{
7352    Mutex::Autolock _l(mLock);
7353    int size = (int)mEffects.size();
7354    int i;
7355    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7356
7357    for (i = 0; i < size; i++) {
7358        if (effect == mEffects[i]) {
7359            // calling stop here will remove pre-processing effect from the audio HAL.
7360            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7361            // the middle of a read from audio HAL
7362            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7363                    mEffects[i]->state() == EffectModule::STOPPING) {
7364                mEffects[i]->stop();
7365            }
7366            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7367                delete[] effect->inBuffer();
7368            } else {
7369                if (i == size - 1 && i != 0) {
7370                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7371                    mEffects[i - 1]->configure();
7372                }
7373            }
7374            mEffects.removeAt(i);
7375            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7376            break;
7377        }
7378    }
7379
7380    return mEffects.size();
7381}
7382
7383// setDevice_l() must be called with PlaybackThread::mLock held
7384void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7385{
7386    size_t size = mEffects.size();
7387    for (size_t i = 0; i < size; i++) {
7388        mEffects[i]->setDevice(device);
7389    }
7390}
7391
7392// setMode_l() must be called with PlaybackThread::mLock held
7393void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode)
7394{
7395    size_t size = mEffects.size();
7396    for (size_t i = 0; i < size; i++) {
7397        mEffects[i]->setMode(mode);
7398    }
7399}
7400
7401// setVolume_l() must be called with PlaybackThread::mLock held
7402bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7403{
7404    uint32_t newLeft = *left;
7405    uint32_t newRight = *right;
7406    bool hasControl = false;
7407    int ctrlIdx = -1;
7408    size_t size = mEffects.size();
7409
7410    // first update volume controller
7411    for (size_t i = size; i > 0; i--) {
7412        if (mEffects[i - 1]->isProcessEnabled() &&
7413            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7414            ctrlIdx = i - 1;
7415            hasControl = true;
7416            break;
7417        }
7418    }
7419
7420    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7421        if (hasControl) {
7422            *left = mNewLeftVolume;
7423            *right = mNewRightVolume;
7424        }
7425        return hasControl;
7426    }
7427
7428    mVolumeCtrlIdx = ctrlIdx;
7429    mLeftVolume = newLeft;
7430    mRightVolume = newRight;
7431
7432    // second get volume update from volume controller
7433    if (ctrlIdx >= 0) {
7434        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7435        mNewLeftVolume = newLeft;
7436        mNewRightVolume = newRight;
7437    }
7438    // then indicate volume to all other effects in chain.
7439    // Pass altered volume to effects before volume controller
7440    // and requested volume to effects after controller
7441    uint32_t lVol = newLeft;
7442    uint32_t rVol = newRight;
7443
7444    for (size_t i = 0; i < size; i++) {
7445        if ((int)i == ctrlIdx) continue;
7446        // this also works for ctrlIdx == -1 when there is no volume controller
7447        if ((int)i > ctrlIdx) {
7448            lVol = *left;
7449            rVol = *right;
7450        }
7451        mEffects[i]->setVolume(&lVol, &rVol, false);
7452    }
7453    *left = newLeft;
7454    *right = newRight;
7455
7456    return hasControl;
7457}
7458
7459status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7460{
7461    const size_t SIZE = 256;
7462    char buffer[SIZE];
7463    String8 result;
7464
7465    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7466    result.append(buffer);
7467
7468    bool locked = tryLock(mLock);
7469    // failed to lock - AudioFlinger is probably deadlocked
7470    if (!locked) {
7471        result.append("\tCould not lock mutex:\n");
7472    }
7473
7474    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7475    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7476            mEffects.size(),
7477            (uint32_t)mInBuffer,
7478            (uint32_t)mOutBuffer,
7479            mActiveTrackCnt);
7480    result.append(buffer);
7481    write(fd, result.string(), result.size());
7482
7483    for (size_t i = 0; i < mEffects.size(); ++i) {
7484        sp<EffectModule> effect = mEffects[i];
7485        if (effect != 0) {
7486            effect->dump(fd, args);
7487        }
7488    }
7489
7490    if (locked) {
7491        mLock.unlock();
7492    }
7493
7494    return NO_ERROR;
7495}
7496
7497// must be called with ThreadBase::mLock held
7498void AudioFlinger::EffectChain::setEffectSuspended_l(
7499        const effect_uuid_t *type, bool suspend)
7500{
7501    sp<SuspendedEffectDesc> desc;
7502    // use effect type UUID timelow as key as there is no real risk of identical
7503    // timeLow fields among effect type UUIDs.
7504    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7505    if (suspend) {
7506        if (index >= 0) {
7507            desc = mSuspendedEffects.valueAt(index);
7508        } else {
7509            desc = new SuspendedEffectDesc();
7510            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7511            mSuspendedEffects.add(type->timeLow, desc);
7512            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7513        }
7514        if (desc->mRefCount++ == 0) {
7515            sp<EffectModule> effect = getEffectIfEnabled(type);
7516            if (effect != 0) {
7517                desc->mEffect = effect;
7518                effect->setSuspended(true);
7519                effect->setEnabled(false);
7520            }
7521        }
7522    } else {
7523        if (index < 0) {
7524            return;
7525        }
7526        desc = mSuspendedEffects.valueAt(index);
7527        if (desc->mRefCount <= 0) {
7528            ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7529            desc->mRefCount = 1;
7530        }
7531        if (--desc->mRefCount == 0) {
7532            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7533            if (desc->mEffect != 0) {
7534                sp<EffectModule> effect = desc->mEffect.promote();
7535                if (effect != 0) {
7536                    effect->setSuspended(false);
7537                    sp<EffectHandle> handle = effect->controlHandle();
7538                    if (handle != 0) {
7539                        effect->setEnabled(handle->enabled());
7540                    }
7541                }
7542                desc->mEffect.clear();
7543            }
7544            mSuspendedEffects.removeItemsAt(index);
7545        }
7546    }
7547}
7548
7549// must be called with ThreadBase::mLock held
7550void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7551{
7552    sp<SuspendedEffectDesc> desc;
7553
7554    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7555    if (suspend) {
7556        if (index >= 0) {
7557            desc = mSuspendedEffects.valueAt(index);
7558        } else {
7559            desc = new SuspendedEffectDesc();
7560            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7561            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7562        }
7563        if (desc->mRefCount++ == 0) {
7564            Vector< sp<EffectModule> > effects;
7565            getSuspendEligibleEffects(effects);
7566            for (size_t i = 0; i < effects.size(); i++) {
7567                setEffectSuspended_l(&effects[i]->desc().type, true);
7568            }
7569        }
7570    } else {
7571        if (index < 0) {
7572            return;
7573        }
7574        desc = mSuspendedEffects.valueAt(index);
7575        if (desc->mRefCount <= 0) {
7576            ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7577            desc->mRefCount = 1;
7578        }
7579        if (--desc->mRefCount == 0) {
7580            Vector<const effect_uuid_t *> types;
7581            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7582                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7583                    continue;
7584                }
7585                types.add(&mSuspendedEffects.valueAt(i)->mType);
7586            }
7587            for (size_t i = 0; i < types.size(); i++) {
7588                setEffectSuspended_l(types[i], false);
7589            }
7590            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7591            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7592        }
7593    }
7594}
7595
7596
7597// The volume effect is used for automated tests only
7598#ifndef OPENSL_ES_H_
7599static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7600                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7601const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7602#endif //OPENSL_ES_H_
7603
7604bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7605{
7606    // auxiliary effects and visualizer are never suspended on output mix
7607    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7608        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7609         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7610         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7611        return false;
7612    }
7613    return true;
7614}
7615
7616void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects)
7617{
7618    effects.clear();
7619    for (size_t i = 0; i < mEffects.size(); i++) {
7620        if (isEffectEligibleForSuspend(mEffects[i]->desc())) {
7621            effects.add(mEffects[i]);
7622        }
7623    }
7624}
7625
7626sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7627                                                            const effect_uuid_t *type)
7628{
7629    sp<EffectModule> effect;
7630    effect = getEffectFromType_l(type);
7631    if (effect != 0 && !effect->isEnabled()) {
7632        effect.clear();
7633    }
7634    return effect;
7635}
7636
7637void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7638                                                            bool enabled)
7639{
7640    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7641    if (enabled) {
7642        if (index < 0) {
7643            // if the effect is not suspend check if all effects are suspended
7644            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7645            if (index < 0) {
7646                return;
7647            }
7648            if (!isEffectEligibleForSuspend(effect->desc())) {
7649                return;
7650            }
7651            setEffectSuspended_l(&effect->desc().type, enabled);
7652            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7653            if (index < 0) {
7654                ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7655                return;
7656            }
7657        }
7658        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7659             effect->desc().type.timeLow);
7660        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7661        // if effect is requested to suspended but was not yet enabled, supend it now.
7662        if (desc->mEffect == 0) {
7663            desc->mEffect = effect;
7664            effect->setEnabled(false);
7665            effect->setSuspended(true);
7666        }
7667    } else {
7668        if (index < 0) {
7669            return;
7670        }
7671        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7672             effect->desc().type.timeLow);
7673        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7674        desc->mEffect.clear();
7675        effect->setSuspended(false);
7676    }
7677}
7678
7679#undef LOG_TAG
7680#define LOG_TAG "AudioFlinger"
7681
7682// ----------------------------------------------------------------------------
7683
7684status_t AudioFlinger::onTransact(
7685        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7686{
7687    return BnAudioFlinger::onTransact(code, data, reply, flags);
7688}
7689
7690}; // namespace android
7691