AudioFlinger.cpp revision 3856b090cd04ba5dd4a59a12430ed724d5995909
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include <math.h>
23#include <signal.h>
24#include <sys/time.h>
25#include <sys/resource.h>
26
27#include <binder/IPCThreadState.h>
28#include <binder/IServiceManager.h>
29#include <utils/Log.h>
30#include <binder/Parcel.h>
31#include <binder/IPCThreadState.h>
32#include <utils/String16.h>
33#include <utils/threads.h>
34#include <utils/Atomic.h>
35
36#include <cutils/bitops.h>
37#include <cutils/properties.h>
38
39#include <media/AudioTrack.h>
40#include <media/AudioRecord.h>
41#include <media/IMediaPlayerService.h>
42
43#include <private/media/AudioTrackShared.h>
44#include <private/media/AudioEffectShared.h>
45
46#include <system/audio.h>
47#include <hardware/audio.h>
48
49#include "AudioMixer.h"
50#include "AudioFlinger.h"
51
52#include <media/EffectsFactoryApi.h>
53#include <audio_effects/effect_visualizer.h>
54#include <audio_effects/effect_ns.h>
55#include <audio_effects/effect_aec.h>
56
57#include <cpustats/ThreadCpuUsage.h>
58#include <powermanager/PowerManager.h>
59// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
60
61// ----------------------------------------------------------------------------
62
63
64namespace android {
65
66static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
67static const char* kHardwareLockedString = "Hardware lock is taken\n";
68
69//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
70static const float MAX_GAIN = 4096.0f;
71static const float MAX_GAIN_INT = 0x1000;
72
73// retry counts for buffer fill timeout
74// 50 * ~20msecs = 1 second
75static const int8_t kMaxTrackRetries = 50;
76static const int8_t kMaxTrackStartupRetries = 50;
77// allow less retry attempts on direct output thread.
78// direct outputs can be a scarce resource in audio hardware and should
79// be released as quickly as possible.
80static const int8_t kMaxTrackRetriesDirect = 2;
81
82static const int kDumpLockRetries = 50;
83static const int kDumpLockSleep = 20000;
84
85static const nsecs_t kWarningThrottle = seconds(5);
86
87// RecordThread loop sleep time upon application overrun or audio HAL read error
88static const int kRecordThreadSleepUs = 5000;
89
90static const nsecs_t kSetParametersTimeout = seconds(2);
91
92// ----------------------------------------------------------------------------
93
94static bool recordingAllowed() {
95    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
96    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
97    if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
98    return ok;
99}
100
101static bool settingsAllowed() {
102    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
103    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
104    if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
105    return ok;
106}
107
108// To collect the amplifier usage
109static void addBatteryData(uint32_t params) {
110    sp<IBinder> binder =
111        defaultServiceManager()->getService(String16("media.player"));
112    sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder);
113    if (service.get() == NULL) {
114        LOGW("Cannot connect to the MediaPlayerService for battery tracking");
115        return;
116    }
117
118    service->addBatteryData(params);
119}
120
121static int load_audio_interface(const char *if_name, const hw_module_t **mod,
122                                audio_hw_device_t **dev)
123{
124    int rc;
125
126    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
127    if (rc)
128        goto out;
129
130    rc = audio_hw_device_open(*mod, dev);
131    LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
132            AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
133    if (rc)
134        goto out;
135
136    return 0;
137
138out:
139    *mod = NULL;
140    *dev = NULL;
141    return rc;
142}
143
144static const char *audio_interfaces[] = {
145    "primary",
146    "a2dp",
147    "usb",
148};
149#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
150
151// ----------------------------------------------------------------------------
152
153AudioFlinger::AudioFlinger()
154    : BnAudioFlinger(),
155        mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
156        mBtNrecIsOff(false)
157{
158}
159
160void AudioFlinger::onFirstRef()
161{
162    int rc = 0;
163
164    Mutex::Autolock _l(mLock);
165
166    /* TODO: move all this work into an Init() function */
167    mHardwareStatus = AUDIO_HW_IDLE;
168
169    for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
170        const hw_module_t *mod;
171        audio_hw_device_t *dev;
172
173        rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
174        if (rc)
175            continue;
176
177        LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
178             mod->name, mod->id);
179        mAudioHwDevs.push(dev);
180
181        if (!mPrimaryHardwareDev) {
182            mPrimaryHardwareDev = dev;
183            LOGI("Using '%s' (%s.%s) as the primary audio interface",
184                 mod->name, mod->id, audio_interfaces[i]);
185        }
186    }
187
188    mHardwareStatus = AUDIO_HW_INIT;
189
190    if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
191        LOGE("Primary audio interface not found");
192        return;
193    }
194
195    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
196        audio_hw_device_t *dev = mAudioHwDevs[i];
197
198        mHardwareStatus = AUDIO_HW_INIT;
199        rc = dev->init_check(dev);
200        if (rc == 0) {
201            AutoMutex lock(mHardwareLock);
202
203            mMode = AUDIO_MODE_NORMAL;
204            mHardwareStatus = AUDIO_HW_SET_MODE;
205            dev->set_mode(dev, mMode);
206            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
207            dev->set_master_volume(dev, 1.0f);
208            mHardwareStatus = AUDIO_HW_IDLE;
209        }
210    }
211}
212
213status_t AudioFlinger::initCheck() const
214{
215    Mutex::Autolock _l(mLock);
216    if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
217        return NO_INIT;
218    return NO_ERROR;
219}
220
221AudioFlinger::~AudioFlinger()
222{
223    int num_devs = mAudioHwDevs.size();
224
225    while (!mRecordThreads.isEmpty()) {
226        // closeInput() will remove first entry from mRecordThreads
227        closeInput(mRecordThreads.keyAt(0));
228    }
229    while (!mPlaybackThreads.isEmpty()) {
230        // closeOutput() will remove first entry from mPlaybackThreads
231        closeOutput(mPlaybackThreads.keyAt(0));
232    }
233
234    for (int i = 0; i < num_devs; i++) {
235        audio_hw_device_t *dev = mAudioHwDevs[i];
236        audio_hw_device_close(dev);
237    }
238    mAudioHwDevs.clear();
239}
240
241audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
242{
243    /* first matching HW device is returned */
244    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
245        audio_hw_device_t *dev = mAudioHwDevs[i];
246        if ((dev->get_supported_devices(dev) & devices) == devices)
247            return dev;
248    }
249    return NULL;
250}
251
252status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
253{
254    const size_t SIZE = 256;
255    char buffer[SIZE];
256    String8 result;
257
258    result.append("Clients:\n");
259    for (size_t i = 0; i < mClients.size(); ++i) {
260        wp<Client> wClient = mClients.valueAt(i);
261        if (wClient != 0) {
262            sp<Client> client = wClient.promote();
263            if (client != 0) {
264                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
265                result.append(buffer);
266            }
267        }
268    }
269
270    result.append("Global session refs:\n");
271    result.append(" session pid cnt\n");
272    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
273        AudioSessionRef *r = mAudioSessionRefs[i];
274        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
275        result.append(buffer);
276    }
277    write(fd, result.string(), result.size());
278    return NO_ERROR;
279}
280
281
282status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
283{
284    const size_t SIZE = 256;
285    char buffer[SIZE];
286    String8 result;
287    int hardwareStatus = mHardwareStatus;
288
289    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
290    result.append(buffer);
291    write(fd, result.string(), result.size());
292    return NO_ERROR;
293}
294
295status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
296{
297    const size_t SIZE = 256;
298    char buffer[SIZE];
299    String8 result;
300    snprintf(buffer, SIZE, "Permission Denial: "
301            "can't dump AudioFlinger from pid=%d, uid=%d\n",
302            IPCThreadState::self()->getCallingPid(),
303            IPCThreadState::self()->getCallingUid());
304    result.append(buffer);
305    write(fd, result.string(), result.size());
306    return NO_ERROR;
307}
308
309static bool tryLock(Mutex& mutex)
310{
311    bool locked = false;
312    for (int i = 0; i < kDumpLockRetries; ++i) {
313        if (mutex.tryLock() == NO_ERROR) {
314            locked = true;
315            break;
316        }
317        usleep(kDumpLockSleep);
318    }
319    return locked;
320}
321
322status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
323{
324    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
325        dumpPermissionDenial(fd, args);
326    } else {
327        // get state of hardware lock
328        bool hardwareLocked = tryLock(mHardwareLock);
329        if (!hardwareLocked) {
330            String8 result(kHardwareLockedString);
331            write(fd, result.string(), result.size());
332        } else {
333            mHardwareLock.unlock();
334        }
335
336        bool locked = tryLock(mLock);
337
338        // failed to lock - AudioFlinger is probably deadlocked
339        if (!locked) {
340            String8 result(kDeadlockedString);
341            write(fd, result.string(), result.size());
342        }
343
344        dumpClients(fd, args);
345        dumpInternals(fd, args);
346
347        // dump playback threads
348        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
349            mPlaybackThreads.valueAt(i)->dump(fd, args);
350        }
351
352        // dump record threads
353        for (size_t i = 0; i < mRecordThreads.size(); i++) {
354            mRecordThreads.valueAt(i)->dump(fd, args);
355        }
356
357        // dump all hardware devs
358        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
359            audio_hw_device_t *dev = mAudioHwDevs[i];
360            dev->dump(dev, fd);
361        }
362        if (locked) mLock.unlock();
363    }
364    return NO_ERROR;
365}
366
367
368// IAudioFlinger interface
369
370
371sp<IAudioTrack> AudioFlinger::createTrack(
372        pid_t pid,
373        int streamType,
374        uint32_t sampleRate,
375        uint32_t format,
376        uint32_t channelMask,
377        int frameCount,
378        uint32_t flags,
379        const sp<IMemory>& sharedBuffer,
380        int output,
381        int *sessionId,
382        status_t *status)
383{
384    sp<PlaybackThread::Track> track;
385    sp<TrackHandle> trackHandle;
386    sp<Client> client;
387    wp<Client> wclient;
388    status_t lStatus;
389    int lSessionId;
390
391    if (streamType >= AUDIO_STREAM_CNT) {
392        LOGE("invalid stream type");
393        lStatus = BAD_VALUE;
394        goto Exit;
395    }
396
397    {
398        Mutex::Autolock _l(mLock);
399        PlaybackThread *thread = checkPlaybackThread_l(output);
400        PlaybackThread *effectThread = NULL;
401        if (thread == NULL) {
402            LOGE("unknown output thread");
403            lStatus = BAD_VALUE;
404            goto Exit;
405        }
406
407        wclient = mClients.valueFor(pid);
408
409        if (wclient != NULL) {
410            client = wclient.promote();
411        } else {
412            client = new Client(this, pid);
413            mClients.add(pid, client);
414        }
415
416        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
417        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
418            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
419                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
420                if (mPlaybackThreads.keyAt(i) != output) {
421                    // prevent same audio session on different output threads
422                    uint32_t sessions = t->hasAudioSession(*sessionId);
423                    if (sessions & PlaybackThread::TRACK_SESSION) {
424                        lStatus = BAD_VALUE;
425                        goto Exit;
426                    }
427                    // check if an effect with same session ID is waiting for a track to be created
428                    if (sessions & PlaybackThread::EFFECT_SESSION) {
429                        effectThread = t.get();
430                    }
431                }
432            }
433            lSessionId = *sessionId;
434        } else {
435            // if no audio session id is provided, create one here
436            lSessionId = nextUniqueId();
437            if (sessionId != NULL) {
438                *sessionId = lSessionId;
439            }
440        }
441        ALOGV("createTrack() lSessionId: %d", lSessionId);
442
443        track = thread->createTrack_l(client, streamType, sampleRate, format,
444                channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
445
446        // move effect chain to this output thread if an effect on same session was waiting
447        // for a track to be created
448        if (lStatus == NO_ERROR && effectThread != NULL) {
449            Mutex::Autolock _dl(thread->mLock);
450            Mutex::Autolock _sl(effectThread->mLock);
451            moveEffectChain_l(lSessionId, effectThread, thread, true);
452        }
453    }
454    if (lStatus == NO_ERROR) {
455        trackHandle = new TrackHandle(track);
456    } else {
457        // remove local strong reference to Client before deleting the Track so that the Client
458        // destructor is called by the TrackBase destructor with mLock held
459        client.clear();
460        track.clear();
461    }
462
463Exit:
464    if(status) {
465        *status = lStatus;
466    }
467    return trackHandle;
468}
469
470uint32_t AudioFlinger::sampleRate(int output) const
471{
472    Mutex::Autolock _l(mLock);
473    PlaybackThread *thread = checkPlaybackThread_l(output);
474    if (thread == NULL) {
475        LOGW("sampleRate() unknown thread %d", output);
476        return 0;
477    }
478    return thread->sampleRate();
479}
480
481int AudioFlinger::channelCount(int output) const
482{
483    Mutex::Autolock _l(mLock);
484    PlaybackThread *thread = checkPlaybackThread_l(output);
485    if (thread == NULL) {
486        LOGW("channelCount() unknown thread %d", output);
487        return 0;
488    }
489    return thread->channelCount();
490}
491
492uint32_t AudioFlinger::format(int output) const
493{
494    Mutex::Autolock _l(mLock);
495    PlaybackThread *thread = checkPlaybackThread_l(output);
496    if (thread == NULL) {
497        LOGW("format() unknown thread %d", output);
498        return 0;
499    }
500    return thread->format();
501}
502
503size_t AudioFlinger::frameCount(int output) const
504{
505    Mutex::Autolock _l(mLock);
506    PlaybackThread *thread = checkPlaybackThread_l(output);
507    if (thread == NULL) {
508        LOGW("frameCount() unknown thread %d", output);
509        return 0;
510    }
511    return thread->frameCount();
512}
513
514uint32_t AudioFlinger::latency(int output) const
515{
516    Mutex::Autolock _l(mLock);
517    PlaybackThread *thread = checkPlaybackThread_l(output);
518    if (thread == NULL) {
519        LOGW("latency() unknown thread %d", output);
520        return 0;
521    }
522    return thread->latency();
523}
524
525status_t AudioFlinger::setMasterVolume(float value)
526{
527    status_t ret = initCheck();
528    if (ret != NO_ERROR) {
529        return ret;
530    }
531
532    // check calling permissions
533    if (!settingsAllowed()) {
534        return PERMISSION_DENIED;
535    }
536
537    // when hw supports master volume, don't scale in sw mixer
538    { // scope for the lock
539        AutoMutex lock(mHardwareLock);
540        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
541        if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
542            value = 1.0f;
543        }
544        mHardwareStatus = AUDIO_HW_IDLE;
545    }
546
547    Mutex::Autolock _l(mLock);
548    mMasterVolume = value;
549    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
550       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
551
552    return NO_ERROR;
553}
554
555status_t AudioFlinger::setMode(int mode)
556{
557    status_t ret = initCheck();
558    if (ret != NO_ERROR) {
559        return ret;
560    }
561
562    // check calling permissions
563    if (!settingsAllowed()) {
564        return PERMISSION_DENIED;
565    }
566    if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) {
567        LOGW("Illegal value: setMode(%d)", mode);
568        return BAD_VALUE;
569    }
570
571    { // scope for the lock
572        AutoMutex lock(mHardwareLock);
573        mHardwareStatus = AUDIO_HW_SET_MODE;
574        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
575        mHardwareStatus = AUDIO_HW_IDLE;
576    }
577
578    if (NO_ERROR == ret) {
579        Mutex::Autolock _l(mLock);
580        mMode = mode;
581        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
582           mPlaybackThreads.valueAt(i)->setMode(mode);
583    }
584
585    return ret;
586}
587
588status_t AudioFlinger::setMicMute(bool state)
589{
590    status_t ret = initCheck();
591    if (ret != NO_ERROR) {
592        return ret;
593    }
594
595    // check calling permissions
596    if (!settingsAllowed()) {
597        return PERMISSION_DENIED;
598    }
599
600    AutoMutex lock(mHardwareLock);
601    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
602    ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
603    mHardwareStatus = AUDIO_HW_IDLE;
604    return ret;
605}
606
607bool AudioFlinger::getMicMute() const
608{
609    status_t ret = initCheck();
610    if (ret != NO_ERROR) {
611        return false;
612    }
613
614    bool state = AUDIO_MODE_INVALID;
615    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
616    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
617    mHardwareStatus = AUDIO_HW_IDLE;
618    return state;
619}
620
621status_t AudioFlinger::setMasterMute(bool muted)
622{
623    // check calling permissions
624    if (!settingsAllowed()) {
625        return PERMISSION_DENIED;
626    }
627
628    Mutex::Autolock _l(mLock);
629    mMasterMute = muted;
630    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
631       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
632
633    return NO_ERROR;
634}
635
636float AudioFlinger::masterVolume() const
637{
638    return mMasterVolume;
639}
640
641bool AudioFlinger::masterMute() const
642{
643    return mMasterMute;
644}
645
646status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
647{
648    // check calling permissions
649    if (!settingsAllowed()) {
650        return PERMISSION_DENIED;
651    }
652
653    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
654        return BAD_VALUE;
655    }
656
657    AutoMutex lock(mLock);
658    PlaybackThread *thread = NULL;
659    if (output) {
660        thread = checkPlaybackThread_l(output);
661        if (thread == NULL) {
662            return BAD_VALUE;
663        }
664    }
665
666    mStreamTypes[stream].volume = value;
667
668    if (thread == NULL) {
669        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
670           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
671        }
672    } else {
673        thread->setStreamVolume(stream, value);
674    }
675
676    return NO_ERROR;
677}
678
679status_t AudioFlinger::setStreamMute(int stream, bool muted)
680{
681    // check calling permissions
682    if (!settingsAllowed()) {
683        return PERMISSION_DENIED;
684    }
685
686    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
687        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
688        return BAD_VALUE;
689    }
690
691    AutoMutex lock(mLock);
692    mStreamTypes[stream].mute = muted;
693    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
694       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
695
696    return NO_ERROR;
697}
698
699float AudioFlinger::streamVolume(int stream, int output) const
700{
701    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
702        return 0.0f;
703    }
704
705    AutoMutex lock(mLock);
706    float volume;
707    if (output) {
708        PlaybackThread *thread = checkPlaybackThread_l(output);
709        if (thread == NULL) {
710            return 0.0f;
711        }
712        volume = thread->streamVolume(stream);
713    } else {
714        volume = mStreamTypes[stream].volume;
715    }
716
717    return volume;
718}
719
720bool AudioFlinger::streamMute(int stream) const
721{
722    if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
723        return true;
724    }
725
726    return mStreamTypes[stream].mute;
727}
728
729status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
730{
731    status_t result;
732
733    ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
734            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
735    // check calling permissions
736    if (!settingsAllowed()) {
737        return PERMISSION_DENIED;
738    }
739
740    // ioHandle == 0 means the parameters are global to the audio hardware interface
741    if (ioHandle == 0) {
742        AutoMutex lock(mHardwareLock);
743        mHardwareStatus = AUDIO_SET_PARAMETER;
744        status_t final_result = NO_ERROR;
745        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
746            audio_hw_device_t *dev = mAudioHwDevs[i];
747            result = dev->set_parameters(dev, keyValuePairs.string());
748            final_result = result ?: final_result;
749        }
750        mHardwareStatus = AUDIO_HW_IDLE;
751        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
752        AudioParameter param = AudioParameter(keyValuePairs);
753        String8 value;
754        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
755            Mutex::Autolock _l(mLock);
756            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
757            if (mBtNrecIsOff != btNrecIsOff) {
758                for (size_t i = 0; i < mRecordThreads.size(); i++) {
759                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
760                    RecordThread::RecordTrack *track = thread->track();
761                    if (track != NULL) {
762                        audio_devices_t device = (audio_devices_t)(
763                                thread->device() & AUDIO_DEVICE_IN_ALL);
764                        bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
765                        thread->setEffectSuspended(FX_IID_AEC,
766                                                   suspend,
767                                                   track->sessionId());
768                        thread->setEffectSuspended(FX_IID_NS,
769                                                   suspend,
770                                                   track->sessionId());
771                    }
772                }
773                mBtNrecIsOff = btNrecIsOff;
774            }
775        }
776        return final_result;
777    }
778
779    // hold a strong ref on thread in case closeOutput() or closeInput() is called
780    // and the thread is exited once the lock is released
781    sp<ThreadBase> thread;
782    {
783        Mutex::Autolock _l(mLock);
784        thread = checkPlaybackThread_l(ioHandle);
785        if (thread == NULL) {
786            thread = checkRecordThread_l(ioHandle);
787        } else if (thread.get() == primaryPlaybackThread_l()) {
788            // indicate output device change to all input threads for pre processing
789            AudioParameter param = AudioParameter(keyValuePairs);
790            int value;
791            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
792                for (size_t i = 0; i < mRecordThreads.size(); i++) {
793                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
794                }
795            }
796        }
797    }
798    if (thread != NULL) {
799        result = thread->setParameters(keyValuePairs);
800        return result;
801    }
802    return BAD_VALUE;
803}
804
805String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
806{
807//    ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
808//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
809
810    if (ioHandle == 0) {
811        String8 out_s8;
812
813        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
814            audio_hw_device_t *dev = mAudioHwDevs[i];
815            char *s = dev->get_parameters(dev, keys.string());
816            out_s8 += String8(s);
817            free(s);
818        }
819        return out_s8;
820    }
821
822    Mutex::Autolock _l(mLock);
823
824    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
825    if (playbackThread != NULL) {
826        return playbackThread->getParameters(keys);
827    }
828    RecordThread *recordThread = checkRecordThread_l(ioHandle);
829    if (recordThread != NULL) {
830        return recordThread->getParameters(keys);
831    }
832    return String8("");
833}
834
835size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
836{
837    status_t ret = initCheck();
838    if (ret != NO_ERROR) {
839        return 0;
840    }
841
842    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
843}
844
845unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
846{
847    if (ioHandle == 0) {
848        return 0;
849    }
850
851    Mutex::Autolock _l(mLock);
852
853    RecordThread *recordThread = checkRecordThread_l(ioHandle);
854    if (recordThread != NULL) {
855        return recordThread->getInputFramesLost();
856    }
857    return 0;
858}
859
860status_t AudioFlinger::setVoiceVolume(float value)
861{
862    status_t ret = initCheck();
863    if (ret != NO_ERROR) {
864        return ret;
865    }
866
867    // check calling permissions
868    if (!settingsAllowed()) {
869        return PERMISSION_DENIED;
870    }
871
872    AutoMutex lock(mHardwareLock);
873    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
874    ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
875    mHardwareStatus = AUDIO_HW_IDLE;
876
877    return ret;
878}
879
880status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
881{
882    status_t status;
883
884    Mutex::Autolock _l(mLock);
885
886    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
887    if (playbackThread != NULL) {
888        return playbackThread->getRenderPosition(halFrames, dspFrames);
889    }
890
891    return BAD_VALUE;
892}
893
894void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
895{
896
897    Mutex::Autolock _l(mLock);
898
899    int pid = IPCThreadState::self()->getCallingPid();
900    if (mNotificationClients.indexOfKey(pid) < 0) {
901        sp<NotificationClient> notificationClient = new NotificationClient(this,
902                                                                            client,
903                                                                            pid);
904        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
905
906        mNotificationClients.add(pid, notificationClient);
907
908        sp<IBinder> binder = client->asBinder();
909        binder->linkToDeath(notificationClient);
910
911        // the config change is always sent from playback or record threads to avoid deadlock
912        // with AudioSystem::gLock
913        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
914            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
915        }
916
917        for (size_t i = 0; i < mRecordThreads.size(); i++) {
918            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
919        }
920    }
921}
922
923void AudioFlinger::removeNotificationClient(pid_t pid)
924{
925    Mutex::Autolock _l(mLock);
926
927    int index = mNotificationClients.indexOfKey(pid);
928    if (index >= 0) {
929        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
930        ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
931        mNotificationClients.removeItem(pid);
932    }
933
934    ALOGV("%d died, releasing its sessions", pid);
935    int num = mAudioSessionRefs.size();
936    bool removed = false;
937    for (int i = 0; i< num; i++) {
938        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
939        ALOGV(" pid %d @ %d", ref->pid, i);
940        if (ref->pid == pid) {
941            ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
942            mAudioSessionRefs.removeAt(i);
943            delete ref;
944            removed = true;
945            i--;
946            num--;
947        }
948    }
949    if (removed) {
950        purgeStaleEffects_l();
951    }
952}
953
954// audioConfigChanged_l() must be called with AudioFlinger::mLock held
955void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
956{
957    size_t size = mNotificationClients.size();
958    for (size_t i = 0; i < size; i++) {
959        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
960    }
961}
962
963// removeClient_l() must be called with AudioFlinger::mLock held
964void AudioFlinger::removeClient_l(pid_t pid)
965{
966    ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
967    mClients.removeItem(pid);
968}
969
970
971// ----------------------------------------------------------------------------
972
973AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
974    :   Thread(false),
975        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
976        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
977        mDevice(device)
978{
979    mDeathRecipient = new PMDeathRecipient(this);
980}
981
982AudioFlinger::ThreadBase::~ThreadBase()
983{
984    mParamCond.broadcast();
985    mNewParameters.clear();
986    // do not lock the mutex in destructor
987    releaseWakeLock_l();
988    if (mPowerManager != 0) {
989        sp<IBinder> binder = mPowerManager->asBinder();
990        binder->unlinkToDeath(mDeathRecipient);
991    }
992}
993
994void AudioFlinger::ThreadBase::exit()
995{
996    // keep a strong ref on ourself so that we wont get
997    // destroyed in the middle of requestExitAndWait()
998    sp <ThreadBase> strongMe = this;
999
1000    ALOGV("ThreadBase::exit");
1001    {
1002        AutoMutex lock(&mLock);
1003        mExiting = true;
1004        requestExit();
1005        mWaitWorkCV.signal();
1006    }
1007    requestExitAndWait();
1008}
1009
1010uint32_t AudioFlinger::ThreadBase::sampleRate() const
1011{
1012    return mSampleRate;
1013}
1014
1015int AudioFlinger::ThreadBase::channelCount() const
1016{
1017    return (int)mChannelCount;
1018}
1019
1020uint32_t AudioFlinger::ThreadBase::format() const
1021{
1022    return mFormat;
1023}
1024
1025size_t AudioFlinger::ThreadBase::frameCount() const
1026{
1027    return mFrameCount;
1028}
1029
1030status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1031{
1032    status_t status;
1033
1034    ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1035    Mutex::Autolock _l(mLock);
1036
1037    mNewParameters.add(keyValuePairs);
1038    mWaitWorkCV.signal();
1039    // wait condition with timeout in case the thread loop has exited
1040    // before the request could be processed
1041    if (mParamCond.waitRelative(mLock, kSetParametersTimeout) == NO_ERROR) {
1042        status = mParamStatus;
1043        mWaitWorkCV.signal();
1044    } else {
1045        status = TIMED_OUT;
1046    }
1047    return status;
1048}
1049
1050void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1051{
1052    Mutex::Autolock _l(mLock);
1053    sendConfigEvent_l(event, param);
1054}
1055
1056// sendConfigEvent_l() must be called with ThreadBase::mLock held
1057void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1058{
1059    ConfigEvent *configEvent = new ConfigEvent();
1060    configEvent->mEvent = event;
1061    configEvent->mParam = param;
1062    mConfigEvents.add(configEvent);
1063    ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1064    mWaitWorkCV.signal();
1065}
1066
1067void AudioFlinger::ThreadBase::processConfigEvents()
1068{
1069    mLock.lock();
1070    while(!mConfigEvents.isEmpty()) {
1071        ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1072        ConfigEvent *configEvent = mConfigEvents[0];
1073        mConfigEvents.removeAt(0);
1074        // release mLock before locking AudioFlinger mLock: lock order is always
1075        // AudioFlinger then ThreadBase to avoid cross deadlock
1076        mLock.unlock();
1077        mAudioFlinger->mLock.lock();
1078        audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
1079        mAudioFlinger->mLock.unlock();
1080        delete configEvent;
1081        mLock.lock();
1082    }
1083    mLock.unlock();
1084}
1085
1086status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1087{
1088    const size_t SIZE = 256;
1089    char buffer[SIZE];
1090    String8 result;
1091
1092    bool locked = tryLock(mLock);
1093    if (!locked) {
1094        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1095        write(fd, buffer, strlen(buffer));
1096    }
1097
1098    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1099    result.append(buffer);
1100    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1101    result.append(buffer);
1102    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1103    result.append(buffer);
1104    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1105    result.append(buffer);
1106    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1107    result.append(buffer);
1108    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1109    result.append(buffer);
1110    snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
1111    result.append(buffer);
1112
1113    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1114    result.append(buffer);
1115    result.append(" Index Command");
1116    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1117        snprintf(buffer, SIZE, "\n %02d    ", i);
1118        result.append(buffer);
1119        result.append(mNewParameters[i]);
1120    }
1121
1122    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1123    result.append(buffer);
1124    snprintf(buffer, SIZE, " Index event param\n");
1125    result.append(buffer);
1126    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1127        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
1128        result.append(buffer);
1129    }
1130    result.append("\n");
1131
1132    write(fd, result.string(), result.size());
1133
1134    if (locked) {
1135        mLock.unlock();
1136    }
1137    return NO_ERROR;
1138}
1139
1140status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1141{
1142    const size_t SIZE = 256;
1143    char buffer[SIZE];
1144    String8 result;
1145
1146    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1147    write(fd, buffer, strlen(buffer));
1148
1149    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1150        sp<EffectChain> chain = mEffectChains[i];
1151        if (chain != 0) {
1152            chain->dump(fd, args);
1153        }
1154    }
1155    return NO_ERROR;
1156}
1157
1158void AudioFlinger::ThreadBase::acquireWakeLock()
1159{
1160    Mutex::Autolock _l(mLock);
1161    acquireWakeLock_l();
1162}
1163
1164void AudioFlinger::ThreadBase::acquireWakeLock_l()
1165{
1166    if (mPowerManager == 0) {
1167        // use checkService() to avoid blocking if power service is not up yet
1168        sp<IBinder> binder =
1169            defaultServiceManager()->checkService(String16("power"));
1170        if (binder == 0) {
1171            LOGW("Thread %s cannot connect to the power manager service", mName);
1172        } else {
1173            mPowerManager = interface_cast<IPowerManager>(binder);
1174            binder->linkToDeath(mDeathRecipient);
1175        }
1176    }
1177    if (mPowerManager != 0) {
1178        sp<IBinder> binder = new BBinder();
1179        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
1180                                                         binder,
1181                                                         String16(mName));
1182        if (status == NO_ERROR) {
1183            mWakeLockToken = binder;
1184        }
1185        ALOGV("acquireWakeLock_l() %s status %d", mName, status);
1186    }
1187}
1188
1189void AudioFlinger::ThreadBase::releaseWakeLock()
1190{
1191    Mutex::Autolock _l(mLock);
1192    releaseWakeLock_l();
1193}
1194
1195void AudioFlinger::ThreadBase::releaseWakeLock_l()
1196{
1197    if (mWakeLockToken != 0) {
1198        ALOGV("releaseWakeLock_l() %s", mName);
1199        if (mPowerManager != 0) {
1200            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
1201        }
1202        mWakeLockToken.clear();
1203    }
1204}
1205
1206void AudioFlinger::ThreadBase::clearPowerManager()
1207{
1208    Mutex::Autolock _l(mLock);
1209    releaseWakeLock_l();
1210    mPowerManager.clear();
1211}
1212
1213void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
1214{
1215    sp<ThreadBase> thread = mThread.promote();
1216    if (thread != 0) {
1217        thread->clearPowerManager();
1218    }
1219    LOGW("power manager service died !!!");
1220}
1221
1222void AudioFlinger::ThreadBase::setEffectSuspended(
1223        const effect_uuid_t *type, bool suspend, int sessionId)
1224{
1225    Mutex::Autolock _l(mLock);
1226    setEffectSuspended_l(type, suspend, sessionId);
1227}
1228
1229void AudioFlinger::ThreadBase::setEffectSuspended_l(
1230        const effect_uuid_t *type, bool suspend, int sessionId)
1231{
1232    sp<EffectChain> chain;
1233    chain = getEffectChain_l(sessionId);
1234    if (chain != 0) {
1235        if (type != NULL) {
1236            chain->setEffectSuspended_l(type, suspend);
1237        } else {
1238            chain->setEffectSuspendedAll_l(suspend);
1239        }
1240    }
1241
1242    updateSuspendedSessions_l(type, suspend, sessionId);
1243}
1244
1245void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1246{
1247    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1248    if (index < 0) {
1249        return;
1250    }
1251
1252    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1253            mSuspendedSessions.editValueAt(index);
1254
1255    for (size_t i = 0; i < sessionEffects.size(); i++) {
1256        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1257        for (int j = 0; j < desc->mRefCount; j++) {
1258            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1259                chain->setEffectSuspendedAll_l(true);
1260            } else {
1261                ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1262                     desc->mType.timeLow);
1263                chain->setEffectSuspended_l(&desc->mType, true);
1264            }
1265        }
1266    }
1267}
1268
1269void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1270                                                         bool suspend,
1271                                                         int sessionId)
1272{
1273    int index = mSuspendedSessions.indexOfKey(sessionId);
1274
1275    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1276
1277    if (suspend) {
1278        if (index >= 0) {
1279            sessionEffects = mSuspendedSessions.editValueAt(index);
1280        } else {
1281            mSuspendedSessions.add(sessionId, sessionEffects);
1282        }
1283    } else {
1284        if (index < 0) {
1285            return;
1286        }
1287        sessionEffects = mSuspendedSessions.editValueAt(index);
1288    }
1289
1290
1291    int key = EffectChain::kKeyForSuspendAll;
1292    if (type != NULL) {
1293        key = type->timeLow;
1294    }
1295    index = sessionEffects.indexOfKey(key);
1296
1297    sp <SuspendedSessionDesc> desc;
1298    if (suspend) {
1299        if (index >= 0) {
1300            desc = sessionEffects.valueAt(index);
1301        } else {
1302            desc = new SuspendedSessionDesc();
1303            if (type != NULL) {
1304                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1305            }
1306            sessionEffects.add(key, desc);
1307            ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1308        }
1309        desc->mRefCount++;
1310    } else {
1311        if (index < 0) {
1312            return;
1313        }
1314        desc = sessionEffects.valueAt(index);
1315        if (--desc->mRefCount == 0) {
1316            ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1317            sessionEffects.removeItemsAt(index);
1318            if (sessionEffects.isEmpty()) {
1319                ALOGV("updateSuspendedSessions_l() restore removing session %d",
1320                                 sessionId);
1321                mSuspendedSessions.removeItem(sessionId);
1322            }
1323        }
1324    }
1325    if (!sessionEffects.isEmpty()) {
1326        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1327    }
1328}
1329
1330void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1331                                                            bool enabled,
1332                                                            int sessionId)
1333{
1334    Mutex::Autolock _l(mLock);
1335    checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1336}
1337
1338void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1339                                                            bool enabled,
1340                                                            int sessionId)
1341{
1342    if (mType != RECORD) {
1343        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1344        // another session. This gives the priority to well behaved effect control panels
1345        // and applications not using global effects.
1346        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1347            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1348        }
1349    }
1350
1351    sp<EffectChain> chain = getEffectChain_l(sessionId);
1352    if (chain != 0) {
1353        chain->checkSuspendOnEffectEnabled(effect, enabled);
1354    }
1355}
1356
1357// ----------------------------------------------------------------------------
1358
1359AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1360                                             AudioStreamOut* output,
1361                                             int id,
1362                                             uint32_t device)
1363    :   ThreadBase(audioFlinger, id, device),
1364        mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
1365        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1366{
1367    snprintf(mName, kNameLength, "AudioOut_%d", id);
1368
1369    readOutputParameters();
1370
1371    mMasterVolume = mAudioFlinger->masterVolume();
1372    mMasterMute = mAudioFlinger->masterMute();
1373
1374    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
1375        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1376        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1377        mStreamTypes[stream].valid = true;
1378    }
1379}
1380
1381AudioFlinger::PlaybackThread::~PlaybackThread()
1382{
1383    delete [] mMixBuffer;
1384}
1385
1386status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1387{
1388    dumpInternals(fd, args);
1389    dumpTracks(fd, args);
1390    dumpEffectChains(fd, args);
1391    return NO_ERROR;
1392}
1393
1394status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1395{
1396    const size_t SIZE = 256;
1397    char buffer[SIZE];
1398    String8 result;
1399
1400    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1401    result.append(buffer);
1402    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1403    for (size_t i = 0; i < mTracks.size(); ++i) {
1404        sp<Track> track = mTracks[i];
1405        if (track != 0) {
1406            track->dump(buffer, SIZE);
1407            result.append(buffer);
1408        }
1409    }
1410
1411    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1412    result.append(buffer);
1413    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1414    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1415        wp<Track> wTrack = mActiveTracks[i];
1416        if (wTrack != 0) {
1417            sp<Track> track = wTrack.promote();
1418            if (track != 0) {
1419                track->dump(buffer, SIZE);
1420                result.append(buffer);
1421            }
1422        }
1423    }
1424    write(fd, result.string(), result.size());
1425    return NO_ERROR;
1426}
1427
1428status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1429{
1430    const size_t SIZE = 256;
1431    char buffer[SIZE];
1432    String8 result;
1433
1434    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1435    result.append(buffer);
1436    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1437    result.append(buffer);
1438    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1439    result.append(buffer);
1440    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1441    result.append(buffer);
1442    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1443    result.append(buffer);
1444    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1445    result.append(buffer);
1446    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1447    result.append(buffer);
1448    write(fd, result.string(), result.size());
1449
1450    dumpBase(fd, args);
1451
1452    return NO_ERROR;
1453}
1454
1455// Thread virtuals
1456status_t AudioFlinger::PlaybackThread::readyToRun()
1457{
1458    status_t status = initCheck();
1459    if (status == NO_ERROR) {
1460        LOGI("AudioFlinger's thread %p ready to run", this);
1461    } else {
1462        LOGE("No working audio driver found.");
1463    }
1464    return status;
1465}
1466
1467void AudioFlinger::PlaybackThread::onFirstRef()
1468{
1469    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1470}
1471
1472// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1473sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1474        const sp<AudioFlinger::Client>& client,
1475        int streamType,
1476        uint32_t sampleRate,
1477        uint32_t format,
1478        uint32_t channelMask,
1479        int frameCount,
1480        const sp<IMemory>& sharedBuffer,
1481        int sessionId,
1482        status_t *status)
1483{
1484    sp<Track> track;
1485    status_t lStatus;
1486
1487    if (mType == DIRECT) {
1488        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1489            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1490                LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1491                        "for output %p with format %d",
1492                        sampleRate, format, channelMask, mOutput, mFormat);
1493                lStatus = BAD_VALUE;
1494                goto Exit;
1495            }
1496        }
1497    } else {
1498        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1499        if (sampleRate > mSampleRate*2) {
1500            LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1501            lStatus = BAD_VALUE;
1502            goto Exit;
1503        }
1504    }
1505
1506    lStatus = initCheck();
1507    if (lStatus != NO_ERROR) {
1508        LOGE("Audio driver not initialized.");
1509        goto Exit;
1510    }
1511
1512    { // scope for mLock
1513        Mutex::Autolock _l(mLock);
1514
1515        // all tracks in same audio session must share the same routing strategy otherwise
1516        // conflicts will happen when tracks are moved from one output to another by audio policy
1517        // manager
1518        uint32_t strategy =
1519                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1520        for (size_t i = 0; i < mTracks.size(); ++i) {
1521            sp<Track> t = mTracks[i];
1522            if (t != 0) {
1523                if (sessionId == t->sessionId() &&
1524                        strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
1525                    lStatus = BAD_VALUE;
1526                    goto Exit;
1527                }
1528            }
1529        }
1530
1531        track = new Track(this, client, streamType, sampleRate, format,
1532                channelMask, frameCount, sharedBuffer, sessionId);
1533        if (track->getCblk() == NULL || track->name() < 0) {
1534            lStatus = NO_MEMORY;
1535            goto Exit;
1536        }
1537        mTracks.add(track);
1538
1539        sp<EffectChain> chain = getEffectChain_l(sessionId);
1540        if (chain != 0) {
1541            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1542            track->setMainBuffer(chain->inBuffer());
1543            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1544            chain->incTrackCnt();
1545        }
1546
1547        // invalidate track immediately if the stream type was moved to another thread since
1548        // createTrack() was called by the client process.
1549        if (!mStreamTypes[streamType].valid) {
1550            LOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1551                 this, streamType);
1552            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1553        }
1554    }
1555    lStatus = NO_ERROR;
1556
1557Exit:
1558    if(status) {
1559        *status = lStatus;
1560    }
1561    return track;
1562}
1563
1564uint32_t AudioFlinger::PlaybackThread::latency() const
1565{
1566    Mutex::Autolock _l(mLock);
1567    if (initCheck() == NO_ERROR) {
1568        return mOutput->stream->get_latency(mOutput->stream);
1569    } else {
1570        return 0;
1571    }
1572}
1573
1574status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1575{
1576    mMasterVolume = value;
1577    return NO_ERROR;
1578}
1579
1580status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1581{
1582    mMasterMute = muted;
1583    return NO_ERROR;
1584}
1585
1586float AudioFlinger::PlaybackThread::masterVolume() const
1587{
1588    return mMasterVolume;
1589}
1590
1591bool AudioFlinger::PlaybackThread::masterMute() const
1592{
1593    return mMasterMute;
1594}
1595
1596status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
1597{
1598    mStreamTypes[stream].volume = value;
1599    return NO_ERROR;
1600}
1601
1602status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
1603{
1604    mStreamTypes[stream].mute = muted;
1605    return NO_ERROR;
1606}
1607
1608float AudioFlinger::PlaybackThread::streamVolume(int stream) const
1609{
1610    return mStreamTypes[stream].volume;
1611}
1612
1613bool AudioFlinger::PlaybackThread::streamMute(int stream) const
1614{
1615    return mStreamTypes[stream].mute;
1616}
1617
1618// addTrack_l() must be called with ThreadBase::mLock held
1619status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1620{
1621    status_t status = ALREADY_EXISTS;
1622
1623    // set retry count for buffer fill
1624    track->mRetryCount = kMaxTrackStartupRetries;
1625    if (mActiveTracks.indexOf(track) < 0) {
1626        // the track is newly added, make sure it fills up all its
1627        // buffers before playing. This is to ensure the client will
1628        // effectively get the latency it requested.
1629        track->mFillingUpStatus = Track::FS_FILLING;
1630        track->mResetDone = false;
1631        mActiveTracks.add(track);
1632        if (track->mainBuffer() != mMixBuffer) {
1633            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1634            if (chain != 0) {
1635                ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1636                chain->incActiveTrackCnt();
1637            }
1638        }
1639
1640        status = NO_ERROR;
1641    }
1642
1643    ALOGV("mWaitWorkCV.broadcast");
1644    mWaitWorkCV.broadcast();
1645
1646    return status;
1647}
1648
1649// destroyTrack_l() must be called with ThreadBase::mLock held
1650void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1651{
1652    track->mState = TrackBase::TERMINATED;
1653    if (mActiveTracks.indexOf(track) < 0) {
1654        removeTrack_l(track);
1655    }
1656}
1657
1658void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1659{
1660    mTracks.remove(track);
1661    deleteTrackName_l(track->name());
1662    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1663    if (chain != 0) {
1664        chain->decTrackCnt();
1665    }
1666}
1667
1668String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1669{
1670    String8 out_s8 = String8("");
1671    char *s;
1672
1673    Mutex::Autolock _l(mLock);
1674    if (initCheck() != NO_ERROR) {
1675        return out_s8;
1676    }
1677
1678    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1679    out_s8 = String8(s);
1680    free(s);
1681    return out_s8;
1682}
1683
1684// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1685void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1686    AudioSystem::OutputDescriptor desc;
1687    void *param2 = 0;
1688
1689    ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1690
1691    switch (event) {
1692    case AudioSystem::OUTPUT_OPENED:
1693    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1694        desc.channels = mChannelMask;
1695        desc.samplingRate = mSampleRate;
1696        desc.format = mFormat;
1697        desc.frameCount = mFrameCount;
1698        desc.latency = latency();
1699        param2 = &desc;
1700        break;
1701
1702    case AudioSystem::STREAM_CONFIG_CHANGED:
1703        param2 = &param;
1704    case AudioSystem::OUTPUT_CLOSED:
1705    default:
1706        break;
1707    }
1708    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1709}
1710
1711void AudioFlinger::PlaybackThread::readOutputParameters()
1712{
1713    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1714    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1715    mChannelCount = (uint16_t)popcount(mChannelMask);
1716    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1717    mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
1718    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1719
1720    // FIXME - Current mixer implementation only supports stereo output: Always
1721    // Allocate a stereo buffer even if HW output is mono.
1722    if (mMixBuffer != NULL) delete[] mMixBuffer;
1723    mMixBuffer = new int16_t[mFrameCount * 2];
1724    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1725
1726    // force reconfiguration of effect chains and engines to take new buffer size and audio
1727    // parameters into account
1728    // Note that mLock is not held when readOutputParameters() is called from the constructor
1729    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1730    // matter.
1731    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1732    Vector< sp<EffectChain> > effectChains = mEffectChains;
1733    for (size_t i = 0; i < effectChains.size(); i ++) {
1734        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1735    }
1736}
1737
1738status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1739{
1740    if (halFrames == 0 || dspFrames == 0) {
1741        return BAD_VALUE;
1742    }
1743    Mutex::Autolock _l(mLock);
1744    if (initCheck() != NO_ERROR) {
1745        return INVALID_OPERATION;
1746    }
1747    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1748
1749    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1750}
1751
1752uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1753{
1754    Mutex::Autolock _l(mLock);
1755    uint32_t result = 0;
1756    if (getEffectChain_l(sessionId) != 0) {
1757        result = EFFECT_SESSION;
1758    }
1759
1760    for (size_t i = 0; i < mTracks.size(); ++i) {
1761        sp<Track> track = mTracks[i];
1762        if (sessionId == track->sessionId() &&
1763                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1764            result |= TRACK_SESSION;
1765            break;
1766        }
1767    }
1768
1769    return result;
1770}
1771
1772uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1773{
1774    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1775    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1776    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1777        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1778    }
1779    for (size_t i = 0; i < mTracks.size(); i++) {
1780        sp<Track> track = mTracks[i];
1781        if (sessionId == track->sessionId() &&
1782                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1783            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1784        }
1785    }
1786    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1787}
1788
1789
1790AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1791{
1792    Mutex::Autolock _l(mLock);
1793    return mOutput;
1794}
1795
1796AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1797{
1798    Mutex::Autolock _l(mLock);
1799    AudioStreamOut *output = mOutput;
1800    mOutput = NULL;
1801    return output;
1802}
1803
1804// this method must always be called either with ThreadBase mLock held or inside the thread loop
1805audio_stream_t* AudioFlinger::PlaybackThread::stream()
1806{
1807    if (mOutput == NULL) {
1808        return NULL;
1809    }
1810    return &mOutput->stream->common;
1811}
1812
1813// ----------------------------------------------------------------------------
1814
1815AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1816    :   PlaybackThread(audioFlinger, output, id, device),
1817        mAudioMixer(0)
1818{
1819    mType = ThreadBase::MIXER;
1820    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1821
1822    // FIXME - Current mixer implementation only supports stereo output
1823    if (mChannelCount == 1) {
1824        LOGE("Invalid audio hardware channel count");
1825    }
1826}
1827
1828AudioFlinger::MixerThread::~MixerThread()
1829{
1830    delete mAudioMixer;
1831}
1832
1833bool AudioFlinger::MixerThread::threadLoop()
1834{
1835    Vector< sp<Track> > tracksToRemove;
1836    uint32_t mixerStatus = MIXER_IDLE;
1837    nsecs_t standbyTime = systemTime();
1838    size_t mixBufferSize = mFrameCount * mFrameSize;
1839    // FIXME: Relaxed timing because of a certain device that can't meet latency
1840    // Should be reduced to 2x after the vendor fixes the driver issue
1841    // increase threshold again due to low power audio mode. The way this warning threshold is
1842    // calculated and its usefulness should be reconsidered anyway.
1843    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1844    nsecs_t lastWarning = 0;
1845    bool longStandbyExit = false;
1846    uint32_t activeSleepTime = activeSleepTimeUs();
1847    uint32_t idleSleepTime = idleSleepTimeUs();
1848    uint32_t sleepTime = idleSleepTime;
1849    Vector< sp<EffectChain> > effectChains;
1850#ifdef DEBUG_CPU_USAGE
1851    ThreadCpuUsage cpu;
1852    const CentralTendencyStatistics& stats = cpu.statistics();
1853#endif
1854
1855    acquireWakeLock();
1856
1857    while (!exitPending())
1858    {
1859#ifdef DEBUG_CPU_USAGE
1860        cpu.sampleAndEnable();
1861        unsigned n = stats.n();
1862        // cpu.elapsed() is expensive, so don't call it every loop
1863        if ((n & 127) == 1) {
1864            long long elapsed = cpu.elapsed();
1865            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1866                double perLoop = elapsed / (double) n;
1867                double perLoop100 = perLoop * 0.01;
1868                double mean = stats.mean();
1869                double stddev = stats.stddev();
1870                double minimum = stats.minimum();
1871                double maximum = stats.maximum();
1872                cpu.resetStatistics();
1873                LOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n  us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n  %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
1874                        elapsed * .000000001, n, perLoop * .000001,
1875                        mean * .001,
1876                        stddev * .001,
1877                        minimum * .001,
1878                        maximum * .001,
1879                        mean / perLoop100,
1880                        stddev / perLoop100,
1881                        minimum / perLoop100,
1882                        maximum / perLoop100);
1883            }
1884        }
1885#endif
1886        processConfigEvents();
1887
1888        mixerStatus = MIXER_IDLE;
1889        { // scope for mLock
1890
1891            Mutex::Autolock _l(mLock);
1892
1893            if (checkForNewParameters_l()) {
1894                mixBufferSize = mFrameCount * mFrameSize;
1895                // FIXME: Relaxed timing because of a certain device that can't meet latency
1896                // Should be reduced to 2x after the vendor fixes the driver issue
1897                // increase threshold again due to low power audio mode. The way this warning
1898                // threshold is calculated and its usefulness should be reconsidered anyway.
1899                maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
1900                activeSleepTime = activeSleepTimeUs();
1901                idleSleepTime = idleSleepTimeUs();
1902            }
1903
1904            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1905
1906            // put audio hardware into standby after short delay
1907            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1908                        mSuspended) {
1909                if (!mStandby) {
1910                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1911                    mOutput->stream->common.standby(&mOutput->stream->common);
1912                    mStandby = true;
1913                    mBytesWritten = 0;
1914                }
1915
1916                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1917                    // we're about to wait, flush the binder command buffer
1918                    IPCThreadState::self()->flushCommands();
1919
1920                    if (exitPending()) break;
1921
1922                    releaseWakeLock_l();
1923                    // wait until we have something to do...
1924                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1925                    mWaitWorkCV.wait(mLock);
1926                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
1927                    acquireWakeLock_l();
1928
1929                    if (mMasterMute == false) {
1930                        char value[PROPERTY_VALUE_MAX];
1931                        property_get("ro.audio.silent", value, "0");
1932                        if (atoi(value)) {
1933                            LOGD("Silence is golden");
1934                            setMasterMute(true);
1935                        }
1936                    }
1937
1938                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1939                    sleepTime = idleSleepTime;
1940                    continue;
1941                }
1942            }
1943
1944            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1945
1946            // prevent any changes in effect chain list and in each effect chain
1947            // during mixing and effect process as the audio buffers could be deleted
1948            // or modified if an effect is created or deleted
1949            lockEffectChains_l(effectChains);
1950       }
1951
1952        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1953            // mix buffers...
1954            mAudioMixer->process();
1955            sleepTime = 0;
1956            standbyTime = systemTime() + kStandbyTimeInNsecs;
1957            //TODO: delay standby when effects have a tail
1958        } else {
1959            // If no tracks are ready, sleep once for the duration of an output
1960            // buffer size, then write 0s to the output
1961            if (sleepTime == 0) {
1962                if (mixerStatus == MIXER_TRACKS_ENABLED) {
1963                    sleepTime = activeSleepTime;
1964                } else {
1965                    sleepTime = idleSleepTime;
1966                }
1967            } else if (mBytesWritten != 0 ||
1968                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
1969                memset (mMixBuffer, 0, mixBufferSize);
1970                sleepTime = 0;
1971                ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
1972            }
1973            // TODO add standby time extension fct of effect tail
1974        }
1975
1976        if (mSuspended) {
1977            sleepTime = suspendSleepTimeUs();
1978        }
1979        // sleepTime == 0 means we must write to audio hardware
1980        if (sleepTime == 0) {
1981             for (size_t i = 0; i < effectChains.size(); i ++) {
1982                 effectChains[i]->process_l();
1983             }
1984             // enable changes in effect chain
1985             unlockEffectChains(effectChains);
1986            mLastWriteTime = systemTime();
1987            mInWrite = true;
1988            mBytesWritten += mixBufferSize;
1989
1990            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
1991            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
1992            mNumWrites++;
1993            mInWrite = false;
1994            nsecs_t now = systemTime();
1995            nsecs_t delta = now - mLastWriteTime;
1996            if (!mStandby && delta > maxPeriod) {
1997                mNumDelayedWrites++;
1998                if ((now - lastWarning) > kWarningThrottle) {
1999                    LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2000                            ns2ms(delta), mNumDelayedWrites, this);
2001                    lastWarning = now;
2002                }
2003                if (mStandby) {
2004                    longStandbyExit = true;
2005                }
2006            }
2007            mStandby = false;
2008        } else {
2009            // enable changes in effect chain
2010            unlockEffectChains(effectChains);
2011            usleep(sleepTime);
2012        }
2013
2014        // finally let go of all our tracks, without the lock held
2015        // since we can't guarantee the destructors won't acquire that
2016        // same lock.
2017        tracksToRemove.clear();
2018
2019        // Effect chains will be actually deleted here if they were removed from
2020        // mEffectChains list during mixing or effects processing
2021        effectChains.clear();
2022    }
2023
2024    if (!mStandby) {
2025        mOutput->stream->common.standby(&mOutput->stream->common);
2026    }
2027
2028    releaseWakeLock();
2029
2030    ALOGV("MixerThread %p exiting", this);
2031    return false;
2032}
2033
2034// prepareTracks_l() must be called with ThreadBase::mLock held
2035uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2036{
2037
2038    uint32_t mixerStatus = MIXER_IDLE;
2039    // find out which tracks need to be processed
2040    size_t count = activeTracks.size();
2041    size_t mixedTracks = 0;
2042    size_t tracksWithEffect = 0;
2043
2044    float masterVolume = mMasterVolume;
2045    bool  masterMute = mMasterMute;
2046
2047    if (masterMute) {
2048        masterVolume = 0;
2049    }
2050    // Delegate master volume control to effect in output mix effect chain if needed
2051    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2052    if (chain != 0) {
2053        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2054        chain->setVolume_l(&v, &v);
2055        masterVolume = (float)((v + (1 << 23)) >> 24);
2056        chain.clear();
2057    }
2058
2059    for (size_t i=0 ; i<count ; i++) {
2060        sp<Track> t = activeTracks[i].promote();
2061        if (t == 0) continue;
2062
2063        Track* const track = t.get();
2064        audio_track_cblk_t* cblk = track->cblk();
2065
2066        // The first time a track is added we wait
2067        // for all its buffers to be filled before processing it
2068        mAudioMixer->setActiveTrack(track->name());
2069        if (cblk->framesReady() && track->isReady() &&
2070                !track->isPaused() && !track->isTerminated())
2071        {
2072            //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
2073
2074            mixedTracks++;
2075
2076            // track->mainBuffer() != mMixBuffer means there is an effect chain
2077            // connected to the track
2078            chain.clear();
2079            if (track->mainBuffer() != mMixBuffer) {
2080                chain = getEffectChain_l(track->sessionId());
2081                // Delegate volume control to effect in track effect chain if needed
2082                if (chain != 0) {
2083                    tracksWithEffect++;
2084                } else {
2085                    LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
2086                            track->name(), track->sessionId());
2087                }
2088            }
2089
2090
2091            int param = AudioMixer::VOLUME;
2092            if (track->mFillingUpStatus == Track::FS_FILLED) {
2093                // no ramp for the first volume setting
2094                track->mFillingUpStatus = Track::FS_ACTIVE;
2095                if (track->mState == TrackBase::RESUMING) {
2096                    track->mState = TrackBase::ACTIVE;
2097                    param = AudioMixer::RAMP_VOLUME;
2098                }
2099                mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2100            } else if (cblk->server != 0) {
2101                // If the track is stopped before the first frame was mixed,
2102                // do not apply ramp
2103                param = AudioMixer::RAMP_VOLUME;
2104            }
2105
2106            // compute volume for this track
2107            uint32_t vl, vr, va;
2108            if (track->isMuted() || track->isPausing() ||
2109                mStreamTypes[track->type()].mute) {
2110                vl = vr = va = 0;
2111                if (track->isPausing()) {
2112                    track->setPaused();
2113                }
2114            } else {
2115
2116                // read original volumes with volume control
2117                float typeVolume = mStreamTypes[track->type()].volume;
2118                float v = masterVolume * typeVolume;
2119                vl = (uint32_t)(v * cblk->volume[0]) << 12;
2120                vr = (uint32_t)(v * cblk->volume[1]) << 12;
2121
2122                va = (uint32_t)(v * cblk->sendLevel);
2123            }
2124            // Delegate volume control to effect in track effect chain if needed
2125            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2126                // Do not ramp volume if volume is controlled by effect
2127                param = AudioMixer::VOLUME;
2128                track->mHasVolumeController = true;
2129            } else {
2130                // force no volume ramp when volume controller was just disabled or removed
2131                // from effect chain to avoid volume spike
2132                if (track->mHasVolumeController) {
2133                    param = AudioMixer::VOLUME;
2134                }
2135                track->mHasVolumeController = false;
2136            }
2137
2138            // Convert volumes from 8.24 to 4.12 format
2139            int16_t left, right, aux;
2140            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2141            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2142            left = int16_t(v_clamped);
2143            v_clamped = (vr + (1 << 11)) >> 12;
2144            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2145            right = int16_t(v_clamped);
2146
2147            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2148            aux = int16_t(va);
2149
2150            // XXX: these things DON'T need to be done each time
2151            mAudioMixer->setBufferProvider(track);
2152            mAudioMixer->enable(AudioMixer::MIXING);
2153
2154            mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
2155            mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
2156            mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
2157            mAudioMixer->setParameter(
2158                AudioMixer::TRACK,
2159                AudioMixer::FORMAT, (void *)track->format());
2160            mAudioMixer->setParameter(
2161                AudioMixer::TRACK,
2162                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2163            mAudioMixer->setParameter(
2164                AudioMixer::RESAMPLE,
2165                AudioMixer::SAMPLE_RATE,
2166                (void *)(cblk->sampleRate));
2167            mAudioMixer->setParameter(
2168                AudioMixer::TRACK,
2169                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2170            mAudioMixer->setParameter(
2171                AudioMixer::TRACK,
2172                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2173
2174            // reset retry count
2175            track->mRetryCount = kMaxTrackRetries;
2176            mixerStatus = MIXER_TRACKS_READY;
2177        } else {
2178            //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
2179            if (track->isStopped()) {
2180                track->reset();
2181            }
2182            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2183                // We have consumed all the buffers of this track.
2184                // Remove it from the list of active tracks.
2185                tracksToRemove->add(track);
2186            } else {
2187                // No buffers for this track. Give it a few chances to
2188                // fill a buffer, then remove it from active list.
2189                if (--(track->mRetryCount) <= 0) {
2190                    ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
2191                    tracksToRemove->add(track);
2192                    // indicate to client process that the track was disabled because of underrun
2193                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2194                } else if (mixerStatus != MIXER_TRACKS_READY) {
2195                    mixerStatus = MIXER_TRACKS_ENABLED;
2196                }
2197            }
2198            mAudioMixer->disable(AudioMixer::MIXING);
2199        }
2200    }
2201
2202    // remove all the tracks that need to be...
2203    count = tracksToRemove->size();
2204    if (UNLIKELY(count)) {
2205        for (size_t i=0 ; i<count ; i++) {
2206            const sp<Track>& track = tracksToRemove->itemAt(i);
2207            mActiveTracks.remove(track);
2208            if (track->mainBuffer() != mMixBuffer) {
2209                chain = getEffectChain_l(track->sessionId());
2210                if (chain != 0) {
2211                    ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2212                    chain->decActiveTrackCnt();
2213                }
2214            }
2215            if (track->isTerminated()) {
2216                removeTrack_l(track);
2217            }
2218        }
2219    }
2220
2221    // mix buffer must be cleared if all tracks are connected to an
2222    // effect chain as in this case the mixer will not write to
2223    // mix buffer and track effects will accumulate into it
2224    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2225        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2226    }
2227
2228    return mixerStatus;
2229}
2230
2231void AudioFlinger::MixerThread::invalidateTracks(int streamType)
2232{
2233    ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2234            this,  streamType, mTracks.size());
2235    Mutex::Autolock _l(mLock);
2236
2237    size_t size = mTracks.size();
2238    for (size_t i = 0; i < size; i++) {
2239        sp<Track> t = mTracks[i];
2240        if (t->type() == streamType) {
2241            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2242            t->mCblk->cv.signal();
2243        }
2244    }
2245}
2246
2247void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid)
2248{
2249    ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2250            this,  streamType, valid);
2251    Mutex::Autolock _l(mLock);
2252
2253    mStreamTypes[streamType].valid = valid;
2254}
2255
2256// getTrackName_l() must be called with ThreadBase::mLock held
2257int AudioFlinger::MixerThread::getTrackName_l()
2258{
2259    return mAudioMixer->getTrackName();
2260}
2261
2262// deleteTrackName_l() must be called with ThreadBase::mLock held
2263void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2264{
2265    ALOGV("remove track (%d) and delete from mixer", name);
2266    mAudioMixer->deleteTrackName(name);
2267}
2268
2269// checkForNewParameters_l() must be called with ThreadBase::mLock held
2270bool AudioFlinger::MixerThread::checkForNewParameters_l()
2271{
2272    bool reconfig = false;
2273
2274    while (!mNewParameters.isEmpty()) {
2275        status_t status = NO_ERROR;
2276        String8 keyValuePair = mNewParameters[0];
2277        AudioParameter param = AudioParameter(keyValuePair);
2278        int value;
2279
2280        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2281            reconfig = true;
2282        }
2283        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2284            if (value != AUDIO_FORMAT_PCM_16_BIT) {
2285                status = BAD_VALUE;
2286            } else {
2287                reconfig = true;
2288            }
2289        }
2290        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2291            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2292                status = BAD_VALUE;
2293            } else {
2294                reconfig = true;
2295            }
2296        }
2297        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2298            // do not accept frame count changes if tracks are open as the track buffer
2299            // size depends on frame count and correct behavior would not be garantied
2300            // if frame count is changed after track creation
2301            if (!mTracks.isEmpty()) {
2302                status = INVALID_OPERATION;
2303            } else {
2304                reconfig = true;
2305            }
2306        }
2307        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2308            // when changing the audio output device, call addBatteryData to notify
2309            // the change
2310            if ((int)mDevice != value) {
2311                uint32_t params = 0;
2312                // check whether speaker is on
2313                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2314                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2315                }
2316
2317                int deviceWithoutSpeaker
2318                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2319                // check if any other device (except speaker) is on
2320                if (value & deviceWithoutSpeaker ) {
2321                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2322                }
2323
2324                if (params != 0) {
2325                    addBatteryData(params);
2326                }
2327            }
2328
2329            // forward device change to effects that have requested to be
2330            // aware of attached audio device.
2331            mDevice = (uint32_t)value;
2332            for (size_t i = 0; i < mEffectChains.size(); i++) {
2333                mEffectChains[i]->setDevice_l(mDevice);
2334            }
2335        }
2336
2337        if (status == NO_ERROR) {
2338            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2339                                                    keyValuePair.string());
2340            if (!mStandby && status == INVALID_OPERATION) {
2341               mOutput->stream->common.standby(&mOutput->stream->common);
2342               mStandby = true;
2343               mBytesWritten = 0;
2344               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2345                                                       keyValuePair.string());
2346            }
2347            if (status == NO_ERROR && reconfig) {
2348                delete mAudioMixer;
2349                readOutputParameters();
2350                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2351                for (size_t i = 0; i < mTracks.size() ; i++) {
2352                    int name = getTrackName_l();
2353                    if (name < 0) break;
2354                    mTracks[i]->mName = name;
2355                    // limit track sample rate to 2 x new output sample rate
2356                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2357                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2358                    }
2359                }
2360                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2361            }
2362        }
2363
2364        mNewParameters.removeAt(0);
2365
2366        mParamStatus = status;
2367        mParamCond.signal();
2368        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2369        // already timed out waiting for the status and will never signal the condition.
2370        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
2371    }
2372    return reconfig;
2373}
2374
2375status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2376{
2377    const size_t SIZE = 256;
2378    char buffer[SIZE];
2379    String8 result;
2380
2381    PlaybackThread::dumpInternals(fd, args);
2382
2383    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2384    result.append(buffer);
2385    write(fd, result.string(), result.size());
2386    return NO_ERROR;
2387}
2388
2389uint32_t AudioFlinger::MixerThread::activeSleepTimeUs()
2390{
2391    return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2392}
2393
2394uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2395{
2396    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2397}
2398
2399uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2400{
2401    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2402}
2403
2404// ----------------------------------------------------------------------------
2405AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2406    :   PlaybackThread(audioFlinger, output, id, device)
2407{
2408    mType = ThreadBase::DIRECT;
2409}
2410
2411AudioFlinger::DirectOutputThread::~DirectOutputThread()
2412{
2413}
2414
2415
2416static inline int16_t clamp16(int32_t sample)
2417{
2418    if ((sample>>15) ^ (sample>>31))
2419        sample = 0x7FFF ^ (sample>>31);
2420    return sample;
2421}
2422
2423static inline
2424int32_t mul(int16_t in, int16_t v)
2425{
2426#if defined(__arm__) && !defined(__thumb__)
2427    int32_t out;
2428    asm( "smulbb %[out], %[in], %[v] \n"
2429         : [out]"=r"(out)
2430         : [in]"%r"(in), [v]"r"(v)
2431         : );
2432    return out;
2433#else
2434    return in * int32_t(v);
2435#endif
2436}
2437
2438void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2439{
2440    // Do not apply volume on compressed audio
2441    if (!audio_is_linear_pcm(mFormat)) {
2442        return;
2443    }
2444
2445    // convert to signed 16 bit before volume calculation
2446    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2447        size_t count = mFrameCount * mChannelCount;
2448        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2449        int16_t *dst = mMixBuffer + count-1;
2450        while(count--) {
2451            *dst-- = (int16_t)(*src--^0x80) << 8;
2452        }
2453    }
2454
2455    size_t frameCount = mFrameCount;
2456    int16_t *out = mMixBuffer;
2457    if (ramp) {
2458        if (mChannelCount == 1) {
2459            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2460            int32_t vlInc = d / (int32_t)frameCount;
2461            int32_t vl = ((int32_t)mLeftVolShort << 16);
2462            do {
2463                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2464                out++;
2465                vl += vlInc;
2466            } while (--frameCount);
2467
2468        } else {
2469            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2470            int32_t vlInc = d / (int32_t)frameCount;
2471            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2472            int32_t vrInc = d / (int32_t)frameCount;
2473            int32_t vl = ((int32_t)mLeftVolShort << 16);
2474            int32_t vr = ((int32_t)mRightVolShort << 16);
2475            do {
2476                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2477                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2478                out += 2;
2479                vl += vlInc;
2480                vr += vrInc;
2481            } while (--frameCount);
2482        }
2483    } else {
2484        if (mChannelCount == 1) {
2485            do {
2486                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2487                out++;
2488            } while (--frameCount);
2489        } else {
2490            do {
2491                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2492                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2493                out += 2;
2494            } while (--frameCount);
2495        }
2496    }
2497
2498    // convert back to unsigned 8 bit after volume calculation
2499    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2500        size_t count = mFrameCount * mChannelCount;
2501        int16_t *src = mMixBuffer;
2502        uint8_t *dst = (uint8_t *)mMixBuffer;
2503        while(count--) {
2504            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2505        }
2506    }
2507
2508    mLeftVolShort = leftVol;
2509    mRightVolShort = rightVol;
2510}
2511
2512bool AudioFlinger::DirectOutputThread::threadLoop()
2513{
2514    uint32_t mixerStatus = MIXER_IDLE;
2515    sp<Track> trackToRemove;
2516    sp<Track> activeTrack;
2517    nsecs_t standbyTime = systemTime();
2518    int8_t *curBuf;
2519    size_t mixBufferSize = mFrameCount*mFrameSize;
2520    uint32_t activeSleepTime = activeSleepTimeUs();
2521    uint32_t idleSleepTime = idleSleepTimeUs();
2522    uint32_t sleepTime = idleSleepTime;
2523    // use shorter standby delay as on normal output to release
2524    // hardware resources as soon as possible
2525    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2526
2527    acquireWakeLock();
2528
2529    while (!exitPending())
2530    {
2531        bool rampVolume;
2532        uint16_t leftVol;
2533        uint16_t rightVol;
2534        Vector< sp<EffectChain> > effectChains;
2535
2536        processConfigEvents();
2537
2538        mixerStatus = MIXER_IDLE;
2539
2540        { // scope for the mLock
2541
2542            Mutex::Autolock _l(mLock);
2543
2544            if (checkForNewParameters_l()) {
2545                mixBufferSize = mFrameCount*mFrameSize;
2546                activeSleepTime = activeSleepTimeUs();
2547                idleSleepTime = idleSleepTimeUs();
2548                standbyDelay = microseconds(activeSleepTime*2);
2549            }
2550
2551            // put audio hardware into standby after short delay
2552            if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2553                        mSuspended) {
2554                // wait until we have something to do...
2555                if (!mStandby) {
2556                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
2557                    mOutput->stream->common.standby(&mOutput->stream->common);
2558                    mStandby = true;
2559                    mBytesWritten = 0;
2560                }
2561
2562                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2563                    // we're about to wait, flush the binder command buffer
2564                    IPCThreadState::self()->flushCommands();
2565
2566                    if (exitPending()) break;
2567
2568                    releaseWakeLock_l();
2569                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2570                    mWaitWorkCV.wait(mLock);
2571                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2572                    acquireWakeLock_l();
2573
2574                    if (mMasterMute == false) {
2575                        char value[PROPERTY_VALUE_MAX];
2576                        property_get("ro.audio.silent", value, "0");
2577                        if (atoi(value)) {
2578                            LOGD("Silence is golden");
2579                            setMasterMute(true);
2580                        }
2581                    }
2582
2583                    standbyTime = systemTime() + standbyDelay;
2584                    sleepTime = idleSleepTime;
2585                    continue;
2586                }
2587            }
2588
2589            effectChains = mEffectChains;
2590
2591            // find out which tracks need to be processed
2592            if (mActiveTracks.size() != 0) {
2593                sp<Track> t = mActiveTracks[0].promote();
2594                if (t == 0) continue;
2595
2596                Track* const track = t.get();
2597                audio_track_cblk_t* cblk = track->cblk();
2598
2599                // The first time a track is added we wait
2600                // for all its buffers to be filled before processing it
2601                if (cblk->framesReady() && track->isReady() &&
2602                        !track->isPaused() && !track->isTerminated())
2603                {
2604                    //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2605
2606                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2607                        track->mFillingUpStatus = Track::FS_ACTIVE;
2608                        mLeftVolFloat = mRightVolFloat = 0;
2609                        mLeftVolShort = mRightVolShort = 0;
2610                        if (track->mState == TrackBase::RESUMING) {
2611                            track->mState = TrackBase::ACTIVE;
2612                            rampVolume = true;
2613                        }
2614                    } else if (cblk->server != 0) {
2615                        // If the track is stopped before the first frame was mixed,
2616                        // do not apply ramp
2617                        rampVolume = true;
2618                    }
2619                    // compute volume for this track
2620                    float left, right;
2621                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2622                        mStreamTypes[track->type()].mute) {
2623                        left = right = 0;
2624                        if (track->isPausing()) {
2625                            track->setPaused();
2626                        }
2627                    } else {
2628                        float typeVolume = mStreamTypes[track->type()].volume;
2629                        float v = mMasterVolume * typeVolume;
2630                        float v_clamped = v * cblk->volume[0];
2631                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2632                        left = v_clamped/MAX_GAIN;
2633                        v_clamped = v * cblk->volume[1];
2634                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2635                        right = v_clamped/MAX_GAIN;
2636                    }
2637
2638                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2639                        mLeftVolFloat = left;
2640                        mRightVolFloat = right;
2641
2642                        // If audio HAL implements volume control,
2643                        // force software volume to nominal value
2644                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2645                            left = 1.0f;
2646                            right = 1.0f;
2647                        }
2648
2649                        // Convert volumes from float to 8.24
2650                        uint32_t vl = (uint32_t)(left * (1 << 24));
2651                        uint32_t vr = (uint32_t)(right * (1 << 24));
2652
2653                        // Delegate volume control to effect in track effect chain if needed
2654                        // only one effect chain can be present on DirectOutputThread, so if
2655                        // there is one, the track is connected to it
2656                        if (!effectChains.isEmpty()) {
2657                            // Do not ramp volume if volume is controlled by effect
2658                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2659                                rampVolume = false;
2660                            }
2661                        }
2662
2663                        // Convert volumes from 8.24 to 4.12 format
2664                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2665                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2666                        leftVol = (uint16_t)v_clamped;
2667                        v_clamped = (vr + (1 << 11)) >> 12;
2668                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2669                        rightVol = (uint16_t)v_clamped;
2670                    } else {
2671                        leftVol = mLeftVolShort;
2672                        rightVol = mRightVolShort;
2673                        rampVolume = false;
2674                    }
2675
2676                    // reset retry count
2677                    track->mRetryCount = kMaxTrackRetriesDirect;
2678                    activeTrack = t;
2679                    mixerStatus = MIXER_TRACKS_READY;
2680                } else {
2681                    //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2682                    if (track->isStopped()) {
2683                        track->reset();
2684                    }
2685                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2686                        // We have consumed all the buffers of this track.
2687                        // Remove it from the list of active tracks.
2688                        trackToRemove = track;
2689                    } else {
2690                        // No buffers for this track. Give it a few chances to
2691                        // fill a buffer, then remove it from active list.
2692                        if (--(track->mRetryCount) <= 0) {
2693                            ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2694                            trackToRemove = track;
2695                        } else {
2696                            mixerStatus = MIXER_TRACKS_ENABLED;
2697                        }
2698                    }
2699                }
2700            }
2701
2702            // remove all the tracks that need to be...
2703            if (UNLIKELY(trackToRemove != 0)) {
2704                mActiveTracks.remove(trackToRemove);
2705                if (!effectChains.isEmpty()) {
2706                    ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2707                            trackToRemove->sessionId());
2708                    effectChains[0]->decActiveTrackCnt();
2709                }
2710                if (trackToRemove->isTerminated()) {
2711                    removeTrack_l(trackToRemove);
2712                }
2713            }
2714
2715            lockEffectChains_l(effectChains);
2716       }
2717
2718        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2719            AudioBufferProvider::Buffer buffer;
2720            size_t frameCount = mFrameCount;
2721            curBuf = (int8_t *)mMixBuffer;
2722            // output audio to hardware
2723            while (frameCount) {
2724                buffer.frameCount = frameCount;
2725                activeTrack->getNextBuffer(&buffer);
2726                if (UNLIKELY(buffer.raw == 0)) {
2727                    memset(curBuf, 0, frameCount * mFrameSize);
2728                    break;
2729                }
2730                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2731                frameCount -= buffer.frameCount;
2732                curBuf += buffer.frameCount * mFrameSize;
2733                activeTrack->releaseBuffer(&buffer);
2734            }
2735            sleepTime = 0;
2736            standbyTime = systemTime() + standbyDelay;
2737        } else {
2738            if (sleepTime == 0) {
2739                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2740                    sleepTime = activeSleepTime;
2741                } else {
2742                    sleepTime = idleSleepTime;
2743                }
2744            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2745                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2746                sleepTime = 0;
2747            }
2748        }
2749
2750        if (mSuspended) {
2751            sleepTime = suspendSleepTimeUs();
2752        }
2753        // sleepTime == 0 means we must write to audio hardware
2754        if (sleepTime == 0) {
2755            if (mixerStatus == MIXER_TRACKS_READY) {
2756                applyVolume(leftVol, rightVol, rampVolume);
2757            }
2758            for (size_t i = 0; i < effectChains.size(); i ++) {
2759                effectChains[i]->process_l();
2760            }
2761            unlockEffectChains(effectChains);
2762
2763            mLastWriteTime = systemTime();
2764            mInWrite = true;
2765            mBytesWritten += mixBufferSize;
2766            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2767            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2768            mNumWrites++;
2769            mInWrite = false;
2770            mStandby = false;
2771        } else {
2772            unlockEffectChains(effectChains);
2773            usleep(sleepTime);
2774        }
2775
2776        // finally let go of removed track, without the lock held
2777        // since we can't guarantee the destructors won't acquire that
2778        // same lock.
2779        trackToRemove.clear();
2780        activeTrack.clear();
2781
2782        // Effect chains will be actually deleted here if they were removed from
2783        // mEffectChains list during mixing or effects processing
2784        effectChains.clear();
2785    }
2786
2787    if (!mStandby) {
2788        mOutput->stream->common.standby(&mOutput->stream->common);
2789    }
2790
2791    releaseWakeLock();
2792
2793    ALOGV("DirectOutputThread %p exiting", this);
2794    return false;
2795}
2796
2797// getTrackName_l() must be called with ThreadBase::mLock held
2798int AudioFlinger::DirectOutputThread::getTrackName_l()
2799{
2800    return 0;
2801}
2802
2803// deleteTrackName_l() must be called with ThreadBase::mLock held
2804void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2805{
2806}
2807
2808// checkForNewParameters_l() must be called with ThreadBase::mLock held
2809bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2810{
2811    bool reconfig = false;
2812
2813    while (!mNewParameters.isEmpty()) {
2814        status_t status = NO_ERROR;
2815        String8 keyValuePair = mNewParameters[0];
2816        AudioParameter param = AudioParameter(keyValuePair);
2817        int value;
2818
2819        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2820            // do not accept frame count changes if tracks are open as the track buffer
2821            // size depends on frame count and correct behavior would not be garantied
2822            // if frame count is changed after track creation
2823            if (!mTracks.isEmpty()) {
2824                status = INVALID_OPERATION;
2825            } else {
2826                reconfig = true;
2827            }
2828        }
2829        if (status == NO_ERROR) {
2830            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2831                                                    keyValuePair.string());
2832            if (!mStandby && status == INVALID_OPERATION) {
2833               mOutput->stream->common.standby(&mOutput->stream->common);
2834               mStandby = true;
2835               mBytesWritten = 0;
2836               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2837                                                       keyValuePair.string());
2838            }
2839            if (status == NO_ERROR && reconfig) {
2840                readOutputParameters();
2841                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2842            }
2843        }
2844
2845        mNewParameters.removeAt(0);
2846
2847        mParamStatus = status;
2848        mParamCond.signal();
2849        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2850        // already timed out waiting for the status and will never signal the condition.
2851        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
2852    }
2853    return reconfig;
2854}
2855
2856uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2857{
2858    uint32_t time;
2859    if (audio_is_linear_pcm(mFormat)) {
2860        time = (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2861    } else {
2862        time = 10000;
2863    }
2864    return time;
2865}
2866
2867uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2868{
2869    uint32_t time;
2870    if (audio_is_linear_pcm(mFormat)) {
2871        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2872    } else {
2873        time = 10000;
2874    }
2875    return time;
2876}
2877
2878uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2879{
2880    uint32_t time;
2881    if (audio_is_linear_pcm(mFormat)) {
2882        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2883    } else {
2884        time = 10000;
2885    }
2886    return time;
2887}
2888
2889
2890// ----------------------------------------------------------------------------
2891
2892AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2893    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2894{
2895    mType = ThreadBase::DUPLICATING;
2896    addOutputTrack(mainThread);
2897}
2898
2899AudioFlinger::DuplicatingThread::~DuplicatingThread()
2900{
2901    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2902        mOutputTracks[i]->destroy();
2903    }
2904    mOutputTracks.clear();
2905}
2906
2907bool AudioFlinger::DuplicatingThread::threadLoop()
2908{
2909    Vector< sp<Track> > tracksToRemove;
2910    uint32_t mixerStatus = MIXER_IDLE;
2911    nsecs_t standbyTime = systemTime();
2912    size_t mixBufferSize = mFrameCount*mFrameSize;
2913    SortedVector< sp<OutputTrack> > outputTracks;
2914    uint32_t writeFrames = 0;
2915    uint32_t activeSleepTime = activeSleepTimeUs();
2916    uint32_t idleSleepTime = idleSleepTimeUs();
2917    uint32_t sleepTime = idleSleepTime;
2918    Vector< sp<EffectChain> > effectChains;
2919
2920    acquireWakeLock();
2921
2922    while (!exitPending())
2923    {
2924        processConfigEvents();
2925
2926        mixerStatus = MIXER_IDLE;
2927        { // scope for the mLock
2928
2929            Mutex::Autolock _l(mLock);
2930
2931            if (checkForNewParameters_l()) {
2932                mixBufferSize = mFrameCount*mFrameSize;
2933                updateWaitTime();
2934                activeSleepTime = activeSleepTimeUs();
2935                idleSleepTime = idleSleepTimeUs();
2936            }
2937
2938            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
2939
2940            for (size_t i = 0; i < mOutputTracks.size(); i++) {
2941                outputTracks.add(mOutputTracks[i]);
2942            }
2943
2944            // put audio hardware into standby after short delay
2945            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
2946                         mSuspended) {
2947                if (!mStandby) {
2948                    for (size_t i = 0; i < outputTracks.size(); i++) {
2949                        outputTracks[i]->stop();
2950                    }
2951                    mStandby = true;
2952                    mBytesWritten = 0;
2953                }
2954
2955                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
2956                    // we're about to wait, flush the binder command buffer
2957                    IPCThreadState::self()->flushCommands();
2958                    outputTracks.clear();
2959
2960                    if (exitPending()) break;
2961
2962                    releaseWakeLock_l();
2963                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
2964                    mWaitWorkCV.wait(mLock);
2965                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
2966                    acquireWakeLock_l();
2967
2968                    if (mMasterMute == false) {
2969                        char value[PROPERTY_VALUE_MAX];
2970                        property_get("ro.audio.silent", value, "0");
2971                        if (atoi(value)) {
2972                            LOGD("Silence is golden");
2973                            setMasterMute(true);
2974                        }
2975                    }
2976
2977                    standbyTime = systemTime() + kStandbyTimeInNsecs;
2978                    sleepTime = idleSleepTime;
2979                    continue;
2980                }
2981            }
2982
2983            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
2984
2985            // prevent any changes in effect chain list and in each effect chain
2986            // during mixing and effect process as the audio buffers could be deleted
2987            // or modified if an effect is created or deleted
2988            lockEffectChains_l(effectChains);
2989        }
2990
2991        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2992            // mix buffers...
2993            if (outputsReady(outputTracks)) {
2994                mAudioMixer->process();
2995            } else {
2996                memset(mMixBuffer, 0, mixBufferSize);
2997            }
2998            sleepTime = 0;
2999            writeFrames = mFrameCount;
3000        } else {
3001            if (sleepTime == 0) {
3002                if (mixerStatus == MIXER_TRACKS_ENABLED) {
3003                    sleepTime = activeSleepTime;
3004                } else {
3005                    sleepTime = idleSleepTime;
3006                }
3007            } else if (mBytesWritten != 0) {
3008                // flush remaining overflow buffers in output tracks
3009                for (size_t i = 0; i < outputTracks.size(); i++) {
3010                    if (outputTracks[i]->isActive()) {
3011                        sleepTime = 0;
3012                        writeFrames = 0;
3013                        memset(mMixBuffer, 0, mixBufferSize);
3014                        break;
3015                    }
3016                }
3017            }
3018        }
3019
3020        if (mSuspended) {
3021            sleepTime = suspendSleepTimeUs();
3022        }
3023        // sleepTime == 0 means we must write to audio hardware
3024        if (sleepTime == 0) {
3025            for (size_t i = 0; i < effectChains.size(); i ++) {
3026                effectChains[i]->process_l();
3027            }
3028            // enable changes in effect chain
3029            unlockEffectChains(effectChains);
3030
3031            standbyTime = systemTime() + kStandbyTimeInNsecs;
3032            for (size_t i = 0; i < outputTracks.size(); i++) {
3033                outputTracks[i]->write(mMixBuffer, writeFrames);
3034            }
3035            mStandby = false;
3036            mBytesWritten += mixBufferSize;
3037        } else {
3038            // enable changes in effect chain
3039            unlockEffectChains(effectChains);
3040            usleep(sleepTime);
3041        }
3042
3043        // finally let go of all our tracks, without the lock held
3044        // since we can't guarantee the destructors won't acquire that
3045        // same lock.
3046        tracksToRemove.clear();
3047        outputTracks.clear();
3048
3049        // Effect chains will be actually deleted here if they were removed from
3050        // mEffectChains list during mixing or effects processing
3051        effectChains.clear();
3052    }
3053
3054    releaseWakeLock();
3055
3056    return false;
3057}
3058
3059void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3060{
3061    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3062    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3063                                            this,
3064                                            mSampleRate,
3065                                            mFormat,
3066                                            mChannelMask,
3067                                            frameCount);
3068    if (outputTrack->cblk() != NULL) {
3069        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3070        mOutputTracks.add(outputTrack);
3071        ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3072        updateWaitTime();
3073    }
3074}
3075
3076void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3077{
3078    Mutex::Autolock _l(mLock);
3079    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3080        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3081            mOutputTracks[i]->destroy();
3082            mOutputTracks.removeAt(i);
3083            updateWaitTime();
3084            return;
3085        }
3086    }
3087    ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3088}
3089
3090void AudioFlinger::DuplicatingThread::updateWaitTime()
3091{
3092    mWaitTimeMs = UINT_MAX;
3093    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3094        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3095        if (strong != NULL) {
3096            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3097            if (waitTimeMs < mWaitTimeMs) {
3098                mWaitTimeMs = waitTimeMs;
3099            }
3100        }
3101    }
3102}
3103
3104
3105bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3106{
3107    for (size_t i = 0; i < outputTracks.size(); i++) {
3108        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3109        if (thread == 0) {
3110            LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3111            return false;
3112        }
3113        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3114        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3115            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3116            return false;
3117        }
3118    }
3119    return true;
3120}
3121
3122uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3123{
3124    return (mWaitTimeMs * 1000) / 2;
3125}
3126
3127// ----------------------------------------------------------------------------
3128
3129// TrackBase constructor must be called with AudioFlinger::mLock held
3130AudioFlinger::ThreadBase::TrackBase::TrackBase(
3131            const wp<ThreadBase>& thread,
3132            const sp<Client>& client,
3133            uint32_t sampleRate,
3134            uint32_t format,
3135            uint32_t channelMask,
3136            int frameCount,
3137            uint32_t flags,
3138            const sp<IMemory>& sharedBuffer,
3139            int sessionId)
3140    :   RefBase(),
3141        mThread(thread),
3142        mClient(client),
3143        mCblk(0),
3144        mFrameCount(0),
3145        mState(IDLE),
3146        mClientTid(-1),
3147        mFormat(format),
3148        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3149        mSessionId(sessionId)
3150{
3151    ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3152
3153    // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3154   size_t size = sizeof(audio_track_cblk_t);
3155   uint8_t channelCount = popcount(channelMask);
3156   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3157   if (sharedBuffer == 0) {
3158       size += bufferSize;
3159   }
3160
3161   if (client != NULL) {
3162        mCblkMemory = client->heap()->allocate(size);
3163        if (mCblkMemory != 0) {
3164            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3165            if (mCblk) { // construct the shared structure in-place.
3166                new(mCblk) audio_track_cblk_t();
3167                // clear all buffers
3168                mCblk->frameCount = frameCount;
3169                mCblk->sampleRate = sampleRate;
3170                mChannelCount = channelCount;
3171                mChannelMask = channelMask;
3172                if (sharedBuffer == 0) {
3173                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3174                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3175                    // Force underrun condition to avoid false underrun callback until first data is
3176                    // written to buffer (other flags are cleared)
3177                    mCblk->flags = CBLK_UNDERRUN_ON;
3178                } else {
3179                    mBuffer = sharedBuffer->pointer();
3180                }
3181                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3182            }
3183        } else {
3184            LOGE("not enough memory for AudioTrack size=%u", size);
3185            client->heap()->dump("AudioTrack");
3186            return;
3187        }
3188   } else {
3189       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3190       if (mCblk) { // construct the shared structure in-place.
3191           new(mCblk) audio_track_cblk_t();
3192           // clear all buffers
3193           mCblk->frameCount = frameCount;
3194           mCblk->sampleRate = sampleRate;
3195           mChannelCount = channelCount;
3196           mChannelMask = channelMask;
3197           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3198           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3199           // Force underrun condition to avoid false underrun callback until first data is
3200           // written to buffer (other flags are cleared)
3201           mCblk->flags = CBLK_UNDERRUN_ON;
3202           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3203       }
3204   }
3205}
3206
3207AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3208{
3209    if (mCblk) {
3210        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3211        if (mClient == NULL) {
3212            delete mCblk;
3213        }
3214    }
3215    mCblkMemory.clear();            // and free the shared memory
3216    if (mClient != NULL) {
3217        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3218        mClient.clear();
3219    }
3220}
3221
3222void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3223{
3224    buffer->raw = 0;
3225    mFrameCount = buffer->frameCount;
3226    step();
3227    buffer->frameCount = 0;
3228}
3229
3230bool AudioFlinger::ThreadBase::TrackBase::step() {
3231    bool result;
3232    audio_track_cblk_t* cblk = this->cblk();
3233
3234    result = cblk->stepServer(mFrameCount);
3235    if (!result) {
3236        ALOGV("stepServer failed acquiring cblk mutex");
3237        mFlags |= STEPSERVER_FAILED;
3238    }
3239    return result;
3240}
3241
3242void AudioFlinger::ThreadBase::TrackBase::reset() {
3243    audio_track_cblk_t* cblk = this->cblk();
3244
3245    cblk->user = 0;
3246    cblk->server = 0;
3247    cblk->userBase = 0;
3248    cblk->serverBase = 0;
3249    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3250    ALOGV("TrackBase::reset");
3251}
3252
3253sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3254{
3255    return mCblkMemory;
3256}
3257
3258int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3259    return (int)mCblk->sampleRate;
3260}
3261
3262int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3263    return (const int)mChannelCount;
3264}
3265
3266uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3267    return mChannelMask;
3268}
3269
3270void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3271    audio_track_cblk_t* cblk = this->cblk();
3272    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
3273    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
3274
3275    // Check validity of returned pointer in case the track control block would have been corrupted.
3276    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3277        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
3278        LOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3279                server %d, serverBase %d, user %d, userBase %d",
3280                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3281                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3282        return 0;
3283    }
3284
3285    return bufferStart;
3286}
3287
3288// ----------------------------------------------------------------------------
3289
3290// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3291AudioFlinger::PlaybackThread::Track::Track(
3292            const wp<ThreadBase>& thread,
3293            const sp<Client>& client,
3294            int streamType,
3295            uint32_t sampleRate,
3296            uint32_t format,
3297            uint32_t channelMask,
3298            int frameCount,
3299            const sp<IMemory>& sharedBuffer,
3300            int sessionId)
3301    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3302    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3303    mAuxEffectId(0), mHasVolumeController(false)
3304{
3305    if (mCblk != NULL) {
3306        sp<ThreadBase> baseThread = thread.promote();
3307        if (baseThread != 0) {
3308            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3309            mName = playbackThread->getTrackName_l();
3310            mMainBuffer = playbackThread->mixBuffer();
3311        }
3312        ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3313        if (mName < 0) {
3314            LOGE("no more track names available");
3315        }
3316        mVolume[0] = 1.0f;
3317        mVolume[1] = 1.0f;
3318        mStreamType = streamType;
3319        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3320        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3321        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3322    }
3323}
3324
3325AudioFlinger::PlaybackThread::Track::~Track()
3326{
3327    ALOGV("PlaybackThread::Track destructor");
3328    sp<ThreadBase> thread = mThread.promote();
3329    if (thread != 0) {
3330        Mutex::Autolock _l(thread->mLock);
3331        mState = TERMINATED;
3332    }
3333}
3334
3335void AudioFlinger::PlaybackThread::Track::destroy()
3336{
3337    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3338    // by removing it from mTracks vector, so there is a risk that this Tracks's
3339    // desctructor is called. As the destructor needs to lock mLock,
3340    // we must acquire a strong reference on this Track before locking mLock
3341    // here so that the destructor is called only when exiting this function.
3342    // On the other hand, as long as Track::destroy() is only called by
3343    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3344    // this Track with its member mTrack.
3345    sp<Track> keep(this);
3346    { // scope for mLock
3347        sp<ThreadBase> thread = mThread.promote();
3348        if (thread != 0) {
3349            if (!isOutputTrack()) {
3350                if (mState == ACTIVE || mState == RESUMING) {
3351                    AudioSystem::stopOutput(thread->id(),
3352                                            (audio_stream_type_t)mStreamType,
3353                                            mSessionId);
3354
3355                    // to track the speaker usage
3356                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3357                }
3358                AudioSystem::releaseOutput(thread->id());
3359            }
3360            Mutex::Autolock _l(thread->mLock);
3361            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3362            playbackThread->destroyTrack_l(this);
3363        }
3364    }
3365}
3366
3367void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3368{
3369    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",
3370            mName - AudioMixer::TRACK0,
3371            (mClient == NULL) ? getpid() : mClient->pid(),
3372            mStreamType,
3373            mFormat,
3374            mChannelMask,
3375            mSessionId,
3376            mFrameCount,
3377            mState,
3378            mMute,
3379            mFillingUpStatus,
3380            mCblk->sampleRate,
3381            mCblk->volume[0],
3382            mCblk->volume[1],
3383            mCblk->server,
3384            mCblk->user,
3385            (int)mMainBuffer,
3386            (int)mAuxBuffer);
3387}
3388
3389status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3390{
3391     audio_track_cblk_t* cblk = this->cblk();
3392     uint32_t framesReady;
3393     uint32_t framesReq = buffer->frameCount;
3394
3395     // Check if last stepServer failed, try to step now
3396     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3397         if (!step())  goto getNextBuffer_exit;
3398         ALOGV("stepServer recovered");
3399         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3400     }
3401
3402     framesReady = cblk->framesReady();
3403
3404     if (LIKELY(framesReady)) {
3405        uint32_t s = cblk->server;
3406        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3407
3408        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3409        if (framesReq > framesReady) {
3410            framesReq = framesReady;
3411        }
3412        if (s + framesReq > bufferEnd) {
3413            framesReq = bufferEnd - s;
3414        }
3415
3416         buffer->raw = getBuffer(s, framesReq);
3417         if (buffer->raw == 0) goto getNextBuffer_exit;
3418
3419         buffer->frameCount = framesReq;
3420        return NO_ERROR;
3421     }
3422
3423getNextBuffer_exit:
3424     buffer->raw = 0;
3425     buffer->frameCount = 0;
3426     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3427     return NOT_ENOUGH_DATA;
3428}
3429
3430bool AudioFlinger::PlaybackThread::Track::isReady() const {
3431    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3432
3433    if (mCblk->framesReady() >= mCblk->frameCount ||
3434            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3435        mFillingUpStatus = FS_FILLED;
3436        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3437        return true;
3438    }
3439    return false;
3440}
3441
3442status_t AudioFlinger::PlaybackThread::Track::start()
3443{
3444    status_t status = NO_ERROR;
3445    ALOGV("start(%d), calling thread %d session %d",
3446            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3447    sp<ThreadBase> thread = mThread.promote();
3448    if (thread != 0) {
3449        Mutex::Autolock _l(thread->mLock);
3450        int state = mState;
3451        // here the track could be either new, or restarted
3452        // in both cases "unstop" the track
3453        if (mState == PAUSED) {
3454            mState = TrackBase::RESUMING;
3455            ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3456        } else {
3457            mState = TrackBase::ACTIVE;
3458            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
3459        }
3460
3461        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3462            thread->mLock.unlock();
3463            status = AudioSystem::startOutput(thread->id(),
3464                                              (audio_stream_type_t)mStreamType,
3465                                              mSessionId);
3466            thread->mLock.lock();
3467
3468            // to track the speaker usage
3469            if (status == NO_ERROR) {
3470                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3471            }
3472        }
3473        if (status == NO_ERROR) {
3474            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3475            playbackThread->addTrack_l(this);
3476        } else {
3477            mState = state;
3478        }
3479    } else {
3480        status = BAD_VALUE;
3481    }
3482    return status;
3483}
3484
3485void AudioFlinger::PlaybackThread::Track::stop()
3486{
3487    ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3488    sp<ThreadBase> thread = mThread.promote();
3489    if (thread != 0) {
3490        Mutex::Autolock _l(thread->mLock);
3491        int state = mState;
3492        if (mState > STOPPED) {
3493            mState = STOPPED;
3494            // If the track is not active (PAUSED and buffers full), flush buffers
3495            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3496            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3497                reset();
3498            }
3499            ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3500        }
3501        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3502            thread->mLock.unlock();
3503            AudioSystem::stopOutput(thread->id(),
3504                                    (audio_stream_type_t)mStreamType,
3505                                    mSessionId);
3506            thread->mLock.lock();
3507
3508            // to track the speaker usage
3509            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3510        }
3511    }
3512}
3513
3514void AudioFlinger::PlaybackThread::Track::pause()
3515{
3516    ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3517    sp<ThreadBase> thread = mThread.promote();
3518    if (thread != 0) {
3519        Mutex::Autolock _l(thread->mLock);
3520        if (mState == ACTIVE || mState == RESUMING) {
3521            mState = PAUSING;
3522            ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3523            if (!isOutputTrack()) {
3524                thread->mLock.unlock();
3525                AudioSystem::stopOutput(thread->id(),
3526                                        (audio_stream_type_t)mStreamType,
3527                                        mSessionId);
3528                thread->mLock.lock();
3529
3530                // to track the speaker usage
3531                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3532            }
3533        }
3534    }
3535}
3536
3537void AudioFlinger::PlaybackThread::Track::flush()
3538{
3539    ALOGV("flush(%d)", mName);
3540    sp<ThreadBase> thread = mThread.promote();
3541    if (thread != 0) {
3542        Mutex::Autolock _l(thread->mLock);
3543        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3544            return;
3545        }
3546        // No point remaining in PAUSED state after a flush => go to
3547        // STOPPED state
3548        mState = STOPPED;
3549
3550        // do not reset the track if it is still in the process of being stopped or paused.
3551        // this will be done by prepareTracks_l() when the track is stopped.
3552        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3553        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3554            reset();
3555        }
3556    }
3557}
3558
3559void AudioFlinger::PlaybackThread::Track::reset()
3560{
3561    // Do not reset twice to avoid discarding data written just after a flush and before
3562    // the audioflinger thread detects the track is stopped.
3563    if (!mResetDone) {
3564        TrackBase::reset();
3565        // Force underrun condition to avoid false underrun callback until first data is
3566        // written to buffer
3567        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3568        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3569        mFillingUpStatus = FS_FILLING;
3570        mResetDone = true;
3571    }
3572}
3573
3574void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3575{
3576    mMute = muted;
3577}
3578
3579void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3580{
3581    mVolume[0] = left;
3582    mVolume[1] = right;
3583}
3584
3585status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3586{
3587    status_t status = DEAD_OBJECT;
3588    sp<ThreadBase> thread = mThread.promote();
3589    if (thread != 0) {
3590       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3591       status = playbackThread->attachAuxEffect(this, EffectId);
3592    }
3593    return status;
3594}
3595
3596void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3597{
3598    mAuxEffectId = EffectId;
3599    mAuxBuffer = buffer;
3600}
3601
3602// ----------------------------------------------------------------------------
3603
3604// RecordTrack constructor must be called with AudioFlinger::mLock held
3605AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3606            const wp<ThreadBase>& thread,
3607            const sp<Client>& client,
3608            uint32_t sampleRate,
3609            uint32_t format,
3610            uint32_t channelMask,
3611            int frameCount,
3612            uint32_t flags,
3613            int sessionId)
3614    :   TrackBase(thread, client, sampleRate, format,
3615                  channelMask, frameCount, flags, 0, sessionId),
3616        mOverflow(false)
3617{
3618    if (mCblk != NULL) {
3619       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3620       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3621           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3622       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3623           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3624       } else {
3625           mCblk->frameSize = sizeof(int8_t);
3626       }
3627    }
3628}
3629
3630AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3631{
3632    sp<ThreadBase> thread = mThread.promote();
3633    if (thread != 0) {
3634        AudioSystem::releaseInput(thread->id());
3635    }
3636}
3637
3638status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3639{
3640    audio_track_cblk_t* cblk = this->cblk();
3641    uint32_t framesAvail;
3642    uint32_t framesReq = buffer->frameCount;
3643
3644     // Check if last stepServer failed, try to step now
3645    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3646        if (!step()) goto getNextBuffer_exit;
3647        ALOGV("stepServer recovered");
3648        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3649    }
3650
3651    framesAvail = cblk->framesAvailable_l();
3652
3653    if (LIKELY(framesAvail)) {
3654        uint32_t s = cblk->server;
3655        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3656
3657        if (framesReq > framesAvail) {
3658            framesReq = framesAvail;
3659        }
3660        if (s + framesReq > bufferEnd) {
3661            framesReq = bufferEnd - s;
3662        }
3663
3664        buffer->raw = getBuffer(s, framesReq);
3665        if (buffer->raw == 0) goto getNextBuffer_exit;
3666
3667        buffer->frameCount = framesReq;
3668        return NO_ERROR;
3669    }
3670
3671getNextBuffer_exit:
3672    buffer->raw = 0;
3673    buffer->frameCount = 0;
3674    return NOT_ENOUGH_DATA;
3675}
3676
3677status_t AudioFlinger::RecordThread::RecordTrack::start()
3678{
3679    sp<ThreadBase> thread = mThread.promote();
3680    if (thread != 0) {
3681        RecordThread *recordThread = (RecordThread *)thread.get();
3682        return recordThread->start(this);
3683    } else {
3684        return BAD_VALUE;
3685    }
3686}
3687
3688void AudioFlinger::RecordThread::RecordTrack::stop()
3689{
3690    sp<ThreadBase> thread = mThread.promote();
3691    if (thread != 0) {
3692        RecordThread *recordThread = (RecordThread *)thread.get();
3693        recordThread->stop(this);
3694        TrackBase::reset();
3695        // Force overerrun condition to avoid false overrun callback until first data is
3696        // read from buffer
3697        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3698    }
3699}
3700
3701void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3702{
3703    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3704            (mClient == NULL) ? getpid() : mClient->pid(),
3705            mFormat,
3706            mChannelMask,
3707            mSessionId,
3708            mFrameCount,
3709            mState,
3710            mCblk->sampleRate,
3711            mCblk->server,
3712            mCblk->user);
3713}
3714
3715
3716// ----------------------------------------------------------------------------
3717
3718AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3719            const wp<ThreadBase>& thread,
3720            DuplicatingThread *sourceThread,
3721            uint32_t sampleRate,
3722            uint32_t format,
3723            uint32_t channelMask,
3724            int frameCount)
3725    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3726    mActive(false), mSourceThread(sourceThread)
3727{
3728
3729    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3730    if (mCblk != NULL) {
3731        mCblk->flags |= CBLK_DIRECTION_OUT;
3732        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3733        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
3734        mOutBuffer.frameCount = 0;
3735        playbackThread->mTracks.add(this);
3736        ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3737                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3738                mCblk, mBuffer, mCblk->buffers,
3739                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3740    } else {
3741        LOGW("Error creating output track on thread %p", playbackThread);
3742    }
3743}
3744
3745AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3746{
3747    clearBufferQueue();
3748}
3749
3750status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3751{
3752    status_t status = Track::start();
3753    if (status != NO_ERROR) {
3754        return status;
3755    }
3756
3757    mActive = true;
3758    mRetryCount = 127;
3759    return status;
3760}
3761
3762void AudioFlinger::PlaybackThread::OutputTrack::stop()
3763{
3764    Track::stop();
3765    clearBufferQueue();
3766    mOutBuffer.frameCount = 0;
3767    mActive = false;
3768}
3769
3770bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3771{
3772    Buffer *pInBuffer;
3773    Buffer inBuffer;
3774    uint32_t channelCount = mChannelCount;
3775    bool outputBufferFull = false;
3776    inBuffer.frameCount = frames;
3777    inBuffer.i16 = data;
3778
3779    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3780
3781    if (!mActive && frames != 0) {
3782        start();
3783        sp<ThreadBase> thread = mThread.promote();
3784        if (thread != 0) {
3785            MixerThread *mixerThread = (MixerThread *)thread.get();
3786            if (mCblk->frameCount > frames){
3787                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3788                    uint32_t startFrames = (mCblk->frameCount - frames);
3789                    pInBuffer = new Buffer;
3790                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3791                    pInBuffer->frameCount = startFrames;
3792                    pInBuffer->i16 = pInBuffer->mBuffer;
3793                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3794                    mBufferQueue.add(pInBuffer);
3795                } else {
3796                    LOGW ("OutputTrack::write() %p no more buffers in queue", this);
3797                }
3798            }
3799        }
3800    }
3801
3802    while (waitTimeLeftMs) {
3803        // First write pending buffers, then new data
3804        if (mBufferQueue.size()) {
3805            pInBuffer = mBufferQueue.itemAt(0);
3806        } else {
3807            pInBuffer = &inBuffer;
3808        }
3809
3810        if (pInBuffer->frameCount == 0) {
3811            break;
3812        }
3813
3814        if (mOutBuffer.frameCount == 0) {
3815            mOutBuffer.frameCount = pInBuffer->frameCount;
3816            nsecs_t startTime = systemTime();
3817            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
3818                ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3819                outputBufferFull = true;
3820                break;
3821            }
3822            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3823            if (waitTimeLeftMs >= waitTimeMs) {
3824                waitTimeLeftMs -= waitTimeMs;
3825            } else {
3826                waitTimeLeftMs = 0;
3827            }
3828        }
3829
3830        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3831        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3832        mCblk->stepUser(outFrames);
3833        pInBuffer->frameCount -= outFrames;
3834        pInBuffer->i16 += outFrames * channelCount;
3835        mOutBuffer.frameCount -= outFrames;
3836        mOutBuffer.i16 += outFrames * channelCount;
3837
3838        if (pInBuffer->frameCount == 0) {
3839            if (mBufferQueue.size()) {
3840                mBufferQueue.removeAt(0);
3841                delete [] pInBuffer->mBuffer;
3842                delete pInBuffer;
3843                ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3844            } else {
3845                break;
3846            }
3847        }
3848    }
3849
3850    // If we could not write all frames, allocate a buffer and queue it for next time.
3851    if (inBuffer.frameCount) {
3852        sp<ThreadBase> thread = mThread.promote();
3853        if (thread != 0 && !thread->standby()) {
3854            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3855                pInBuffer = new Buffer;
3856                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3857                pInBuffer->frameCount = inBuffer.frameCount;
3858                pInBuffer->i16 = pInBuffer->mBuffer;
3859                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3860                mBufferQueue.add(pInBuffer);
3861                ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3862            } else {
3863                LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3864            }
3865        }
3866    }
3867
3868    // Calling write() with a 0 length buffer, means that no more data will be written:
3869    // If no more buffers are pending, fill output track buffer to make sure it is started
3870    // by output mixer.
3871    if (frames == 0 && mBufferQueue.size() == 0) {
3872        if (mCblk->user < mCblk->frameCount) {
3873            frames = mCblk->frameCount - mCblk->user;
3874            pInBuffer = new Buffer;
3875            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3876            pInBuffer->frameCount = frames;
3877            pInBuffer->i16 = pInBuffer->mBuffer;
3878            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3879            mBufferQueue.add(pInBuffer);
3880        } else if (mActive) {
3881            stop();
3882        }
3883    }
3884
3885    return outputBufferFull;
3886}
3887
3888status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3889{
3890    int active;
3891    status_t result;
3892    audio_track_cblk_t* cblk = mCblk;
3893    uint32_t framesReq = buffer->frameCount;
3894
3895//    ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3896    buffer->frameCount  = 0;
3897
3898    uint32_t framesAvail = cblk->framesAvailable();
3899
3900
3901    if (framesAvail == 0) {
3902        Mutex::Autolock _l(cblk->lock);
3903        goto start_loop_here;
3904        while (framesAvail == 0) {
3905            active = mActive;
3906            if (UNLIKELY(!active)) {
3907                ALOGV("Not active and NO_MORE_BUFFERS");
3908                return AudioTrack::NO_MORE_BUFFERS;
3909            }
3910            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
3911            if (result != NO_ERROR) {
3912                return AudioTrack::NO_MORE_BUFFERS;
3913            }
3914            // read the server count again
3915        start_loop_here:
3916            framesAvail = cblk->framesAvailable_l();
3917        }
3918    }
3919
3920//    if (framesAvail < framesReq) {
3921//        return AudioTrack::NO_MORE_BUFFERS;
3922//    }
3923
3924    if (framesReq > framesAvail) {
3925        framesReq = framesAvail;
3926    }
3927
3928    uint32_t u = cblk->user;
3929    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
3930
3931    if (u + framesReq > bufferEnd) {
3932        framesReq = bufferEnd - u;
3933    }
3934
3935    buffer->frameCount  = framesReq;
3936    buffer->raw         = (void *)cblk->buffer(u);
3937    return NO_ERROR;
3938}
3939
3940
3941void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
3942{
3943    size_t size = mBufferQueue.size();
3944    Buffer *pBuffer;
3945
3946    for (size_t i = 0; i < size; i++) {
3947        pBuffer = mBufferQueue.itemAt(i);
3948        delete [] pBuffer->mBuffer;
3949        delete pBuffer;
3950    }
3951    mBufferQueue.clear();
3952}
3953
3954// ----------------------------------------------------------------------------
3955
3956AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
3957    :   RefBase(),
3958        mAudioFlinger(audioFlinger),
3959        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
3960        mPid(pid)
3961{
3962    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
3963}
3964
3965// Client destructor must be called with AudioFlinger::mLock held
3966AudioFlinger::Client::~Client()
3967{
3968    mAudioFlinger->removeClient_l(mPid);
3969}
3970
3971const sp<MemoryDealer>& AudioFlinger::Client::heap() const
3972{
3973    return mMemoryDealer;
3974}
3975
3976// ----------------------------------------------------------------------------
3977
3978AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
3979                                                     const sp<IAudioFlingerClient>& client,
3980                                                     pid_t pid)
3981    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
3982{
3983}
3984
3985AudioFlinger::NotificationClient::~NotificationClient()
3986{
3987    mClient.clear();
3988}
3989
3990void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
3991{
3992    sp<NotificationClient> keep(this);
3993    {
3994        mAudioFlinger->removeNotificationClient(mPid);
3995    }
3996}
3997
3998// ----------------------------------------------------------------------------
3999
4000AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
4001    : BnAudioTrack(),
4002      mTrack(track)
4003{
4004}
4005
4006AudioFlinger::TrackHandle::~TrackHandle() {
4007    // just stop the track on deletion, associated resources
4008    // will be freed from the main thread once all pending buffers have
4009    // been played. Unless it's not in the active track list, in which
4010    // case we free everything now...
4011    mTrack->destroy();
4012}
4013
4014status_t AudioFlinger::TrackHandle::start() {
4015    return mTrack->start();
4016}
4017
4018void AudioFlinger::TrackHandle::stop() {
4019    mTrack->stop();
4020}
4021
4022void AudioFlinger::TrackHandle::flush() {
4023    mTrack->flush();
4024}
4025
4026void AudioFlinger::TrackHandle::mute(bool e) {
4027    mTrack->mute(e);
4028}
4029
4030void AudioFlinger::TrackHandle::pause() {
4031    mTrack->pause();
4032}
4033
4034void AudioFlinger::TrackHandle::setVolume(float left, float right) {
4035    mTrack->setVolume(left, right);
4036}
4037
4038sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4039    return mTrack->getCblk();
4040}
4041
4042status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4043{
4044    return mTrack->attachAuxEffect(EffectId);
4045}
4046
4047status_t AudioFlinger::TrackHandle::onTransact(
4048    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4049{
4050    return BnAudioTrack::onTransact(code, data, reply, flags);
4051}
4052
4053// ----------------------------------------------------------------------------
4054
4055sp<IAudioRecord> AudioFlinger::openRecord(
4056        pid_t pid,
4057        int input,
4058        uint32_t sampleRate,
4059        uint32_t format,
4060        uint32_t channelMask,
4061        int frameCount,
4062        uint32_t flags,
4063        int *sessionId,
4064        status_t *status)
4065{
4066    sp<RecordThread::RecordTrack> recordTrack;
4067    sp<RecordHandle> recordHandle;
4068    sp<Client> client;
4069    wp<Client> wclient;
4070    status_t lStatus;
4071    RecordThread *thread;
4072    size_t inFrameCount;
4073    int lSessionId;
4074
4075    // check calling permissions
4076    if (!recordingAllowed()) {
4077        lStatus = PERMISSION_DENIED;
4078        goto Exit;
4079    }
4080
4081    // add client to list
4082    { // scope for mLock
4083        Mutex::Autolock _l(mLock);
4084        thread = checkRecordThread_l(input);
4085        if (thread == NULL) {
4086            lStatus = BAD_VALUE;
4087            goto Exit;
4088        }
4089
4090        wclient = mClients.valueFor(pid);
4091        if (wclient != NULL) {
4092            client = wclient.promote();
4093        } else {
4094            client = new Client(this, pid);
4095            mClients.add(pid, client);
4096        }
4097
4098        // If no audio session id is provided, create one here
4099        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4100            lSessionId = *sessionId;
4101        } else {
4102            lSessionId = nextUniqueId();
4103            if (sessionId != NULL) {
4104                *sessionId = lSessionId;
4105            }
4106        }
4107        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4108        recordTrack = thread->createRecordTrack_l(client,
4109                                                sampleRate,
4110                                                format,
4111                                                channelMask,
4112                                                frameCount,
4113                                                flags,
4114                                                lSessionId,
4115                                                &lStatus);
4116    }
4117    if (lStatus != NO_ERROR) {
4118        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4119        // destructor is called by the TrackBase destructor with mLock held
4120        client.clear();
4121        recordTrack.clear();
4122        goto Exit;
4123    }
4124
4125    // return to handle to client
4126    recordHandle = new RecordHandle(recordTrack);
4127    lStatus = NO_ERROR;
4128
4129Exit:
4130    if (status) {
4131        *status = lStatus;
4132    }
4133    return recordHandle;
4134}
4135
4136// ----------------------------------------------------------------------------
4137
4138AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4139    : BnAudioRecord(),
4140    mRecordTrack(recordTrack)
4141{
4142}
4143
4144AudioFlinger::RecordHandle::~RecordHandle() {
4145    stop();
4146}
4147
4148status_t AudioFlinger::RecordHandle::start() {
4149    ALOGV("RecordHandle::start()");
4150    return mRecordTrack->start();
4151}
4152
4153void AudioFlinger::RecordHandle::stop() {
4154    ALOGV("RecordHandle::stop()");
4155    mRecordTrack->stop();
4156}
4157
4158sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4159    return mRecordTrack->getCblk();
4160}
4161
4162status_t AudioFlinger::RecordHandle::onTransact(
4163    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4164{
4165    return BnAudioRecord::onTransact(code, data, reply, flags);
4166}
4167
4168// ----------------------------------------------------------------------------
4169
4170AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4171                                         AudioStreamIn *input,
4172                                         uint32_t sampleRate,
4173                                         uint32_t channels,
4174                                         int id,
4175                                         uint32_t device) :
4176    ThreadBase(audioFlinger, id, device),
4177    mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
4178{
4179    mType = ThreadBase::RECORD;
4180
4181    snprintf(mName, kNameLength, "AudioIn_%d", id);
4182
4183    mReqChannelCount = popcount(channels);
4184    mReqSampleRate = sampleRate;
4185    readInputParameters();
4186}
4187
4188
4189AudioFlinger::RecordThread::~RecordThread()
4190{
4191    delete[] mRsmpInBuffer;
4192    if (mResampler != 0) {
4193        delete mResampler;
4194        delete[] mRsmpOutBuffer;
4195    }
4196}
4197
4198void AudioFlinger::RecordThread::onFirstRef()
4199{
4200    run(mName, PRIORITY_URGENT_AUDIO);
4201}
4202
4203status_t AudioFlinger::RecordThread::readyToRun()
4204{
4205    status_t status = initCheck();
4206    LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4207    return status;
4208}
4209
4210bool AudioFlinger::RecordThread::threadLoop()
4211{
4212    AudioBufferProvider::Buffer buffer;
4213    sp<RecordTrack> activeTrack;
4214    Vector< sp<EffectChain> > effectChains;
4215
4216    nsecs_t lastWarning = 0;
4217
4218    acquireWakeLock();
4219
4220    // start recording
4221    while (!exitPending()) {
4222
4223        processConfigEvents();
4224
4225        { // scope for mLock
4226            Mutex::Autolock _l(mLock);
4227            checkForNewParameters_l();
4228            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4229                if (!mStandby) {
4230                    mInput->stream->common.standby(&mInput->stream->common);
4231                    mStandby = true;
4232                }
4233
4234                if (exitPending()) break;
4235
4236                releaseWakeLock_l();
4237                ALOGV("RecordThread: loop stopping");
4238                // go to sleep
4239                mWaitWorkCV.wait(mLock);
4240                ALOGV("RecordThread: loop starting");
4241                acquireWakeLock_l();
4242                continue;
4243            }
4244            if (mActiveTrack != 0) {
4245                if (mActiveTrack->mState == TrackBase::PAUSING) {
4246                    if (!mStandby) {
4247                        mInput->stream->common.standby(&mInput->stream->common);
4248                        mStandby = true;
4249                    }
4250                    mActiveTrack.clear();
4251                    mStartStopCond.broadcast();
4252                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4253                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4254                        mActiveTrack.clear();
4255                        mStartStopCond.broadcast();
4256                    } else if (mBytesRead != 0) {
4257                        // record start succeeds only if first read from audio input
4258                        // succeeds
4259                        if (mBytesRead > 0) {
4260                            mActiveTrack->mState = TrackBase::ACTIVE;
4261                        } else {
4262                            mActiveTrack.clear();
4263                        }
4264                        mStartStopCond.broadcast();
4265                    }
4266                    mStandby = false;
4267                }
4268            }
4269            lockEffectChains_l(effectChains);
4270        }
4271
4272        if (mActiveTrack != 0) {
4273            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4274                mActiveTrack->mState != TrackBase::RESUMING) {
4275                unlockEffectChains(effectChains);
4276                usleep(kRecordThreadSleepUs);
4277                continue;
4278            }
4279            for (size_t i = 0; i < effectChains.size(); i ++) {
4280                effectChains[i]->process_l();
4281            }
4282
4283            buffer.frameCount = mFrameCount;
4284            if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4285                size_t framesOut = buffer.frameCount;
4286                if (mResampler == 0) {
4287                    // no resampling
4288                    while (framesOut) {
4289                        size_t framesIn = mFrameCount - mRsmpInIndex;
4290                        if (framesIn) {
4291                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4292                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4293                            if (framesIn > framesOut)
4294                                framesIn = framesOut;
4295                            mRsmpInIndex += framesIn;
4296                            framesOut -= framesIn;
4297                            if ((int)mChannelCount == mReqChannelCount ||
4298                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4299                                memcpy(dst, src, framesIn * mFrameSize);
4300                            } else {
4301                                int16_t *src16 = (int16_t *)src;
4302                                int16_t *dst16 = (int16_t *)dst;
4303                                if (mChannelCount == 1) {
4304                                    while (framesIn--) {
4305                                        *dst16++ = *src16;
4306                                        *dst16++ = *src16++;
4307                                    }
4308                                } else {
4309                                    while (framesIn--) {
4310                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4311                                        src16 += 2;
4312                                    }
4313                                }
4314                            }
4315                        }
4316                        if (framesOut && mFrameCount == mRsmpInIndex) {
4317                            if (framesOut == mFrameCount &&
4318                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4319                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4320                                framesOut = 0;
4321                            } else {
4322                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4323                                mRsmpInIndex = 0;
4324                            }
4325                            if (mBytesRead < 0) {
4326                                LOGE("Error reading audio input");
4327                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4328                                    // Force input into standby so that it tries to
4329                                    // recover at next read attempt
4330                                    mInput->stream->common.standby(&mInput->stream->common);
4331                                    usleep(kRecordThreadSleepUs);
4332                                }
4333                                mRsmpInIndex = mFrameCount;
4334                                framesOut = 0;
4335                                buffer.frameCount = 0;
4336                            }
4337                        }
4338                    }
4339                } else {
4340                    // resampling
4341
4342                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4343                    // alter output frame count as if we were expecting stereo samples
4344                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4345                        framesOut >>= 1;
4346                    }
4347                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4348                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4349                    // are 32 bit aligned which should be always true.
4350                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4351                        AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4352                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4353                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4354                        int16_t *dst = buffer.i16;
4355                        while (framesOut--) {
4356                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4357                            src += 2;
4358                        }
4359                    } else {
4360                        AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4361                    }
4362
4363                }
4364                mActiveTrack->releaseBuffer(&buffer);
4365                mActiveTrack->overflow();
4366            }
4367            // client isn't retrieving buffers fast enough
4368            else {
4369                if (!mActiveTrack->setOverflow()) {
4370                    nsecs_t now = systemTime();
4371                    if ((now - lastWarning) > kWarningThrottle) {
4372                        LOGW("RecordThread: buffer overflow");
4373                        lastWarning = now;
4374                    }
4375                }
4376                // Release the processor for a while before asking for a new buffer.
4377                // This will give the application more chance to read from the buffer and
4378                // clear the overflow.
4379                usleep(kRecordThreadSleepUs);
4380            }
4381        }
4382        // enable changes in effect chain
4383        unlockEffectChains(effectChains);
4384        effectChains.clear();
4385    }
4386
4387    if (!mStandby) {
4388        mInput->stream->common.standby(&mInput->stream->common);
4389    }
4390    mActiveTrack.clear();
4391
4392    mStartStopCond.broadcast();
4393
4394    releaseWakeLock();
4395
4396    ALOGV("RecordThread %p exiting", this);
4397    return false;
4398}
4399
4400
4401sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4402        const sp<AudioFlinger::Client>& client,
4403        uint32_t sampleRate,
4404        int format,
4405        int channelMask,
4406        int frameCount,
4407        uint32_t flags,
4408        int sessionId,
4409        status_t *status)
4410{
4411    sp<RecordTrack> track;
4412    status_t lStatus;
4413
4414    lStatus = initCheck();
4415    if (lStatus != NO_ERROR) {
4416        LOGE("Audio driver not initialized.");
4417        goto Exit;
4418    }
4419
4420    { // scope for mLock
4421        Mutex::Autolock _l(mLock);
4422
4423        track = new RecordTrack(this, client, sampleRate,
4424                      format, channelMask, frameCount, flags, sessionId);
4425
4426        if (track->getCblk() == NULL) {
4427            lStatus = NO_MEMORY;
4428            goto Exit;
4429        }
4430
4431        mTrack = track.get();
4432        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4433        bool suspend = audio_is_bluetooth_sco_device(
4434                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4435        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4436        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4437    }
4438    lStatus = NO_ERROR;
4439
4440Exit:
4441    if (status) {
4442        *status = lStatus;
4443    }
4444    return track;
4445}
4446
4447status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4448{
4449    ALOGV("RecordThread::start");
4450    sp <ThreadBase> strongMe = this;
4451    status_t status = NO_ERROR;
4452    {
4453        AutoMutex lock(&mLock);
4454        if (mActiveTrack != 0) {
4455            if (recordTrack != mActiveTrack.get()) {
4456                status = -EBUSY;
4457            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4458                mActiveTrack->mState = TrackBase::ACTIVE;
4459            }
4460            return status;
4461        }
4462
4463        recordTrack->mState = TrackBase::IDLE;
4464        mActiveTrack = recordTrack;
4465        mLock.unlock();
4466        status_t status = AudioSystem::startInput(mId);
4467        mLock.lock();
4468        if (status != NO_ERROR) {
4469            mActiveTrack.clear();
4470            return status;
4471        }
4472        mRsmpInIndex = mFrameCount;
4473        mBytesRead = 0;
4474        if (mResampler != NULL) {
4475            mResampler->reset();
4476        }
4477        mActiveTrack->mState = TrackBase::RESUMING;
4478        // signal thread to start
4479        ALOGV("Signal record thread");
4480        mWaitWorkCV.signal();
4481        // do not wait for mStartStopCond if exiting
4482        if (mExiting) {
4483            mActiveTrack.clear();
4484            status = INVALID_OPERATION;
4485            goto startError;
4486        }
4487        mStartStopCond.wait(mLock);
4488        if (mActiveTrack == 0) {
4489            ALOGV("Record failed to start");
4490            status = BAD_VALUE;
4491            goto startError;
4492        }
4493        ALOGV("Record started OK");
4494        return status;
4495    }
4496startError:
4497    AudioSystem::stopInput(mId);
4498    return status;
4499}
4500
4501void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4502    ALOGV("RecordThread::stop");
4503    sp <ThreadBase> strongMe = this;
4504    {
4505        AutoMutex lock(&mLock);
4506        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4507            mActiveTrack->mState = TrackBase::PAUSING;
4508            // do not wait for mStartStopCond if exiting
4509            if (mExiting) {
4510                return;
4511            }
4512            mStartStopCond.wait(mLock);
4513            // if we have been restarted, recordTrack == mActiveTrack.get() here
4514            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4515                mLock.unlock();
4516                AudioSystem::stopInput(mId);
4517                mLock.lock();
4518                ALOGV("Record stopped OK");
4519            }
4520        }
4521    }
4522}
4523
4524status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4525{
4526    const size_t SIZE = 256;
4527    char buffer[SIZE];
4528    String8 result;
4529    pid_t pid = 0;
4530
4531    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4532    result.append(buffer);
4533
4534    if (mActiveTrack != 0) {
4535        result.append("Active Track:\n");
4536        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4537        mActiveTrack->dump(buffer, SIZE);
4538        result.append(buffer);
4539
4540        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4541        result.append(buffer);
4542        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4543        result.append(buffer);
4544        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
4545        result.append(buffer);
4546        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4547        result.append(buffer);
4548        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4549        result.append(buffer);
4550
4551
4552    } else {
4553        result.append("No record client\n");
4554    }
4555    write(fd, result.string(), result.size());
4556
4557    dumpBase(fd, args);
4558    dumpEffectChains(fd, args);
4559
4560    return NO_ERROR;
4561}
4562
4563status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4564{
4565    size_t framesReq = buffer->frameCount;
4566    size_t framesReady = mFrameCount - mRsmpInIndex;
4567    int channelCount;
4568
4569    if (framesReady == 0) {
4570        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4571        if (mBytesRead < 0) {
4572            LOGE("RecordThread::getNextBuffer() Error reading audio input");
4573            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4574                // Force input into standby so that it tries to
4575                // recover at next read attempt
4576                mInput->stream->common.standby(&mInput->stream->common);
4577                usleep(kRecordThreadSleepUs);
4578            }
4579            buffer->raw = 0;
4580            buffer->frameCount = 0;
4581            return NOT_ENOUGH_DATA;
4582        }
4583        mRsmpInIndex = 0;
4584        framesReady = mFrameCount;
4585    }
4586
4587    if (framesReq > framesReady) {
4588        framesReq = framesReady;
4589    }
4590
4591    if (mChannelCount == 1 && mReqChannelCount == 2) {
4592        channelCount = 1;
4593    } else {
4594        channelCount = 2;
4595    }
4596    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4597    buffer->frameCount = framesReq;
4598    return NO_ERROR;
4599}
4600
4601void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4602{
4603    mRsmpInIndex += buffer->frameCount;
4604    buffer->frameCount = 0;
4605}
4606
4607bool AudioFlinger::RecordThread::checkForNewParameters_l()
4608{
4609    bool reconfig = false;
4610
4611    while (!mNewParameters.isEmpty()) {
4612        status_t status = NO_ERROR;
4613        String8 keyValuePair = mNewParameters[0];
4614        AudioParameter param = AudioParameter(keyValuePair);
4615        int value;
4616        int reqFormat = mFormat;
4617        int reqSamplingRate = mReqSampleRate;
4618        int reqChannelCount = mReqChannelCount;
4619
4620        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4621            reqSamplingRate = value;
4622            reconfig = true;
4623        }
4624        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4625            reqFormat = value;
4626            reconfig = true;
4627        }
4628        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4629            reqChannelCount = popcount(value);
4630            reconfig = true;
4631        }
4632        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4633            // do not accept frame count changes if tracks are open as the track buffer
4634            // size depends on frame count and correct behavior would not be garantied
4635            // if frame count is changed after track creation
4636            if (mActiveTrack != 0) {
4637                status = INVALID_OPERATION;
4638            } else {
4639                reconfig = true;
4640            }
4641        }
4642        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4643            // forward device change to effects that have requested to be
4644            // aware of attached audio device.
4645            for (size_t i = 0; i < mEffectChains.size(); i++) {
4646                mEffectChains[i]->setDevice_l(value);
4647            }
4648            // store input device and output device but do not forward output device to audio HAL.
4649            // Note that status is ignored by the caller for output device
4650            // (see AudioFlinger::setParameters()
4651            if (value & AUDIO_DEVICE_OUT_ALL) {
4652                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4653                status = BAD_VALUE;
4654            } else {
4655                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4656                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4657                if (mTrack != NULL) {
4658                    bool suspend = audio_is_bluetooth_sco_device(
4659                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4660                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4661                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4662                }
4663            }
4664            mDevice |= (uint32_t)value;
4665        }
4666        if (status == NO_ERROR) {
4667            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4668            if (status == INVALID_OPERATION) {
4669               mInput->stream->common.standby(&mInput->stream->common);
4670               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4671            }
4672            if (reconfig) {
4673                if (status == BAD_VALUE &&
4674                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4675                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4676                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4677                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4678                    (reqChannelCount < 3)) {
4679                    status = NO_ERROR;
4680                }
4681                if (status == NO_ERROR) {
4682                    readInputParameters();
4683                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4684                }
4685            }
4686        }
4687
4688        mNewParameters.removeAt(0);
4689
4690        mParamStatus = status;
4691        mParamCond.signal();
4692        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4693        // already timed out waiting for the status and will never signal the condition.
4694        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
4695    }
4696    return reconfig;
4697}
4698
4699String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4700{
4701    char *s;
4702    String8 out_s8 = String8();
4703
4704    Mutex::Autolock _l(mLock);
4705    if (initCheck() != NO_ERROR) {
4706        return out_s8;
4707    }
4708
4709    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4710    out_s8 = String8(s);
4711    free(s);
4712    return out_s8;
4713}
4714
4715void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4716    AudioSystem::OutputDescriptor desc;
4717    void *param2 = 0;
4718
4719    switch (event) {
4720    case AudioSystem::INPUT_OPENED:
4721    case AudioSystem::INPUT_CONFIG_CHANGED:
4722        desc.channels = mChannelMask;
4723        desc.samplingRate = mSampleRate;
4724        desc.format = mFormat;
4725        desc.frameCount = mFrameCount;
4726        desc.latency = 0;
4727        param2 = &desc;
4728        break;
4729
4730    case AudioSystem::INPUT_CLOSED:
4731    default:
4732        break;
4733    }
4734    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4735}
4736
4737void AudioFlinger::RecordThread::readInputParameters()
4738{
4739    if (mRsmpInBuffer) delete mRsmpInBuffer;
4740    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4741    if (mResampler) delete mResampler;
4742    mResampler = 0;
4743
4744    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4745    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4746    mChannelCount = (uint16_t)popcount(mChannelMask);
4747    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4748    mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
4749    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4750    mFrameCount = mInputBytes / mFrameSize;
4751    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4752
4753    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4754    {
4755        int channelCount;
4756         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4757         // stereo to mono post process as the resampler always outputs stereo.
4758        if (mChannelCount == 1 && mReqChannelCount == 2) {
4759            channelCount = 1;
4760        } else {
4761            channelCount = 2;
4762        }
4763        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4764        mResampler->setSampleRate(mSampleRate);
4765        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4766        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4767
4768        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4769        if (mChannelCount == 1 && mReqChannelCount == 1) {
4770            mFrameCount >>= 1;
4771        }
4772
4773    }
4774    mRsmpInIndex = mFrameCount;
4775}
4776
4777unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4778{
4779    Mutex::Autolock _l(mLock);
4780    if (initCheck() != NO_ERROR) {
4781        return 0;
4782    }
4783
4784    return mInput->stream->get_input_frames_lost(mInput->stream);
4785}
4786
4787uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4788{
4789    Mutex::Autolock _l(mLock);
4790    uint32_t result = 0;
4791    if (getEffectChain_l(sessionId) != 0) {
4792        result = EFFECT_SESSION;
4793    }
4794
4795    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4796        result |= TRACK_SESSION;
4797    }
4798
4799    return result;
4800}
4801
4802AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4803{
4804    Mutex::Autolock _l(mLock);
4805    return mTrack;
4806}
4807
4808AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
4809{
4810    Mutex::Autolock _l(mLock);
4811    return mInput;
4812}
4813
4814AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4815{
4816    Mutex::Autolock _l(mLock);
4817    AudioStreamIn *input = mInput;
4818    mInput = NULL;
4819    return input;
4820}
4821
4822// this method must always be called either with ThreadBase mLock held or inside the thread loop
4823audio_stream_t* AudioFlinger::RecordThread::stream()
4824{
4825    if (mInput == NULL) {
4826        return NULL;
4827    }
4828    return &mInput->stream->common;
4829}
4830
4831
4832// ----------------------------------------------------------------------------
4833
4834int AudioFlinger::openOutput(uint32_t *pDevices,
4835                                uint32_t *pSamplingRate,
4836                                uint32_t *pFormat,
4837                                uint32_t *pChannels,
4838                                uint32_t *pLatencyMs,
4839                                uint32_t flags)
4840{
4841    status_t status;
4842    PlaybackThread *thread = NULL;
4843    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4844    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4845    uint32_t format = pFormat ? *pFormat : 0;
4846    uint32_t channels = pChannels ? *pChannels : 0;
4847    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4848    audio_stream_out_t *outStream;
4849    audio_hw_device_t *outHwDev;
4850
4851    ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4852            pDevices ? *pDevices : 0,
4853            samplingRate,
4854            format,
4855            channels,
4856            flags);
4857
4858    if (pDevices == NULL || *pDevices == 0) {
4859        return 0;
4860    }
4861
4862    Mutex::Autolock _l(mLock);
4863
4864    outHwDev = findSuitableHwDev_l(*pDevices);
4865    if (outHwDev == NULL)
4866        return 0;
4867
4868    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
4869                                          &channels, &samplingRate, &outStream);
4870    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4871            outStream,
4872            samplingRate,
4873            format,
4874            channels,
4875            status);
4876
4877    mHardwareStatus = AUDIO_HW_IDLE;
4878    if (outStream != NULL) {
4879        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4880        int id = nextUniqueId();
4881
4882        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4883            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4884            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4885            thread = new DirectOutputThread(this, output, id, *pDevices);
4886            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4887        } else {
4888            thread = new MixerThread(this, output, id, *pDevices);
4889            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4890        }
4891        mPlaybackThreads.add(id, thread);
4892
4893        if (pSamplingRate) *pSamplingRate = samplingRate;
4894        if (pFormat) *pFormat = format;
4895        if (pChannels) *pChannels = channels;
4896        if (pLatencyMs) *pLatencyMs = thread->latency();
4897
4898        // notify client processes of the new output creation
4899        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4900        return id;
4901    }
4902
4903    return 0;
4904}
4905
4906int AudioFlinger::openDuplicateOutput(int output1, int output2)
4907{
4908    Mutex::Autolock _l(mLock);
4909    MixerThread *thread1 = checkMixerThread_l(output1);
4910    MixerThread *thread2 = checkMixerThread_l(output2);
4911
4912    if (thread1 == NULL || thread2 == NULL) {
4913        LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
4914        return 0;
4915    }
4916
4917    int id = nextUniqueId();
4918    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
4919    thread->addOutputTrack(thread2);
4920    mPlaybackThreads.add(id, thread);
4921    // notify client processes of the new output creation
4922    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4923    return id;
4924}
4925
4926status_t AudioFlinger::closeOutput(int output)
4927{
4928    // keep strong reference on the playback thread so that
4929    // it is not destroyed while exit() is executed
4930    sp <PlaybackThread> thread;
4931    {
4932        Mutex::Autolock _l(mLock);
4933        thread = checkPlaybackThread_l(output);
4934        if (thread == NULL) {
4935            return BAD_VALUE;
4936        }
4937
4938        ALOGV("closeOutput() %d", output);
4939
4940        if (thread->type() == ThreadBase::MIXER) {
4941            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4942                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
4943                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
4944                    dupThread->removeOutputTrack((MixerThread *)thread.get());
4945                }
4946            }
4947        }
4948        void *param2 = 0;
4949        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
4950        mPlaybackThreads.removeItem(output);
4951    }
4952    thread->exit();
4953
4954    if (thread->type() != ThreadBase::DUPLICATING) {
4955        AudioStreamOut *out = thread->clearOutput();
4956        // from now on thread->mOutput is NULL
4957        out->hwDev->close_output_stream(out->hwDev, out->stream);
4958        delete out;
4959    }
4960    return NO_ERROR;
4961}
4962
4963status_t AudioFlinger::suspendOutput(int output)
4964{
4965    Mutex::Autolock _l(mLock);
4966    PlaybackThread *thread = checkPlaybackThread_l(output);
4967
4968    if (thread == NULL) {
4969        return BAD_VALUE;
4970    }
4971
4972    ALOGV("suspendOutput() %d", output);
4973    thread->suspend();
4974
4975    return NO_ERROR;
4976}
4977
4978status_t AudioFlinger::restoreOutput(int output)
4979{
4980    Mutex::Autolock _l(mLock);
4981    PlaybackThread *thread = checkPlaybackThread_l(output);
4982
4983    if (thread == NULL) {
4984        return BAD_VALUE;
4985    }
4986
4987    ALOGV("restoreOutput() %d", output);
4988
4989    thread->restore();
4990
4991    return NO_ERROR;
4992}
4993
4994int AudioFlinger::openInput(uint32_t *pDevices,
4995                                uint32_t *pSamplingRate,
4996                                uint32_t *pFormat,
4997                                uint32_t *pChannels,
4998                                uint32_t acoustics)
4999{
5000    status_t status;
5001    RecordThread *thread = NULL;
5002    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
5003    uint32_t format = pFormat ? *pFormat : 0;
5004    uint32_t channels = pChannels ? *pChannels : 0;
5005    uint32_t reqSamplingRate = samplingRate;
5006    uint32_t reqFormat = format;
5007    uint32_t reqChannels = channels;
5008    audio_stream_in_t *inStream;
5009    audio_hw_device_t *inHwDev;
5010
5011    if (pDevices == NULL || *pDevices == 0) {
5012        return 0;
5013    }
5014
5015    Mutex::Autolock _l(mLock);
5016
5017    inHwDev = findSuitableHwDev_l(*pDevices);
5018    if (inHwDev == NULL)
5019        return 0;
5020
5021    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5022                                        &channels, &samplingRate,
5023                                        (audio_in_acoustics_t)acoustics,
5024                                        &inStream);
5025    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5026            inStream,
5027            samplingRate,
5028            format,
5029            channels,
5030            acoustics,
5031            status);
5032
5033    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5034    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5035    // or stereo to mono conversions on 16 bit PCM inputs.
5036    if (inStream == NULL && status == BAD_VALUE &&
5037        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5038        (samplingRate <= 2 * reqSamplingRate) &&
5039        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5040        ALOGV("openInput() reopening with proposed sampling rate and channels");
5041        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5042                                            &channels, &samplingRate,
5043                                            (audio_in_acoustics_t)acoustics,
5044                                            &inStream);
5045    }
5046
5047    if (inStream != NULL) {
5048        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5049
5050        int id = nextUniqueId();
5051        // Start record thread
5052        // RecorThread require both input and output device indication to forward to audio
5053        // pre processing modules
5054        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5055        thread = new RecordThread(this,
5056                                  input,
5057                                  reqSamplingRate,
5058                                  reqChannels,
5059                                  id,
5060                                  device);
5061        mRecordThreads.add(id, thread);
5062        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
5063        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5064        if (pFormat) *pFormat = format;
5065        if (pChannels) *pChannels = reqChannels;
5066
5067        input->stream->common.standby(&input->stream->common);
5068
5069        // notify client processes of the new input creation
5070        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5071        return id;
5072    }
5073
5074    return 0;
5075}
5076
5077status_t AudioFlinger::closeInput(int input)
5078{
5079    // keep strong reference on the record thread so that
5080    // it is not destroyed while exit() is executed
5081    sp <RecordThread> thread;
5082    {
5083        Mutex::Autolock _l(mLock);
5084        thread = checkRecordThread_l(input);
5085        if (thread == NULL) {
5086            return BAD_VALUE;
5087        }
5088
5089        ALOGV("closeInput() %d", input);
5090        void *param2 = 0;
5091        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5092        mRecordThreads.removeItem(input);
5093    }
5094    thread->exit();
5095
5096    AudioStreamIn *in = thread->clearInput();
5097    // from now on thread->mInput is NULL
5098    in->hwDev->close_input_stream(in->hwDev, in->stream);
5099    delete in;
5100
5101    return NO_ERROR;
5102}
5103
5104status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
5105{
5106    Mutex::Autolock _l(mLock);
5107    MixerThread *dstThread = checkMixerThread_l(output);
5108    if (dstThread == NULL) {
5109        LOGW("setStreamOutput() bad output id %d", output);
5110        return BAD_VALUE;
5111    }
5112
5113    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
5114    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5115
5116    dstThread->setStreamValid(stream, true);
5117
5118    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5119        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5120        if (thread != dstThread &&
5121            thread->type() != ThreadBase::DIRECT) {
5122            MixerThread *srcThread = (MixerThread *)thread;
5123            srcThread->setStreamValid(stream, false);
5124            srcThread->invalidateTracks(stream);
5125        }
5126    }
5127
5128    return NO_ERROR;
5129}
5130
5131
5132int AudioFlinger::newAudioSessionId()
5133{
5134    return nextUniqueId();
5135}
5136
5137void AudioFlinger::acquireAudioSessionId(int audioSession)
5138{
5139    Mutex::Autolock _l(mLock);
5140    int caller = IPCThreadState::self()->getCallingPid();
5141    ALOGV("acquiring %d from %d", audioSession, caller);
5142    int num = mAudioSessionRefs.size();
5143    for (int i = 0; i< num; i++) {
5144        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5145        if (ref->sessionid == audioSession && ref->pid == caller) {
5146            ref->cnt++;
5147            ALOGV(" incremented refcount to %d", ref->cnt);
5148            return;
5149        }
5150    }
5151    AudioSessionRef *ref = new AudioSessionRef();
5152    ref->sessionid = audioSession;
5153    ref->pid = caller;
5154    ref->cnt = 1;
5155    mAudioSessionRefs.push(ref);
5156    ALOGV(" added new entry for %d", ref->sessionid);
5157}
5158
5159void AudioFlinger::releaseAudioSessionId(int audioSession)
5160{
5161    Mutex::Autolock _l(mLock);
5162    int caller = IPCThreadState::self()->getCallingPid();
5163    ALOGV("releasing %d from %d", audioSession, caller);
5164    int num = mAudioSessionRefs.size();
5165    for (int i = 0; i< num; i++) {
5166        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5167        if (ref->sessionid == audioSession && ref->pid == caller) {
5168            ref->cnt--;
5169            ALOGV(" decremented refcount to %d", ref->cnt);
5170            if (ref->cnt == 0) {
5171                mAudioSessionRefs.removeAt(i);
5172                delete ref;
5173                purgeStaleEffects_l();
5174            }
5175            return;
5176        }
5177    }
5178    LOGW("session id %d not found for pid %d", audioSession, caller);
5179}
5180
5181void AudioFlinger::purgeStaleEffects_l() {
5182
5183    ALOGV("purging stale effects");
5184
5185    Vector< sp<EffectChain> > chains;
5186
5187    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5188        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5189        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5190            sp<EffectChain> ec = t->mEffectChains[j];
5191            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5192                chains.push(ec);
5193            }
5194        }
5195    }
5196    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5197        sp<RecordThread> t = mRecordThreads.valueAt(i);
5198        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5199            sp<EffectChain> ec = t->mEffectChains[j];
5200            chains.push(ec);
5201        }
5202    }
5203
5204    for (size_t i = 0; i < chains.size(); i++) {
5205        sp<EffectChain> ec = chains[i];
5206        int sessionid = ec->sessionId();
5207        sp<ThreadBase> t = ec->mThread.promote();
5208        if (t == 0) {
5209            continue;
5210        }
5211        size_t numsessionrefs = mAudioSessionRefs.size();
5212        bool found = false;
5213        for (size_t k = 0; k < numsessionrefs; k++) {
5214            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5215            if (ref->sessionid == sessionid) {
5216                ALOGV(" session %d still exists for %d with %d refs",
5217                     sessionid, ref->pid, ref->cnt);
5218                found = true;
5219                break;
5220            }
5221        }
5222        if (!found) {
5223            // remove all effects from the chain
5224            while (ec->mEffects.size()) {
5225                sp<EffectModule> effect = ec->mEffects[0];
5226                effect->unPin();
5227                Mutex::Autolock _l (t->mLock);
5228                t->removeEffect_l(effect);
5229                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5230                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5231                    if (handle != 0) {
5232                        handle->mEffect.clear();
5233                        if (handle->mHasControl && handle->mEnabled) {
5234                            t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
5235                        }
5236                    }
5237                }
5238                AudioSystem::unregisterEffect(effect->id());
5239            }
5240        }
5241    }
5242    return;
5243}
5244
5245// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5246AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5247{
5248    PlaybackThread *thread = NULL;
5249    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5250        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5251    }
5252    return thread;
5253}
5254
5255// checkMixerThread_l() must be called with AudioFlinger::mLock held
5256AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5257{
5258    PlaybackThread *thread = checkPlaybackThread_l(output);
5259    if (thread != NULL) {
5260        if (thread->type() == ThreadBase::DIRECT) {
5261            thread = NULL;
5262        }
5263    }
5264    return (MixerThread *)thread;
5265}
5266
5267// checkRecordThread_l() must be called with AudioFlinger::mLock held
5268AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5269{
5270    RecordThread *thread = NULL;
5271    if (mRecordThreads.indexOfKey(input) >= 0) {
5272        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5273    }
5274    return thread;
5275}
5276
5277uint32_t AudioFlinger::nextUniqueId()
5278{
5279    return android_atomic_inc(&mNextUniqueId);
5280}
5281
5282AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5283{
5284    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5285        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5286        AudioStreamOut *output = thread->getOutput();
5287        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5288            return thread;
5289        }
5290    }
5291    return NULL;
5292}
5293
5294uint32_t AudioFlinger::primaryOutputDevice_l()
5295{
5296    PlaybackThread *thread = primaryPlaybackThread_l();
5297
5298    if (thread == NULL) {
5299        return 0;
5300    }
5301
5302    return thread->device();
5303}
5304
5305
5306// ----------------------------------------------------------------------------
5307//  Effect management
5308// ----------------------------------------------------------------------------
5309
5310
5311status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5312{
5313    Mutex::Autolock _l(mLock);
5314    return EffectQueryNumberEffects(numEffects);
5315}
5316
5317status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5318{
5319    Mutex::Autolock _l(mLock);
5320    return EffectQueryEffect(index, descriptor);
5321}
5322
5323status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5324{
5325    Mutex::Autolock _l(mLock);
5326    return EffectGetDescriptor(pUuid, descriptor);
5327}
5328
5329
5330sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5331        effect_descriptor_t *pDesc,
5332        const sp<IEffectClient>& effectClient,
5333        int32_t priority,
5334        int io,
5335        int sessionId,
5336        status_t *status,
5337        int *id,
5338        int *enabled)
5339{
5340    status_t lStatus = NO_ERROR;
5341    sp<EffectHandle> handle;
5342    effect_descriptor_t desc;
5343    sp<Client> client;
5344    wp<Client> wclient;
5345
5346    ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5347            pid, effectClient.get(), priority, sessionId, io);
5348
5349    if (pDesc == NULL) {
5350        lStatus = BAD_VALUE;
5351        goto Exit;
5352    }
5353
5354    // check audio settings permission for global effects
5355    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5356        lStatus = PERMISSION_DENIED;
5357        goto Exit;
5358    }
5359
5360    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5361    // that can only be created by audio policy manager (running in same process)
5362    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5363        lStatus = PERMISSION_DENIED;
5364        goto Exit;
5365    }
5366
5367    if (io == 0) {
5368        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5369            // output must be specified by AudioPolicyManager when using session
5370            // AUDIO_SESSION_OUTPUT_STAGE
5371            lStatus = BAD_VALUE;
5372            goto Exit;
5373        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5374            // if the output returned by getOutputForEffect() is removed before we lock the
5375            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5376            // and we will exit safely
5377            io = AudioSystem::getOutputForEffect(&desc);
5378        }
5379    }
5380
5381    {
5382        Mutex::Autolock _l(mLock);
5383
5384
5385        if (!EffectIsNullUuid(&pDesc->uuid)) {
5386            // if uuid is specified, request effect descriptor
5387            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5388            if (lStatus < 0) {
5389                LOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5390                goto Exit;
5391            }
5392        } else {
5393            // if uuid is not specified, look for an available implementation
5394            // of the required type in effect factory
5395            if (EffectIsNullUuid(&pDesc->type)) {
5396                LOGW("createEffect() no effect type");
5397                lStatus = BAD_VALUE;
5398                goto Exit;
5399            }
5400            uint32_t numEffects = 0;
5401            effect_descriptor_t d;
5402            d.flags = 0; // prevent compiler warning
5403            bool found = false;
5404
5405            lStatus = EffectQueryNumberEffects(&numEffects);
5406            if (lStatus < 0) {
5407                LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5408                goto Exit;
5409            }
5410            for (uint32_t i = 0; i < numEffects; i++) {
5411                lStatus = EffectQueryEffect(i, &desc);
5412                if (lStatus < 0) {
5413                    LOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5414                    continue;
5415                }
5416                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5417                    // If matching type found save effect descriptor. If the session is
5418                    // 0 and the effect is not auxiliary, continue enumeration in case
5419                    // an auxiliary version of this effect type is available
5420                    found = true;
5421                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5422                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5423                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5424                        break;
5425                    }
5426                }
5427            }
5428            if (!found) {
5429                lStatus = BAD_VALUE;
5430                LOGW("createEffect() effect not found");
5431                goto Exit;
5432            }
5433            // For same effect type, chose auxiliary version over insert version if
5434            // connect to output mix (Compliance to OpenSL ES)
5435            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5436                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5437                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5438            }
5439        }
5440
5441        // Do not allow auxiliary effects on a session different from 0 (output mix)
5442        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5443             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5444            lStatus = INVALID_OPERATION;
5445            goto Exit;
5446        }
5447
5448        // check recording permission for visualizer
5449        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5450            !recordingAllowed()) {
5451            lStatus = PERMISSION_DENIED;
5452            goto Exit;
5453        }
5454
5455        // return effect descriptor
5456        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5457
5458        // If output is not specified try to find a matching audio session ID in one of the
5459        // output threads.
5460        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5461        // because of code checking output when entering the function.
5462        // Note: io is never 0 when creating an effect on an input
5463        if (io == 0) {
5464             // look for the thread where the specified audio session is present
5465            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5466                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5467                    io = mPlaybackThreads.keyAt(i);
5468                    break;
5469                }
5470            }
5471            if (io == 0) {
5472               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5473                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5474                       io = mRecordThreads.keyAt(i);
5475                       break;
5476                   }
5477               }
5478            }
5479            // If no output thread contains the requested session ID, default to
5480            // first output. The effect chain will be moved to the correct output
5481            // thread when a track with the same session ID is created
5482            if (io == 0 && mPlaybackThreads.size()) {
5483                io = mPlaybackThreads.keyAt(0);
5484            }
5485            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
5486        }
5487        ThreadBase *thread = checkRecordThread_l(io);
5488        if (thread == NULL) {
5489            thread = checkPlaybackThread_l(io);
5490            if (thread == NULL) {
5491                LOGE("createEffect() unknown output thread");
5492                lStatus = BAD_VALUE;
5493                goto Exit;
5494            }
5495        }
5496
5497        wclient = mClients.valueFor(pid);
5498
5499        if (wclient != NULL) {
5500            client = wclient.promote();
5501        } else {
5502            client = new Client(this, pid);
5503            mClients.add(pid, client);
5504        }
5505
5506        // create effect on selected output thread
5507        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5508                &desc, enabled, &lStatus);
5509        if (handle != 0 && id != NULL) {
5510            *id = handle->id();
5511        }
5512    }
5513
5514Exit:
5515    if(status) {
5516        *status = lStatus;
5517    }
5518    return handle;
5519}
5520
5521status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5522{
5523    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5524            sessionId, srcOutput, dstOutput);
5525    Mutex::Autolock _l(mLock);
5526    if (srcOutput == dstOutput) {
5527        LOGW("moveEffects() same dst and src outputs %d", dstOutput);
5528        return NO_ERROR;
5529    }
5530    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5531    if (srcThread == NULL) {
5532        LOGW("moveEffects() bad srcOutput %d", srcOutput);
5533        return BAD_VALUE;
5534    }
5535    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5536    if (dstThread == NULL) {
5537        LOGW("moveEffects() bad dstOutput %d", dstOutput);
5538        return BAD_VALUE;
5539    }
5540
5541    Mutex::Autolock _dl(dstThread->mLock);
5542    Mutex::Autolock _sl(srcThread->mLock);
5543    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5544
5545    return NO_ERROR;
5546}
5547
5548// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5549status_t AudioFlinger::moveEffectChain_l(int sessionId,
5550                                   AudioFlinger::PlaybackThread *srcThread,
5551                                   AudioFlinger::PlaybackThread *dstThread,
5552                                   bool reRegister)
5553{
5554    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5555            sessionId, srcThread, dstThread);
5556
5557    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5558    if (chain == 0) {
5559        LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5560                sessionId, srcThread);
5561        return INVALID_OPERATION;
5562    }
5563
5564    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5565    // so that a new chain is created with correct parameters when first effect is added. This is
5566    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
5567    // removed.
5568    srcThread->removeEffectChain_l(chain);
5569
5570    // transfer all effects one by one so that new effect chain is created on new thread with
5571    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5572    int dstOutput = dstThread->id();
5573    sp<EffectChain> dstChain;
5574    uint32_t strategy = 0; // prevent compiler warning
5575    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5576    while (effect != 0) {
5577        srcThread->removeEffect_l(effect);
5578        dstThread->addEffect_l(effect);
5579        // removeEffect_l() has stopped the effect if it was active so it must be restarted
5580        if (effect->state() == EffectModule::ACTIVE ||
5581                effect->state() == EffectModule::STOPPING) {
5582            effect->start();
5583        }
5584        // if the move request is not received from audio policy manager, the effect must be
5585        // re-registered with the new strategy and output
5586        if (dstChain == 0) {
5587            dstChain = effect->chain().promote();
5588            if (dstChain == 0) {
5589                LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5590                srcThread->addEffect_l(effect);
5591                return NO_INIT;
5592            }
5593            strategy = dstChain->strategy();
5594        }
5595        if (reRegister) {
5596            AudioSystem::unregisterEffect(effect->id());
5597            AudioSystem::registerEffect(&effect->desc(),
5598                                        dstOutput,
5599                                        strategy,
5600                                        sessionId,
5601                                        effect->id());
5602        }
5603        effect = chain->getEffectFromId_l(0);
5604    }
5605
5606    return NO_ERROR;
5607}
5608
5609
5610// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5611sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5612        const sp<AudioFlinger::Client>& client,
5613        const sp<IEffectClient>& effectClient,
5614        int32_t priority,
5615        int sessionId,
5616        effect_descriptor_t *desc,
5617        int *enabled,
5618        status_t *status
5619        )
5620{
5621    sp<EffectModule> effect;
5622    sp<EffectHandle> handle;
5623    status_t lStatus;
5624    sp<EffectChain> chain;
5625    bool chainCreated = false;
5626    bool effectCreated = false;
5627    bool effectRegistered = false;
5628
5629    lStatus = initCheck();
5630    if (lStatus != NO_ERROR) {
5631        LOGW("createEffect_l() Audio driver not initialized.");
5632        goto Exit;
5633    }
5634
5635    // Do not allow effects with session ID 0 on direct output or duplicating threads
5636    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5637    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5638        LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5639                desc->name, sessionId);
5640        lStatus = BAD_VALUE;
5641        goto Exit;
5642    }
5643    // Only Pre processor effects are allowed on input threads and only on input threads
5644    if ((mType == RECORD &&
5645            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5646            (mType != RECORD &&
5647                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5648        LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5649                desc->name, desc->flags, mType);
5650        lStatus = BAD_VALUE;
5651        goto Exit;
5652    }
5653
5654    ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5655
5656    { // scope for mLock
5657        Mutex::Autolock _l(mLock);
5658
5659        // check for existing effect chain with the requested audio session
5660        chain = getEffectChain_l(sessionId);
5661        if (chain == 0) {
5662            // create a new chain for this session
5663            ALOGV("createEffect_l() new effect chain for session %d", sessionId);
5664            chain = new EffectChain(this, sessionId);
5665            addEffectChain_l(chain);
5666            chain->setStrategy(getStrategyForSession_l(sessionId));
5667            chainCreated = true;
5668        } else {
5669            effect = chain->getEffectFromDesc_l(desc);
5670        }
5671
5672        ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
5673
5674        if (effect == 0) {
5675            int id = mAudioFlinger->nextUniqueId();
5676            // Check CPU and memory usage
5677            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5678            if (lStatus != NO_ERROR) {
5679                goto Exit;
5680            }
5681            effectRegistered = true;
5682            // create a new effect module if none present in the chain
5683            effect = new EffectModule(this, chain, desc, id, sessionId);
5684            lStatus = effect->status();
5685            if (lStatus != NO_ERROR) {
5686                goto Exit;
5687            }
5688            lStatus = chain->addEffect_l(effect);
5689            if (lStatus != NO_ERROR) {
5690                goto Exit;
5691            }
5692            effectCreated = true;
5693
5694            effect->setDevice(mDevice);
5695            effect->setMode(mAudioFlinger->getMode());
5696        }
5697        // create effect handle and connect it to effect module
5698        handle = new EffectHandle(effect, client, effectClient, priority);
5699        lStatus = effect->addHandle(handle);
5700        if (enabled) {
5701            *enabled = (int)effect->isEnabled();
5702        }
5703    }
5704
5705Exit:
5706    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5707        Mutex::Autolock _l(mLock);
5708        if (effectCreated) {
5709            chain->removeEffect_l(effect);
5710        }
5711        if (effectRegistered) {
5712            AudioSystem::unregisterEffect(effect->id());
5713        }
5714        if (chainCreated) {
5715            removeEffectChain_l(chain);
5716        }
5717        handle.clear();
5718    }
5719
5720    if(status) {
5721        *status = lStatus;
5722    }
5723    return handle;
5724}
5725
5726sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5727{
5728    sp<EffectModule> effect;
5729
5730    sp<EffectChain> chain = getEffectChain_l(sessionId);
5731    if (chain != 0) {
5732        effect = chain->getEffectFromId_l(effectId);
5733    }
5734    return effect;
5735}
5736
5737// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5738// PlaybackThread::mLock held
5739status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5740{
5741    // check for existing effect chain with the requested audio session
5742    int sessionId = effect->sessionId();
5743    sp<EffectChain> chain = getEffectChain_l(sessionId);
5744    bool chainCreated = false;
5745
5746    if (chain == 0) {
5747        // create a new chain for this session
5748        ALOGV("addEffect_l() new effect chain for session %d", sessionId);
5749        chain = new EffectChain(this, sessionId);
5750        addEffectChain_l(chain);
5751        chain->setStrategy(getStrategyForSession_l(sessionId));
5752        chainCreated = true;
5753    }
5754    ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5755
5756    if (chain->getEffectFromId_l(effect->id()) != 0) {
5757        LOGW("addEffect_l() %p effect %s already present in chain %p",
5758                this, effect->desc().name, chain.get());
5759        return BAD_VALUE;
5760    }
5761
5762    status_t status = chain->addEffect_l(effect);
5763    if (status != NO_ERROR) {
5764        if (chainCreated) {
5765            removeEffectChain_l(chain);
5766        }
5767        return status;
5768    }
5769
5770    effect->setDevice(mDevice);
5771    effect->setMode(mAudioFlinger->getMode());
5772    return NO_ERROR;
5773}
5774
5775void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5776
5777    ALOGV("removeEffect_l() %p effect %p", this, effect.get());
5778    effect_descriptor_t desc = effect->desc();
5779    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5780        detachAuxEffect_l(effect->id());
5781    }
5782
5783    sp<EffectChain> chain = effect->chain().promote();
5784    if (chain != 0) {
5785        // remove effect chain if removing last effect
5786        if (chain->removeEffect_l(effect) == 0) {
5787            removeEffectChain_l(chain);
5788        }
5789    } else {
5790        LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5791    }
5792}
5793
5794void AudioFlinger::ThreadBase::lockEffectChains_l(
5795        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5796{
5797    effectChains = mEffectChains;
5798    for (size_t i = 0; i < mEffectChains.size(); i++) {
5799        mEffectChains[i]->lock();
5800    }
5801}
5802
5803void AudioFlinger::ThreadBase::unlockEffectChains(
5804        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5805{
5806    for (size_t i = 0; i < effectChains.size(); i++) {
5807        effectChains[i]->unlock();
5808    }
5809}
5810
5811sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5812{
5813    Mutex::Autolock _l(mLock);
5814    return getEffectChain_l(sessionId);
5815}
5816
5817sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5818{
5819    sp<EffectChain> chain;
5820
5821    size_t size = mEffectChains.size();
5822    for (size_t i = 0; i < size; i++) {
5823        if (mEffectChains[i]->sessionId() == sessionId) {
5824            chain = mEffectChains[i];
5825            break;
5826        }
5827    }
5828    return chain;
5829}
5830
5831void AudioFlinger::ThreadBase::setMode(uint32_t mode)
5832{
5833    Mutex::Autolock _l(mLock);
5834    size_t size = mEffectChains.size();
5835    for (size_t i = 0; i < size; i++) {
5836        mEffectChains[i]->setMode_l(mode);
5837    }
5838}
5839
5840void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5841                                                    const wp<EffectHandle>& handle,
5842                                                    bool unpiniflast) {
5843
5844    Mutex::Autolock _l(mLock);
5845    ALOGV("disconnectEffect() %p effect %p", this, effect.get());
5846    // delete the effect module if removing last handle on it
5847    if (effect->removeHandle(handle) == 0) {
5848        if (!effect->isPinned() || unpiniflast) {
5849            removeEffect_l(effect);
5850            AudioSystem::unregisterEffect(effect->id());
5851        }
5852    }
5853}
5854
5855status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5856{
5857    int session = chain->sessionId();
5858    int16_t *buffer = mMixBuffer;
5859    bool ownsBuffer = false;
5860
5861    ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5862    if (session > 0) {
5863        // Only one effect chain can be present in direct output thread and it uses
5864        // the mix buffer as input
5865        if (mType != DIRECT) {
5866            size_t numSamples = mFrameCount * mChannelCount;
5867            buffer = new int16_t[numSamples];
5868            memset(buffer, 0, numSamples * sizeof(int16_t));
5869            ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5870            ownsBuffer = true;
5871        }
5872
5873        // Attach all tracks with same session ID to this chain.
5874        for (size_t i = 0; i < mTracks.size(); ++i) {
5875            sp<Track> track = mTracks[i];
5876            if (session == track->sessionId()) {
5877                ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5878                track->setMainBuffer(buffer);
5879                chain->incTrackCnt();
5880            }
5881        }
5882
5883        // indicate all active tracks in the chain
5884        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5885            sp<Track> track = mActiveTracks[i].promote();
5886            if (track == 0) continue;
5887            if (session == track->sessionId()) {
5888                ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5889                chain->incActiveTrackCnt();
5890            }
5891        }
5892    }
5893
5894    chain->setInBuffer(buffer, ownsBuffer);
5895    chain->setOutBuffer(mMixBuffer);
5896    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5897    // chains list in order to be processed last as it contains output stage effects
5898    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5899    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5900    // after track specific effects and before output stage
5901    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5902    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5903    // Effect chain for other sessions are inserted at beginning of effect
5904    // chains list to be processed before output mix effects. Relative order between other
5905    // sessions is not important
5906    size_t size = mEffectChains.size();
5907    size_t i = 0;
5908    for (i = 0; i < size; i++) {
5909        if (mEffectChains[i]->sessionId() < session) break;
5910    }
5911    mEffectChains.insertAt(chain, i);
5912    checkSuspendOnAddEffectChain_l(chain);
5913
5914    return NO_ERROR;
5915}
5916
5917size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
5918{
5919    int session = chain->sessionId();
5920
5921    ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
5922
5923    for (size_t i = 0; i < mEffectChains.size(); i++) {
5924        if (chain == mEffectChains[i]) {
5925            mEffectChains.removeAt(i);
5926            // detach all active tracks from the chain
5927            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5928                sp<Track> track = mActiveTracks[i].promote();
5929                if (track == 0) continue;
5930                if (session == track->sessionId()) {
5931                    ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
5932                            chain.get(), session);
5933                    chain->decActiveTrackCnt();
5934                }
5935            }
5936
5937            // detach all tracks with same session ID from this chain
5938            for (size_t i = 0; i < mTracks.size(); ++i) {
5939                sp<Track> track = mTracks[i];
5940                if (session == track->sessionId()) {
5941                    track->setMainBuffer(mMixBuffer);
5942                    chain->decTrackCnt();
5943                }
5944            }
5945            break;
5946        }
5947    }
5948    return mEffectChains.size();
5949}
5950
5951status_t AudioFlinger::PlaybackThread::attachAuxEffect(
5952        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
5953{
5954    Mutex::Autolock _l(mLock);
5955    return attachAuxEffect_l(track, EffectId);
5956}
5957
5958status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
5959        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
5960{
5961    status_t status = NO_ERROR;
5962
5963    if (EffectId == 0) {
5964        track->setAuxBuffer(0, NULL);
5965    } else {
5966        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
5967        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
5968        if (effect != 0) {
5969            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5970                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
5971            } else {
5972                status = INVALID_OPERATION;
5973            }
5974        } else {
5975            status = BAD_VALUE;
5976        }
5977    }
5978    return status;
5979}
5980
5981void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
5982{
5983     for (size_t i = 0; i < mTracks.size(); ++i) {
5984        sp<Track> track = mTracks[i];
5985        if (track->auxEffectId() == effectId) {
5986            attachAuxEffect_l(track, 0);
5987        }
5988    }
5989}
5990
5991status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
5992{
5993    // only one chain per input thread
5994    if (mEffectChains.size() != 0) {
5995        return INVALID_OPERATION;
5996    }
5997    ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
5998
5999    chain->setInBuffer(NULL);
6000    chain->setOutBuffer(NULL);
6001
6002    checkSuspendOnAddEffectChain_l(chain);
6003
6004    mEffectChains.add(chain);
6005
6006    return NO_ERROR;
6007}
6008
6009size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
6010{
6011    ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
6012    LOGW_IF(mEffectChains.size() != 1,
6013            "removeEffectChain_l() %p invalid chain size %d on thread %p",
6014            chain.get(), mEffectChains.size(), this);
6015    if (mEffectChains.size() == 1) {
6016        mEffectChains.removeAt(0);
6017    }
6018    return 0;
6019}
6020
6021// ----------------------------------------------------------------------------
6022//  EffectModule implementation
6023// ----------------------------------------------------------------------------
6024
6025#undef LOG_TAG
6026#define LOG_TAG "AudioFlinger::EffectModule"
6027
6028AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6029                                        const wp<AudioFlinger::EffectChain>& chain,
6030                                        effect_descriptor_t *desc,
6031                                        int id,
6032                                        int sessionId)
6033    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6034      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6035{
6036    ALOGV("Constructor %p", this);
6037    int lStatus;
6038    sp<ThreadBase> thread = mThread.promote();
6039    if (thread == 0) {
6040        return;
6041    }
6042
6043    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6044
6045    // create effect engine from effect factory
6046    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6047
6048    if (mStatus != NO_ERROR) {
6049        return;
6050    }
6051    lStatus = init();
6052    if (lStatus < 0) {
6053        mStatus = lStatus;
6054        goto Error;
6055    }
6056
6057    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6058        mPinned = true;
6059    }
6060    ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6061    return;
6062Error:
6063    EffectRelease(mEffectInterface);
6064    mEffectInterface = NULL;
6065    ALOGV("Constructor Error %d", mStatus);
6066}
6067
6068AudioFlinger::EffectModule::~EffectModule()
6069{
6070    ALOGV("Destructor %p", this);
6071    if (mEffectInterface != NULL) {
6072        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6073                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6074            sp<ThreadBase> thread = mThread.promote();
6075            if (thread != 0) {
6076                audio_stream_t *stream = thread->stream();
6077                if (stream != NULL) {
6078                    stream->remove_audio_effect(stream, mEffectInterface);
6079                }
6080            }
6081        }
6082        // release effect engine
6083        EffectRelease(mEffectInterface);
6084    }
6085}
6086
6087status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6088{
6089    status_t status;
6090
6091    Mutex::Autolock _l(mLock);
6092    // First handle in mHandles has highest priority and controls the effect module
6093    int priority = handle->priority();
6094    size_t size = mHandles.size();
6095    sp<EffectHandle> h;
6096    size_t i;
6097    for (i = 0; i < size; i++) {
6098        h = mHandles[i].promote();
6099        if (h == 0) continue;
6100        if (h->priority() <= priority) break;
6101    }
6102    // if inserted in first place, move effect control from previous owner to this handle
6103    if (i == 0) {
6104        bool enabled = false;
6105        if (h != 0) {
6106            enabled = h->enabled();
6107            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6108        }
6109        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6110        status = NO_ERROR;
6111    } else {
6112        status = ALREADY_EXISTS;
6113    }
6114    ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6115    mHandles.insertAt(handle, i);
6116    return status;
6117}
6118
6119size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6120{
6121    Mutex::Autolock _l(mLock);
6122    size_t size = mHandles.size();
6123    size_t i;
6124    for (i = 0; i < size; i++) {
6125        if (mHandles[i] == handle) break;
6126    }
6127    if (i == size) {
6128        return size;
6129    }
6130    ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6131
6132    bool enabled = false;
6133    EffectHandle *hdl = handle.unsafe_get();
6134    if (hdl) {
6135        ALOGV("removeHandle() unsafe_get OK");
6136        enabled = hdl->enabled();
6137    }
6138    mHandles.removeAt(i);
6139    size = mHandles.size();
6140    // if removed from first place, move effect control from this handle to next in line
6141    if (i == 0 && size != 0) {
6142        sp<EffectHandle> h = mHandles[0].promote();
6143        if (h != 0) {
6144            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6145        }
6146    }
6147
6148    // Prevent calls to process() and other functions on effect interface from now on.
6149    // The effect engine will be released by the destructor when the last strong reference on
6150    // this object is released which can happen after next process is called.
6151    if (size == 0 && !mPinned) {
6152        mState = DESTROYED;
6153    }
6154
6155    return size;
6156}
6157
6158sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6159{
6160    Mutex::Autolock _l(mLock);
6161    sp<EffectHandle> handle;
6162    if (mHandles.size() != 0) {
6163        handle = mHandles[0].promote();
6164    }
6165    return handle;
6166}
6167
6168void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6169{
6170    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6171    // keep a strong reference on this EffectModule to avoid calling the
6172    // destructor before we exit
6173    sp<EffectModule> keep(this);
6174    {
6175        sp<ThreadBase> thread = mThread.promote();
6176        if (thread != 0) {
6177            thread->disconnectEffect(keep, handle, unpiniflast);
6178        }
6179    }
6180}
6181
6182void AudioFlinger::EffectModule::updateState() {
6183    Mutex::Autolock _l(mLock);
6184
6185    switch (mState) {
6186    case RESTART:
6187        reset_l();
6188        // FALL THROUGH
6189
6190    case STARTING:
6191        // clear auxiliary effect input buffer for next accumulation
6192        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6193            memset(mConfig.inputCfg.buffer.raw,
6194                   0,
6195                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6196        }
6197        start_l();
6198        mState = ACTIVE;
6199        break;
6200    case STOPPING:
6201        stop_l();
6202        mDisableWaitCnt = mMaxDisableWaitCnt;
6203        mState = STOPPED;
6204        break;
6205    case STOPPED:
6206        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6207        // turn off sequence.
6208        if (--mDisableWaitCnt == 0) {
6209            reset_l();
6210            mState = IDLE;
6211        }
6212        break;
6213    default: //IDLE , ACTIVE, DESTROYED
6214        break;
6215    }
6216}
6217
6218void AudioFlinger::EffectModule::process()
6219{
6220    Mutex::Autolock _l(mLock);
6221
6222    if (mState == DESTROYED || mEffectInterface == NULL ||
6223            mConfig.inputCfg.buffer.raw == NULL ||
6224            mConfig.outputCfg.buffer.raw == NULL) {
6225        return;
6226    }
6227
6228    if (isProcessEnabled()) {
6229        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6230        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6231            AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
6232                                        mConfig.inputCfg.buffer.s32,
6233                                        mConfig.inputCfg.buffer.frameCount/2);
6234        }
6235
6236        // do the actual processing in the effect engine
6237        int ret = (*mEffectInterface)->process(mEffectInterface,
6238                                               &mConfig.inputCfg.buffer,
6239                                               &mConfig.outputCfg.buffer);
6240
6241        // force transition to IDLE state when engine is ready
6242        if (mState == STOPPED && ret == -ENODATA) {
6243            mDisableWaitCnt = 1;
6244        }
6245
6246        // clear auxiliary effect input buffer for next accumulation
6247        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6248            memset(mConfig.inputCfg.buffer.raw, 0,
6249                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6250        }
6251    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6252                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6253        // If an insert effect is idle and input buffer is different from output buffer,
6254        // accumulate input onto output
6255        sp<EffectChain> chain = mChain.promote();
6256        if (chain != 0 && chain->activeTrackCnt() != 0) {
6257            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6258            int16_t *in = mConfig.inputCfg.buffer.s16;
6259            int16_t *out = mConfig.outputCfg.buffer.s16;
6260            for (size_t i = 0; i < frameCnt; i++) {
6261                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6262            }
6263        }
6264    }
6265}
6266
6267void AudioFlinger::EffectModule::reset_l()
6268{
6269    if (mEffectInterface == NULL) {
6270        return;
6271    }
6272    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6273}
6274
6275status_t AudioFlinger::EffectModule::configure()
6276{
6277    uint32_t channels;
6278    if (mEffectInterface == NULL) {
6279        return NO_INIT;
6280    }
6281
6282    sp<ThreadBase> thread = mThread.promote();
6283    if (thread == 0) {
6284        return DEAD_OBJECT;
6285    }
6286
6287    // TODO: handle configuration of effects replacing track process
6288    if (thread->channelCount() == 1) {
6289        channels = AUDIO_CHANNEL_OUT_MONO;
6290    } else {
6291        channels = AUDIO_CHANNEL_OUT_STEREO;
6292    }
6293
6294    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6295        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6296    } else {
6297        mConfig.inputCfg.channels = channels;
6298    }
6299    mConfig.outputCfg.channels = channels;
6300    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6301    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6302    mConfig.inputCfg.samplingRate = thread->sampleRate();
6303    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6304    mConfig.inputCfg.bufferProvider.cookie = NULL;
6305    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6306    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6307    mConfig.outputCfg.bufferProvider.cookie = NULL;
6308    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6309    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6310    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6311    // Insert effect:
6312    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6313    // always overwrites output buffer: input buffer == output buffer
6314    // - in other sessions:
6315    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6316    //      other effect: overwrites output buffer: input buffer == output buffer
6317    // Auxiliary effect:
6318    //      accumulates in output buffer: input buffer != output buffer
6319    // Therefore: accumulate <=> input buffer != output buffer
6320    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6321        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6322    } else {
6323        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6324    }
6325    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6326    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6327    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6328    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6329
6330    ALOGV("configure() %p thread %p buffer %p framecount %d",
6331            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6332
6333    status_t cmdStatus;
6334    uint32_t size = sizeof(int);
6335    status_t status = (*mEffectInterface)->command(mEffectInterface,
6336                                                   EFFECT_CMD_CONFIGURE,
6337                                                   sizeof(effect_config_t),
6338                                                   &mConfig,
6339                                                   &size,
6340                                                   &cmdStatus);
6341    if (status == 0) {
6342        status = cmdStatus;
6343    }
6344
6345    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6346            (1000 * mConfig.outputCfg.buffer.frameCount);
6347
6348    return status;
6349}
6350
6351status_t AudioFlinger::EffectModule::init()
6352{
6353    Mutex::Autolock _l(mLock);
6354    if (mEffectInterface == NULL) {
6355        return NO_INIT;
6356    }
6357    status_t cmdStatus;
6358    uint32_t size = sizeof(status_t);
6359    status_t status = (*mEffectInterface)->command(mEffectInterface,
6360                                                   EFFECT_CMD_INIT,
6361                                                   0,
6362                                                   NULL,
6363                                                   &size,
6364                                                   &cmdStatus);
6365    if (status == 0) {
6366        status = cmdStatus;
6367    }
6368    return status;
6369}
6370
6371status_t AudioFlinger::EffectModule::start()
6372{
6373    Mutex::Autolock _l(mLock);
6374    return start_l();
6375}
6376
6377status_t AudioFlinger::EffectModule::start_l()
6378{
6379    if (mEffectInterface == NULL) {
6380        return NO_INIT;
6381    }
6382    status_t cmdStatus;
6383    uint32_t size = sizeof(status_t);
6384    status_t status = (*mEffectInterface)->command(mEffectInterface,
6385                                                   EFFECT_CMD_ENABLE,
6386                                                   0,
6387                                                   NULL,
6388                                                   &size,
6389                                                   &cmdStatus);
6390    if (status == 0) {
6391        status = cmdStatus;
6392    }
6393    if (status == 0 &&
6394            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6395             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6396        sp<ThreadBase> thread = mThread.promote();
6397        if (thread != 0) {
6398            audio_stream_t *stream = thread->stream();
6399            if (stream != NULL) {
6400                stream->add_audio_effect(stream, mEffectInterface);
6401            }
6402        }
6403    }
6404    return status;
6405}
6406
6407status_t AudioFlinger::EffectModule::stop()
6408{
6409    Mutex::Autolock _l(mLock);
6410    return stop_l();
6411}
6412
6413status_t AudioFlinger::EffectModule::stop_l()
6414{
6415    if (mEffectInterface == NULL) {
6416        return NO_INIT;
6417    }
6418    status_t cmdStatus;
6419    uint32_t size = sizeof(status_t);
6420    status_t status = (*mEffectInterface)->command(mEffectInterface,
6421                                                   EFFECT_CMD_DISABLE,
6422                                                   0,
6423                                                   NULL,
6424                                                   &size,
6425                                                   &cmdStatus);
6426    if (status == 0) {
6427        status = cmdStatus;
6428    }
6429    if (status == 0 &&
6430            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6431             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6432        sp<ThreadBase> thread = mThread.promote();
6433        if (thread != 0) {
6434            audio_stream_t *stream = thread->stream();
6435            if (stream != NULL) {
6436                stream->remove_audio_effect(stream, mEffectInterface);
6437            }
6438        }
6439    }
6440    return status;
6441}
6442
6443status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6444                                             uint32_t cmdSize,
6445                                             void *pCmdData,
6446                                             uint32_t *replySize,
6447                                             void *pReplyData)
6448{
6449    Mutex::Autolock _l(mLock);
6450//    ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6451
6452    if (mState == DESTROYED || mEffectInterface == NULL) {
6453        return NO_INIT;
6454    }
6455    status_t status = (*mEffectInterface)->command(mEffectInterface,
6456                                                   cmdCode,
6457                                                   cmdSize,
6458                                                   pCmdData,
6459                                                   replySize,
6460                                                   pReplyData);
6461    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6462        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6463        for (size_t i = 1; i < mHandles.size(); i++) {
6464            sp<EffectHandle> h = mHandles[i].promote();
6465            if (h != 0) {
6466                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6467            }
6468        }
6469    }
6470    return status;
6471}
6472
6473status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6474{
6475
6476    Mutex::Autolock _l(mLock);
6477    ALOGV("setEnabled %p enabled %d", this, enabled);
6478
6479    if (enabled != isEnabled()) {
6480        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6481        if (enabled && status != NO_ERROR) {
6482            return status;
6483        }
6484
6485        switch (mState) {
6486        // going from disabled to enabled
6487        case IDLE:
6488            mState = STARTING;
6489            break;
6490        case STOPPED:
6491            mState = RESTART;
6492            break;
6493        case STOPPING:
6494            mState = ACTIVE;
6495            break;
6496
6497        // going from enabled to disabled
6498        case RESTART:
6499            mState = STOPPED;
6500            break;
6501        case STARTING:
6502            mState = IDLE;
6503            break;
6504        case ACTIVE:
6505            mState = STOPPING;
6506            break;
6507        case DESTROYED:
6508            return NO_ERROR; // simply ignore as we are being destroyed
6509        }
6510        for (size_t i = 1; i < mHandles.size(); i++) {
6511            sp<EffectHandle> h = mHandles[i].promote();
6512            if (h != 0) {
6513                h->setEnabled(enabled);
6514            }
6515        }
6516    }
6517    return NO_ERROR;
6518}
6519
6520bool AudioFlinger::EffectModule::isEnabled()
6521{
6522    switch (mState) {
6523    case RESTART:
6524    case STARTING:
6525    case ACTIVE:
6526        return true;
6527    case IDLE:
6528    case STOPPING:
6529    case STOPPED:
6530    case DESTROYED:
6531    default:
6532        return false;
6533    }
6534}
6535
6536bool AudioFlinger::EffectModule::isProcessEnabled()
6537{
6538    switch (mState) {
6539    case RESTART:
6540    case ACTIVE:
6541    case STOPPING:
6542    case STOPPED:
6543        return true;
6544    case IDLE:
6545    case STARTING:
6546    case DESTROYED:
6547    default:
6548        return false;
6549    }
6550}
6551
6552status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6553{
6554    Mutex::Autolock _l(mLock);
6555    status_t status = NO_ERROR;
6556
6557    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6558    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6559    if (isProcessEnabled() &&
6560            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6561            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6562        status_t cmdStatus;
6563        uint32_t volume[2];
6564        uint32_t *pVolume = NULL;
6565        uint32_t size = sizeof(volume);
6566        volume[0] = *left;
6567        volume[1] = *right;
6568        if (controller) {
6569            pVolume = volume;
6570        }
6571        status = (*mEffectInterface)->command(mEffectInterface,
6572                                              EFFECT_CMD_SET_VOLUME,
6573                                              size,
6574                                              volume,
6575                                              &size,
6576                                              pVolume);
6577        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6578            *left = volume[0];
6579            *right = volume[1];
6580        }
6581    }
6582    return status;
6583}
6584
6585status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6586{
6587    Mutex::Autolock _l(mLock);
6588    status_t status = NO_ERROR;
6589    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6590        // audio pre processing modules on RecordThread can receive both output and
6591        // input device indication in the same call
6592        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6593        if (dev) {
6594            status_t cmdStatus;
6595            uint32_t size = sizeof(status_t);
6596
6597            status = (*mEffectInterface)->command(mEffectInterface,
6598                                                  EFFECT_CMD_SET_DEVICE,
6599                                                  sizeof(uint32_t),
6600                                                  &dev,
6601                                                  &size,
6602                                                  &cmdStatus);
6603            if (status == NO_ERROR) {
6604                status = cmdStatus;
6605            }
6606        }
6607        dev = device & AUDIO_DEVICE_IN_ALL;
6608        if (dev) {
6609            status_t cmdStatus;
6610            uint32_t size = sizeof(status_t);
6611
6612            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6613                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6614                                                  sizeof(uint32_t),
6615                                                  &dev,
6616                                                  &size,
6617                                                  &cmdStatus);
6618            if (status2 == NO_ERROR) {
6619                status2 = cmdStatus;
6620            }
6621            if (status == NO_ERROR) {
6622                status = status2;
6623            }
6624        }
6625    }
6626    return status;
6627}
6628
6629status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
6630{
6631    Mutex::Autolock _l(mLock);
6632    status_t status = NO_ERROR;
6633    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6634        status_t cmdStatus;
6635        uint32_t size = sizeof(status_t);
6636        status = (*mEffectInterface)->command(mEffectInterface,
6637                                              EFFECT_CMD_SET_AUDIO_MODE,
6638                                              sizeof(int),
6639                                              &mode,
6640                                              &size,
6641                                              &cmdStatus);
6642        if (status == NO_ERROR) {
6643            status = cmdStatus;
6644        }
6645    }
6646    return status;
6647}
6648
6649void AudioFlinger::EffectModule::setSuspended(bool suspended)
6650{
6651    Mutex::Autolock _l(mLock);
6652    mSuspended = suspended;
6653}
6654bool AudioFlinger::EffectModule::suspended()
6655{
6656    Mutex::Autolock _l(mLock);
6657    return mSuspended;
6658}
6659
6660status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6661{
6662    const size_t SIZE = 256;
6663    char buffer[SIZE];
6664    String8 result;
6665
6666    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6667    result.append(buffer);
6668
6669    bool locked = tryLock(mLock);
6670    // failed to lock - AudioFlinger is probably deadlocked
6671    if (!locked) {
6672        result.append("\t\tCould not lock Fx mutex:\n");
6673    }
6674
6675    result.append("\t\tSession Status State Engine:\n");
6676    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6677            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6678    result.append(buffer);
6679
6680    result.append("\t\tDescriptor:\n");
6681    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6682            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6683            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6684            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6685    result.append(buffer);
6686    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6687                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6688                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6689                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6690    result.append(buffer);
6691    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6692            mDescriptor.apiVersion,
6693            mDescriptor.flags);
6694    result.append(buffer);
6695    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6696            mDescriptor.name);
6697    result.append(buffer);
6698    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6699            mDescriptor.implementor);
6700    result.append(buffer);
6701
6702    result.append("\t\t- Input configuration:\n");
6703    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6704    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6705            (uint32_t)mConfig.inputCfg.buffer.raw,
6706            mConfig.inputCfg.buffer.frameCount,
6707            mConfig.inputCfg.samplingRate,
6708            mConfig.inputCfg.channels,
6709            mConfig.inputCfg.format);
6710    result.append(buffer);
6711
6712    result.append("\t\t- Output configuration:\n");
6713    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6714    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6715            (uint32_t)mConfig.outputCfg.buffer.raw,
6716            mConfig.outputCfg.buffer.frameCount,
6717            mConfig.outputCfg.samplingRate,
6718            mConfig.outputCfg.channels,
6719            mConfig.outputCfg.format);
6720    result.append(buffer);
6721
6722    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6723    result.append(buffer);
6724    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6725    for (size_t i = 0; i < mHandles.size(); ++i) {
6726        sp<EffectHandle> handle = mHandles[i].promote();
6727        if (handle != 0) {
6728            handle->dump(buffer, SIZE);
6729            result.append(buffer);
6730        }
6731    }
6732
6733    result.append("\n");
6734
6735    write(fd, result.string(), result.length());
6736
6737    if (locked) {
6738        mLock.unlock();
6739    }
6740
6741    return NO_ERROR;
6742}
6743
6744// ----------------------------------------------------------------------------
6745//  EffectHandle implementation
6746// ----------------------------------------------------------------------------
6747
6748#undef LOG_TAG
6749#define LOG_TAG "AudioFlinger::EffectHandle"
6750
6751AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6752                                        const sp<AudioFlinger::Client>& client,
6753                                        const sp<IEffectClient>& effectClient,
6754                                        int32_t priority)
6755    : BnEffect(),
6756    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6757    mPriority(priority), mHasControl(false), mEnabled(false)
6758{
6759    ALOGV("constructor %p", this);
6760
6761    if (client == 0) {
6762        return;
6763    }
6764    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6765    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6766    if (mCblkMemory != 0) {
6767        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6768
6769        if (mCblk) {
6770            new(mCblk) effect_param_cblk_t();
6771            mBuffer = (uint8_t *)mCblk + bufOffset;
6772         }
6773    } else {
6774        LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6775        return;
6776    }
6777}
6778
6779AudioFlinger::EffectHandle::~EffectHandle()
6780{
6781    ALOGV("Destructor %p", this);
6782    disconnect(false);
6783    ALOGV("Destructor DONE %p", this);
6784}
6785
6786status_t AudioFlinger::EffectHandle::enable()
6787{
6788    ALOGV("enable %p", this);
6789    if (!mHasControl) return INVALID_OPERATION;
6790    if (mEffect == 0) return DEAD_OBJECT;
6791
6792    if (mEnabled) {
6793        return NO_ERROR;
6794    }
6795
6796    mEnabled = true;
6797
6798    sp<ThreadBase> thread = mEffect->thread().promote();
6799    if (thread != 0) {
6800        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6801    }
6802
6803    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6804    if (mEffect->suspended()) {
6805        return NO_ERROR;
6806    }
6807
6808    status_t status = mEffect->setEnabled(true);
6809    if (status != NO_ERROR) {
6810        if (thread != 0) {
6811            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6812        }
6813        mEnabled = false;
6814    }
6815    return status;
6816}
6817
6818status_t AudioFlinger::EffectHandle::disable()
6819{
6820    ALOGV("disable %p", this);
6821    if (!mHasControl) return INVALID_OPERATION;
6822    if (mEffect == 0) return DEAD_OBJECT;
6823
6824    if (!mEnabled) {
6825        return NO_ERROR;
6826    }
6827    mEnabled = false;
6828
6829    if (mEffect->suspended()) {
6830        return NO_ERROR;
6831    }
6832
6833    status_t status = mEffect->setEnabled(false);
6834
6835    sp<ThreadBase> thread = mEffect->thread().promote();
6836    if (thread != 0) {
6837        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6838    }
6839
6840    return status;
6841}
6842
6843void AudioFlinger::EffectHandle::disconnect()
6844{
6845    disconnect(true);
6846}
6847
6848void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6849{
6850    ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6851    if (mEffect == 0) {
6852        return;
6853    }
6854    mEffect->disconnect(this, unpiniflast);
6855
6856    if (mHasControl && mEnabled) {
6857        sp<ThreadBase> thread = mEffect->thread().promote();
6858        if (thread != 0) {
6859            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6860        }
6861    }
6862
6863    // release sp on module => module destructor can be called now
6864    mEffect.clear();
6865    if (mClient != 0) {
6866        if (mCblk) {
6867            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6868        }
6869        mCblkMemory.clear();            // and free the shared memory
6870        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6871        mClient.clear();
6872    }
6873}
6874
6875status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6876                                             uint32_t cmdSize,
6877                                             void *pCmdData,
6878                                             uint32_t *replySize,
6879                                             void *pReplyData)
6880{
6881//    ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6882//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6883
6884    // only get parameter command is permitted for applications not controlling the effect
6885    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6886        return INVALID_OPERATION;
6887    }
6888    if (mEffect == 0) return DEAD_OBJECT;
6889    if (mClient == 0) return INVALID_OPERATION;
6890
6891    // handle commands that are not forwarded transparently to effect engine
6892    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6893        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6894        // no risk to block the whole media server process or mixer threads is we are stuck here
6895        Mutex::Autolock _l(mCblk->lock);
6896        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6897            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6898            mCblk->serverIndex = 0;
6899            mCblk->clientIndex = 0;
6900            return BAD_VALUE;
6901        }
6902        status_t status = NO_ERROR;
6903        while (mCblk->serverIndex < mCblk->clientIndex) {
6904            int reply;
6905            uint32_t rsize = sizeof(int);
6906            int *p = (int *)(mBuffer + mCblk->serverIndex);
6907            int size = *p++;
6908            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
6909                LOGW("command(): invalid parameter block size");
6910                break;
6911            }
6912            effect_param_t *param = (effect_param_t *)p;
6913            if (param->psize == 0 || param->vsize == 0) {
6914                LOGW("command(): null parameter or value size");
6915                mCblk->serverIndex += size;
6916                continue;
6917            }
6918            uint32_t psize = sizeof(effect_param_t) +
6919                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
6920                             param->vsize;
6921            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
6922                                            psize,
6923                                            p,
6924                                            &rsize,
6925                                            &reply);
6926            // stop at first error encountered
6927            if (ret != NO_ERROR) {
6928                status = ret;
6929                *(int *)pReplyData = reply;
6930                break;
6931            } else if (reply != NO_ERROR) {
6932                *(int *)pReplyData = reply;
6933                break;
6934            }
6935            mCblk->serverIndex += size;
6936        }
6937        mCblk->serverIndex = 0;
6938        mCblk->clientIndex = 0;
6939        return status;
6940    } else if (cmdCode == EFFECT_CMD_ENABLE) {
6941        *(int *)pReplyData = NO_ERROR;
6942        return enable();
6943    } else if (cmdCode == EFFECT_CMD_DISABLE) {
6944        *(int *)pReplyData = NO_ERROR;
6945        return disable();
6946    }
6947
6948    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
6949}
6950
6951sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
6952    return mCblkMemory;
6953}
6954
6955void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
6956{
6957    ALOGV("setControl %p control %d", this, hasControl);
6958
6959    mHasControl = hasControl;
6960    mEnabled = enabled;
6961
6962    if (signal && mEffectClient != 0) {
6963        mEffectClient->controlStatusChanged(hasControl);
6964    }
6965}
6966
6967void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
6968                                                 uint32_t cmdSize,
6969                                                 void *pCmdData,
6970                                                 uint32_t replySize,
6971                                                 void *pReplyData)
6972{
6973    if (mEffectClient != 0) {
6974        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
6975    }
6976}
6977
6978
6979
6980void AudioFlinger::EffectHandle::setEnabled(bool enabled)
6981{
6982    if (mEffectClient != 0) {
6983        mEffectClient->enableStatusChanged(enabled);
6984    }
6985}
6986
6987status_t AudioFlinger::EffectHandle::onTransact(
6988    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
6989{
6990    return BnEffect::onTransact(code, data, reply, flags);
6991}
6992
6993
6994void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
6995{
6996    bool locked = mCblk ? tryLock(mCblk->lock) : false;
6997
6998    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
6999            (mClient == NULL) ? getpid() : mClient->pid(),
7000            mPriority,
7001            mHasControl,
7002            !locked,
7003            mCblk ? mCblk->clientIndex : 0,
7004            mCblk ? mCblk->serverIndex : 0
7005            );
7006
7007    if (locked) {
7008        mCblk->lock.unlock();
7009    }
7010}
7011
7012#undef LOG_TAG
7013#define LOG_TAG "AudioFlinger::EffectChain"
7014
7015AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
7016                                        int sessionId)
7017    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0),
7018      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
7019      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
7020{
7021    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
7022}
7023
7024AudioFlinger::EffectChain::~EffectChain()
7025{
7026    if (mOwnInBuffer) {
7027        delete mInBuffer;
7028    }
7029
7030}
7031
7032// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7033sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7034{
7035    sp<EffectModule> effect;
7036    size_t size = mEffects.size();
7037
7038    for (size_t i = 0; i < size; i++) {
7039        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7040            effect = mEffects[i];
7041            break;
7042        }
7043    }
7044    return effect;
7045}
7046
7047// getEffectFromId_l() must be called with ThreadBase::mLock held
7048sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7049{
7050    sp<EffectModule> effect;
7051    size_t size = mEffects.size();
7052
7053    for (size_t i = 0; i < size; i++) {
7054        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7055        if (id == 0 || mEffects[i]->id() == id) {
7056            effect = mEffects[i];
7057            break;
7058        }
7059    }
7060    return effect;
7061}
7062
7063// getEffectFromType_l() must be called with ThreadBase::mLock held
7064sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7065        const effect_uuid_t *type)
7066{
7067    sp<EffectModule> effect;
7068    size_t size = mEffects.size();
7069
7070    for (size_t i = 0; i < size; i++) {
7071        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7072            effect = mEffects[i];
7073            break;
7074        }
7075    }
7076    return effect;
7077}
7078
7079// Must be called with EffectChain::mLock locked
7080void AudioFlinger::EffectChain::process_l()
7081{
7082    sp<ThreadBase> thread = mThread.promote();
7083    if (thread == 0) {
7084        LOGW("process_l(): cannot promote mixer thread");
7085        return;
7086    }
7087    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7088            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7089    bool tracksOnSession = false;
7090    if (!isGlobalSession) {
7091        tracksOnSession = (trackCnt() != 0);
7092    }
7093
7094    // if no track is active, input buffer must be cleared here as the mixer process
7095    // will not do it
7096    if (tracksOnSession &&
7097            activeTrackCnt() == 0) {
7098        size_t numSamples = thread->frameCount() * thread->channelCount();
7099        memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7100    }
7101
7102    size_t size = mEffects.size();
7103    // do not process effect if no track is present in same audio session
7104    if (isGlobalSession || tracksOnSession) {
7105        for (size_t i = 0; i < size; i++) {
7106            mEffects[i]->process();
7107        }
7108    }
7109    for (size_t i = 0; i < size; i++) {
7110        mEffects[i]->updateState();
7111    }
7112}
7113
7114// addEffect_l() must be called with PlaybackThread::mLock held
7115status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7116{
7117    effect_descriptor_t desc = effect->desc();
7118    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7119
7120    Mutex::Autolock _l(mLock);
7121    effect->setChain(this);
7122    sp<ThreadBase> thread = mThread.promote();
7123    if (thread == 0) {
7124        return NO_INIT;
7125    }
7126    effect->setThread(thread);
7127
7128    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7129        // Auxiliary effects are inserted at the beginning of mEffects vector as
7130        // they are processed first and accumulated in chain input buffer
7131        mEffects.insertAt(effect, 0);
7132
7133        // the input buffer for auxiliary effect contains mono samples in
7134        // 32 bit format. This is to avoid saturation in AudoMixer
7135        // accumulation stage. Saturation is done in EffectModule::process() before
7136        // calling the process in effect engine
7137        size_t numSamples = thread->frameCount();
7138        int32_t *buffer = new int32_t[numSamples];
7139        memset(buffer, 0, numSamples * sizeof(int32_t));
7140        effect->setInBuffer((int16_t *)buffer);
7141        // auxiliary effects output samples to chain input buffer for further processing
7142        // by insert effects
7143        effect->setOutBuffer(mInBuffer);
7144    } else {
7145        // Insert effects are inserted at the end of mEffects vector as they are processed
7146        //  after track and auxiliary effects.
7147        // Insert effect order as a function of indicated preference:
7148        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7149        //  another effect is present
7150        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7151        //  last effect claiming first position
7152        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7153        //  first effect claiming last position
7154        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7155        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7156        // already present
7157
7158        int size = (int)mEffects.size();
7159        int idx_insert = size;
7160        int idx_insert_first = -1;
7161        int idx_insert_last = -1;
7162
7163        for (int i = 0; i < size; i++) {
7164            effect_descriptor_t d = mEffects[i]->desc();
7165            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7166            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7167            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7168                // check invalid effect chaining combinations
7169                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7170                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7171                    LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7172                    return INVALID_OPERATION;
7173                }
7174                // remember position of first insert effect and by default
7175                // select this as insert position for new effect
7176                if (idx_insert == size) {
7177                    idx_insert = i;
7178                }
7179                // remember position of last insert effect claiming
7180                // first position
7181                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7182                    idx_insert_first = i;
7183                }
7184                // remember position of first insert effect claiming
7185                // last position
7186                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7187                    idx_insert_last == -1) {
7188                    idx_insert_last = i;
7189                }
7190            }
7191        }
7192
7193        // modify idx_insert from first position if needed
7194        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7195            if (idx_insert_last != -1) {
7196                idx_insert = idx_insert_last;
7197            } else {
7198                idx_insert = size;
7199            }
7200        } else {
7201            if (idx_insert_first != -1) {
7202                idx_insert = idx_insert_first + 1;
7203            }
7204        }
7205
7206        // always read samples from chain input buffer
7207        effect->setInBuffer(mInBuffer);
7208
7209        // if last effect in the chain, output samples to chain
7210        // output buffer, otherwise to chain input buffer
7211        if (idx_insert == size) {
7212            if (idx_insert != 0) {
7213                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7214                mEffects[idx_insert-1]->configure();
7215            }
7216            effect->setOutBuffer(mOutBuffer);
7217        } else {
7218            effect->setOutBuffer(mInBuffer);
7219        }
7220        mEffects.insertAt(effect, idx_insert);
7221
7222        ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7223    }
7224    effect->configure();
7225    return NO_ERROR;
7226}
7227
7228// removeEffect_l() must be called with PlaybackThread::mLock held
7229size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7230{
7231    Mutex::Autolock _l(mLock);
7232    int size = (int)mEffects.size();
7233    int i;
7234    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7235
7236    for (i = 0; i < size; i++) {
7237        if (effect == mEffects[i]) {
7238            // calling stop here will remove pre-processing effect from the audio HAL.
7239            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7240            // the middle of a read from audio HAL
7241            if (mEffects[i]->state() == EffectModule::ACTIVE ||
7242                    mEffects[i]->state() == EffectModule::STOPPING) {
7243                mEffects[i]->stop();
7244            }
7245            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7246                delete[] effect->inBuffer();
7247            } else {
7248                if (i == size - 1 && i != 0) {
7249                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7250                    mEffects[i - 1]->configure();
7251                }
7252            }
7253            mEffects.removeAt(i);
7254            ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7255            break;
7256        }
7257    }
7258
7259    return mEffects.size();
7260}
7261
7262// setDevice_l() must be called with PlaybackThread::mLock held
7263void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7264{
7265    size_t size = mEffects.size();
7266    for (size_t i = 0; i < size; i++) {
7267        mEffects[i]->setDevice(device);
7268    }
7269}
7270
7271// setMode_l() must be called with PlaybackThread::mLock held
7272void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
7273{
7274    size_t size = mEffects.size();
7275    for (size_t i = 0; i < size; i++) {
7276        mEffects[i]->setMode(mode);
7277    }
7278}
7279
7280// setVolume_l() must be called with PlaybackThread::mLock held
7281bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7282{
7283    uint32_t newLeft = *left;
7284    uint32_t newRight = *right;
7285    bool hasControl = false;
7286    int ctrlIdx = -1;
7287    size_t size = mEffects.size();
7288
7289    // first update volume controller
7290    for (size_t i = size; i > 0; i--) {
7291        if (mEffects[i - 1]->isProcessEnabled() &&
7292            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7293            ctrlIdx = i - 1;
7294            hasControl = true;
7295            break;
7296        }
7297    }
7298
7299    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7300        if (hasControl) {
7301            *left = mNewLeftVolume;
7302            *right = mNewRightVolume;
7303        }
7304        return hasControl;
7305    }
7306
7307    mVolumeCtrlIdx = ctrlIdx;
7308    mLeftVolume = newLeft;
7309    mRightVolume = newRight;
7310
7311    // second get volume update from volume controller
7312    if (ctrlIdx >= 0) {
7313        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7314        mNewLeftVolume = newLeft;
7315        mNewRightVolume = newRight;
7316    }
7317    // then indicate volume to all other effects in chain.
7318    // Pass altered volume to effects before volume controller
7319    // and requested volume to effects after controller
7320    uint32_t lVol = newLeft;
7321    uint32_t rVol = newRight;
7322
7323    for (size_t i = 0; i < size; i++) {
7324        if ((int)i == ctrlIdx) continue;
7325        // this also works for ctrlIdx == -1 when there is no volume controller
7326        if ((int)i > ctrlIdx) {
7327            lVol = *left;
7328            rVol = *right;
7329        }
7330        mEffects[i]->setVolume(&lVol, &rVol, false);
7331    }
7332    *left = newLeft;
7333    *right = newRight;
7334
7335    return hasControl;
7336}
7337
7338status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7339{
7340    const size_t SIZE = 256;
7341    char buffer[SIZE];
7342    String8 result;
7343
7344    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7345    result.append(buffer);
7346
7347    bool locked = tryLock(mLock);
7348    // failed to lock - AudioFlinger is probably deadlocked
7349    if (!locked) {
7350        result.append("\tCould not lock mutex:\n");
7351    }
7352
7353    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7354    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7355            mEffects.size(),
7356            (uint32_t)mInBuffer,
7357            (uint32_t)mOutBuffer,
7358            mActiveTrackCnt);
7359    result.append(buffer);
7360    write(fd, result.string(), result.size());
7361
7362    for (size_t i = 0; i < mEffects.size(); ++i) {
7363        sp<EffectModule> effect = mEffects[i];
7364        if (effect != 0) {
7365            effect->dump(fd, args);
7366        }
7367    }
7368
7369    if (locked) {
7370        mLock.unlock();
7371    }
7372
7373    return NO_ERROR;
7374}
7375
7376// must be called with ThreadBase::mLock held
7377void AudioFlinger::EffectChain::setEffectSuspended_l(
7378        const effect_uuid_t *type, bool suspend)
7379{
7380    sp<SuspendedEffectDesc> desc;
7381    // use effect type UUID timelow as key as there is no real risk of identical
7382    // timeLow fields among effect type UUIDs.
7383    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7384    if (suspend) {
7385        if (index >= 0) {
7386            desc = mSuspendedEffects.valueAt(index);
7387        } else {
7388            desc = new SuspendedEffectDesc();
7389            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7390            mSuspendedEffects.add(type->timeLow, desc);
7391            ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7392        }
7393        if (desc->mRefCount++ == 0) {
7394            sp<EffectModule> effect = getEffectIfEnabled(type);
7395            if (effect != 0) {
7396                desc->mEffect = effect;
7397                effect->setSuspended(true);
7398                effect->setEnabled(false);
7399            }
7400        }
7401    } else {
7402        if (index < 0) {
7403            return;
7404        }
7405        desc = mSuspendedEffects.valueAt(index);
7406        if (desc->mRefCount <= 0) {
7407            LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7408            desc->mRefCount = 1;
7409        }
7410        if (--desc->mRefCount == 0) {
7411            ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7412            if (desc->mEffect != 0) {
7413                sp<EffectModule> effect = desc->mEffect.promote();
7414                if (effect != 0) {
7415                    effect->setSuspended(false);
7416                    sp<EffectHandle> handle = effect->controlHandle();
7417                    if (handle != 0) {
7418                        effect->setEnabled(handle->enabled());
7419                    }
7420                }
7421                desc->mEffect.clear();
7422            }
7423            mSuspendedEffects.removeItemsAt(index);
7424        }
7425    }
7426}
7427
7428// must be called with ThreadBase::mLock held
7429void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7430{
7431    sp<SuspendedEffectDesc> desc;
7432
7433    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7434    if (suspend) {
7435        if (index >= 0) {
7436            desc = mSuspendedEffects.valueAt(index);
7437        } else {
7438            desc = new SuspendedEffectDesc();
7439            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7440            ALOGV("setEffectSuspendedAll_l() add entry for 0");
7441        }
7442        if (desc->mRefCount++ == 0) {
7443            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7444            for (size_t i = 0; i < effects.size(); i++) {
7445                setEffectSuspended_l(&effects[i]->desc().type, true);
7446            }
7447        }
7448    } else {
7449        if (index < 0) {
7450            return;
7451        }
7452        desc = mSuspendedEffects.valueAt(index);
7453        if (desc->mRefCount <= 0) {
7454            LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7455            desc->mRefCount = 1;
7456        }
7457        if (--desc->mRefCount == 0) {
7458            Vector<const effect_uuid_t *> types;
7459            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7460                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7461                    continue;
7462                }
7463                types.add(&mSuspendedEffects.valueAt(i)->mType);
7464            }
7465            for (size_t i = 0; i < types.size(); i++) {
7466                setEffectSuspended_l(types[i], false);
7467            }
7468            ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7469            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7470        }
7471    }
7472}
7473
7474
7475// The volume effect is used for automated tests only
7476#ifndef OPENSL_ES_H_
7477static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7478                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7479const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7480#endif //OPENSL_ES_H_
7481
7482bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7483{
7484    // auxiliary effects and visualizer are never suspended on output mix
7485    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7486        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7487         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7488         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7489        return false;
7490    }
7491    return true;
7492}
7493
7494Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7495{
7496    Vector< sp<EffectModule> > effects;
7497    for (size_t i = 0; i < mEffects.size(); i++) {
7498        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7499            continue;
7500        }
7501        effects.add(mEffects[i]);
7502    }
7503    return effects;
7504}
7505
7506sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7507                                                            const effect_uuid_t *type)
7508{
7509    sp<EffectModule> effect;
7510    effect = getEffectFromType_l(type);
7511    if (effect != 0 && !effect->isEnabled()) {
7512        effect.clear();
7513    }
7514    return effect;
7515}
7516
7517void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7518                                                            bool enabled)
7519{
7520    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7521    if (enabled) {
7522        if (index < 0) {
7523            // if the effect is not suspend check if all effects are suspended
7524            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7525            if (index < 0) {
7526                return;
7527            }
7528            if (!isEffectEligibleForSuspend(effect->desc())) {
7529                return;
7530            }
7531            setEffectSuspended_l(&effect->desc().type, enabled);
7532            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7533            if (index < 0) {
7534                LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7535                return;
7536            }
7537        }
7538        ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7539             effect->desc().type.timeLow);
7540        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7541        // if effect is requested to suspended but was not yet enabled, supend it now.
7542        if (desc->mEffect == 0) {
7543            desc->mEffect = effect;
7544            effect->setEnabled(false);
7545            effect->setSuspended(true);
7546        }
7547    } else {
7548        if (index < 0) {
7549            return;
7550        }
7551        ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7552             effect->desc().type.timeLow);
7553        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7554        desc->mEffect.clear();
7555        effect->setSuspended(false);
7556    }
7557}
7558
7559#undef LOG_TAG
7560#define LOG_TAG "AudioFlinger"
7561
7562// ----------------------------------------------------------------------------
7563
7564status_t AudioFlinger::onTransact(
7565        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7566{
7567    return BnAudioFlinger::onTransact(code, data, reply, flags);
7568}
7569
7570}; // namespace android
7571