AudioFlinger.cpp revision 6bffdb8b598a2399e57f6ca48660fb7bdb2490b5 
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        LOGV("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        LOGV("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    LOGV("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//    LOGV("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        LOGV("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        LOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
931        mNotificationClients.removeItem(pid);
932    }
933
934    LOGV("%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        LOGV(" pid %d @ %d", ref->pid, i);
940        if (ref->pid == pid) {
941            LOGV(" 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    LOGV("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}
989
990void AudioFlinger::ThreadBase::exit()
991{
992    // keep a strong ref on ourself so that we wont get
993    // destroyed in the middle of requestExitAndWait()
994    sp <ThreadBase> strongMe = this;
995
996    LOGV("ThreadBase::exit");
997    {
998        AutoMutex lock(&mLock);
999        mExiting = true;
1000        requestExit();
1001        mWaitWorkCV.signal();
1002    }
1003    requestExitAndWait();
1004}
1005
1006uint32_t AudioFlinger::ThreadBase::sampleRate() const
1007{
1008    return mSampleRate;
1009}
1010
1011int AudioFlinger::ThreadBase::channelCount() const
1012{
1013    return (int)mChannelCount;
1014}
1015
1016uint32_t AudioFlinger::ThreadBase::format() const
1017{
1018    return mFormat;
1019}
1020
1021size_t AudioFlinger::ThreadBase::frameCount() const
1022{
1023    return mFrameCount;
1024}
1025
1026status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
1027{
1028    status_t status;
1029
1030    LOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
1031    Mutex::Autolock _l(mLock);
1032
1033    mNewParameters.add(keyValuePairs);
1034    mWaitWorkCV.signal();
1035    // wait condition with timeout in case the thread loop has exited
1036    // before the request could be processed
1037    if (mParamCond.waitRelative(mLock, kSetParametersTimeout) == NO_ERROR) {
1038        status = mParamStatus;
1039        mWaitWorkCV.signal();
1040    } else {
1041        status = TIMED_OUT;
1042    }
1043    return status;
1044}
1045
1046void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
1047{
1048    Mutex::Autolock _l(mLock);
1049    sendConfigEvent_l(event, param);
1050}
1051
1052// sendConfigEvent_l() must be called with ThreadBase::mLock held
1053void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
1054{
1055    ConfigEvent *configEvent = new ConfigEvent();
1056    configEvent->mEvent = event;
1057    configEvent->mParam = param;
1058    mConfigEvents.add(configEvent);
1059    LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
1060    mWaitWorkCV.signal();
1061}
1062
1063void AudioFlinger::ThreadBase::processConfigEvents()
1064{
1065    mLock.lock();
1066    while(!mConfigEvents.isEmpty()) {
1067        LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
1068        ConfigEvent *configEvent = mConfigEvents[0];
1069        mConfigEvents.removeAt(0);
1070        // release mLock before locking AudioFlinger mLock: lock order is always
1071        // AudioFlinger then ThreadBase to avoid cross deadlock
1072        mLock.unlock();
1073        mAudioFlinger->mLock.lock();
1074        audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
1075        mAudioFlinger->mLock.unlock();
1076        delete configEvent;
1077        mLock.lock();
1078    }
1079    mLock.unlock();
1080}
1081
1082status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
1083{
1084    const size_t SIZE = 256;
1085    char buffer[SIZE];
1086    String8 result;
1087
1088    bool locked = tryLock(mLock);
1089    if (!locked) {
1090        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1091        write(fd, buffer, strlen(buffer));
1092    }
1093
1094    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1095    result.append(buffer);
1096    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1097    result.append(buffer);
1098    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1099    result.append(buffer);
1100    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1101    result.append(buffer);
1102    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1103    result.append(buffer);
1104    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1105    result.append(buffer);
1106    snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
1107    result.append(buffer);
1108
1109    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1110    result.append(buffer);
1111    result.append(" Index Command");
1112    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1113        snprintf(buffer, SIZE, "\n %02d    ", i);
1114        result.append(buffer);
1115        result.append(mNewParameters[i]);
1116    }
1117
1118    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1119    result.append(buffer);
1120    snprintf(buffer, SIZE, " Index event param\n");
1121    result.append(buffer);
1122    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1123        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
1124        result.append(buffer);
1125    }
1126    result.append("\n");
1127
1128    write(fd, result.string(), result.size());
1129
1130    if (locked) {
1131        mLock.unlock();
1132    }
1133    return NO_ERROR;
1134}
1135
1136status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1137{
1138    const size_t SIZE = 256;
1139    char buffer[SIZE];
1140    String8 result;
1141
1142    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1143    write(fd, buffer, strlen(buffer));
1144
1145    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1146        sp<EffectChain> chain = mEffectChains[i];
1147        if (chain != 0) {
1148            chain->dump(fd, args);
1149        }
1150    }
1151    return NO_ERROR;
1152}
1153
1154void AudioFlinger::ThreadBase::acquireWakeLock()
1155{
1156    Mutex::Autolock _l(mLock);
1157    acquireWakeLock_l();
1158}
1159
1160void AudioFlinger::ThreadBase::acquireWakeLock_l()
1161{
1162    if (mPowerManager == 0) {
1163        // use checkService() to avoid blocking if power service is not up yet
1164        sp<IBinder> binder =
1165            defaultServiceManager()->checkService(String16("power"));
1166        if (binder == 0) {
1167            LOGW("Thread %s cannot connect to the power manager service", mName);
1168        } else {
1169            mPowerManager = interface_cast<IPowerManager>(binder);
1170            binder->linkToDeath(mDeathRecipient);
1171        }
1172    }
1173    if (mPowerManager != 0) {
1174        sp<IBinder> binder = new BBinder();
1175        status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
1176                                                         binder,
1177                                                         String16(mName));
1178        if (status == NO_ERROR) {
1179            mWakeLockToken = binder;
1180        }
1181        LOGV("acquireWakeLock_l() %s status %d", mName, status);
1182    }
1183}
1184
1185void AudioFlinger::ThreadBase::releaseWakeLock()
1186{
1187    Mutex::Autolock _l(mLock);
1188    releaseWakeLock_l();
1189}
1190
1191void AudioFlinger::ThreadBase::releaseWakeLock_l()
1192{
1193    if (mWakeLockToken != 0) {
1194        LOGV("releaseWakeLock_l() %s", mName);
1195        if (mPowerManager != 0) {
1196            mPowerManager->releaseWakeLock(mWakeLockToken, 0);
1197        }
1198        mWakeLockToken.clear();
1199    }
1200}
1201
1202void AudioFlinger::ThreadBase::clearPowerManager()
1203{
1204    Mutex::Autolock _l(mLock);
1205    releaseWakeLock_l();
1206    mPowerManager.clear();
1207}
1208
1209void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
1210{
1211    sp<ThreadBase> thread = mThread.promote();
1212    if (thread != 0) {
1213        thread->clearPowerManager();
1214    }
1215    LOGW("power manager service died !!!");
1216}
1217
1218void AudioFlinger::ThreadBase::setEffectSuspended(
1219        const effect_uuid_t *type, bool suspend, int sessionId)
1220{
1221    Mutex::Autolock _l(mLock);
1222    setEffectSuspended_l(type, suspend, sessionId);
1223}
1224
1225void AudioFlinger::ThreadBase::setEffectSuspended_l(
1226        const effect_uuid_t *type, bool suspend, int sessionId)
1227{
1228    sp<EffectChain> chain;
1229    chain = getEffectChain_l(sessionId);
1230    if (chain != 0) {
1231        if (type != NULL) {
1232            chain->setEffectSuspended_l(type, suspend);
1233        } else {
1234            chain->setEffectSuspendedAll_l(suspend);
1235        }
1236    }
1237
1238    updateSuspendedSessions_l(type, suspend, sessionId);
1239}
1240
1241void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1242{
1243    int index = mSuspendedSessions.indexOfKey(chain->sessionId());
1244    if (index < 0) {
1245        return;
1246    }
1247
1248    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
1249            mSuspendedSessions.editValueAt(index);
1250
1251    for (size_t i = 0; i < sessionEffects.size(); i++) {
1252        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1253        for (int j = 0; j < desc->mRefCount; j++) {
1254            if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1255                chain->setEffectSuspendedAll_l(true);
1256            } else {
1257                LOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1258                     desc->mType.timeLow);
1259                chain->setEffectSuspended_l(&desc->mType, true);
1260            }
1261        }
1262    }
1263}
1264
1265void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1266                                                         bool suspend,
1267                                                         int sessionId)
1268{
1269    int index = mSuspendedSessions.indexOfKey(sessionId);
1270
1271    KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1272
1273    if (suspend) {
1274        if (index >= 0) {
1275            sessionEffects = mSuspendedSessions.editValueAt(index);
1276        } else {
1277            mSuspendedSessions.add(sessionId, sessionEffects);
1278        }
1279    } else {
1280        if (index < 0) {
1281            return;
1282        }
1283        sessionEffects = mSuspendedSessions.editValueAt(index);
1284    }
1285
1286
1287    int key = EffectChain::kKeyForSuspendAll;
1288    if (type != NULL) {
1289        key = type->timeLow;
1290    }
1291    index = sessionEffects.indexOfKey(key);
1292
1293    sp <SuspendedSessionDesc> desc;
1294    if (suspend) {
1295        if (index >= 0) {
1296            desc = sessionEffects.valueAt(index);
1297        } else {
1298            desc = new SuspendedSessionDesc();
1299            if (type != NULL) {
1300                memcpy(&desc->mType, type, sizeof(effect_uuid_t));
1301            }
1302            sessionEffects.add(key, desc);
1303            LOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1304        }
1305        desc->mRefCount++;
1306    } else {
1307        if (index < 0) {
1308            return;
1309        }
1310        desc = sessionEffects.valueAt(index);
1311        if (--desc->mRefCount == 0) {
1312            LOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1313            sessionEffects.removeItemsAt(index);
1314            if (sessionEffects.isEmpty()) {
1315                LOGV("updateSuspendedSessions_l() restore removing session %d",
1316                                 sessionId);
1317                mSuspendedSessions.removeItem(sessionId);
1318            }
1319        }
1320    }
1321    if (!sessionEffects.isEmpty()) {
1322        mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1323    }
1324}
1325
1326void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1327                                                            bool enabled,
1328                                                            int sessionId)
1329{
1330    Mutex::Autolock _l(mLock);
1331
1332    if (mType != RECORD) {
1333        // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1334        // another session. This gives the priority to well behaved effect control panels
1335        // and applications not using global effects.
1336        if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1337            setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1338        }
1339    }
1340
1341    sp<EffectChain> chain = getEffectChain_l(sessionId);
1342    if (chain != 0) {
1343        chain->checkSuspendOnEffectEnabled(effect, enabled);
1344    }
1345}
1346
1347// ----------------------------------------------------------------------------
1348
1349AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1350                                             AudioStreamOut* output,
1351                                             int id,
1352                                             uint32_t device)
1353    :   ThreadBase(audioFlinger, id, device),
1354        mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
1355        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1356{
1357    snprintf(mName, kNameLength, "AudioOut_%d", id);
1358
1359    readOutputParameters();
1360
1361    mMasterVolume = mAudioFlinger->masterVolume();
1362    mMasterMute = mAudioFlinger->masterMute();
1363
1364    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
1365        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1366        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1367        mStreamTypes[stream].valid = true;
1368    }
1369}
1370
1371AudioFlinger::PlaybackThread::~PlaybackThread()
1372{
1373    delete [] mMixBuffer;
1374}
1375
1376status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1377{
1378    dumpInternals(fd, args);
1379    dumpTracks(fd, args);
1380    dumpEffectChains(fd, args);
1381    return NO_ERROR;
1382}
1383
1384status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1385{
1386    const size_t SIZE = 256;
1387    char buffer[SIZE];
1388    String8 result;
1389
1390    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1391    result.append(buffer);
1392    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1393    for (size_t i = 0; i < mTracks.size(); ++i) {
1394        sp<Track> track = mTracks[i];
1395        if (track != 0) {
1396            track->dump(buffer, SIZE);
1397            result.append(buffer);
1398        }
1399    }
1400
1401    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1402    result.append(buffer);
1403    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1404    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1405        wp<Track> wTrack = mActiveTracks[i];
1406        if (wTrack != 0) {
1407            sp<Track> track = wTrack.promote();
1408            if (track != 0) {
1409                track->dump(buffer, SIZE);
1410                result.append(buffer);
1411            }
1412        }
1413    }
1414    write(fd, result.string(), result.size());
1415    return NO_ERROR;
1416}
1417
1418status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1419{
1420    const size_t SIZE = 256;
1421    char buffer[SIZE];
1422    String8 result;
1423
1424    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1425    result.append(buffer);
1426    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1427    result.append(buffer);
1428    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1429    result.append(buffer);
1430    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1431    result.append(buffer);
1432    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1433    result.append(buffer);
1434    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1435    result.append(buffer);
1436    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1437    result.append(buffer);
1438    write(fd, result.string(), result.size());
1439
1440    dumpBase(fd, args);
1441
1442    return NO_ERROR;
1443}
1444
1445// Thread virtuals
1446status_t AudioFlinger::PlaybackThread::readyToRun()
1447{
1448    status_t status = initCheck();
1449    if (status == NO_ERROR) {
1450        LOGI("AudioFlinger's thread %p ready to run", this);
1451    } else {
1452        LOGE("No working audio driver found.");
1453    }
1454    return status;
1455}
1456
1457void AudioFlinger::PlaybackThread::onFirstRef()
1458{
1459    run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1460}
1461
1462// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1463sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1464        const sp<AudioFlinger::Client>& client,
1465        int streamType,
1466        uint32_t sampleRate,
1467        uint32_t format,
1468        uint32_t channelMask,
1469        int frameCount,
1470        const sp<IMemory>& sharedBuffer,
1471        int sessionId,
1472        status_t *status)
1473{
1474    sp<Track> track;
1475    status_t lStatus;
1476
1477    if (mType == DIRECT) {
1478        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1479            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1480                LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1481                        "for output %p with format %d",
1482                        sampleRate, format, channelMask, mOutput, mFormat);
1483                lStatus = BAD_VALUE;
1484                goto Exit;
1485            }
1486        }
1487    } else {
1488        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1489        if (sampleRate > mSampleRate*2) {
1490            LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1491            lStatus = BAD_VALUE;
1492            goto Exit;
1493        }
1494    }
1495
1496    lStatus = initCheck();
1497    if (lStatus != NO_ERROR) {
1498        LOGE("Audio driver not initialized.");
1499        goto Exit;
1500    }
1501
1502    { // scope for mLock
1503        Mutex::Autolock _l(mLock);
1504
1505        // all tracks in same audio session must share the same routing strategy otherwise
1506        // conflicts will happen when tracks are moved from one output to another by audio policy
1507        // manager
1508        uint32_t strategy =
1509                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1510        for (size_t i = 0; i < mTracks.size(); ++i) {
1511            sp<Track> t = mTracks[i];
1512            if (t != 0) {
1513                if (sessionId == t->sessionId() &&
1514                        strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
1515                    lStatus = BAD_VALUE;
1516                    goto Exit;
1517                }
1518            }
1519        }
1520
1521        track = new Track(this, client, streamType, sampleRate, format,
1522                channelMask, frameCount, sharedBuffer, sessionId);
1523        if (track->getCblk() == NULL || track->name() < 0) {
1524            lStatus = NO_MEMORY;
1525            goto Exit;
1526        }
1527        mTracks.add(track);
1528
1529        sp<EffectChain> chain = getEffectChain_l(sessionId);
1530        if (chain != 0) {
1531            LOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1532            track->setMainBuffer(chain->inBuffer());
1533            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1534            chain->incTrackCnt();
1535        }
1536
1537        // invalidate track immediately if the stream type was moved to another thread since
1538        // createTrack() was called by the client process.
1539        if (!mStreamTypes[streamType].valid) {
1540            LOGW("createTrack_l() on thread %p: invalidating track on stream %d",
1541                 this, streamType);
1542            android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
1543        }
1544    }
1545    lStatus = NO_ERROR;
1546
1547Exit:
1548    if(status) {
1549        *status = lStatus;
1550    }
1551    return track;
1552}
1553
1554uint32_t AudioFlinger::PlaybackThread::latency() const
1555{
1556    Mutex::Autolock _l(mLock);
1557    if (initCheck() == NO_ERROR) {
1558        return mOutput->stream->get_latency(mOutput->stream);
1559    } else {
1560        return 0;
1561    }
1562}
1563
1564status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1565{
1566    mMasterVolume = value;
1567    return NO_ERROR;
1568}
1569
1570status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1571{
1572    mMasterMute = muted;
1573    return NO_ERROR;
1574}
1575
1576float AudioFlinger::PlaybackThread::masterVolume() const
1577{
1578    return mMasterVolume;
1579}
1580
1581bool AudioFlinger::PlaybackThread::masterMute() const
1582{
1583    return mMasterMute;
1584}
1585
1586status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
1587{
1588    mStreamTypes[stream].volume = value;
1589    return NO_ERROR;
1590}
1591
1592status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
1593{
1594    mStreamTypes[stream].mute = muted;
1595    return NO_ERROR;
1596}
1597
1598float AudioFlinger::PlaybackThread::streamVolume(int stream) const
1599{
1600    return mStreamTypes[stream].volume;
1601}
1602
1603bool AudioFlinger::PlaybackThread::streamMute(int stream) const
1604{
1605    return mStreamTypes[stream].mute;
1606}
1607
1608// addTrack_l() must be called with ThreadBase::mLock held
1609status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1610{
1611    status_t status = ALREADY_EXISTS;
1612
1613    // set retry count for buffer fill
1614    track->mRetryCount = kMaxTrackStartupRetries;
1615    if (mActiveTracks.indexOf(track) < 0) {
1616        // the track is newly added, make sure it fills up all its
1617        // buffers before playing. This is to ensure the client will
1618        // effectively get the latency it requested.
1619        track->mFillingUpStatus = Track::FS_FILLING;
1620        track->mResetDone = false;
1621        mActiveTracks.add(track);
1622        if (track->mainBuffer() != mMixBuffer) {
1623            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1624            if (chain != 0) {
1625                LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1626                chain->incActiveTrackCnt();
1627            }
1628        }
1629
1630        status = NO_ERROR;
1631    }
1632
1633    LOGV("mWaitWorkCV.broadcast");
1634    mWaitWorkCV.broadcast();
1635
1636    return status;
1637}
1638
1639// destroyTrack_l() must be called with ThreadBase::mLock held
1640void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1641{
1642    track->mState = TrackBase::TERMINATED;
1643    if (mActiveTracks.indexOf(track) < 0) {
1644        removeTrack_l(track);
1645    }
1646}
1647
1648void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1649{
1650    mTracks.remove(track);
1651    deleteTrackName_l(track->name());
1652    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1653    if (chain != 0) {
1654        chain->decTrackCnt();
1655    }
1656}
1657
1658String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1659{
1660    String8 out_s8 = String8("");
1661    char *s;
1662
1663    Mutex::Autolock _l(mLock);
1664    if (initCheck() != NO_ERROR) {
1665        return out_s8;
1666    }
1667
1668    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1669    out_s8 = String8(s);
1670    free(s);
1671    return out_s8;
1672}
1673
1674// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1675void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1676    AudioSystem::OutputDescriptor desc;
1677    void *param2 = 0;
1678
1679    LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1680
1681    switch (event) {
1682    case AudioSystem::OUTPUT_OPENED:
1683    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1684        desc.channels = mChannelMask;
1685        desc.samplingRate = mSampleRate;
1686        desc.format = mFormat;
1687        desc.frameCount = mFrameCount;
1688        desc.latency = latency();
1689        param2 = &desc;
1690        break;
1691
1692    case AudioSystem::STREAM_CONFIG_CHANGED:
1693        param2 = &param;
1694    case AudioSystem::OUTPUT_CLOSED:
1695    default:
1696        break;
1697    }
1698    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1699}
1700
1701void AudioFlinger::PlaybackThread::readOutputParameters()
1702{
1703    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1704    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1705    mChannelCount = (uint16_t)popcount(mChannelMask);
1706    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1707    mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
1708    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1709
1710    // FIXME - Current mixer implementation only supports stereo output: Always
1711    // Allocate a stereo buffer even if HW output is mono.
1712    if (mMixBuffer != NULL) delete[] mMixBuffer;
1713    mMixBuffer = new int16_t[mFrameCount * 2];
1714    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1715
1716    // force reconfiguration of effect chains and engines to take new buffer size and audio
1717    // parameters into account
1718    // Note that mLock is not held when readOutputParameters() is called from the constructor
1719    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1720    // matter.
1721    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1722    Vector< sp<EffectChain> > effectChains = mEffectChains;
1723    for (size_t i = 0; i < effectChains.size(); i ++) {
1724        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1725    }
1726}
1727
1728status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1729{
1730    if (halFrames == 0 || dspFrames == 0) {
1731        return BAD_VALUE;
1732    }
1733    Mutex::Autolock _l(mLock);
1734    if (initCheck() != NO_ERROR) {
1735        return INVALID_OPERATION;
1736    }
1737    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1738
1739    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1740}
1741
1742uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1743{
1744    Mutex::Autolock _l(mLock);
1745    uint32_t result = 0;
1746    if (getEffectChain_l(sessionId) != 0) {
1747        result = EFFECT_SESSION;
1748    }
1749
1750    for (size_t i = 0; i < mTracks.size(); ++i) {
1751        sp<Track> track = mTracks[i];
1752        if (sessionId == track->sessionId() &&
1753                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1754            result |= TRACK_SESSION;
1755            break;
1756        }
1757    }
1758
1759    return result;
1760}
1761
1762uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1763{
1764    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1765    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1766    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1767        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1768    }
1769    for (size_t i = 0; i < mTracks.size(); i++) {
1770        sp<Track> track = mTracks[i];
1771        if (sessionId == track->sessionId() &&
1772                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1773            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1774        }
1775    }
1776    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1777}
1778
1779
1780AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
1781{
1782    Mutex::Autolock _l(mLock);
1783    return mOutput;
1784}
1785
1786AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1787{
1788    Mutex::Autolock _l(mLock);
1789    AudioStreamOut *output = mOutput;
1790    mOutput = NULL;
1791    return output;
1792}
1793
1794// this method must always be called either with ThreadBase mLock held or inside the thread loop
1795audio_stream_t* AudioFlinger::PlaybackThread::stream()
1796{
1797    if (mOutput == NULL) {
1798        return NULL;
1799    }
1800    return &mOutput->stream->common;
1801}
1802
1803// ----------------------------------------------------------------------------
1804
1805AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1806    :   PlaybackThread(audioFlinger, output, id, device),
1807        mAudioMixer(0)
1808{
1809    mType = ThreadBase::MIXER;
1810    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1811
1812    // FIXME - Current mixer implementation only supports stereo output
1813    if (mChannelCount == 1) {
1814        LOGE("Invalid audio hardware channel count");
1815    }
1816}
1817
1818AudioFlinger::MixerThread::~MixerThread()
1819{
1820    delete mAudioMixer;
1821}
1822
1823bool AudioFlinger::MixerThread::threadLoop()
1824{
1825    Vector< sp<Track> > tracksToRemove;
1826    uint32_t mixerStatus = MIXER_IDLE;
1827    nsecs_t standbyTime = systemTime();
1828    size_t mixBufferSize = mFrameCount * mFrameSize;
1829    // FIXME: Relaxed timing because of a certain device that can't meet latency
1830    // Should be reduced to 2x after the vendor fixes the driver issue
1831    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
1832    nsecs_t lastWarning = 0;
1833    bool longStandbyExit = false;
1834    uint32_t activeSleepTime = activeSleepTimeUs();
1835    uint32_t idleSleepTime = idleSleepTimeUs();
1836    uint32_t sleepTime = idleSleepTime;
1837    Vector< sp<EffectChain> > effectChains;
1838#ifdef DEBUG_CPU_USAGE
1839    ThreadCpuUsage cpu;
1840    const CentralTendencyStatistics& stats = cpu.statistics();
1841#endif
1842
1843    acquireWakeLock();
1844
1845    while (!exitPending())
1846    {
1847#ifdef DEBUG_CPU_USAGE
1848        cpu.sampleAndEnable();
1849        unsigned n = stats.n();
1850        // cpu.elapsed() is expensive, so don't call it every loop
1851        if ((n & 127) == 1) {
1852            long long elapsed = cpu.elapsed();
1853            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1854                double perLoop = elapsed / (double) n;
1855                double perLoop100 = perLoop * 0.01;
1856                double mean = stats.mean();
1857                double stddev = stats.stddev();
1858                double minimum = stats.minimum();
1859                double maximum = stats.maximum();
1860                cpu.resetStatistics();
1861                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",
1862                        elapsed * .000000001, n, perLoop * .000001,
1863                        mean * .001,
1864                        stddev * .001,
1865                        minimum * .001,
1866                        maximum * .001,
1867                        mean / perLoop100,
1868                        stddev / perLoop100,
1869                        minimum / perLoop100,
1870                        maximum / perLoop100);
1871            }
1872        }
1873#endif
1874        processConfigEvents();
1875
1876        mixerStatus = MIXER_IDLE;
1877        { // scope for mLock
1878
1879            Mutex::Autolock _l(mLock);
1880
1881            if (checkForNewParameters_l()) {
1882                mixBufferSize = mFrameCount * mFrameSize;
1883                // FIXME: Relaxed timing because of a certain device that can't meet latency
1884                // Should be reduced to 2x after the vendor fixes the driver issue
1885                maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
1886                activeSleepTime = activeSleepTimeUs();
1887                idleSleepTime = idleSleepTimeUs();
1888            }
1889
1890            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1891
1892            // put audio hardware into standby after short delay
1893            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1894                        mSuspended) {
1895                if (!mStandby) {
1896                    LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1897                    mOutput->stream->common.standby(&mOutput->stream->common);
1898                    mStandby = true;
1899                    mBytesWritten = 0;
1900                }
1901
1902                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1903                    // we're about to wait, flush the binder command buffer
1904                    IPCThreadState::self()->flushCommands();
1905
1906                    if (exitPending()) break;
1907
1908                    releaseWakeLock_l();
1909                    // wait until we have something to do...
1910                    LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1911                    mWaitWorkCV.wait(mLock);
1912                    LOGV("MixerThread %p TID %d waking up\n", this, gettid());
1913                    acquireWakeLock_l();
1914
1915                    if (mMasterMute == false) {
1916                        char value[PROPERTY_VALUE_MAX];
1917                        property_get("ro.audio.silent", value, "0");
1918                        if (atoi(value)) {
1919                            LOGD("Silence is golden");
1920                            setMasterMute(true);
1921                        }
1922                    }
1923
1924                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1925                    sleepTime = idleSleepTime;
1926                    continue;
1927                }
1928            }
1929
1930            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1931
1932            // prevent any changes in effect chain list and in each effect chain
1933            // during mixing and effect process as the audio buffers could be deleted
1934            // or modified if an effect is created or deleted
1935            lockEffectChains_l(effectChains);
1936       }
1937
1938        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1939            // mix buffers...
1940            mAudioMixer->process();
1941            sleepTime = 0;
1942            standbyTime = systemTime() + kStandbyTimeInNsecs;
1943            //TODO: delay standby when effects have a tail
1944        } else {
1945            // If no tracks are ready, sleep once for the duration of an output
1946            // buffer size, then write 0s to the output
1947            if (sleepTime == 0) {
1948                if (mixerStatus == MIXER_TRACKS_ENABLED) {
1949                    sleepTime = activeSleepTime;
1950                } else {
1951                    sleepTime = idleSleepTime;
1952                }
1953            } else if (mBytesWritten != 0 ||
1954                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
1955                memset (mMixBuffer, 0, mixBufferSize);
1956                sleepTime = 0;
1957                LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
1958            }
1959            // TODO add standby time extension fct of effect tail
1960        }
1961
1962        if (mSuspended) {
1963            sleepTime = suspendSleepTimeUs();
1964        }
1965        // sleepTime == 0 means we must write to audio hardware
1966        if (sleepTime == 0) {
1967             for (size_t i = 0; i < effectChains.size(); i ++) {
1968                 effectChains[i]->process_l();
1969             }
1970             // enable changes in effect chain
1971             unlockEffectChains(effectChains);
1972            mLastWriteTime = systemTime();
1973            mInWrite = true;
1974            mBytesWritten += mixBufferSize;
1975
1976            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
1977            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
1978            mNumWrites++;
1979            mInWrite = false;
1980            nsecs_t now = systemTime();
1981            nsecs_t delta = now - mLastWriteTime;
1982            if (delta > maxPeriod) {
1983                mNumDelayedWrites++;
1984                if ((now - lastWarning) > kWarningThrottle) {
1985                    LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
1986                            ns2ms(delta), mNumDelayedWrites, this);
1987                    lastWarning = now;
1988                }
1989                if (mStandby) {
1990                    longStandbyExit = true;
1991                }
1992            }
1993            mStandby = false;
1994        } else {
1995            // enable changes in effect chain
1996            unlockEffectChains(effectChains);
1997            usleep(sleepTime);
1998        }
1999
2000        // finally let go of all our tracks, without the lock held
2001        // since we can't guarantee the destructors won't acquire that
2002        // same lock.
2003        tracksToRemove.clear();
2004
2005        // Effect chains will be actually deleted here if they were removed from
2006        // mEffectChains list during mixing or effects processing
2007        effectChains.clear();
2008    }
2009
2010    if (!mStandby) {
2011        mOutput->stream->common.standby(&mOutput->stream->common);
2012    }
2013
2014    releaseWakeLock();
2015
2016    LOGV("MixerThread %p exiting", this);
2017    return false;
2018}
2019
2020// prepareTracks_l() must be called with ThreadBase::mLock held
2021uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
2022{
2023
2024    uint32_t mixerStatus = MIXER_IDLE;
2025    // find out which tracks need to be processed
2026    size_t count = activeTracks.size();
2027    size_t mixedTracks = 0;
2028    size_t tracksWithEffect = 0;
2029
2030    float masterVolume = mMasterVolume;
2031    bool  masterMute = mMasterMute;
2032
2033    if (masterMute) {
2034        masterVolume = 0;
2035    }
2036    // Delegate master volume control to effect in output mix effect chain if needed
2037    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2038    if (chain != 0) {
2039        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2040        chain->setVolume_l(&v, &v);
2041        masterVolume = (float)((v + (1 << 23)) >> 24);
2042        chain.clear();
2043    }
2044
2045    for (size_t i=0 ; i<count ; i++) {
2046        sp<Track> t = activeTracks[i].promote();
2047        if (t == 0) continue;
2048
2049        Track* const track = t.get();
2050        audio_track_cblk_t* cblk = track->cblk();
2051
2052        // The first time a track is added we wait
2053        // for all its buffers to be filled before processing it
2054        mAudioMixer->setActiveTrack(track->name());
2055        if (cblk->framesReady() && track->isReady() &&
2056                !track->isPaused() && !track->isTerminated())
2057        {
2058            //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
2059
2060            mixedTracks++;
2061
2062            // track->mainBuffer() != mMixBuffer means there is an effect chain
2063            // connected to the track
2064            chain.clear();
2065            if (track->mainBuffer() != mMixBuffer) {
2066                chain = getEffectChain_l(track->sessionId());
2067                // Delegate volume control to effect in track effect chain if needed
2068                if (chain != 0) {
2069                    tracksWithEffect++;
2070                } else {
2071                    LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
2072                            track->name(), track->sessionId());
2073                }
2074            }
2075
2076
2077            int param = AudioMixer::VOLUME;
2078            if (track->mFillingUpStatus == Track::FS_FILLED) {
2079                // no ramp for the first volume setting
2080                track->mFillingUpStatus = Track::FS_ACTIVE;
2081                if (track->mState == TrackBase::RESUMING) {
2082                    track->mState = TrackBase::ACTIVE;
2083                    param = AudioMixer::RAMP_VOLUME;
2084                }
2085                mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2086            } else if (cblk->server != 0) {
2087                // If the track is stopped before the first frame was mixed,
2088                // do not apply ramp
2089                param = AudioMixer::RAMP_VOLUME;
2090            }
2091
2092            // compute volume for this track
2093            uint32_t vl, vr, va;
2094            if (track->isMuted() || track->isPausing() ||
2095                mStreamTypes[track->type()].mute) {
2096                vl = vr = va = 0;
2097                if (track->isPausing()) {
2098                    track->setPaused();
2099                }
2100            } else {
2101
2102                // read original volumes with volume control
2103                float typeVolume = mStreamTypes[track->type()].volume;
2104                float v = masterVolume * typeVolume;
2105                vl = (uint32_t)(v * cblk->volume[0]) << 12;
2106                vr = (uint32_t)(v * cblk->volume[1]) << 12;
2107
2108                va = (uint32_t)(v * cblk->sendLevel);
2109            }
2110            // Delegate volume control to effect in track effect chain if needed
2111            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2112                // Do not ramp volume if volume is controlled by effect
2113                param = AudioMixer::VOLUME;
2114                track->mHasVolumeController = true;
2115            } else {
2116                // force no volume ramp when volume controller was just disabled or removed
2117                // from effect chain to avoid volume spike
2118                if (track->mHasVolumeController) {
2119                    param = AudioMixer::VOLUME;
2120                }
2121                track->mHasVolumeController = false;
2122            }
2123
2124            // Convert volumes from 8.24 to 4.12 format
2125            int16_t left, right, aux;
2126            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2127            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2128            left = int16_t(v_clamped);
2129            v_clamped = (vr + (1 << 11)) >> 12;
2130            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2131            right = int16_t(v_clamped);
2132
2133            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
2134            aux = int16_t(va);
2135
2136            // XXX: these things DON'T need to be done each time
2137            mAudioMixer->setBufferProvider(track);
2138            mAudioMixer->enable(AudioMixer::MIXING);
2139
2140            mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
2141            mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
2142            mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
2143            mAudioMixer->setParameter(
2144                AudioMixer::TRACK,
2145                AudioMixer::FORMAT, (void *)track->format());
2146            mAudioMixer->setParameter(
2147                AudioMixer::TRACK,
2148                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
2149            mAudioMixer->setParameter(
2150                AudioMixer::RESAMPLE,
2151                AudioMixer::SAMPLE_RATE,
2152                (void *)(cblk->sampleRate));
2153            mAudioMixer->setParameter(
2154                AudioMixer::TRACK,
2155                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2156            mAudioMixer->setParameter(
2157                AudioMixer::TRACK,
2158                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2159
2160            // reset retry count
2161            track->mRetryCount = kMaxTrackRetries;
2162            mixerStatus = MIXER_TRACKS_READY;
2163        } else {
2164            //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
2165            if (track->isStopped()) {
2166                track->reset();
2167            }
2168            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2169                // We have consumed all the buffers of this track.
2170                // Remove it from the list of active tracks.
2171                tracksToRemove->add(track);
2172            } else {
2173                // No buffers for this track. Give it a few chances to
2174                // fill a buffer, then remove it from active list.
2175                if (--(track->mRetryCount) <= 0) {
2176                    LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
2177                    tracksToRemove->add(track);
2178                    // indicate to client process that the track was disabled because of underrun
2179                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
2180                } else if (mixerStatus != MIXER_TRACKS_READY) {
2181                    mixerStatus = MIXER_TRACKS_ENABLED;
2182                }
2183            }
2184            mAudioMixer->disable(AudioMixer::MIXING);
2185        }
2186    }
2187
2188    // remove all the tracks that need to be...
2189    count = tracksToRemove->size();
2190    if (UNLIKELY(count)) {
2191        for (size_t i=0 ; i<count ; i++) {
2192            const sp<Track>& track = tracksToRemove->itemAt(i);
2193            mActiveTracks.remove(track);
2194            if (track->mainBuffer() != mMixBuffer) {
2195                chain = getEffectChain_l(track->sessionId());
2196                if (chain != 0) {
2197                    LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
2198                    chain->decActiveTrackCnt();
2199                }
2200            }
2201            if (track->isTerminated()) {
2202                removeTrack_l(track);
2203            }
2204        }
2205    }
2206
2207    // mix buffer must be cleared if all tracks are connected to an
2208    // effect chain as in this case the mixer will not write to
2209    // mix buffer and track effects will accumulate into it
2210    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
2211        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
2212    }
2213
2214    return mixerStatus;
2215}
2216
2217void AudioFlinger::MixerThread::invalidateTracks(int streamType)
2218{
2219    LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
2220            this,  streamType, mTracks.size());
2221    Mutex::Autolock _l(mLock);
2222
2223    size_t size = mTracks.size();
2224    for (size_t i = 0; i < size; i++) {
2225        sp<Track> t = mTracks[i];
2226        if (t->type() == streamType) {
2227            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
2228            t->mCblk->cv.signal();
2229        }
2230    }
2231}
2232
2233void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid)
2234{
2235    LOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
2236            this,  streamType, valid);
2237    Mutex::Autolock _l(mLock);
2238
2239    mStreamTypes[streamType].valid = valid;
2240}
2241
2242// getTrackName_l() must be called with ThreadBase::mLock held
2243int AudioFlinger::MixerThread::getTrackName_l()
2244{
2245    return mAudioMixer->getTrackName();
2246}
2247
2248// deleteTrackName_l() must be called with ThreadBase::mLock held
2249void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2250{
2251    LOGV("remove track (%d) and delete from mixer", name);
2252    mAudioMixer->deleteTrackName(name);
2253}
2254
2255// checkForNewParameters_l() must be called with ThreadBase::mLock held
2256bool AudioFlinger::MixerThread::checkForNewParameters_l()
2257{
2258    bool reconfig = false;
2259
2260    while (!mNewParameters.isEmpty()) {
2261        status_t status = NO_ERROR;
2262        String8 keyValuePair = mNewParameters[0];
2263        AudioParameter param = AudioParameter(keyValuePair);
2264        int value;
2265
2266        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2267            reconfig = true;
2268        }
2269        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2270            if (value != AUDIO_FORMAT_PCM_16_BIT) {
2271                status = BAD_VALUE;
2272            } else {
2273                reconfig = true;
2274            }
2275        }
2276        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2277            if (value != AUDIO_CHANNEL_OUT_STEREO) {
2278                status = BAD_VALUE;
2279            } else {
2280                reconfig = true;
2281            }
2282        }
2283        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2284            // do not accept frame count changes if tracks are open as the track buffer
2285            // size depends on frame count and correct behavior would not be garantied
2286            // if frame count is changed after track creation
2287            if (!mTracks.isEmpty()) {
2288                status = INVALID_OPERATION;
2289            } else {
2290                reconfig = true;
2291            }
2292        }
2293        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2294            // when changing the audio output device, call addBatteryData to notify
2295            // the change
2296            if ((int)mDevice != value) {
2297                uint32_t params = 0;
2298                // check whether speaker is on
2299                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2300                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2301                }
2302
2303                int deviceWithoutSpeaker
2304                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2305                // check if any other device (except speaker) is on
2306                if (value & deviceWithoutSpeaker ) {
2307                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2308                }
2309
2310                if (params != 0) {
2311                    addBatteryData(params);
2312                }
2313            }
2314
2315            // forward device change to effects that have requested to be
2316            // aware of attached audio device.
2317            mDevice = (uint32_t)value;
2318            for (size_t i = 0; i < mEffectChains.size(); i++) {
2319                mEffectChains[i]->setDevice_l(mDevice);
2320            }
2321        }
2322
2323        if (status == NO_ERROR) {
2324            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2325                                                    keyValuePair.string());
2326            if (!mStandby && status == INVALID_OPERATION) {
2327               mOutput->stream->common.standby(&mOutput->stream->common);
2328               mStandby = true;
2329               mBytesWritten = 0;
2330               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2331                                                       keyValuePair.string());
2332            }
2333            if (status == NO_ERROR && reconfig) {
2334                delete mAudioMixer;
2335                readOutputParameters();
2336                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2337                for (size_t i = 0; i < mTracks.size() ; i++) {
2338                    int name = getTrackName_l();
2339                    if (name < 0) break;
2340                    mTracks[i]->mName = name;
2341                    // limit track sample rate to 2 x new output sample rate
2342                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2343                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2344                    }
2345                }
2346                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2347            }
2348        }
2349
2350        mNewParameters.removeAt(0);
2351
2352        mParamStatus = status;
2353        mParamCond.signal();
2354        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2355        // already timed out waiting for the status and will never signal the condition.
2356        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
2357    }
2358    return reconfig;
2359}
2360
2361status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2362{
2363    const size_t SIZE = 256;
2364    char buffer[SIZE];
2365    String8 result;
2366
2367    PlaybackThread::dumpInternals(fd, args);
2368
2369    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2370    result.append(buffer);
2371    write(fd, result.string(), result.size());
2372    return NO_ERROR;
2373}
2374
2375uint32_t AudioFlinger::MixerThread::activeSleepTimeUs()
2376{
2377    return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2378}
2379
2380uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2381{
2382    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2383}
2384
2385uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2386{
2387    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2388}
2389
2390// ----------------------------------------------------------------------------
2391AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2392    :   PlaybackThread(audioFlinger, output, id, device)
2393{
2394    mType = ThreadBase::DIRECT;
2395}
2396
2397AudioFlinger::DirectOutputThread::~DirectOutputThread()
2398{
2399}
2400
2401
2402static inline int16_t clamp16(int32_t sample)
2403{
2404    if ((sample>>15) ^ (sample>>31))
2405        sample = 0x7FFF ^ (sample>>31);
2406    return sample;
2407}
2408
2409static inline
2410int32_t mul(int16_t in, int16_t v)
2411{
2412#if defined(__arm__) && !defined(__thumb__)
2413    int32_t out;
2414    asm( "smulbb %[out], %[in], %[v] \n"
2415         : [out]"=r"(out)
2416         : [in]"%r"(in), [v]"r"(v)
2417         : );
2418    return out;
2419#else
2420    return in * int32_t(v);
2421#endif
2422}
2423
2424void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2425{
2426    // Do not apply volume on compressed audio
2427    if (!audio_is_linear_pcm(mFormat)) {
2428        return;
2429    }
2430
2431    // convert to signed 16 bit before volume calculation
2432    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2433        size_t count = mFrameCount * mChannelCount;
2434        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2435        int16_t *dst = mMixBuffer + count-1;
2436        while(count--) {
2437            *dst-- = (int16_t)(*src--^0x80) << 8;
2438        }
2439    }
2440
2441    size_t frameCount = mFrameCount;
2442    int16_t *out = mMixBuffer;
2443    if (ramp) {
2444        if (mChannelCount == 1) {
2445            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2446            int32_t vlInc = d / (int32_t)frameCount;
2447            int32_t vl = ((int32_t)mLeftVolShort << 16);
2448            do {
2449                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2450                out++;
2451                vl += vlInc;
2452            } while (--frameCount);
2453
2454        } else {
2455            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2456            int32_t vlInc = d / (int32_t)frameCount;
2457            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2458            int32_t vrInc = d / (int32_t)frameCount;
2459            int32_t vl = ((int32_t)mLeftVolShort << 16);
2460            int32_t vr = ((int32_t)mRightVolShort << 16);
2461            do {
2462                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2463                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2464                out += 2;
2465                vl += vlInc;
2466                vr += vrInc;
2467            } while (--frameCount);
2468        }
2469    } else {
2470        if (mChannelCount == 1) {
2471            do {
2472                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2473                out++;
2474            } while (--frameCount);
2475        } else {
2476            do {
2477                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2478                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2479                out += 2;
2480            } while (--frameCount);
2481        }
2482    }
2483
2484    // convert back to unsigned 8 bit after volume calculation
2485    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2486        size_t count = mFrameCount * mChannelCount;
2487        int16_t *src = mMixBuffer;
2488        uint8_t *dst = (uint8_t *)mMixBuffer;
2489        while(count--) {
2490            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2491        }
2492    }
2493
2494    mLeftVolShort = leftVol;
2495    mRightVolShort = rightVol;
2496}
2497
2498bool AudioFlinger::DirectOutputThread::threadLoop()
2499{
2500    uint32_t mixerStatus = MIXER_IDLE;
2501    sp<Track> trackToRemove;
2502    sp<Track> activeTrack;
2503    nsecs_t standbyTime = systemTime();
2504    int8_t *curBuf;
2505    size_t mixBufferSize = mFrameCount*mFrameSize;
2506    uint32_t activeSleepTime = activeSleepTimeUs();
2507    uint32_t idleSleepTime = idleSleepTimeUs();
2508    uint32_t sleepTime = idleSleepTime;
2509    // use shorter standby delay as on normal output to release
2510    // hardware resources as soon as possible
2511    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2512
2513    acquireWakeLock();
2514
2515    while (!exitPending())
2516    {
2517        bool rampVolume;
2518        uint16_t leftVol;
2519        uint16_t rightVol;
2520        Vector< sp<EffectChain> > effectChains;
2521
2522        processConfigEvents();
2523
2524        mixerStatus = MIXER_IDLE;
2525
2526        { // scope for the mLock
2527
2528            Mutex::Autolock _l(mLock);
2529
2530            if (checkForNewParameters_l()) {
2531                mixBufferSize = mFrameCount*mFrameSize;
2532                activeSleepTime = activeSleepTimeUs();
2533                idleSleepTime = idleSleepTimeUs();
2534                standbyDelay = microseconds(activeSleepTime*2);
2535            }
2536
2537            // put audio hardware into standby after short delay
2538            if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2539                        mSuspended) {
2540                // wait until we have something to do...
2541                if (!mStandby) {
2542                    LOGV("Audio hardware entering standby, mixer %p\n", this);
2543                    mOutput->stream->common.standby(&mOutput->stream->common);
2544                    mStandby = true;
2545                    mBytesWritten = 0;
2546                }
2547
2548                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2549                    // we're about to wait, flush the binder command buffer
2550                    IPCThreadState::self()->flushCommands();
2551
2552                    if (exitPending()) break;
2553
2554                    releaseWakeLock_l();
2555                    LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2556                    mWaitWorkCV.wait(mLock);
2557                    LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2558                    acquireWakeLock_l();
2559
2560                    if (mMasterMute == false) {
2561                        char value[PROPERTY_VALUE_MAX];
2562                        property_get("ro.audio.silent", value, "0");
2563                        if (atoi(value)) {
2564                            LOGD("Silence is golden");
2565                            setMasterMute(true);
2566                        }
2567                    }
2568
2569                    standbyTime = systemTime() + standbyDelay;
2570                    sleepTime = idleSleepTime;
2571                    continue;
2572                }
2573            }
2574
2575            effectChains = mEffectChains;
2576
2577            // find out which tracks need to be processed
2578            if (mActiveTracks.size() != 0) {
2579                sp<Track> t = mActiveTracks[0].promote();
2580                if (t == 0) continue;
2581
2582                Track* const track = t.get();
2583                audio_track_cblk_t* cblk = track->cblk();
2584
2585                // The first time a track is added we wait
2586                // for all its buffers to be filled before processing it
2587                if (cblk->framesReady() && track->isReady() &&
2588                        !track->isPaused() && !track->isTerminated())
2589                {
2590                    //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2591
2592                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2593                        track->mFillingUpStatus = Track::FS_ACTIVE;
2594                        mLeftVolFloat = mRightVolFloat = 0;
2595                        mLeftVolShort = mRightVolShort = 0;
2596                        if (track->mState == TrackBase::RESUMING) {
2597                            track->mState = TrackBase::ACTIVE;
2598                            rampVolume = true;
2599                        }
2600                    } else if (cblk->server != 0) {
2601                        // If the track is stopped before the first frame was mixed,
2602                        // do not apply ramp
2603                        rampVolume = true;
2604                    }
2605                    // compute volume for this track
2606                    float left, right;
2607                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2608                        mStreamTypes[track->type()].mute) {
2609                        left = right = 0;
2610                        if (track->isPausing()) {
2611                            track->setPaused();
2612                        }
2613                    } else {
2614                        float typeVolume = mStreamTypes[track->type()].volume;
2615                        float v = mMasterVolume * typeVolume;
2616                        float v_clamped = v * cblk->volume[0];
2617                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2618                        left = v_clamped/MAX_GAIN;
2619                        v_clamped = v * cblk->volume[1];
2620                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2621                        right = v_clamped/MAX_GAIN;
2622                    }
2623
2624                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2625                        mLeftVolFloat = left;
2626                        mRightVolFloat = right;
2627
2628                        // If audio HAL implements volume control,
2629                        // force software volume to nominal value
2630                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2631                            left = 1.0f;
2632                            right = 1.0f;
2633                        }
2634
2635                        // Convert volumes from float to 8.24
2636                        uint32_t vl = (uint32_t)(left * (1 << 24));
2637                        uint32_t vr = (uint32_t)(right * (1 << 24));
2638
2639                        // Delegate volume control to effect in track effect chain if needed
2640                        // only one effect chain can be present on DirectOutputThread, so if
2641                        // there is one, the track is connected to it
2642                        if (!effectChains.isEmpty()) {
2643                            // Do not ramp volume if volume is controlled by effect
2644                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2645                                rampVolume = false;
2646                            }
2647                        }
2648
2649                        // Convert volumes from 8.24 to 4.12 format
2650                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2651                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2652                        leftVol = (uint16_t)v_clamped;
2653                        v_clamped = (vr + (1 << 11)) >> 12;
2654                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2655                        rightVol = (uint16_t)v_clamped;
2656                    } else {
2657                        leftVol = mLeftVolShort;
2658                        rightVol = mRightVolShort;
2659                        rampVolume = false;
2660                    }
2661
2662                    // reset retry count
2663                    track->mRetryCount = kMaxTrackRetriesDirect;
2664                    activeTrack = t;
2665                    mixerStatus = MIXER_TRACKS_READY;
2666                } else {
2667                    //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2668                    if (track->isStopped()) {
2669                        track->reset();
2670                    }
2671                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2672                        // We have consumed all the buffers of this track.
2673                        // Remove it from the list of active tracks.
2674                        trackToRemove = track;
2675                    } else {
2676                        // No buffers for this track. Give it a few chances to
2677                        // fill a buffer, then remove it from active list.
2678                        if (--(track->mRetryCount) <= 0) {
2679                            LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2680                            trackToRemove = track;
2681                        } else {
2682                            mixerStatus = MIXER_TRACKS_ENABLED;
2683                        }
2684                    }
2685                }
2686            }
2687
2688            // remove all the tracks that need to be...
2689            if (UNLIKELY(trackToRemove != 0)) {
2690                mActiveTracks.remove(trackToRemove);
2691                if (!effectChains.isEmpty()) {
2692                    LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2693                            trackToRemove->sessionId());
2694                    effectChains[0]->decActiveTrackCnt();
2695                }
2696                if (trackToRemove->isTerminated()) {
2697                    removeTrack_l(trackToRemove);
2698                }
2699            }
2700
2701            lockEffectChains_l(effectChains);
2702       }
2703
2704        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2705            AudioBufferProvider::Buffer buffer;
2706            size_t frameCount = mFrameCount;
2707            curBuf = (int8_t *)mMixBuffer;
2708            // output audio to hardware
2709            while (frameCount) {
2710                buffer.frameCount = frameCount;
2711                activeTrack->getNextBuffer(&buffer);
2712                if (UNLIKELY(buffer.raw == 0)) {
2713                    memset(curBuf, 0, frameCount * mFrameSize);
2714                    break;
2715                }
2716                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2717                frameCount -= buffer.frameCount;
2718                curBuf += buffer.frameCount * mFrameSize;
2719                activeTrack->releaseBuffer(&buffer);
2720            }
2721            sleepTime = 0;
2722            standbyTime = systemTime() + standbyDelay;
2723        } else {
2724            if (sleepTime == 0) {
2725                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2726                    sleepTime = activeSleepTime;
2727                } else {
2728                    sleepTime = idleSleepTime;
2729                }
2730            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2731                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2732                sleepTime = 0;
2733            }
2734        }
2735
2736        if (mSuspended) {
2737            sleepTime = suspendSleepTimeUs();
2738        }
2739        // sleepTime == 0 means we must write to audio hardware
2740        if (sleepTime == 0) {
2741            if (mixerStatus == MIXER_TRACKS_READY) {
2742                applyVolume(leftVol, rightVol, rampVolume);
2743            }
2744            for (size_t i = 0; i < effectChains.size(); i ++) {
2745                effectChains[i]->process_l();
2746            }
2747            unlockEffectChains(effectChains);
2748
2749            mLastWriteTime = systemTime();
2750            mInWrite = true;
2751            mBytesWritten += mixBufferSize;
2752            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2753            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2754            mNumWrites++;
2755            mInWrite = false;
2756            mStandby = false;
2757        } else {
2758            unlockEffectChains(effectChains);
2759            usleep(sleepTime);
2760        }
2761
2762        // finally let go of removed track, without the lock held
2763        // since we can't guarantee the destructors won't acquire that
2764        // same lock.
2765        trackToRemove.clear();
2766        activeTrack.clear();
2767
2768        // Effect chains will be actually deleted here if they were removed from
2769        // mEffectChains list during mixing or effects processing
2770        effectChains.clear();
2771    }
2772
2773    if (!mStandby) {
2774        mOutput->stream->common.standby(&mOutput->stream->common);
2775    }
2776
2777    releaseWakeLock();
2778
2779    LOGV("DirectOutputThread %p exiting", this);
2780    return false;
2781}
2782
2783// getTrackName_l() must be called with ThreadBase::mLock held
2784int AudioFlinger::DirectOutputThread::getTrackName_l()
2785{
2786    return 0;
2787}
2788
2789// deleteTrackName_l() must be called with ThreadBase::mLock held
2790void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2791{
2792}
2793
2794// checkForNewParameters_l() must be called with ThreadBase::mLock held
2795bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2796{
2797    bool reconfig = false;
2798
2799    while (!mNewParameters.isEmpty()) {
2800        status_t status = NO_ERROR;
2801        String8 keyValuePair = mNewParameters[0];
2802        AudioParameter param = AudioParameter(keyValuePair);
2803        int value;
2804
2805        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2806            // do not accept frame count changes if tracks are open as the track buffer
2807            // size depends on frame count and correct behavior would not be garantied
2808            // if frame count is changed after track creation
2809            if (!mTracks.isEmpty()) {
2810                status = INVALID_OPERATION;
2811            } else {
2812                reconfig = true;
2813            }
2814        }
2815        if (status == NO_ERROR) {
2816            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2817                                                    keyValuePair.string());
2818            if (!mStandby && status == INVALID_OPERATION) {
2819               mOutput->stream->common.standby(&mOutput->stream->common);
2820               mStandby = true;
2821               mBytesWritten = 0;
2822               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2823                                                       keyValuePair.string());
2824            }
2825            if (status == NO_ERROR && reconfig) {
2826                readOutputParameters();
2827                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2828            }
2829        }
2830
2831        mNewParameters.removeAt(0);
2832
2833        mParamStatus = status;
2834        mParamCond.signal();
2835        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
2836        // already timed out waiting for the status and will never signal the condition.
2837        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
2838    }
2839    return reconfig;
2840}
2841
2842uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2843{
2844    uint32_t time;
2845    if (audio_is_linear_pcm(mFormat)) {
2846        time = (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2847    } else {
2848        time = 10000;
2849    }
2850    return time;
2851}
2852
2853uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2854{
2855    uint32_t time;
2856    if (audio_is_linear_pcm(mFormat)) {
2857        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2858    } else {
2859        time = 10000;
2860    }
2861    return time;
2862}
2863
2864uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2865{
2866    uint32_t time;
2867    if (audio_is_linear_pcm(mFormat)) {
2868        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2869    } else {
2870        time = 10000;
2871    }
2872    return time;
2873}
2874
2875
2876// ----------------------------------------------------------------------------
2877
2878AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2879    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2880{
2881    mType = ThreadBase::DUPLICATING;
2882    addOutputTrack(mainThread);
2883}
2884
2885AudioFlinger::DuplicatingThread::~DuplicatingThread()
2886{
2887    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2888        mOutputTracks[i]->destroy();
2889    }
2890    mOutputTracks.clear();
2891}
2892
2893bool AudioFlinger::DuplicatingThread::threadLoop()
2894{
2895    Vector< sp<Track> > tracksToRemove;
2896    uint32_t mixerStatus = MIXER_IDLE;
2897    nsecs_t standbyTime = systemTime();
2898    size_t mixBufferSize = mFrameCount*mFrameSize;
2899    SortedVector< sp<OutputTrack> > outputTracks;
2900    uint32_t writeFrames = 0;
2901    uint32_t activeSleepTime = activeSleepTimeUs();
2902    uint32_t idleSleepTime = idleSleepTimeUs();
2903    uint32_t sleepTime = idleSleepTime;
2904    Vector< sp<EffectChain> > effectChains;
2905
2906    acquireWakeLock();
2907
2908    while (!exitPending())
2909    {
2910        processConfigEvents();
2911
2912        mixerStatus = MIXER_IDLE;
2913        { // scope for the mLock
2914
2915            Mutex::Autolock _l(mLock);
2916
2917            if (checkForNewParameters_l()) {
2918                mixBufferSize = mFrameCount*mFrameSize;
2919                updateWaitTime();
2920                activeSleepTime = activeSleepTimeUs();
2921                idleSleepTime = idleSleepTimeUs();
2922            }
2923
2924            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
2925
2926            for (size_t i = 0; i < mOutputTracks.size(); i++) {
2927                outputTracks.add(mOutputTracks[i]);
2928            }
2929
2930            // put audio hardware into standby after short delay
2931            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
2932                         mSuspended) {
2933                if (!mStandby) {
2934                    for (size_t i = 0; i < outputTracks.size(); i++) {
2935                        outputTracks[i]->stop();
2936                    }
2937                    mStandby = true;
2938                    mBytesWritten = 0;
2939                }
2940
2941                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
2942                    // we're about to wait, flush the binder command buffer
2943                    IPCThreadState::self()->flushCommands();
2944                    outputTracks.clear();
2945
2946                    if (exitPending()) break;
2947
2948                    releaseWakeLock_l();
2949                    LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
2950                    mWaitWorkCV.wait(mLock);
2951                    LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
2952                    acquireWakeLock_l();
2953
2954                    if (mMasterMute == false) {
2955                        char value[PROPERTY_VALUE_MAX];
2956                        property_get("ro.audio.silent", value, "0");
2957                        if (atoi(value)) {
2958                            LOGD("Silence is golden");
2959                            setMasterMute(true);
2960                        }
2961                    }
2962
2963                    standbyTime = systemTime() + kStandbyTimeInNsecs;
2964                    sleepTime = idleSleepTime;
2965                    continue;
2966                }
2967            }
2968
2969            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
2970
2971            // prevent any changes in effect chain list and in each effect chain
2972            // during mixing and effect process as the audio buffers could be deleted
2973            // or modified if an effect is created or deleted
2974            lockEffectChains_l(effectChains);
2975        }
2976
2977        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2978            // mix buffers...
2979            if (outputsReady(outputTracks)) {
2980                mAudioMixer->process();
2981            } else {
2982                memset(mMixBuffer, 0, mixBufferSize);
2983            }
2984            sleepTime = 0;
2985            writeFrames = mFrameCount;
2986        } else {
2987            if (sleepTime == 0) {
2988                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2989                    sleepTime = activeSleepTime;
2990                } else {
2991                    sleepTime = idleSleepTime;
2992                }
2993            } else if (mBytesWritten != 0) {
2994                // flush remaining overflow buffers in output tracks
2995                for (size_t i = 0; i < outputTracks.size(); i++) {
2996                    if (outputTracks[i]->isActive()) {
2997                        sleepTime = 0;
2998                        writeFrames = 0;
2999                        memset(mMixBuffer, 0, mixBufferSize);
3000                        break;
3001                    }
3002                }
3003            }
3004        }
3005
3006        if (mSuspended) {
3007            sleepTime = suspendSleepTimeUs();
3008        }
3009        // sleepTime == 0 means we must write to audio hardware
3010        if (sleepTime == 0) {
3011            for (size_t i = 0; i < effectChains.size(); i ++) {
3012                effectChains[i]->process_l();
3013            }
3014            // enable changes in effect chain
3015            unlockEffectChains(effectChains);
3016
3017            standbyTime = systemTime() + kStandbyTimeInNsecs;
3018            for (size_t i = 0; i < outputTracks.size(); i++) {
3019                outputTracks[i]->write(mMixBuffer, writeFrames);
3020            }
3021            mStandby = false;
3022            mBytesWritten += mixBufferSize;
3023        } else {
3024            // enable changes in effect chain
3025            unlockEffectChains(effectChains);
3026            usleep(sleepTime);
3027        }
3028
3029        // finally let go of all our tracks, without the lock held
3030        // since we can't guarantee the destructors won't acquire that
3031        // same lock.
3032        tracksToRemove.clear();
3033        outputTracks.clear();
3034
3035        // Effect chains will be actually deleted here if they were removed from
3036        // mEffectChains list during mixing or effects processing
3037        effectChains.clear();
3038    }
3039
3040    releaseWakeLock();
3041
3042    return false;
3043}
3044
3045void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3046{
3047    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
3048    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
3049                                            this,
3050                                            mSampleRate,
3051                                            mFormat,
3052                                            mChannelMask,
3053                                            frameCount);
3054    if (outputTrack->cblk() != NULL) {
3055        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3056        mOutputTracks.add(outputTrack);
3057        LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3058        updateWaitTime();
3059    }
3060}
3061
3062void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3063{
3064    Mutex::Autolock _l(mLock);
3065    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3066        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
3067            mOutputTracks[i]->destroy();
3068            mOutputTracks.removeAt(i);
3069            updateWaitTime();
3070            return;
3071        }
3072    }
3073    LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3074}
3075
3076void AudioFlinger::DuplicatingThread::updateWaitTime()
3077{
3078    mWaitTimeMs = UINT_MAX;
3079    for (size_t i = 0; i < mOutputTracks.size(); i++) {
3080        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3081        if (strong != NULL) {
3082            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3083            if (waitTimeMs < mWaitTimeMs) {
3084                mWaitTimeMs = waitTimeMs;
3085            }
3086        }
3087    }
3088}
3089
3090
3091bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
3092{
3093    for (size_t i = 0; i < outputTracks.size(); i++) {
3094        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
3095        if (thread == 0) {
3096            LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
3097            return false;
3098        }
3099        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3100        if (playbackThread->standby() && !playbackThread->isSuspended()) {
3101            LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
3102            return false;
3103        }
3104    }
3105    return true;
3106}
3107
3108uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
3109{
3110    return (mWaitTimeMs * 1000) / 2;
3111}
3112
3113// ----------------------------------------------------------------------------
3114
3115// TrackBase constructor must be called with AudioFlinger::mLock held
3116AudioFlinger::ThreadBase::TrackBase::TrackBase(
3117            const wp<ThreadBase>& thread,
3118            const sp<Client>& client,
3119            uint32_t sampleRate,
3120            uint32_t format,
3121            uint32_t channelMask,
3122            int frameCount,
3123            uint32_t flags,
3124            const sp<IMemory>& sharedBuffer,
3125            int sessionId)
3126    :   RefBase(),
3127        mThread(thread),
3128        mClient(client),
3129        mCblk(0),
3130        mFrameCount(0),
3131        mState(IDLE),
3132        mClientTid(-1),
3133        mFormat(format),
3134        mFlags(flags & ~SYSTEM_FLAGS_MASK),
3135        mSessionId(sessionId)
3136{
3137    LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
3138
3139    // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
3140   size_t size = sizeof(audio_track_cblk_t);
3141   uint8_t channelCount = popcount(channelMask);
3142   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
3143   if (sharedBuffer == 0) {
3144       size += bufferSize;
3145   }
3146
3147   if (client != NULL) {
3148        mCblkMemory = client->heap()->allocate(size);
3149        if (mCblkMemory != 0) {
3150            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
3151            if (mCblk) { // construct the shared structure in-place.
3152                new(mCblk) audio_track_cblk_t();
3153                // clear all buffers
3154                mCblk->frameCount = frameCount;
3155                mCblk->sampleRate = sampleRate;
3156                mChannelCount = channelCount;
3157                mChannelMask = channelMask;
3158                if (sharedBuffer == 0) {
3159                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3160                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3161                    // Force underrun condition to avoid false underrun callback until first data is
3162                    // written to buffer (other flags are cleared)
3163                    mCblk->flags = CBLK_UNDERRUN_ON;
3164                } else {
3165                    mBuffer = sharedBuffer->pointer();
3166                }
3167                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3168            }
3169        } else {
3170            LOGE("not enough memory for AudioTrack size=%u", size);
3171            client->heap()->dump("AudioTrack");
3172            return;
3173        }
3174   } else {
3175       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
3176       if (mCblk) { // construct the shared structure in-place.
3177           new(mCblk) audio_track_cblk_t();
3178           // clear all buffers
3179           mCblk->frameCount = frameCount;
3180           mCblk->sampleRate = sampleRate;
3181           mChannelCount = channelCount;
3182           mChannelMask = channelMask;
3183           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
3184           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
3185           // Force underrun condition to avoid false underrun callback until first data is
3186           // written to buffer (other flags are cleared)
3187           mCblk->flags = CBLK_UNDERRUN_ON;
3188           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
3189       }
3190   }
3191}
3192
3193AudioFlinger::ThreadBase::TrackBase::~TrackBase()
3194{
3195    if (mCblk) {
3196        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
3197        if (mClient == NULL) {
3198            delete mCblk;
3199        }
3200    }
3201    mCblkMemory.clear();            // and free the shared memory
3202    if (mClient != NULL) {
3203        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
3204        mClient.clear();
3205    }
3206}
3207
3208void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
3209{
3210    buffer->raw = 0;
3211    mFrameCount = buffer->frameCount;
3212    step();
3213    buffer->frameCount = 0;
3214}
3215
3216bool AudioFlinger::ThreadBase::TrackBase::step() {
3217    bool result;
3218    audio_track_cblk_t* cblk = this->cblk();
3219
3220    result = cblk->stepServer(mFrameCount);
3221    if (!result) {
3222        LOGV("stepServer failed acquiring cblk mutex");
3223        mFlags |= STEPSERVER_FAILED;
3224    }
3225    return result;
3226}
3227
3228void AudioFlinger::ThreadBase::TrackBase::reset() {
3229    audio_track_cblk_t* cblk = this->cblk();
3230
3231    cblk->user = 0;
3232    cblk->server = 0;
3233    cblk->userBase = 0;
3234    cblk->serverBase = 0;
3235    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
3236    LOGV("TrackBase::reset");
3237}
3238
3239sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
3240{
3241    return mCblkMemory;
3242}
3243
3244int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
3245    return (int)mCblk->sampleRate;
3246}
3247
3248int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
3249    return (const int)mChannelCount;
3250}
3251
3252uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
3253    return mChannelMask;
3254}
3255
3256void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
3257    audio_track_cblk_t* cblk = this->cblk();
3258    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
3259    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
3260
3261    // Check validity of returned pointer in case the track control block would have been corrupted.
3262    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
3263        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
3264        LOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
3265                server %d, serverBase %d, user %d, userBase %d",
3266                bufferStart, bufferEnd, mBuffer, mBufferEnd,
3267                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
3268        return 0;
3269    }
3270
3271    return bufferStart;
3272}
3273
3274// ----------------------------------------------------------------------------
3275
3276// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
3277AudioFlinger::PlaybackThread::Track::Track(
3278            const wp<ThreadBase>& thread,
3279            const sp<Client>& client,
3280            int streamType,
3281            uint32_t sampleRate,
3282            uint32_t format,
3283            uint32_t channelMask,
3284            int frameCount,
3285            const sp<IMemory>& sharedBuffer,
3286            int sessionId)
3287    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
3288    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
3289    mAuxEffectId(0), mHasVolumeController(false)
3290{
3291    if (mCblk != NULL) {
3292        sp<ThreadBase> baseThread = thread.promote();
3293        if (baseThread != 0) {
3294            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
3295            mName = playbackThread->getTrackName_l();
3296            mMainBuffer = playbackThread->mixBuffer();
3297        }
3298        LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3299        if (mName < 0) {
3300            LOGE("no more track names available");
3301        }
3302        mVolume[0] = 1.0f;
3303        mVolume[1] = 1.0f;
3304        mStreamType = streamType;
3305        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
3306        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
3307        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
3308    }
3309}
3310
3311AudioFlinger::PlaybackThread::Track::~Track()
3312{
3313    LOGV("PlaybackThread::Track destructor");
3314    sp<ThreadBase> thread = mThread.promote();
3315    if (thread != 0) {
3316        Mutex::Autolock _l(thread->mLock);
3317        mState = TERMINATED;
3318    }
3319}
3320
3321void AudioFlinger::PlaybackThread::Track::destroy()
3322{
3323    // NOTE: destroyTrack_l() can remove a strong reference to this Track
3324    // by removing it from mTracks vector, so there is a risk that this Tracks's
3325    // desctructor is called. As the destructor needs to lock mLock,
3326    // we must acquire a strong reference on this Track before locking mLock
3327    // here so that the destructor is called only when exiting this function.
3328    // On the other hand, as long as Track::destroy() is only called by
3329    // TrackHandle destructor, the TrackHandle still holds a strong ref on
3330    // this Track with its member mTrack.
3331    sp<Track> keep(this);
3332    { // scope for mLock
3333        sp<ThreadBase> thread = mThread.promote();
3334        if (thread != 0) {
3335            if (!isOutputTrack()) {
3336                if (mState == ACTIVE || mState == RESUMING) {
3337                    AudioSystem::stopOutput(thread->id(),
3338                                            (audio_stream_type_t)mStreamType,
3339                                            mSessionId);
3340
3341                    // to track the speaker usage
3342                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3343                }
3344                AudioSystem::releaseOutput(thread->id());
3345            }
3346            Mutex::Autolock _l(thread->mLock);
3347            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3348            playbackThread->destroyTrack_l(this);
3349        }
3350    }
3351}
3352
3353void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3354{
3355    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",
3356            mName - AudioMixer::TRACK0,
3357            (mClient == NULL) ? getpid() : mClient->pid(),
3358            mStreamType,
3359            mFormat,
3360            mChannelMask,
3361            mSessionId,
3362            mFrameCount,
3363            mState,
3364            mMute,
3365            mFillingUpStatus,
3366            mCblk->sampleRate,
3367            mCblk->volume[0],
3368            mCblk->volume[1],
3369            mCblk->server,
3370            mCblk->user,
3371            (int)mMainBuffer,
3372            (int)mAuxBuffer);
3373}
3374
3375status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3376{
3377     audio_track_cblk_t* cblk = this->cblk();
3378     uint32_t framesReady;
3379     uint32_t framesReq = buffer->frameCount;
3380
3381     // Check if last stepServer failed, try to step now
3382     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3383         if (!step())  goto getNextBuffer_exit;
3384         LOGV("stepServer recovered");
3385         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3386     }
3387
3388     framesReady = cblk->framesReady();
3389
3390     if (LIKELY(framesReady)) {
3391        uint32_t s = cblk->server;
3392        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3393
3394        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3395        if (framesReq > framesReady) {
3396            framesReq = framesReady;
3397        }
3398        if (s + framesReq > bufferEnd) {
3399            framesReq = bufferEnd - s;
3400        }
3401
3402         buffer->raw = getBuffer(s, framesReq);
3403         if (buffer->raw == 0) goto getNextBuffer_exit;
3404
3405         buffer->frameCount = framesReq;
3406        return NO_ERROR;
3407     }
3408
3409getNextBuffer_exit:
3410     buffer->raw = 0;
3411     buffer->frameCount = 0;
3412     LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3413     return NOT_ENOUGH_DATA;
3414}
3415
3416bool AudioFlinger::PlaybackThread::Track::isReady() const {
3417    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3418
3419    if (mCblk->framesReady() >= mCblk->frameCount ||
3420            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3421        mFillingUpStatus = FS_FILLED;
3422        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3423        return true;
3424    }
3425    return false;
3426}
3427
3428status_t AudioFlinger::PlaybackThread::Track::start()
3429{
3430    status_t status = NO_ERROR;
3431    LOGV("start(%d), calling thread %d session %d",
3432            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3433    sp<ThreadBase> thread = mThread.promote();
3434    if (thread != 0) {
3435        Mutex::Autolock _l(thread->mLock);
3436        int state = mState;
3437        // here the track could be either new, or restarted
3438        // in both cases "unstop" the track
3439        if (mState == PAUSED) {
3440            mState = TrackBase::RESUMING;
3441            LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3442        } else {
3443            mState = TrackBase::ACTIVE;
3444            LOGV("? => ACTIVE (%d) on thread %p", mName, this);
3445        }
3446
3447        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3448            thread->mLock.unlock();
3449            status = AudioSystem::startOutput(thread->id(),
3450                                              (audio_stream_type_t)mStreamType,
3451                                              mSessionId);
3452            thread->mLock.lock();
3453
3454            // to track the speaker usage
3455            if (status == NO_ERROR) {
3456                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3457            }
3458        }
3459        if (status == NO_ERROR) {
3460            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3461            playbackThread->addTrack_l(this);
3462        } else {
3463            mState = state;
3464        }
3465    } else {
3466        status = BAD_VALUE;
3467    }
3468    return status;
3469}
3470
3471void AudioFlinger::PlaybackThread::Track::stop()
3472{
3473    LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3474    sp<ThreadBase> thread = mThread.promote();
3475    if (thread != 0) {
3476        Mutex::Autolock _l(thread->mLock);
3477        int state = mState;
3478        if (mState > STOPPED) {
3479            mState = STOPPED;
3480            // If the track is not active (PAUSED and buffers full), flush buffers
3481            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3482            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3483                reset();
3484            }
3485            LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3486        }
3487        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3488            thread->mLock.unlock();
3489            AudioSystem::stopOutput(thread->id(),
3490                                    (audio_stream_type_t)mStreamType,
3491                                    mSessionId);
3492            thread->mLock.lock();
3493
3494            // to track the speaker usage
3495            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3496        }
3497    }
3498}
3499
3500void AudioFlinger::PlaybackThread::Track::pause()
3501{
3502    LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3503    sp<ThreadBase> thread = mThread.promote();
3504    if (thread != 0) {
3505        Mutex::Autolock _l(thread->mLock);
3506        if (mState == ACTIVE || mState == RESUMING) {
3507            mState = PAUSING;
3508            LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3509            if (!isOutputTrack()) {
3510                thread->mLock.unlock();
3511                AudioSystem::stopOutput(thread->id(),
3512                                        (audio_stream_type_t)mStreamType,
3513                                        mSessionId);
3514                thread->mLock.lock();
3515
3516                // to track the speaker usage
3517                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3518            }
3519        }
3520    }
3521}
3522
3523void AudioFlinger::PlaybackThread::Track::flush()
3524{
3525    LOGV("flush(%d)", mName);
3526    sp<ThreadBase> thread = mThread.promote();
3527    if (thread != 0) {
3528        Mutex::Autolock _l(thread->mLock);
3529        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3530            return;
3531        }
3532        // No point remaining in PAUSED state after a flush => go to
3533        // STOPPED state
3534        mState = STOPPED;
3535
3536        // do not reset the track if it is still in the process of being stopped or paused.
3537        // this will be done by prepareTracks_l() when the track is stopped.
3538        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3539        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3540            reset();
3541        }
3542    }
3543}
3544
3545void AudioFlinger::PlaybackThread::Track::reset()
3546{
3547    // Do not reset twice to avoid discarding data written just after a flush and before
3548    // the audioflinger thread detects the track is stopped.
3549    if (!mResetDone) {
3550        TrackBase::reset();
3551        // Force underrun condition to avoid false underrun callback until first data is
3552        // written to buffer
3553        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3554        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3555        mFillingUpStatus = FS_FILLING;
3556        mResetDone = true;
3557    }
3558}
3559
3560void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3561{
3562    mMute = muted;
3563}
3564
3565void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3566{
3567    mVolume[0] = left;
3568    mVolume[1] = right;
3569}
3570
3571status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3572{
3573    status_t status = DEAD_OBJECT;
3574    sp<ThreadBase> thread = mThread.promote();
3575    if (thread != 0) {
3576       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3577       status = playbackThread->attachAuxEffect(this, EffectId);
3578    }
3579    return status;
3580}
3581
3582void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3583{
3584    mAuxEffectId = EffectId;
3585    mAuxBuffer = buffer;
3586}
3587
3588// ----------------------------------------------------------------------------
3589
3590// RecordTrack constructor must be called with AudioFlinger::mLock held
3591AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3592            const wp<ThreadBase>& thread,
3593            const sp<Client>& client,
3594            uint32_t sampleRate,
3595            uint32_t format,
3596            uint32_t channelMask,
3597            int frameCount,
3598            uint32_t flags,
3599            int sessionId)
3600    :   TrackBase(thread, client, sampleRate, format,
3601                  channelMask, frameCount, flags, 0, sessionId),
3602        mOverflow(false)
3603{
3604    if (mCblk != NULL) {
3605       LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3606       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3607           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3608       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3609           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3610       } else {
3611           mCblk->frameSize = sizeof(int8_t);
3612       }
3613    }
3614}
3615
3616AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3617{
3618    sp<ThreadBase> thread = mThread.promote();
3619    if (thread != 0) {
3620        AudioSystem::releaseInput(thread->id());
3621    }
3622}
3623
3624status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3625{
3626    audio_track_cblk_t* cblk = this->cblk();
3627    uint32_t framesAvail;
3628    uint32_t framesReq = buffer->frameCount;
3629
3630     // Check if last stepServer failed, try to step now
3631    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3632        if (!step()) goto getNextBuffer_exit;
3633        LOGV("stepServer recovered");
3634        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3635    }
3636
3637    framesAvail = cblk->framesAvailable_l();
3638
3639    if (LIKELY(framesAvail)) {
3640        uint32_t s = cblk->server;
3641        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3642
3643        if (framesReq > framesAvail) {
3644            framesReq = framesAvail;
3645        }
3646        if (s + framesReq > bufferEnd) {
3647            framesReq = bufferEnd - s;
3648        }
3649
3650        buffer->raw = getBuffer(s, framesReq);
3651        if (buffer->raw == 0) goto getNextBuffer_exit;
3652
3653        buffer->frameCount = framesReq;
3654        return NO_ERROR;
3655    }
3656
3657getNextBuffer_exit:
3658    buffer->raw = 0;
3659    buffer->frameCount = 0;
3660    return NOT_ENOUGH_DATA;
3661}
3662
3663status_t AudioFlinger::RecordThread::RecordTrack::start()
3664{
3665    sp<ThreadBase> thread = mThread.promote();
3666    if (thread != 0) {
3667        RecordThread *recordThread = (RecordThread *)thread.get();
3668        return recordThread->start(this);
3669    } else {
3670        return BAD_VALUE;
3671    }
3672}
3673
3674void AudioFlinger::RecordThread::RecordTrack::stop()
3675{
3676    sp<ThreadBase> thread = mThread.promote();
3677    if (thread != 0) {
3678        RecordThread *recordThread = (RecordThread *)thread.get();
3679        recordThread->stop(this);
3680        TrackBase::reset();
3681        // Force overerrun condition to avoid false overrun callback until first data is
3682        // read from buffer
3683        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3684    }
3685}
3686
3687void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3688{
3689    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3690            (mClient == NULL) ? getpid() : mClient->pid(),
3691            mFormat,
3692            mChannelMask,
3693            mSessionId,
3694            mFrameCount,
3695            mState,
3696            mCblk->sampleRate,
3697            mCblk->server,
3698            mCblk->user);
3699}
3700
3701
3702// ----------------------------------------------------------------------------
3703
3704AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3705            const wp<ThreadBase>& thread,
3706            DuplicatingThread *sourceThread,
3707            uint32_t sampleRate,
3708            uint32_t format,
3709            uint32_t channelMask,
3710            int frameCount)
3711    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3712    mActive(false), mSourceThread(sourceThread)
3713{
3714
3715    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3716    if (mCblk != NULL) {
3717        mCblk->flags |= CBLK_DIRECTION_OUT;
3718        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3719        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
3720        mOutBuffer.frameCount = 0;
3721        playbackThread->mTracks.add(this);
3722        LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3723                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3724                mCblk, mBuffer, mCblk->buffers,
3725                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3726    } else {
3727        LOGW("Error creating output track on thread %p", playbackThread);
3728    }
3729}
3730
3731AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3732{
3733    clearBufferQueue();
3734}
3735
3736status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3737{
3738    status_t status = Track::start();
3739    if (status != NO_ERROR) {
3740        return status;
3741    }
3742
3743    mActive = true;
3744    mRetryCount = 127;
3745    return status;
3746}
3747
3748void AudioFlinger::PlaybackThread::OutputTrack::stop()
3749{
3750    Track::stop();
3751    clearBufferQueue();
3752    mOutBuffer.frameCount = 0;
3753    mActive = false;
3754}
3755
3756bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3757{
3758    Buffer *pInBuffer;
3759    Buffer inBuffer;
3760    uint32_t channelCount = mChannelCount;
3761    bool outputBufferFull = false;
3762    inBuffer.frameCount = frames;
3763    inBuffer.i16 = data;
3764
3765    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3766
3767    if (!mActive && frames != 0) {
3768        start();
3769        sp<ThreadBase> thread = mThread.promote();
3770        if (thread != 0) {
3771            MixerThread *mixerThread = (MixerThread *)thread.get();
3772            if (mCblk->frameCount > frames){
3773                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3774                    uint32_t startFrames = (mCblk->frameCount - frames);
3775                    pInBuffer = new Buffer;
3776                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3777                    pInBuffer->frameCount = startFrames;
3778                    pInBuffer->i16 = pInBuffer->mBuffer;
3779                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3780                    mBufferQueue.add(pInBuffer);
3781                } else {
3782                    LOGW ("OutputTrack::write() %p no more buffers in queue", this);
3783                }
3784            }
3785        }
3786    }
3787
3788    while (waitTimeLeftMs) {
3789        // First write pending buffers, then new data
3790        if (mBufferQueue.size()) {
3791            pInBuffer = mBufferQueue.itemAt(0);
3792        } else {
3793            pInBuffer = &inBuffer;
3794        }
3795
3796        if (pInBuffer->frameCount == 0) {
3797            break;
3798        }
3799
3800        if (mOutBuffer.frameCount == 0) {
3801            mOutBuffer.frameCount = pInBuffer->frameCount;
3802            nsecs_t startTime = systemTime();
3803            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
3804                LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3805                outputBufferFull = true;
3806                break;
3807            }
3808            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3809            if (waitTimeLeftMs >= waitTimeMs) {
3810                waitTimeLeftMs -= waitTimeMs;
3811            } else {
3812                waitTimeLeftMs = 0;
3813            }
3814        }
3815
3816        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3817        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3818        mCblk->stepUser(outFrames);
3819        pInBuffer->frameCount -= outFrames;
3820        pInBuffer->i16 += outFrames * channelCount;
3821        mOutBuffer.frameCount -= outFrames;
3822        mOutBuffer.i16 += outFrames * channelCount;
3823
3824        if (pInBuffer->frameCount == 0) {
3825            if (mBufferQueue.size()) {
3826                mBufferQueue.removeAt(0);
3827                delete [] pInBuffer->mBuffer;
3828                delete pInBuffer;
3829                LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3830            } else {
3831                break;
3832            }
3833        }
3834    }
3835
3836    // If we could not write all frames, allocate a buffer and queue it for next time.
3837    if (inBuffer.frameCount) {
3838        sp<ThreadBase> thread = mThread.promote();
3839        if (thread != 0 && !thread->standby()) {
3840            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3841                pInBuffer = new Buffer;
3842                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3843                pInBuffer->frameCount = inBuffer.frameCount;
3844                pInBuffer->i16 = pInBuffer->mBuffer;
3845                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3846                mBufferQueue.add(pInBuffer);
3847                LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3848            } else {
3849                LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3850            }
3851        }
3852    }
3853
3854    // Calling write() with a 0 length buffer, means that no more data will be written:
3855    // If no more buffers are pending, fill output track buffer to make sure it is started
3856    // by output mixer.
3857    if (frames == 0 && mBufferQueue.size() == 0) {
3858        if (mCblk->user < mCblk->frameCount) {
3859            frames = mCblk->frameCount - mCblk->user;
3860            pInBuffer = new Buffer;
3861            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3862            pInBuffer->frameCount = frames;
3863            pInBuffer->i16 = pInBuffer->mBuffer;
3864            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3865            mBufferQueue.add(pInBuffer);
3866        } else if (mActive) {
3867            stop();
3868        }
3869    }
3870
3871    return outputBufferFull;
3872}
3873
3874status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3875{
3876    int active;
3877    status_t result;
3878    audio_track_cblk_t* cblk = mCblk;
3879    uint32_t framesReq = buffer->frameCount;
3880
3881//    LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3882    buffer->frameCount  = 0;
3883
3884    uint32_t framesAvail = cblk->framesAvailable();
3885
3886
3887    if (framesAvail == 0) {
3888        Mutex::Autolock _l(cblk->lock);
3889        goto start_loop_here;
3890        while (framesAvail == 0) {
3891            active = mActive;
3892            if (UNLIKELY(!active)) {
3893                LOGV("Not active and NO_MORE_BUFFERS");
3894                return AudioTrack::NO_MORE_BUFFERS;
3895            }
3896            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
3897            if (result != NO_ERROR) {
3898                return AudioTrack::NO_MORE_BUFFERS;
3899            }
3900            // read the server count again
3901        start_loop_here:
3902            framesAvail = cblk->framesAvailable_l();
3903        }
3904    }
3905
3906//    if (framesAvail < framesReq) {
3907//        return AudioTrack::NO_MORE_BUFFERS;
3908//    }
3909
3910    if (framesReq > framesAvail) {
3911        framesReq = framesAvail;
3912    }
3913
3914    uint32_t u = cblk->user;
3915    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
3916
3917    if (u + framesReq > bufferEnd) {
3918        framesReq = bufferEnd - u;
3919    }
3920
3921    buffer->frameCount  = framesReq;
3922    buffer->raw         = (void *)cblk->buffer(u);
3923    return NO_ERROR;
3924}
3925
3926
3927void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
3928{
3929    size_t size = mBufferQueue.size();
3930    Buffer *pBuffer;
3931
3932    for (size_t i = 0; i < size; i++) {
3933        pBuffer = mBufferQueue.itemAt(i);
3934        delete [] pBuffer->mBuffer;
3935        delete pBuffer;
3936    }
3937    mBufferQueue.clear();
3938}
3939
3940// ----------------------------------------------------------------------------
3941
3942AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
3943    :   RefBase(),
3944        mAudioFlinger(audioFlinger),
3945        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
3946        mPid(pid)
3947{
3948    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
3949}
3950
3951// Client destructor must be called with AudioFlinger::mLock held
3952AudioFlinger::Client::~Client()
3953{
3954    mAudioFlinger->removeClient_l(mPid);
3955}
3956
3957const sp<MemoryDealer>& AudioFlinger::Client::heap() const
3958{
3959    return mMemoryDealer;
3960}
3961
3962// ----------------------------------------------------------------------------
3963
3964AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
3965                                                     const sp<IAudioFlingerClient>& client,
3966                                                     pid_t pid)
3967    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
3968{
3969}
3970
3971AudioFlinger::NotificationClient::~NotificationClient()
3972{
3973    mClient.clear();
3974}
3975
3976void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
3977{
3978    sp<NotificationClient> keep(this);
3979    {
3980        mAudioFlinger->removeNotificationClient(mPid);
3981    }
3982}
3983
3984// ----------------------------------------------------------------------------
3985
3986AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
3987    : BnAudioTrack(),
3988      mTrack(track)
3989{
3990}
3991
3992AudioFlinger::TrackHandle::~TrackHandle() {
3993    // just stop the track on deletion, associated resources
3994    // will be freed from the main thread once all pending buffers have
3995    // been played. Unless it's not in the active track list, in which
3996    // case we free everything now...
3997    mTrack->destroy();
3998}
3999
4000status_t AudioFlinger::TrackHandle::start() {
4001    return mTrack->start();
4002}
4003
4004void AudioFlinger::TrackHandle::stop() {
4005    mTrack->stop();
4006}
4007
4008void AudioFlinger::TrackHandle::flush() {
4009    mTrack->flush();
4010}
4011
4012void AudioFlinger::TrackHandle::mute(bool e) {
4013    mTrack->mute(e);
4014}
4015
4016void AudioFlinger::TrackHandle::pause() {
4017    mTrack->pause();
4018}
4019
4020void AudioFlinger::TrackHandle::setVolume(float left, float right) {
4021    mTrack->setVolume(left, right);
4022}
4023
4024sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
4025    return mTrack->getCblk();
4026}
4027
4028status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
4029{
4030    return mTrack->attachAuxEffect(EffectId);
4031}
4032
4033status_t AudioFlinger::TrackHandle::onTransact(
4034    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4035{
4036    return BnAudioTrack::onTransact(code, data, reply, flags);
4037}
4038
4039// ----------------------------------------------------------------------------
4040
4041sp<IAudioRecord> AudioFlinger::openRecord(
4042        pid_t pid,
4043        int input,
4044        uint32_t sampleRate,
4045        uint32_t format,
4046        uint32_t channelMask,
4047        int frameCount,
4048        uint32_t flags,
4049        int *sessionId,
4050        status_t *status)
4051{
4052    sp<RecordThread::RecordTrack> recordTrack;
4053    sp<RecordHandle> recordHandle;
4054    sp<Client> client;
4055    wp<Client> wclient;
4056    status_t lStatus;
4057    RecordThread *thread;
4058    size_t inFrameCount;
4059    int lSessionId;
4060
4061    // check calling permissions
4062    if (!recordingAllowed()) {
4063        lStatus = PERMISSION_DENIED;
4064        goto Exit;
4065    }
4066
4067    // add client to list
4068    { // scope for mLock
4069        Mutex::Autolock _l(mLock);
4070        thread = checkRecordThread_l(input);
4071        if (thread == NULL) {
4072            lStatus = BAD_VALUE;
4073            goto Exit;
4074        }
4075
4076        wclient = mClients.valueFor(pid);
4077        if (wclient != NULL) {
4078            client = wclient.promote();
4079        } else {
4080            client = new Client(this, pid);
4081            mClients.add(pid, client);
4082        }
4083
4084        // If no audio session id is provided, create one here
4085        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
4086            lSessionId = *sessionId;
4087        } else {
4088            lSessionId = nextUniqueId();
4089            if (sessionId != NULL) {
4090                *sessionId = lSessionId;
4091            }
4092        }
4093        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
4094        recordTrack = thread->createRecordTrack_l(client,
4095                                                sampleRate,
4096                                                format,
4097                                                channelMask,
4098                                                frameCount,
4099                                                flags,
4100                                                lSessionId,
4101                                                &lStatus);
4102    }
4103    if (lStatus != NO_ERROR) {
4104        // remove local strong reference to Client before deleting the RecordTrack so that the Client
4105        // destructor is called by the TrackBase destructor with mLock held
4106        client.clear();
4107        recordTrack.clear();
4108        goto Exit;
4109    }
4110
4111    // return to handle to client
4112    recordHandle = new RecordHandle(recordTrack);
4113    lStatus = NO_ERROR;
4114
4115Exit:
4116    if (status) {
4117        *status = lStatus;
4118    }
4119    return recordHandle;
4120}
4121
4122// ----------------------------------------------------------------------------
4123
4124AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
4125    : BnAudioRecord(),
4126    mRecordTrack(recordTrack)
4127{
4128}
4129
4130AudioFlinger::RecordHandle::~RecordHandle() {
4131    stop();
4132}
4133
4134status_t AudioFlinger::RecordHandle::start() {
4135    LOGV("RecordHandle::start()");
4136    return mRecordTrack->start();
4137}
4138
4139void AudioFlinger::RecordHandle::stop() {
4140    LOGV("RecordHandle::stop()");
4141    mRecordTrack->stop();
4142}
4143
4144sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
4145    return mRecordTrack->getCblk();
4146}
4147
4148status_t AudioFlinger::RecordHandle::onTransact(
4149    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
4150{
4151    return BnAudioRecord::onTransact(code, data, reply, flags);
4152}
4153
4154// ----------------------------------------------------------------------------
4155
4156AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
4157                                         AudioStreamIn *input,
4158                                         uint32_t sampleRate,
4159                                         uint32_t channels,
4160                                         int id,
4161                                         uint32_t device) :
4162    ThreadBase(audioFlinger, id, device),
4163    mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
4164{
4165    mType = ThreadBase::RECORD;
4166
4167    snprintf(mName, kNameLength, "AudioIn_%d", id);
4168
4169    mReqChannelCount = popcount(channels);
4170    mReqSampleRate = sampleRate;
4171    readInputParameters();
4172}
4173
4174
4175AudioFlinger::RecordThread::~RecordThread()
4176{
4177    delete[] mRsmpInBuffer;
4178    if (mResampler != 0) {
4179        delete mResampler;
4180        delete[] mRsmpOutBuffer;
4181    }
4182}
4183
4184void AudioFlinger::RecordThread::onFirstRef()
4185{
4186    run(mName, PRIORITY_URGENT_AUDIO);
4187}
4188
4189status_t AudioFlinger::RecordThread::readyToRun()
4190{
4191    status_t status = initCheck();
4192    LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
4193    return status;
4194}
4195
4196bool AudioFlinger::RecordThread::threadLoop()
4197{
4198    AudioBufferProvider::Buffer buffer;
4199    sp<RecordTrack> activeTrack;
4200    Vector< sp<EffectChain> > effectChains;
4201
4202    nsecs_t lastWarning = 0;
4203
4204    acquireWakeLock();
4205
4206    // start recording
4207    while (!exitPending()) {
4208
4209        processConfigEvents();
4210
4211        { // scope for mLock
4212            Mutex::Autolock _l(mLock);
4213            checkForNewParameters_l();
4214            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
4215                if (!mStandby) {
4216                    mInput->stream->common.standby(&mInput->stream->common);
4217                    mStandby = true;
4218                }
4219
4220                if (exitPending()) break;
4221
4222                releaseWakeLock_l();
4223                LOGV("RecordThread: loop stopping");
4224                // go to sleep
4225                mWaitWorkCV.wait(mLock);
4226                LOGV("RecordThread: loop starting");
4227                acquireWakeLock_l();
4228                continue;
4229            }
4230            if (mActiveTrack != 0) {
4231                if (mActiveTrack->mState == TrackBase::PAUSING) {
4232                    if (!mStandby) {
4233                        mInput->stream->common.standby(&mInput->stream->common);
4234                        mStandby = true;
4235                    }
4236                    mActiveTrack.clear();
4237                    mStartStopCond.broadcast();
4238                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
4239                    if (mReqChannelCount != mActiveTrack->channelCount()) {
4240                        mActiveTrack.clear();
4241                        mStartStopCond.broadcast();
4242                    } else if (mBytesRead != 0) {
4243                        // record start succeeds only if first read from audio input
4244                        // succeeds
4245                        if (mBytesRead > 0) {
4246                            mActiveTrack->mState = TrackBase::ACTIVE;
4247                        } else {
4248                            mActiveTrack.clear();
4249                        }
4250                        mStartStopCond.broadcast();
4251                    }
4252                    mStandby = false;
4253                }
4254            }
4255            lockEffectChains_l(effectChains);
4256        }
4257
4258        if (mActiveTrack != 0) {
4259            if (mActiveTrack->mState != TrackBase::ACTIVE &&
4260                mActiveTrack->mState != TrackBase::RESUMING) {
4261                unlockEffectChains(effectChains);
4262                usleep(kRecordThreadSleepUs);
4263                continue;
4264            }
4265            for (size_t i = 0; i < effectChains.size(); i ++) {
4266                effectChains[i]->process_l();
4267            }
4268
4269            buffer.frameCount = mFrameCount;
4270            if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
4271                size_t framesOut = buffer.frameCount;
4272                if (mResampler == 0) {
4273                    // no resampling
4274                    while (framesOut) {
4275                        size_t framesIn = mFrameCount - mRsmpInIndex;
4276                        if (framesIn) {
4277                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
4278                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
4279                            if (framesIn > framesOut)
4280                                framesIn = framesOut;
4281                            mRsmpInIndex += framesIn;
4282                            framesOut -= framesIn;
4283                            if ((int)mChannelCount == mReqChannelCount ||
4284                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
4285                                memcpy(dst, src, framesIn * mFrameSize);
4286                            } else {
4287                                int16_t *src16 = (int16_t *)src;
4288                                int16_t *dst16 = (int16_t *)dst;
4289                                if (mChannelCount == 1) {
4290                                    while (framesIn--) {
4291                                        *dst16++ = *src16;
4292                                        *dst16++ = *src16++;
4293                                    }
4294                                } else {
4295                                    while (framesIn--) {
4296                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
4297                                        src16 += 2;
4298                                    }
4299                                }
4300                            }
4301                        }
4302                        if (framesOut && mFrameCount == mRsmpInIndex) {
4303                            if (framesOut == mFrameCount &&
4304                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
4305                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
4306                                framesOut = 0;
4307                            } else {
4308                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4309                                mRsmpInIndex = 0;
4310                            }
4311                            if (mBytesRead < 0) {
4312                                LOGE("Error reading audio input");
4313                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
4314                                    // Force input into standby so that it tries to
4315                                    // recover at next read attempt
4316                                    mInput->stream->common.standby(&mInput->stream->common);
4317                                    usleep(kRecordThreadSleepUs);
4318                                }
4319                                mRsmpInIndex = mFrameCount;
4320                                framesOut = 0;
4321                                buffer.frameCount = 0;
4322                            }
4323                        }
4324                    }
4325                } else {
4326                    // resampling
4327
4328                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
4329                    // alter output frame count as if we were expecting stereo samples
4330                    if (mChannelCount == 1 && mReqChannelCount == 1) {
4331                        framesOut >>= 1;
4332                    }
4333                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
4334                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
4335                    // are 32 bit aligned which should be always true.
4336                    if (mChannelCount == 2 && mReqChannelCount == 1) {
4337                        AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
4338                        // the resampler always outputs stereo samples: do post stereo to mono conversion
4339                        int16_t *src = (int16_t *)mRsmpOutBuffer;
4340                        int16_t *dst = buffer.i16;
4341                        while (framesOut--) {
4342                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
4343                            src += 2;
4344                        }
4345                    } else {
4346                        AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
4347                    }
4348
4349                }
4350                mActiveTrack->releaseBuffer(&buffer);
4351                mActiveTrack->overflow();
4352            }
4353            // client isn't retrieving buffers fast enough
4354            else {
4355                if (!mActiveTrack->setOverflow()) {
4356                    nsecs_t now = systemTime();
4357                    if ((now - lastWarning) > kWarningThrottle) {
4358                        LOGW("RecordThread: buffer overflow");
4359                        lastWarning = now;
4360                    }
4361                }
4362                // Release the processor for a while before asking for a new buffer.
4363                // This will give the application more chance to read from the buffer and
4364                // clear the overflow.
4365                usleep(kRecordThreadSleepUs);
4366            }
4367        }
4368        // enable changes in effect chain
4369        unlockEffectChains(effectChains);
4370        effectChains.clear();
4371    }
4372
4373    if (!mStandby) {
4374        mInput->stream->common.standby(&mInput->stream->common);
4375    }
4376    mActiveTrack.clear();
4377
4378    mStartStopCond.broadcast();
4379
4380    releaseWakeLock();
4381
4382    LOGV("RecordThread %p exiting", this);
4383    return false;
4384}
4385
4386
4387sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4388        const sp<AudioFlinger::Client>& client,
4389        uint32_t sampleRate,
4390        int format,
4391        int channelMask,
4392        int frameCount,
4393        uint32_t flags,
4394        int sessionId,
4395        status_t *status)
4396{
4397    sp<RecordTrack> track;
4398    status_t lStatus;
4399
4400    lStatus = initCheck();
4401    if (lStatus != NO_ERROR) {
4402        LOGE("Audio driver not initialized.");
4403        goto Exit;
4404    }
4405
4406    { // scope for mLock
4407        Mutex::Autolock _l(mLock);
4408
4409        track = new RecordTrack(this, client, sampleRate,
4410                      format, channelMask, frameCount, flags, sessionId);
4411
4412        if (track->getCblk() == NULL) {
4413            lStatus = NO_MEMORY;
4414            goto Exit;
4415        }
4416
4417        mTrack = track.get();
4418        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4419        bool suspend = audio_is_bluetooth_sco_device(
4420                (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
4421        setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
4422        setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
4423    }
4424    lStatus = NO_ERROR;
4425
4426Exit:
4427    if (status) {
4428        *status = lStatus;
4429    }
4430    return track;
4431}
4432
4433status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4434{
4435    LOGV("RecordThread::start");
4436    sp <ThreadBase> strongMe = this;
4437    status_t status = NO_ERROR;
4438    {
4439        AutoMutex lock(&mLock);
4440        if (mActiveTrack != 0) {
4441            if (recordTrack != mActiveTrack.get()) {
4442                status = -EBUSY;
4443            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4444                mActiveTrack->mState = TrackBase::ACTIVE;
4445            }
4446            return status;
4447        }
4448
4449        recordTrack->mState = TrackBase::IDLE;
4450        mActiveTrack = recordTrack;
4451        mLock.unlock();
4452        status_t status = AudioSystem::startInput(mId);
4453        mLock.lock();
4454        if (status != NO_ERROR) {
4455            mActiveTrack.clear();
4456            return status;
4457        }
4458        mRsmpInIndex = mFrameCount;
4459        mBytesRead = 0;
4460        if (mResampler != NULL) {
4461            mResampler->reset();
4462        }
4463        mActiveTrack->mState = TrackBase::RESUMING;
4464        // signal thread to start
4465        LOGV("Signal record thread");
4466        mWaitWorkCV.signal();
4467        // do not wait for mStartStopCond if exiting
4468        if (mExiting) {
4469            mActiveTrack.clear();
4470            status = INVALID_OPERATION;
4471            goto startError;
4472        }
4473        mStartStopCond.wait(mLock);
4474        if (mActiveTrack == 0) {
4475            LOGV("Record failed to start");
4476            status = BAD_VALUE;
4477            goto startError;
4478        }
4479        LOGV("Record started OK");
4480        return status;
4481    }
4482startError:
4483    AudioSystem::stopInput(mId);
4484    return status;
4485}
4486
4487void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4488    LOGV("RecordThread::stop");
4489    sp <ThreadBase> strongMe = this;
4490    {
4491        AutoMutex lock(&mLock);
4492        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4493            mActiveTrack->mState = TrackBase::PAUSING;
4494            // do not wait for mStartStopCond if exiting
4495            if (mExiting) {
4496                return;
4497            }
4498            mStartStopCond.wait(mLock);
4499            // if we have been restarted, recordTrack == mActiveTrack.get() here
4500            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4501                mLock.unlock();
4502                AudioSystem::stopInput(mId);
4503                mLock.lock();
4504                LOGV("Record stopped OK");
4505            }
4506        }
4507    }
4508}
4509
4510status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4511{
4512    const size_t SIZE = 256;
4513    char buffer[SIZE];
4514    String8 result;
4515    pid_t pid = 0;
4516
4517    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4518    result.append(buffer);
4519
4520    if (mActiveTrack != 0) {
4521        result.append("Active Track:\n");
4522        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4523        mActiveTrack->dump(buffer, SIZE);
4524        result.append(buffer);
4525
4526        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4527        result.append(buffer);
4528        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4529        result.append(buffer);
4530        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
4531        result.append(buffer);
4532        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4533        result.append(buffer);
4534        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4535        result.append(buffer);
4536
4537
4538    } else {
4539        result.append("No record client\n");
4540    }
4541    write(fd, result.string(), result.size());
4542
4543    dumpBase(fd, args);
4544    dumpEffectChains(fd, args);
4545
4546    return NO_ERROR;
4547}
4548
4549status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4550{
4551    size_t framesReq = buffer->frameCount;
4552    size_t framesReady = mFrameCount - mRsmpInIndex;
4553    int channelCount;
4554
4555    if (framesReady == 0) {
4556        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4557        if (mBytesRead < 0) {
4558            LOGE("RecordThread::getNextBuffer() Error reading audio input");
4559            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4560                // Force input into standby so that it tries to
4561                // recover at next read attempt
4562                mInput->stream->common.standby(&mInput->stream->common);
4563                usleep(kRecordThreadSleepUs);
4564            }
4565            buffer->raw = 0;
4566            buffer->frameCount = 0;
4567            return NOT_ENOUGH_DATA;
4568        }
4569        mRsmpInIndex = 0;
4570        framesReady = mFrameCount;
4571    }
4572
4573    if (framesReq > framesReady) {
4574        framesReq = framesReady;
4575    }
4576
4577    if (mChannelCount == 1 && mReqChannelCount == 2) {
4578        channelCount = 1;
4579    } else {
4580        channelCount = 2;
4581    }
4582    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4583    buffer->frameCount = framesReq;
4584    return NO_ERROR;
4585}
4586
4587void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4588{
4589    mRsmpInIndex += buffer->frameCount;
4590    buffer->frameCount = 0;
4591}
4592
4593bool AudioFlinger::RecordThread::checkForNewParameters_l()
4594{
4595    bool reconfig = false;
4596
4597    while (!mNewParameters.isEmpty()) {
4598        status_t status = NO_ERROR;
4599        String8 keyValuePair = mNewParameters[0];
4600        AudioParameter param = AudioParameter(keyValuePair);
4601        int value;
4602        int reqFormat = mFormat;
4603        int reqSamplingRate = mReqSampleRate;
4604        int reqChannelCount = mReqChannelCount;
4605
4606        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4607            reqSamplingRate = value;
4608            reconfig = true;
4609        }
4610        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4611            reqFormat = value;
4612            reconfig = true;
4613        }
4614        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4615            reqChannelCount = popcount(value);
4616            reconfig = true;
4617        }
4618        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4619            // do not accept frame count changes if tracks are open as the track buffer
4620            // size depends on frame count and correct behavior would not be garantied
4621            // if frame count is changed after track creation
4622            if (mActiveTrack != 0) {
4623                status = INVALID_OPERATION;
4624            } else {
4625                reconfig = true;
4626            }
4627        }
4628        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4629            // forward device change to effects that have requested to be
4630            // aware of attached audio device.
4631            for (size_t i = 0; i < mEffectChains.size(); i++) {
4632                mEffectChains[i]->setDevice_l(value);
4633            }
4634            // store input device and output device but do not forward output device to audio HAL.
4635            // Note that status is ignored by the caller for output device
4636            // (see AudioFlinger::setParameters()
4637            if (value & AUDIO_DEVICE_OUT_ALL) {
4638                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4639                status = BAD_VALUE;
4640            } else {
4641                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4642                // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
4643                if (mTrack != NULL) {
4644                    bool suspend = audio_is_bluetooth_sco_device(
4645                            (audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
4646                    setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
4647                    setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
4648                }
4649            }
4650            mDevice |= (uint32_t)value;
4651        }
4652        if (status == NO_ERROR) {
4653            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4654            if (status == INVALID_OPERATION) {
4655               mInput->stream->common.standby(&mInput->stream->common);
4656               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4657            }
4658            if (reconfig) {
4659                if (status == BAD_VALUE &&
4660                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4661                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4662                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4663                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4664                    (reqChannelCount < 3)) {
4665                    status = NO_ERROR;
4666                }
4667                if (status == NO_ERROR) {
4668                    readInputParameters();
4669                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4670                }
4671            }
4672        }
4673
4674        mNewParameters.removeAt(0);
4675
4676        mParamStatus = status;
4677        mParamCond.signal();
4678        // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4679        // already timed out waiting for the status and will never signal the condition.
4680        mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
4681    }
4682    return reconfig;
4683}
4684
4685String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4686{
4687    char *s;
4688    String8 out_s8 = String8();
4689
4690    Mutex::Autolock _l(mLock);
4691    if (initCheck() != NO_ERROR) {
4692        return out_s8;
4693    }
4694
4695    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4696    out_s8 = String8(s);
4697    free(s);
4698    return out_s8;
4699}
4700
4701void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4702    AudioSystem::OutputDescriptor desc;
4703    void *param2 = 0;
4704
4705    switch (event) {
4706    case AudioSystem::INPUT_OPENED:
4707    case AudioSystem::INPUT_CONFIG_CHANGED:
4708        desc.channels = mChannelMask;
4709        desc.samplingRate = mSampleRate;
4710        desc.format = mFormat;
4711        desc.frameCount = mFrameCount;
4712        desc.latency = 0;
4713        param2 = &desc;
4714        break;
4715
4716    case AudioSystem::INPUT_CLOSED:
4717    default:
4718        break;
4719    }
4720    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4721}
4722
4723void AudioFlinger::RecordThread::readInputParameters()
4724{
4725    if (mRsmpInBuffer) delete mRsmpInBuffer;
4726    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4727    if (mResampler) delete mResampler;
4728    mResampler = 0;
4729
4730    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4731    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4732    mChannelCount = (uint16_t)popcount(mChannelMask);
4733    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4734    mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
4735    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4736    mFrameCount = mInputBytes / mFrameSize;
4737    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4738
4739    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4740    {
4741        int channelCount;
4742         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4743         // stereo to mono post process as the resampler always outputs stereo.
4744        if (mChannelCount == 1 && mReqChannelCount == 2) {
4745            channelCount = 1;
4746        } else {
4747            channelCount = 2;
4748        }
4749        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4750        mResampler->setSampleRate(mSampleRate);
4751        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4752        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4753
4754        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4755        if (mChannelCount == 1 && mReqChannelCount == 1) {
4756            mFrameCount >>= 1;
4757        }
4758
4759    }
4760    mRsmpInIndex = mFrameCount;
4761}
4762
4763unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4764{
4765    Mutex::Autolock _l(mLock);
4766    if (initCheck() != NO_ERROR) {
4767        return 0;
4768    }
4769
4770    return mInput->stream->get_input_frames_lost(mInput->stream);
4771}
4772
4773uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4774{
4775    Mutex::Autolock _l(mLock);
4776    uint32_t result = 0;
4777    if (getEffectChain_l(sessionId) != 0) {
4778        result = EFFECT_SESSION;
4779    }
4780
4781    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4782        result |= TRACK_SESSION;
4783    }
4784
4785    return result;
4786}
4787
4788AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
4789{
4790    Mutex::Autolock _l(mLock);
4791    return mTrack;
4792}
4793
4794AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
4795{
4796    Mutex::Autolock _l(mLock);
4797    return mInput;
4798}
4799
4800AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4801{
4802    Mutex::Autolock _l(mLock);
4803    AudioStreamIn *input = mInput;
4804    mInput = NULL;
4805    return input;
4806}
4807
4808// this method must always be called either with ThreadBase mLock held or inside the thread loop
4809audio_stream_t* AudioFlinger::RecordThread::stream()
4810{
4811    if (mInput == NULL) {
4812        return NULL;
4813    }
4814    return &mInput->stream->common;
4815}
4816
4817
4818// ----------------------------------------------------------------------------
4819
4820int AudioFlinger::openOutput(uint32_t *pDevices,
4821                                uint32_t *pSamplingRate,
4822                                uint32_t *pFormat,
4823                                uint32_t *pChannels,
4824                                uint32_t *pLatencyMs,
4825                                uint32_t flags)
4826{
4827    status_t status;
4828    PlaybackThread *thread = NULL;
4829    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4830    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4831    uint32_t format = pFormat ? *pFormat : 0;
4832    uint32_t channels = pChannels ? *pChannels : 0;
4833    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4834    audio_stream_out_t *outStream;
4835    audio_hw_device_t *outHwDev;
4836
4837    LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4838            pDevices ? *pDevices : 0,
4839            samplingRate,
4840            format,
4841            channels,
4842            flags);
4843
4844    if (pDevices == NULL || *pDevices == 0) {
4845        return 0;
4846    }
4847
4848    Mutex::Autolock _l(mLock);
4849
4850    outHwDev = findSuitableHwDev_l(*pDevices);
4851    if (outHwDev == NULL)
4852        return 0;
4853
4854    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
4855                                          &channels, &samplingRate, &outStream);
4856    LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4857            outStream,
4858            samplingRate,
4859            format,
4860            channels,
4861            status);
4862
4863    mHardwareStatus = AUDIO_HW_IDLE;
4864    if (outStream != NULL) {
4865        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4866        int id = nextUniqueId();
4867
4868        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4869            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4870            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4871            thread = new DirectOutputThread(this, output, id, *pDevices);
4872            LOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4873        } else {
4874            thread = new MixerThread(this, output, id, *pDevices);
4875            LOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4876        }
4877        mPlaybackThreads.add(id, thread);
4878
4879        if (pSamplingRate) *pSamplingRate = samplingRate;
4880        if (pFormat) *pFormat = format;
4881        if (pChannels) *pChannels = channels;
4882        if (pLatencyMs) *pLatencyMs = thread->latency();
4883
4884        // notify client processes of the new output creation
4885        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4886        return id;
4887    }
4888
4889    return 0;
4890}
4891
4892int AudioFlinger::openDuplicateOutput(int output1, int output2)
4893{
4894    Mutex::Autolock _l(mLock);
4895    MixerThread *thread1 = checkMixerThread_l(output1);
4896    MixerThread *thread2 = checkMixerThread_l(output2);
4897
4898    if (thread1 == NULL || thread2 == NULL) {
4899        LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
4900        return 0;
4901    }
4902
4903    int id = nextUniqueId();
4904    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
4905    thread->addOutputTrack(thread2);
4906    mPlaybackThreads.add(id, thread);
4907    // notify client processes of the new output creation
4908    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4909    return id;
4910}
4911
4912status_t AudioFlinger::closeOutput(int output)
4913{
4914    // keep strong reference on the playback thread so that
4915    // it is not destroyed while exit() is executed
4916    sp <PlaybackThread> thread;
4917    {
4918        Mutex::Autolock _l(mLock);
4919        thread = checkPlaybackThread_l(output);
4920        if (thread == NULL) {
4921            return BAD_VALUE;
4922        }
4923
4924        LOGV("closeOutput() %d", output);
4925
4926        if (thread->type() == ThreadBase::MIXER) {
4927            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4928                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
4929                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
4930                    dupThread->removeOutputTrack((MixerThread *)thread.get());
4931                }
4932            }
4933        }
4934        void *param2 = 0;
4935        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
4936        mPlaybackThreads.removeItem(output);
4937    }
4938    thread->exit();
4939
4940    if (thread->type() != ThreadBase::DUPLICATING) {
4941        AudioStreamOut *out = thread->clearOutput();
4942        // from now on thread->mOutput is NULL
4943        out->hwDev->close_output_stream(out->hwDev, out->stream);
4944        delete out;
4945    }
4946    return NO_ERROR;
4947}
4948
4949status_t AudioFlinger::suspendOutput(int output)
4950{
4951    Mutex::Autolock _l(mLock);
4952    PlaybackThread *thread = checkPlaybackThread_l(output);
4953
4954    if (thread == NULL) {
4955        return BAD_VALUE;
4956    }
4957
4958    LOGV("suspendOutput() %d", output);
4959    thread->suspend();
4960
4961    return NO_ERROR;
4962}
4963
4964status_t AudioFlinger::restoreOutput(int output)
4965{
4966    Mutex::Autolock _l(mLock);
4967    PlaybackThread *thread = checkPlaybackThread_l(output);
4968
4969    if (thread == NULL) {
4970        return BAD_VALUE;
4971    }
4972
4973    LOGV("restoreOutput() %d", output);
4974
4975    thread->restore();
4976
4977    return NO_ERROR;
4978}
4979
4980int AudioFlinger::openInput(uint32_t *pDevices,
4981                                uint32_t *pSamplingRate,
4982                                uint32_t *pFormat,
4983                                uint32_t *pChannels,
4984                                uint32_t acoustics)
4985{
4986    status_t status;
4987    RecordThread *thread = NULL;
4988    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4989    uint32_t format = pFormat ? *pFormat : 0;
4990    uint32_t channels = pChannels ? *pChannels : 0;
4991    uint32_t reqSamplingRate = samplingRate;
4992    uint32_t reqFormat = format;
4993    uint32_t reqChannels = channels;
4994    audio_stream_in_t *inStream;
4995    audio_hw_device_t *inHwDev;
4996
4997    if (pDevices == NULL || *pDevices == 0) {
4998        return 0;
4999    }
5000
5001    Mutex::Autolock _l(mLock);
5002
5003    inHwDev = findSuitableHwDev_l(*pDevices);
5004    if (inHwDev == NULL)
5005        return 0;
5006
5007    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5008                                        &channels, &samplingRate,
5009                                        (audio_in_acoustics_t)acoustics,
5010                                        &inStream);
5011    LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
5012            inStream,
5013            samplingRate,
5014            format,
5015            channels,
5016            acoustics,
5017            status);
5018
5019    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
5020    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
5021    // or stereo to mono conversions on 16 bit PCM inputs.
5022    if (inStream == NULL && status == BAD_VALUE &&
5023        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
5024        (samplingRate <= 2 * reqSamplingRate) &&
5025        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
5026        LOGV("openInput() reopening with proposed sampling rate and channels");
5027        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
5028                                            &channels, &samplingRate,
5029                                            (audio_in_acoustics_t)acoustics,
5030                                            &inStream);
5031    }
5032
5033    if (inStream != NULL) {
5034        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
5035
5036        int id = nextUniqueId();
5037        // Start record thread
5038        // RecorThread require both input and output device indication to forward to audio
5039        // pre processing modules
5040        uint32_t device = (*pDevices) | primaryOutputDevice_l();
5041        thread = new RecordThread(this,
5042                                  input,
5043                                  reqSamplingRate,
5044                                  reqChannels,
5045                                  id,
5046                                  device);
5047        mRecordThreads.add(id, thread);
5048        LOGV("openInput() created record thread: ID %d thread %p", id, thread);
5049        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
5050        if (pFormat) *pFormat = format;
5051        if (pChannels) *pChannels = reqChannels;
5052
5053        input->stream->common.standby(&input->stream->common);
5054
5055        // notify client processes of the new input creation
5056        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
5057        return id;
5058    }
5059
5060    return 0;
5061}
5062
5063status_t AudioFlinger::closeInput(int input)
5064{
5065    // keep strong reference on the record thread so that
5066    // it is not destroyed while exit() is executed
5067    sp <RecordThread> thread;
5068    {
5069        Mutex::Autolock _l(mLock);
5070        thread = checkRecordThread_l(input);
5071        if (thread == NULL) {
5072            return BAD_VALUE;
5073        }
5074
5075        LOGV("closeInput() %d", input);
5076        void *param2 = 0;
5077        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
5078        mRecordThreads.removeItem(input);
5079    }
5080    thread->exit();
5081
5082    AudioStreamIn *in = thread->clearInput();
5083    // from now on thread->mInput is NULL
5084    in->hwDev->close_input_stream(in->hwDev, in->stream);
5085    delete in;
5086
5087    return NO_ERROR;
5088}
5089
5090status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
5091{
5092    Mutex::Autolock _l(mLock);
5093    MixerThread *dstThread = checkMixerThread_l(output);
5094    if (dstThread == NULL) {
5095        LOGW("setStreamOutput() bad output id %d", output);
5096        return BAD_VALUE;
5097    }
5098
5099    LOGV("setStreamOutput() stream %d to output %d", stream, output);
5100    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
5101
5102    dstThread->setStreamValid(stream, true);
5103
5104    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5105        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5106        if (thread != dstThread &&
5107            thread->type() != ThreadBase::DIRECT) {
5108            MixerThread *srcThread = (MixerThread *)thread;
5109            srcThread->setStreamValid(stream, false);
5110            srcThread->invalidateTracks(stream);
5111        }
5112    }
5113
5114    return NO_ERROR;
5115}
5116
5117
5118int AudioFlinger::newAudioSessionId()
5119{
5120    return nextUniqueId();
5121}
5122
5123void AudioFlinger::acquireAudioSessionId(int audioSession)
5124{
5125    Mutex::Autolock _l(mLock);
5126    int caller = IPCThreadState::self()->getCallingPid();
5127    LOGV("acquiring %d from %d", audioSession, caller);
5128    int num = mAudioSessionRefs.size();
5129    for (int i = 0; i< num; i++) {
5130        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
5131        if (ref->sessionid == audioSession && ref->pid == caller) {
5132            ref->cnt++;
5133            LOGV(" incremented refcount to %d", ref->cnt);
5134            return;
5135        }
5136    }
5137    AudioSessionRef *ref = new AudioSessionRef();
5138    ref->sessionid = audioSession;
5139    ref->pid = caller;
5140    ref->cnt = 1;
5141    mAudioSessionRefs.push(ref);
5142    LOGV(" added new entry for %d", ref->sessionid);
5143}
5144
5145void AudioFlinger::releaseAudioSessionId(int audioSession)
5146{
5147    Mutex::Autolock _l(mLock);
5148    int caller = IPCThreadState::self()->getCallingPid();
5149    LOGV("releasing %d from %d", audioSession, caller);
5150    int num = mAudioSessionRefs.size();
5151    for (int i = 0; i< num; i++) {
5152        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
5153        if (ref->sessionid == audioSession && ref->pid == caller) {
5154            ref->cnt--;
5155            LOGV(" decremented refcount to %d", ref->cnt);
5156            if (ref->cnt == 0) {
5157                mAudioSessionRefs.removeAt(i);
5158                delete ref;
5159                purgeStaleEffects_l();
5160            }
5161            return;
5162        }
5163    }
5164    LOGW("session id %d not found for pid %d", audioSession, caller);
5165}
5166
5167void AudioFlinger::purgeStaleEffects_l() {
5168
5169    LOGV("purging stale effects");
5170
5171    Vector< sp<EffectChain> > chains;
5172
5173    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5174        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
5175        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5176            sp<EffectChain> ec = t->mEffectChains[j];
5177            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
5178                chains.push(ec);
5179            }
5180        }
5181    }
5182    for (size_t i = 0; i < mRecordThreads.size(); i++) {
5183        sp<RecordThread> t = mRecordThreads.valueAt(i);
5184        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
5185            sp<EffectChain> ec = t->mEffectChains[j];
5186            chains.push(ec);
5187        }
5188    }
5189
5190    for (size_t i = 0; i < chains.size(); i++) {
5191        sp<EffectChain> ec = chains[i];
5192        int sessionid = ec->sessionId();
5193        sp<ThreadBase> t = ec->mThread.promote();
5194        if (t == 0) {
5195            continue;
5196        }
5197        size_t numsessionrefs = mAudioSessionRefs.size();
5198        bool found = false;
5199        for (size_t k = 0; k < numsessionrefs; k++) {
5200            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
5201            if (ref->sessionid == sessionid) {
5202                LOGV(" session %d still exists for %d with %d refs",
5203                     sessionid, ref->pid, ref->cnt);
5204                found = true;
5205                break;
5206            }
5207        }
5208        if (!found) {
5209            // remove all effects from the chain
5210            while (ec->mEffects.size()) {
5211                sp<EffectModule> effect = ec->mEffects[0];
5212                effect->unPin();
5213                Mutex::Autolock _l (t->mLock);
5214                t->removeEffect_l(effect);
5215                for (size_t j = 0; j < effect->mHandles.size(); j++) {
5216                    sp<EffectHandle> handle = effect->mHandles[j].promote();
5217                    if (handle != 0) {
5218                        handle->mEffect.clear();
5219                    }
5220                }
5221                AudioSystem::unregisterEffect(effect->id());
5222            }
5223        }
5224    }
5225    return;
5226}
5227
5228// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
5229AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
5230{
5231    PlaybackThread *thread = NULL;
5232    if (mPlaybackThreads.indexOfKey(output) >= 0) {
5233        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
5234    }
5235    return thread;
5236}
5237
5238// checkMixerThread_l() must be called with AudioFlinger::mLock held
5239AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
5240{
5241    PlaybackThread *thread = checkPlaybackThread_l(output);
5242    if (thread != NULL) {
5243        if (thread->type() == ThreadBase::DIRECT) {
5244            thread = NULL;
5245        }
5246    }
5247    return (MixerThread *)thread;
5248}
5249
5250// checkRecordThread_l() must be called with AudioFlinger::mLock held
5251AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
5252{
5253    RecordThread *thread = NULL;
5254    if (mRecordThreads.indexOfKey(input) >= 0) {
5255        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
5256    }
5257    return thread;
5258}
5259
5260uint32_t AudioFlinger::nextUniqueId()
5261{
5262    return android_atomic_inc(&mNextUniqueId);
5263}
5264
5265AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
5266{
5267    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5268        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
5269        AudioStreamOut *output = thread->getOutput();
5270        if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
5271            return thread;
5272        }
5273    }
5274    return NULL;
5275}
5276
5277uint32_t AudioFlinger::primaryOutputDevice_l()
5278{
5279    PlaybackThread *thread = primaryPlaybackThread_l();
5280
5281    if (thread == NULL) {
5282        return 0;
5283    }
5284
5285    return thread->device();
5286}
5287
5288
5289// ----------------------------------------------------------------------------
5290//  Effect management
5291// ----------------------------------------------------------------------------
5292
5293
5294status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
5295{
5296    Mutex::Autolock _l(mLock);
5297    return EffectQueryNumberEffects(numEffects);
5298}
5299
5300status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
5301{
5302    Mutex::Autolock _l(mLock);
5303    return EffectQueryEffect(index, descriptor);
5304}
5305
5306status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
5307{
5308    Mutex::Autolock _l(mLock);
5309    return EffectGetDescriptor(pUuid, descriptor);
5310}
5311
5312
5313sp<IEffect> AudioFlinger::createEffect(pid_t pid,
5314        effect_descriptor_t *pDesc,
5315        const sp<IEffectClient>& effectClient,
5316        int32_t priority,
5317        int io,
5318        int sessionId,
5319        status_t *status,
5320        int *id,
5321        int *enabled)
5322{
5323    status_t lStatus = NO_ERROR;
5324    sp<EffectHandle> handle;
5325    effect_descriptor_t desc;
5326    sp<Client> client;
5327    wp<Client> wclient;
5328
5329    LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
5330            pid, effectClient.get(), priority, sessionId, io);
5331
5332    if (pDesc == NULL) {
5333        lStatus = BAD_VALUE;
5334        goto Exit;
5335    }
5336
5337    // check audio settings permission for global effects
5338    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
5339        lStatus = PERMISSION_DENIED;
5340        goto Exit;
5341    }
5342
5343    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
5344    // that can only be created by audio policy manager (running in same process)
5345    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
5346        lStatus = PERMISSION_DENIED;
5347        goto Exit;
5348    }
5349
5350    if (io == 0) {
5351        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
5352            // output must be specified by AudioPolicyManager when using session
5353            // AUDIO_SESSION_OUTPUT_STAGE
5354            lStatus = BAD_VALUE;
5355            goto Exit;
5356        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
5357            // if the output returned by getOutputForEffect() is removed before we lock the
5358            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
5359            // and we will exit safely
5360            io = AudioSystem::getOutputForEffect(&desc);
5361        }
5362    }
5363
5364    {
5365        Mutex::Autolock _l(mLock);
5366
5367
5368        if (!EffectIsNullUuid(&pDesc->uuid)) {
5369            // if uuid is specified, request effect descriptor
5370            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
5371            if (lStatus < 0) {
5372                LOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
5373                goto Exit;
5374            }
5375        } else {
5376            // if uuid is not specified, look for an available implementation
5377            // of the required type in effect factory
5378            if (EffectIsNullUuid(&pDesc->type)) {
5379                LOGW("createEffect() no effect type");
5380                lStatus = BAD_VALUE;
5381                goto Exit;
5382            }
5383            uint32_t numEffects = 0;
5384            effect_descriptor_t d;
5385            d.flags = 0; // prevent compiler warning
5386            bool found = false;
5387
5388            lStatus = EffectQueryNumberEffects(&numEffects);
5389            if (lStatus < 0) {
5390                LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
5391                goto Exit;
5392            }
5393            for (uint32_t i = 0; i < numEffects; i++) {
5394                lStatus = EffectQueryEffect(i, &desc);
5395                if (lStatus < 0) {
5396                    LOGW("createEffect() error %d from EffectQueryEffect", lStatus);
5397                    continue;
5398                }
5399                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
5400                    // If matching type found save effect descriptor. If the session is
5401                    // 0 and the effect is not auxiliary, continue enumeration in case
5402                    // an auxiliary version of this effect type is available
5403                    found = true;
5404                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
5405                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
5406                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5407                        break;
5408                    }
5409                }
5410            }
5411            if (!found) {
5412                lStatus = BAD_VALUE;
5413                LOGW("createEffect() effect not found");
5414                goto Exit;
5415            }
5416            // For same effect type, chose auxiliary version over insert version if
5417            // connect to output mix (Compliance to OpenSL ES)
5418            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
5419                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
5420                memcpy(&desc, &d, sizeof(effect_descriptor_t));
5421            }
5422        }
5423
5424        // Do not allow auxiliary effects on a session different from 0 (output mix)
5425        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
5426             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5427            lStatus = INVALID_OPERATION;
5428            goto Exit;
5429        }
5430
5431        // check recording permission for visualizer
5432        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
5433            !recordingAllowed()) {
5434            lStatus = PERMISSION_DENIED;
5435            goto Exit;
5436        }
5437
5438        // return effect descriptor
5439        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
5440
5441        // If output is not specified try to find a matching audio session ID in one of the
5442        // output threads.
5443        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
5444        // because of code checking output when entering the function.
5445        // Note: io is never 0 when creating an effect on an input
5446        if (io == 0) {
5447             // look for the thread where the specified audio session is present
5448            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
5449                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5450                    io = mPlaybackThreads.keyAt(i);
5451                    break;
5452                }
5453            }
5454            if (io == 0) {
5455               for (size_t i = 0; i < mRecordThreads.size(); i++) {
5456                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
5457                       io = mRecordThreads.keyAt(i);
5458                       break;
5459                   }
5460               }
5461            }
5462            // If no output thread contains the requested session ID, default to
5463            // first output. The effect chain will be moved to the correct output
5464            // thread when a track with the same session ID is created
5465            if (io == 0 && mPlaybackThreads.size()) {
5466                io = mPlaybackThreads.keyAt(0);
5467            }
5468            LOGV("createEffect() got io %d for effect %s", io, desc.name);
5469        }
5470        ThreadBase *thread = checkRecordThread_l(io);
5471        if (thread == NULL) {
5472            thread = checkPlaybackThread_l(io);
5473            if (thread == NULL) {
5474                LOGE("createEffect() unknown output thread");
5475                lStatus = BAD_VALUE;
5476                goto Exit;
5477            }
5478        }
5479
5480        wclient = mClients.valueFor(pid);
5481
5482        if (wclient != NULL) {
5483            client = wclient.promote();
5484        } else {
5485            client = new Client(this, pid);
5486            mClients.add(pid, client);
5487        }
5488
5489        // create effect on selected output thread
5490        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
5491                &desc, enabled, &lStatus);
5492        if (handle != 0 && id != NULL) {
5493            *id = handle->id();
5494        }
5495    }
5496
5497Exit:
5498    if(status) {
5499        *status = lStatus;
5500    }
5501    return handle;
5502}
5503
5504status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
5505{
5506    LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
5507            sessionId, srcOutput, dstOutput);
5508    Mutex::Autolock _l(mLock);
5509    if (srcOutput == dstOutput) {
5510        LOGW("moveEffects() same dst and src outputs %d", dstOutput);
5511        return NO_ERROR;
5512    }
5513    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
5514    if (srcThread == NULL) {
5515        LOGW("moveEffects() bad srcOutput %d", srcOutput);
5516        return BAD_VALUE;
5517    }
5518    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5519    if (dstThread == NULL) {
5520        LOGW("moveEffects() bad dstOutput %d", dstOutput);
5521        return BAD_VALUE;
5522    }
5523
5524    Mutex::Autolock _dl(dstThread->mLock);
5525    Mutex::Autolock _sl(srcThread->mLock);
5526    moveEffectChain_l(sessionId, srcThread, dstThread, false);
5527
5528    return NO_ERROR;
5529}
5530
5531// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
5532status_t AudioFlinger::moveEffectChain_l(int sessionId,
5533                                   AudioFlinger::PlaybackThread *srcThread,
5534                                   AudioFlinger::PlaybackThread *dstThread,
5535                                   bool reRegister)
5536{
5537    LOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5538            sessionId, srcThread, dstThread);
5539
5540    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
5541    if (chain == 0) {
5542        LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5543                sessionId, srcThread);
5544        return INVALID_OPERATION;
5545    }
5546
5547    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5548    // so that a new chain is created with correct parameters when first effect is added. This is
5549    // otherwise unecessary as removeEffect_l() will remove the chain when last effect is
5550    // removed.
5551    srcThread->removeEffectChain_l(chain);
5552
5553    // transfer all effects one by one so that new effect chain is created on new thread with
5554    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5555    int dstOutput = dstThread->id();
5556    sp<EffectChain> dstChain;
5557    uint32_t strategy = 0; // prevent compiler warning
5558    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5559    while (effect != 0) {
5560        srcThread->removeEffect_l(effect);
5561        dstThread->addEffect_l(effect);
5562        // if the move request is not received from audio policy manager, the effect must be
5563        // re-registered with the new strategy and output
5564        if (dstChain == 0) {
5565            dstChain = effect->chain().promote();
5566            if (dstChain == 0) {
5567                LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5568                srcThread->addEffect_l(effect);
5569                return NO_INIT;
5570            }
5571            strategy = dstChain->strategy();
5572        }
5573        if (reRegister) {
5574            AudioSystem::unregisterEffect(effect->id());
5575            AudioSystem::registerEffect(&effect->desc(),
5576                                        dstOutput,
5577                                        strategy,
5578                                        sessionId,
5579                                        effect->id());
5580        }
5581        effect = chain->getEffectFromId_l(0);
5582    }
5583
5584    return NO_ERROR;
5585}
5586
5587
5588// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5589sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5590        const sp<AudioFlinger::Client>& client,
5591        const sp<IEffectClient>& effectClient,
5592        int32_t priority,
5593        int sessionId,
5594        effect_descriptor_t *desc,
5595        int *enabled,
5596        status_t *status
5597        )
5598{
5599    sp<EffectModule> effect;
5600    sp<EffectHandle> handle;
5601    status_t lStatus;
5602    sp<EffectChain> chain;
5603    bool chainCreated = false;
5604    bool effectCreated = false;
5605    bool effectRegistered = false;
5606
5607    lStatus = initCheck();
5608    if (lStatus != NO_ERROR) {
5609        LOGW("createEffect_l() Audio driver not initialized.");
5610        goto Exit;
5611    }
5612
5613    // Do not allow effects with session ID 0 on direct output or duplicating threads
5614    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5615    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5616        LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5617                desc->name, sessionId);
5618        lStatus = BAD_VALUE;
5619        goto Exit;
5620    }
5621    // Only Pre processor effects are allowed on input threads and only on input threads
5622    if ((mType == RECORD &&
5623            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5624            (mType != RECORD &&
5625                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5626        LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5627                desc->name, desc->flags, mType);
5628        lStatus = BAD_VALUE;
5629        goto Exit;
5630    }
5631
5632    LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5633
5634    { // scope for mLock
5635        Mutex::Autolock _l(mLock);
5636
5637        // check for existing effect chain with the requested audio session
5638        chain = getEffectChain_l(sessionId);
5639        if (chain == 0) {
5640            // create a new chain for this session
5641            LOGV("createEffect_l() new effect chain for session %d", sessionId);
5642            chain = new EffectChain(this, sessionId);
5643            addEffectChain_l(chain);
5644            chain->setStrategy(getStrategyForSession_l(sessionId));
5645            chainCreated = true;
5646        } else {
5647            effect = chain->getEffectFromDesc_l(desc);
5648        }
5649
5650        LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
5651
5652        if (effect == 0) {
5653            int id = mAudioFlinger->nextUniqueId();
5654            // Check CPU and memory usage
5655            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5656            if (lStatus != NO_ERROR) {
5657                goto Exit;
5658            }
5659            effectRegistered = true;
5660            // create a new effect module if none present in the chain
5661            effect = new EffectModule(this, chain, desc, id, sessionId);
5662            lStatus = effect->status();
5663            if (lStatus != NO_ERROR) {
5664                goto Exit;
5665            }
5666            lStatus = chain->addEffect_l(effect);
5667            if (lStatus != NO_ERROR) {
5668                goto Exit;
5669            }
5670            effectCreated = true;
5671
5672            effect->setDevice(mDevice);
5673            effect->setMode(mAudioFlinger->getMode());
5674        }
5675        // create effect handle and connect it to effect module
5676        handle = new EffectHandle(effect, client, effectClient, priority);
5677        lStatus = effect->addHandle(handle);
5678        if (enabled) {
5679            *enabled = (int)effect->isEnabled();
5680        }
5681    }
5682
5683Exit:
5684    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5685        Mutex::Autolock _l(mLock);
5686        if (effectCreated) {
5687            chain->removeEffect_l(effect);
5688        }
5689        if (effectRegistered) {
5690            AudioSystem::unregisterEffect(effect->id());
5691        }
5692        if (chainCreated) {
5693            removeEffectChain_l(chain);
5694        }
5695        handle.clear();
5696    }
5697
5698    if(status) {
5699        *status = lStatus;
5700    }
5701    return handle;
5702}
5703
5704sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5705{
5706    sp<EffectModule> effect;
5707
5708    sp<EffectChain> chain = getEffectChain_l(sessionId);
5709    if (chain != 0) {
5710        effect = chain->getEffectFromId_l(effectId);
5711    }
5712    return effect;
5713}
5714
5715// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5716// PlaybackThread::mLock held
5717status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5718{
5719    // check for existing effect chain with the requested audio session
5720    int sessionId = effect->sessionId();
5721    sp<EffectChain> chain = getEffectChain_l(sessionId);
5722    bool chainCreated = false;
5723
5724    if (chain == 0) {
5725        // create a new chain for this session
5726        LOGV("addEffect_l() new effect chain for session %d", sessionId);
5727        chain = new EffectChain(this, sessionId);
5728        addEffectChain_l(chain);
5729        chain->setStrategy(getStrategyForSession_l(sessionId));
5730        chainCreated = true;
5731    }
5732    LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5733
5734    if (chain->getEffectFromId_l(effect->id()) != 0) {
5735        LOGW("addEffect_l() %p effect %s already present in chain %p",
5736                this, effect->desc().name, chain.get());
5737        return BAD_VALUE;
5738    }
5739
5740    status_t status = chain->addEffect_l(effect);
5741    if (status != NO_ERROR) {
5742        if (chainCreated) {
5743            removeEffectChain_l(chain);
5744        }
5745        return status;
5746    }
5747
5748    effect->setDevice(mDevice);
5749    effect->setMode(mAudioFlinger->getMode());
5750    return NO_ERROR;
5751}
5752
5753void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5754
5755    LOGV("removeEffect_l() %p effect %p", this, effect.get());
5756    effect_descriptor_t desc = effect->desc();
5757    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5758        detachAuxEffect_l(effect->id());
5759    }
5760
5761    sp<EffectChain> chain = effect->chain().promote();
5762    if (chain != 0) {
5763        // remove effect chain if removing last effect
5764        if (chain->removeEffect_l(effect) == 0) {
5765            removeEffectChain_l(chain);
5766        }
5767    } else {
5768        LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5769    }
5770}
5771
5772void AudioFlinger::ThreadBase::lockEffectChains_l(
5773        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5774{
5775    effectChains = mEffectChains;
5776    for (size_t i = 0; i < mEffectChains.size(); i++) {
5777        mEffectChains[i]->lock();
5778    }
5779}
5780
5781void AudioFlinger::ThreadBase::unlockEffectChains(
5782        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5783{
5784    for (size_t i = 0; i < effectChains.size(); i++) {
5785        effectChains[i]->unlock();
5786    }
5787}
5788
5789sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5790{
5791    Mutex::Autolock _l(mLock);
5792    return getEffectChain_l(sessionId);
5793}
5794
5795sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5796{
5797    sp<EffectChain> chain;
5798
5799    size_t size = mEffectChains.size();
5800    for (size_t i = 0; i < size; i++) {
5801        if (mEffectChains[i]->sessionId() == sessionId) {
5802            chain = mEffectChains[i];
5803            break;
5804        }
5805    }
5806    return chain;
5807}
5808
5809void AudioFlinger::ThreadBase::setMode(uint32_t mode)
5810{
5811    Mutex::Autolock _l(mLock);
5812    size_t size = mEffectChains.size();
5813    for (size_t i = 0; i < size; i++) {
5814        mEffectChains[i]->setMode_l(mode);
5815    }
5816}
5817
5818void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5819                                                    const wp<EffectHandle>& handle,
5820                                                    bool unpiniflast) {
5821
5822    Mutex::Autolock _l(mLock);
5823    LOGV("disconnectEffect() %p effect %p", this, effect.get());
5824    // delete the effect module if removing last handle on it
5825    if (effect->removeHandle(handle) == 0) {
5826        if (!effect->isPinned() || unpiniflast) {
5827            removeEffect_l(effect);
5828            AudioSystem::unregisterEffect(effect->id());
5829        }
5830    }
5831}
5832
5833status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5834{
5835    int session = chain->sessionId();
5836    int16_t *buffer = mMixBuffer;
5837    bool ownsBuffer = false;
5838
5839    LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5840    if (session > 0) {
5841        // Only one effect chain can be present in direct output thread and it uses
5842        // the mix buffer as input
5843        if (mType != DIRECT) {
5844            size_t numSamples = mFrameCount * mChannelCount;
5845            buffer = new int16_t[numSamples];
5846            memset(buffer, 0, numSamples * sizeof(int16_t));
5847            LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5848            ownsBuffer = true;
5849        }
5850
5851        // Attach all tracks with same session ID to this chain.
5852        for (size_t i = 0; i < mTracks.size(); ++i) {
5853            sp<Track> track = mTracks[i];
5854            if (session == track->sessionId()) {
5855                LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5856                track->setMainBuffer(buffer);
5857                chain->incTrackCnt();
5858            }
5859        }
5860
5861        // indicate all active tracks in the chain
5862        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5863            sp<Track> track = mActiveTracks[i].promote();
5864            if (track == 0) continue;
5865            if (session == track->sessionId()) {
5866                LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5867                chain->incActiveTrackCnt();
5868            }
5869        }
5870    }
5871
5872    chain->setInBuffer(buffer, ownsBuffer);
5873    chain->setOutBuffer(mMixBuffer);
5874    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5875    // chains list in order to be processed last as it contains output stage effects
5876    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5877    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5878    // after track specific effects and before output stage
5879    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5880    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5881    // Effect chain for other sessions are inserted at beginning of effect
5882    // chains list to be processed before output mix effects. Relative order between other
5883    // sessions is not important
5884    size_t size = mEffectChains.size();
5885    size_t i = 0;
5886    for (i = 0; i < size; i++) {
5887        if (mEffectChains[i]->sessionId() < session) break;
5888    }
5889    mEffectChains.insertAt(chain, i);
5890    checkSuspendOnAddEffectChain_l(chain);
5891
5892    return NO_ERROR;
5893}
5894
5895size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
5896{
5897    int session = chain->sessionId();
5898
5899    LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
5900
5901    for (size_t i = 0; i < mEffectChains.size(); i++) {
5902        if (chain == mEffectChains[i]) {
5903            mEffectChains.removeAt(i);
5904            // detach all active tracks from the chain
5905            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5906                sp<Track> track = mActiveTracks[i].promote();
5907                if (track == 0) continue;
5908                if (session == track->sessionId()) {
5909                    LOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
5910                            chain.get(), session);
5911                    chain->decActiveTrackCnt();
5912                }
5913            }
5914
5915            // detach all tracks with same session ID from this chain
5916            for (size_t i = 0; i < mTracks.size(); ++i) {
5917                sp<Track> track = mTracks[i];
5918                if (session == track->sessionId()) {
5919                    track->setMainBuffer(mMixBuffer);
5920                    chain->decTrackCnt();
5921                }
5922            }
5923            break;
5924        }
5925    }
5926    return mEffectChains.size();
5927}
5928
5929status_t AudioFlinger::PlaybackThread::attachAuxEffect(
5930        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
5931{
5932    Mutex::Autolock _l(mLock);
5933    return attachAuxEffect_l(track, EffectId);
5934}
5935
5936status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
5937        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
5938{
5939    status_t status = NO_ERROR;
5940
5941    if (EffectId == 0) {
5942        track->setAuxBuffer(0, NULL);
5943    } else {
5944        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
5945        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
5946        if (effect != 0) {
5947            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5948                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
5949            } else {
5950                status = INVALID_OPERATION;
5951            }
5952        } else {
5953            status = BAD_VALUE;
5954        }
5955    }
5956    return status;
5957}
5958
5959void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
5960{
5961     for (size_t i = 0; i < mTracks.size(); ++i) {
5962        sp<Track> track = mTracks[i];
5963        if (track->auxEffectId() == effectId) {
5964            attachAuxEffect_l(track, 0);
5965        }
5966    }
5967}
5968
5969status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
5970{
5971    // only one chain per input thread
5972    if (mEffectChains.size() != 0) {
5973        return INVALID_OPERATION;
5974    }
5975    LOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
5976
5977    chain->setInBuffer(NULL);
5978    chain->setOutBuffer(NULL);
5979
5980    checkSuspendOnAddEffectChain_l(chain);
5981
5982    mEffectChains.add(chain);
5983
5984    return NO_ERROR;
5985}
5986
5987size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
5988{
5989    LOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
5990    LOGW_IF(mEffectChains.size() != 1,
5991            "removeEffectChain_l() %p invalid chain size %d on thread %p",
5992            chain.get(), mEffectChains.size(), this);
5993    if (mEffectChains.size() == 1) {
5994        mEffectChains.removeAt(0);
5995    }
5996    return 0;
5997}
5998
5999// ----------------------------------------------------------------------------
6000//  EffectModule implementation
6001// ----------------------------------------------------------------------------
6002
6003#undef LOG_TAG
6004#define LOG_TAG "AudioFlinger::EffectModule"
6005
6006AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
6007                                        const wp<AudioFlinger::EffectChain>& chain,
6008                                        effect_descriptor_t *desc,
6009                                        int id,
6010                                        int sessionId)
6011    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
6012      mStatus(NO_INIT), mState(IDLE), mSuspended(false)
6013{
6014    LOGV("Constructor %p", this);
6015    int lStatus;
6016    sp<ThreadBase> thread = mThread.promote();
6017    if (thread == 0) {
6018        return;
6019    }
6020
6021    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
6022
6023    // create effect engine from effect factory
6024    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
6025
6026    if (mStatus != NO_ERROR) {
6027        return;
6028    }
6029    lStatus = init();
6030    if (lStatus < 0) {
6031        mStatus = lStatus;
6032        goto Error;
6033    }
6034
6035    if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
6036        mPinned = true;
6037    }
6038    LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
6039    return;
6040Error:
6041    EffectRelease(mEffectInterface);
6042    mEffectInterface = NULL;
6043    LOGV("Constructor Error %d", mStatus);
6044}
6045
6046AudioFlinger::EffectModule::~EffectModule()
6047{
6048    LOGV("Destructor %p", this);
6049    if (mEffectInterface != NULL) {
6050        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6051                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
6052            sp<ThreadBase> thread = mThread.promote();
6053            if (thread != 0) {
6054                audio_stream_t *stream = thread->stream();
6055                if (stream != NULL) {
6056                    stream->remove_audio_effect(stream, mEffectInterface);
6057                }
6058            }
6059        }
6060        // release effect engine
6061        EffectRelease(mEffectInterface);
6062    }
6063}
6064
6065status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
6066{
6067    status_t status;
6068
6069    Mutex::Autolock _l(mLock);
6070    // First handle in mHandles has highest priority and controls the effect module
6071    int priority = handle->priority();
6072    size_t size = mHandles.size();
6073    sp<EffectHandle> h;
6074    size_t i;
6075    for (i = 0; i < size; i++) {
6076        h = mHandles[i].promote();
6077        if (h == 0) continue;
6078        if (h->priority() <= priority) break;
6079    }
6080    // if inserted in first place, move effect control from previous owner to this handle
6081    if (i == 0) {
6082        bool enabled = false;
6083        if (h != 0) {
6084            enabled = h->enabled();
6085            h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
6086        }
6087        handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
6088        status = NO_ERROR;
6089    } else {
6090        status = ALREADY_EXISTS;
6091    }
6092    LOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
6093    mHandles.insertAt(handle, i);
6094    return status;
6095}
6096
6097size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
6098{
6099    Mutex::Autolock _l(mLock);
6100    size_t size = mHandles.size();
6101    size_t i;
6102    for (i = 0; i < size; i++) {
6103        if (mHandles[i] == handle) break;
6104    }
6105    if (i == size) {
6106        return size;
6107    }
6108    LOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
6109
6110    bool enabled = false;
6111    EffectHandle *hdl = handle.unsafe_get();
6112    if (hdl) {
6113        LOGV("removeHandle() unsafe_get OK");
6114        enabled = hdl->enabled();
6115    }
6116    mHandles.removeAt(i);
6117    size = mHandles.size();
6118    // if removed from first place, move effect control from this handle to next in line
6119    if (i == 0 && size != 0) {
6120        sp<EffectHandle> h = mHandles[0].promote();
6121        if (h != 0) {
6122            h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
6123        }
6124    }
6125
6126    // Prevent calls to process() and other functions on effect interface from now on.
6127    // The effect engine will be released by the destructor when the last strong reference on
6128    // this object is released which can happen after next process is called.
6129    if (size == 0 && !mPinned) {
6130        mState = DESTROYED;
6131    }
6132
6133    return size;
6134}
6135
6136sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
6137{
6138    Mutex::Autolock _l(mLock);
6139    sp<EffectHandle> handle;
6140    if (mHandles.size() != 0) {
6141        handle = mHandles[0].promote();
6142    }
6143    return handle;
6144}
6145
6146void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
6147{
6148    LOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
6149    // keep a strong reference on this EffectModule to avoid calling the
6150    // destructor before we exit
6151    sp<EffectModule> keep(this);
6152    {
6153        sp<ThreadBase> thread = mThread.promote();
6154        if (thread != 0) {
6155            thread->disconnectEffect(keep, handle, unpiniflast);
6156        }
6157    }
6158}
6159
6160void AudioFlinger::EffectModule::updateState() {
6161    Mutex::Autolock _l(mLock);
6162
6163    switch (mState) {
6164    case RESTART:
6165        reset_l();
6166        // FALL THROUGH
6167
6168    case STARTING:
6169        // clear auxiliary effect input buffer for next accumulation
6170        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6171            memset(mConfig.inputCfg.buffer.raw,
6172                   0,
6173                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6174        }
6175        start_l();
6176        mState = ACTIVE;
6177        break;
6178    case STOPPING:
6179        stop_l();
6180        mDisableWaitCnt = mMaxDisableWaitCnt;
6181        mState = STOPPED;
6182        break;
6183    case STOPPED:
6184        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
6185        // turn off sequence.
6186        if (--mDisableWaitCnt == 0) {
6187            reset_l();
6188            mState = IDLE;
6189        }
6190        break;
6191    default: //IDLE , ACTIVE, DESTROYED
6192        break;
6193    }
6194}
6195
6196void AudioFlinger::EffectModule::process()
6197{
6198    Mutex::Autolock _l(mLock);
6199
6200    if (mState == DESTROYED || mEffectInterface == NULL ||
6201            mConfig.inputCfg.buffer.raw == NULL ||
6202            mConfig.outputCfg.buffer.raw == NULL) {
6203        return;
6204    }
6205
6206    if (isProcessEnabled()) {
6207        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
6208        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6209            AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
6210                                        mConfig.inputCfg.buffer.s32,
6211                                        mConfig.inputCfg.buffer.frameCount/2);
6212        }
6213
6214        // do the actual processing in the effect engine
6215        int ret = (*mEffectInterface)->process(mEffectInterface,
6216                                               &mConfig.inputCfg.buffer,
6217                                               &mConfig.outputCfg.buffer);
6218
6219        // force transition to IDLE state when engine is ready
6220        if (mState == STOPPED && ret == -ENODATA) {
6221            mDisableWaitCnt = 1;
6222        }
6223
6224        // clear auxiliary effect input buffer for next accumulation
6225        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6226            memset(mConfig.inputCfg.buffer.raw, 0,
6227                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
6228        }
6229    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
6230                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6231        // If an insert effect is idle and input buffer is different from output buffer,
6232        // accumulate input onto output
6233        sp<EffectChain> chain = mChain.promote();
6234        if (chain != 0 && chain->activeTrackCnt() != 0) {
6235            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
6236            int16_t *in = mConfig.inputCfg.buffer.s16;
6237            int16_t *out = mConfig.outputCfg.buffer.s16;
6238            for (size_t i = 0; i < frameCnt; i++) {
6239                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
6240            }
6241        }
6242    }
6243}
6244
6245void AudioFlinger::EffectModule::reset_l()
6246{
6247    if (mEffectInterface == NULL) {
6248        return;
6249    }
6250    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
6251}
6252
6253status_t AudioFlinger::EffectModule::configure()
6254{
6255    uint32_t channels;
6256    if (mEffectInterface == NULL) {
6257        return NO_INIT;
6258    }
6259
6260    sp<ThreadBase> thread = mThread.promote();
6261    if (thread == 0) {
6262        return DEAD_OBJECT;
6263    }
6264
6265    // TODO: handle configuration of effects replacing track process
6266    if (thread->channelCount() == 1) {
6267        channels = AUDIO_CHANNEL_OUT_MONO;
6268    } else {
6269        channels = AUDIO_CHANNEL_OUT_STEREO;
6270    }
6271
6272    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6273        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
6274    } else {
6275        mConfig.inputCfg.channels = channels;
6276    }
6277    mConfig.outputCfg.channels = channels;
6278    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6279    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
6280    mConfig.inputCfg.samplingRate = thread->sampleRate();
6281    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
6282    mConfig.inputCfg.bufferProvider.cookie = NULL;
6283    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
6284    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
6285    mConfig.outputCfg.bufferProvider.cookie = NULL;
6286    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
6287    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
6288    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
6289    // Insert effect:
6290    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
6291    // always overwrites output buffer: input buffer == output buffer
6292    // - in other sessions:
6293    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
6294    //      other effect: overwrites output buffer: input buffer == output buffer
6295    // Auxiliary effect:
6296    //      accumulates in output buffer: input buffer != output buffer
6297    // Therefore: accumulate <=> input buffer != output buffer
6298    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
6299        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
6300    } else {
6301        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
6302    }
6303    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
6304    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
6305    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
6306    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
6307
6308    LOGV("configure() %p thread %p buffer %p framecount %d",
6309            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
6310
6311    status_t cmdStatus;
6312    uint32_t size = sizeof(int);
6313    status_t status = (*mEffectInterface)->command(mEffectInterface,
6314                                                   EFFECT_CMD_CONFIGURE,
6315                                                   sizeof(effect_config_t),
6316                                                   &mConfig,
6317                                                   &size,
6318                                                   &cmdStatus);
6319    if (status == 0) {
6320        status = cmdStatus;
6321    }
6322
6323    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
6324            (1000 * mConfig.outputCfg.buffer.frameCount);
6325
6326    return status;
6327}
6328
6329status_t AudioFlinger::EffectModule::init()
6330{
6331    Mutex::Autolock _l(mLock);
6332    if (mEffectInterface == NULL) {
6333        return NO_INIT;
6334    }
6335    status_t cmdStatus;
6336    uint32_t size = sizeof(status_t);
6337    status_t status = (*mEffectInterface)->command(mEffectInterface,
6338                                                   EFFECT_CMD_INIT,
6339                                                   0,
6340                                                   NULL,
6341                                                   &size,
6342                                                   &cmdStatus);
6343    if (status == 0) {
6344        status = cmdStatus;
6345    }
6346    return status;
6347}
6348
6349status_t AudioFlinger::EffectModule::start_l()
6350{
6351    if (mEffectInterface == NULL) {
6352        return NO_INIT;
6353    }
6354    status_t cmdStatus;
6355    uint32_t size = sizeof(status_t);
6356    status_t status = (*mEffectInterface)->command(mEffectInterface,
6357                                                   EFFECT_CMD_ENABLE,
6358                                                   0,
6359                                                   NULL,
6360                                                   &size,
6361                                                   &cmdStatus);
6362    if (status == 0) {
6363        status = cmdStatus;
6364    }
6365    if (status == 0 &&
6366            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6367             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6368        sp<ThreadBase> thread = mThread.promote();
6369        if (thread != 0) {
6370            audio_stream_t *stream = thread->stream();
6371            if (stream != NULL) {
6372                stream->add_audio_effect(stream, mEffectInterface);
6373            }
6374        }
6375    }
6376    return status;
6377}
6378
6379status_t AudioFlinger::EffectModule::stop()
6380{
6381    Mutex::Autolock _l(mLock);
6382    return stop_l();
6383}
6384
6385status_t AudioFlinger::EffectModule::stop_l()
6386{
6387    if (mEffectInterface == NULL) {
6388        return NO_INIT;
6389    }
6390    status_t cmdStatus;
6391    uint32_t size = sizeof(status_t);
6392    status_t status = (*mEffectInterface)->command(mEffectInterface,
6393                                                   EFFECT_CMD_DISABLE,
6394                                                   0,
6395                                                   NULL,
6396                                                   &size,
6397                                                   &cmdStatus);
6398    if (status == 0) {
6399        status = cmdStatus;
6400    }
6401    if (status == 0 &&
6402            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
6403             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
6404        sp<ThreadBase> thread = mThread.promote();
6405        if (thread != 0) {
6406            audio_stream_t *stream = thread->stream();
6407            if (stream != NULL) {
6408                stream->remove_audio_effect(stream, mEffectInterface);
6409            }
6410        }
6411    }
6412    return status;
6413}
6414
6415status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
6416                                             uint32_t cmdSize,
6417                                             void *pCmdData,
6418                                             uint32_t *replySize,
6419                                             void *pReplyData)
6420{
6421    Mutex::Autolock _l(mLock);
6422//    LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
6423
6424    if (mState == DESTROYED || mEffectInterface == NULL) {
6425        return NO_INIT;
6426    }
6427    status_t status = (*mEffectInterface)->command(mEffectInterface,
6428                                                   cmdCode,
6429                                                   cmdSize,
6430                                                   pCmdData,
6431                                                   replySize,
6432                                                   pReplyData);
6433    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
6434        uint32_t size = (replySize == NULL) ? 0 : *replySize;
6435        for (size_t i = 1; i < mHandles.size(); i++) {
6436            sp<EffectHandle> h = mHandles[i].promote();
6437            if (h != 0) {
6438                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
6439            }
6440        }
6441    }
6442    return status;
6443}
6444
6445status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
6446{
6447
6448    Mutex::Autolock _l(mLock);
6449    LOGV("setEnabled %p enabled %d", this, enabled);
6450
6451    if (enabled != isEnabled()) {
6452        status_t status = AudioSystem::setEffectEnabled(mId, enabled);
6453        if (enabled && status != NO_ERROR) {
6454            return status;
6455        }
6456
6457        switch (mState) {
6458        // going from disabled to enabled
6459        case IDLE:
6460            mState = STARTING;
6461            break;
6462        case STOPPED:
6463            mState = RESTART;
6464            break;
6465        case STOPPING:
6466            mState = ACTIVE;
6467            break;
6468
6469        // going from enabled to disabled
6470        case RESTART:
6471            mState = STOPPED;
6472            break;
6473        case STARTING:
6474            mState = IDLE;
6475            break;
6476        case ACTIVE:
6477            mState = STOPPING;
6478            break;
6479        case DESTROYED:
6480            return NO_ERROR; // simply ignore as we are being destroyed
6481        }
6482        for (size_t i = 1; i < mHandles.size(); i++) {
6483            sp<EffectHandle> h = mHandles[i].promote();
6484            if (h != 0) {
6485                h->setEnabled(enabled);
6486            }
6487        }
6488    }
6489    return NO_ERROR;
6490}
6491
6492bool AudioFlinger::EffectModule::isEnabled()
6493{
6494    switch (mState) {
6495    case RESTART:
6496    case STARTING:
6497    case ACTIVE:
6498        return true;
6499    case IDLE:
6500    case STOPPING:
6501    case STOPPED:
6502    case DESTROYED:
6503    default:
6504        return false;
6505    }
6506}
6507
6508bool AudioFlinger::EffectModule::isProcessEnabled()
6509{
6510    switch (mState) {
6511    case RESTART:
6512    case ACTIVE:
6513    case STOPPING:
6514    case STOPPED:
6515        return true;
6516    case IDLE:
6517    case STARTING:
6518    case DESTROYED:
6519    default:
6520        return false;
6521    }
6522}
6523
6524status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
6525{
6526    Mutex::Autolock _l(mLock);
6527    status_t status = NO_ERROR;
6528
6529    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
6530    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
6531    if (isProcessEnabled() &&
6532            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
6533            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
6534        status_t cmdStatus;
6535        uint32_t volume[2];
6536        uint32_t *pVolume = NULL;
6537        uint32_t size = sizeof(volume);
6538        volume[0] = *left;
6539        volume[1] = *right;
6540        if (controller) {
6541            pVolume = volume;
6542        }
6543        status = (*mEffectInterface)->command(mEffectInterface,
6544                                              EFFECT_CMD_SET_VOLUME,
6545                                              size,
6546                                              volume,
6547                                              &size,
6548                                              pVolume);
6549        if (controller && status == NO_ERROR && size == sizeof(volume)) {
6550            *left = volume[0];
6551            *right = volume[1];
6552        }
6553    }
6554    return status;
6555}
6556
6557status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
6558{
6559    Mutex::Autolock _l(mLock);
6560    status_t status = NO_ERROR;
6561    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
6562        // audio pre processing modules on RecordThread can receive both output and
6563        // input device indication in the same call
6564        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
6565        if (dev) {
6566            status_t cmdStatus;
6567            uint32_t size = sizeof(status_t);
6568
6569            status = (*mEffectInterface)->command(mEffectInterface,
6570                                                  EFFECT_CMD_SET_DEVICE,
6571                                                  sizeof(uint32_t),
6572                                                  &dev,
6573                                                  &size,
6574                                                  &cmdStatus);
6575            if (status == NO_ERROR) {
6576                status = cmdStatus;
6577            }
6578        }
6579        dev = device & AUDIO_DEVICE_IN_ALL;
6580        if (dev) {
6581            status_t cmdStatus;
6582            uint32_t size = sizeof(status_t);
6583
6584            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6585                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6586                                                  sizeof(uint32_t),
6587                                                  &dev,
6588                                                  &size,
6589                                                  &cmdStatus);
6590            if (status2 == NO_ERROR) {
6591                status2 = cmdStatus;
6592            }
6593            if (status == NO_ERROR) {
6594                status = status2;
6595            }
6596        }
6597    }
6598    return status;
6599}
6600
6601status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
6602{
6603    Mutex::Autolock _l(mLock);
6604    status_t status = NO_ERROR;
6605    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6606        status_t cmdStatus;
6607        uint32_t size = sizeof(status_t);
6608        status = (*mEffectInterface)->command(mEffectInterface,
6609                                              EFFECT_CMD_SET_AUDIO_MODE,
6610                                              sizeof(int),
6611                                              &mode,
6612                                              &size,
6613                                              &cmdStatus);
6614        if (status == NO_ERROR) {
6615            status = cmdStatus;
6616        }
6617    }
6618    return status;
6619}
6620
6621void AudioFlinger::EffectModule::setSuspended(bool suspended)
6622{
6623    Mutex::Autolock _l(mLock);
6624    mSuspended = suspended;
6625}
6626bool AudioFlinger::EffectModule::suspended()
6627{
6628    Mutex::Autolock _l(mLock);
6629    return mSuspended;
6630}
6631
6632status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6633{
6634    const size_t SIZE = 256;
6635    char buffer[SIZE];
6636    String8 result;
6637
6638    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6639    result.append(buffer);
6640
6641    bool locked = tryLock(mLock);
6642    // failed to lock - AudioFlinger is probably deadlocked
6643    if (!locked) {
6644        result.append("\t\tCould not lock Fx mutex:\n");
6645    }
6646
6647    result.append("\t\tSession Status State Engine:\n");
6648    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6649            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6650    result.append(buffer);
6651
6652    result.append("\t\tDescriptor:\n");
6653    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6654            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6655            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6656            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6657    result.append(buffer);
6658    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6659                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6660                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6661                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6662    result.append(buffer);
6663    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6664            mDescriptor.apiVersion,
6665            mDescriptor.flags);
6666    result.append(buffer);
6667    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6668            mDescriptor.name);
6669    result.append(buffer);
6670    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6671            mDescriptor.implementor);
6672    result.append(buffer);
6673
6674    result.append("\t\t- Input configuration:\n");
6675    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6676    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6677            (uint32_t)mConfig.inputCfg.buffer.raw,
6678            mConfig.inputCfg.buffer.frameCount,
6679            mConfig.inputCfg.samplingRate,
6680            mConfig.inputCfg.channels,
6681            mConfig.inputCfg.format);
6682    result.append(buffer);
6683
6684    result.append("\t\t- Output configuration:\n");
6685    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6686    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6687            (uint32_t)mConfig.outputCfg.buffer.raw,
6688            mConfig.outputCfg.buffer.frameCount,
6689            mConfig.outputCfg.samplingRate,
6690            mConfig.outputCfg.channels,
6691            mConfig.outputCfg.format);
6692    result.append(buffer);
6693
6694    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6695    result.append(buffer);
6696    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6697    for (size_t i = 0; i < mHandles.size(); ++i) {
6698        sp<EffectHandle> handle = mHandles[i].promote();
6699        if (handle != 0) {
6700            handle->dump(buffer, SIZE);
6701            result.append(buffer);
6702        }
6703    }
6704
6705    result.append("\n");
6706
6707    write(fd, result.string(), result.length());
6708
6709    if (locked) {
6710        mLock.unlock();
6711    }
6712
6713    return NO_ERROR;
6714}
6715
6716// ----------------------------------------------------------------------------
6717//  EffectHandle implementation
6718// ----------------------------------------------------------------------------
6719
6720#undef LOG_TAG
6721#define LOG_TAG "AudioFlinger::EffectHandle"
6722
6723AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6724                                        const sp<AudioFlinger::Client>& client,
6725                                        const sp<IEffectClient>& effectClient,
6726                                        int32_t priority)
6727    : BnEffect(),
6728    mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
6729    mPriority(priority), mHasControl(false), mEnabled(false)
6730{
6731    LOGV("constructor %p", this);
6732
6733    if (client == 0) {
6734        return;
6735    }
6736    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6737    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6738    if (mCblkMemory != 0) {
6739        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6740
6741        if (mCblk) {
6742            new(mCblk) effect_param_cblk_t();
6743            mBuffer = (uint8_t *)mCblk + bufOffset;
6744         }
6745    } else {
6746        LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6747        return;
6748    }
6749}
6750
6751AudioFlinger::EffectHandle::~EffectHandle()
6752{
6753    LOGV("Destructor %p", this);
6754    disconnect(false);
6755    LOGV("Destructor DONE %p", this);
6756}
6757
6758status_t AudioFlinger::EffectHandle::enable()
6759{
6760    LOGV("enable %p", this);
6761    if (!mHasControl) return INVALID_OPERATION;
6762    if (mEffect == 0) return DEAD_OBJECT;
6763
6764    if (mEnabled) {
6765        return NO_ERROR;
6766    }
6767
6768    mEnabled = true;
6769
6770    sp<ThreadBase> thread = mEffect->thread().promote();
6771    if (thread != 0) {
6772        thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
6773    }
6774
6775    // checkSuspendOnEffectEnabled() can suspend this same effect when enabled
6776    if (mEffect->suspended()) {
6777        return NO_ERROR;
6778    }
6779
6780    status_t status = mEffect->setEnabled(true);
6781    if (status != NO_ERROR) {
6782        if (thread != 0) {
6783            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6784        }
6785        mEnabled = false;
6786    }
6787    return status;
6788}
6789
6790status_t AudioFlinger::EffectHandle::disable()
6791{
6792    LOGV("disable %p", this);
6793    if (!mHasControl) return INVALID_OPERATION;
6794    if (mEffect == 0) return DEAD_OBJECT;
6795
6796    if (!mEnabled) {
6797        return NO_ERROR;
6798    }
6799    mEnabled = false;
6800
6801    if (mEffect->suspended()) {
6802        return NO_ERROR;
6803    }
6804
6805    status_t status = mEffect->setEnabled(false);
6806
6807    sp<ThreadBase> thread = mEffect->thread().promote();
6808    if (thread != 0) {
6809        thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6810    }
6811
6812    return status;
6813}
6814
6815void AudioFlinger::EffectHandle::disconnect()
6816{
6817    disconnect(true);
6818}
6819
6820void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
6821{
6822    LOGV("disconnect(%s)", unpiniflast ? "true" : "false");
6823    if (mEffect == 0) {
6824        return;
6825    }
6826    mEffect->disconnect(this, unpiniflast);
6827
6828    if (mEnabled) {
6829        sp<ThreadBase> thread = mEffect->thread().promote();
6830        if (thread != 0) {
6831            thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
6832        }
6833    }
6834
6835    // release sp on module => module destructor can be called now
6836    mEffect.clear();
6837    if (mClient != 0) {
6838        if (mCblk) {
6839            mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6840        }
6841        mCblkMemory.clear();            // and free the shared memory
6842        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6843        mClient.clear();
6844    }
6845}
6846
6847status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6848                                             uint32_t cmdSize,
6849                                             void *pCmdData,
6850                                             uint32_t *replySize,
6851                                             void *pReplyData)
6852{
6853//    LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6854//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6855
6856    // only get parameter command is permitted for applications not controlling the effect
6857    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6858        return INVALID_OPERATION;
6859    }
6860    if (mEffect == 0) return DEAD_OBJECT;
6861    if (mClient == 0) return INVALID_OPERATION;
6862
6863    // handle commands that are not forwarded transparently to effect engine
6864    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6865        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6866        // no risk to block the whole media server process or mixer threads is we are stuck here
6867        Mutex::Autolock _l(mCblk->lock);
6868        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6869            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6870            mCblk->serverIndex = 0;
6871            mCblk->clientIndex = 0;
6872            return BAD_VALUE;
6873        }
6874        status_t status = NO_ERROR;
6875        while (mCblk->serverIndex < mCblk->clientIndex) {
6876            int reply;
6877            uint32_t rsize = sizeof(int);
6878            int *p = (int *)(mBuffer + mCblk->serverIndex);
6879            int size = *p++;
6880            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
6881                LOGW("command(): invalid parameter block size");
6882                break;
6883            }
6884            effect_param_t *param = (effect_param_t *)p;
6885            if (param->psize == 0 || param->vsize == 0) {
6886                LOGW("command(): null parameter or value size");
6887                mCblk->serverIndex += size;
6888                continue;
6889            }
6890            uint32_t psize = sizeof(effect_param_t) +
6891                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
6892                             param->vsize;
6893            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
6894                                            psize,
6895                                            p,
6896                                            &rsize,
6897                                            &reply);
6898            // stop at first error encountered
6899            if (ret != NO_ERROR) {
6900                status = ret;
6901                *(int *)pReplyData = reply;
6902                break;
6903            } else if (reply != NO_ERROR) {
6904                *(int *)pReplyData = reply;
6905                break;
6906            }
6907            mCblk->serverIndex += size;
6908        }
6909        mCblk->serverIndex = 0;
6910        mCblk->clientIndex = 0;
6911        return status;
6912    } else if (cmdCode == EFFECT_CMD_ENABLE) {
6913        *(int *)pReplyData = NO_ERROR;
6914        return enable();
6915    } else if (cmdCode == EFFECT_CMD_DISABLE) {
6916        *(int *)pReplyData = NO_ERROR;
6917        return disable();
6918    }
6919
6920    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
6921}
6922
6923sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
6924    return mCblkMemory;
6925}
6926
6927void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
6928{
6929    LOGV("setControl %p control %d", this, hasControl);
6930
6931    mHasControl = hasControl;
6932    mEnabled = enabled;
6933
6934    if (signal && mEffectClient != 0) {
6935        mEffectClient->controlStatusChanged(hasControl);
6936    }
6937}
6938
6939void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
6940                                                 uint32_t cmdSize,
6941                                                 void *pCmdData,
6942                                                 uint32_t replySize,
6943                                                 void *pReplyData)
6944{
6945    if (mEffectClient != 0) {
6946        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
6947    }
6948}
6949
6950
6951
6952void AudioFlinger::EffectHandle::setEnabled(bool enabled)
6953{
6954    if (mEffectClient != 0) {
6955        mEffectClient->enableStatusChanged(enabled);
6956    }
6957}
6958
6959status_t AudioFlinger::EffectHandle::onTransact(
6960    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
6961{
6962    return BnEffect::onTransact(code, data, reply, flags);
6963}
6964
6965
6966void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
6967{
6968    bool locked = mCblk ? tryLock(mCblk->lock) : false;
6969
6970    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
6971            (mClient == NULL) ? getpid() : mClient->pid(),
6972            mPriority,
6973            mHasControl,
6974            !locked,
6975            mCblk ? mCblk->clientIndex : 0,
6976            mCblk ? mCblk->serverIndex : 0
6977            );
6978
6979    if (locked) {
6980        mCblk->lock.unlock();
6981    }
6982}
6983
6984#undef LOG_TAG
6985#define LOG_TAG "AudioFlinger::EffectChain"
6986
6987AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
6988                                        int sessionId)
6989    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0),
6990      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
6991      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
6992{
6993    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
6994}
6995
6996AudioFlinger::EffectChain::~EffectChain()
6997{
6998    if (mOwnInBuffer) {
6999        delete mInBuffer;
7000    }
7001
7002}
7003
7004// getEffectFromDesc_l() must be called with ThreadBase::mLock held
7005sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
7006{
7007    sp<EffectModule> effect;
7008    size_t size = mEffects.size();
7009
7010    for (size_t i = 0; i < size; i++) {
7011        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
7012            effect = mEffects[i];
7013            break;
7014        }
7015    }
7016    return effect;
7017}
7018
7019// getEffectFromId_l() must be called with ThreadBase::mLock held
7020sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
7021{
7022    sp<EffectModule> effect;
7023    size_t size = mEffects.size();
7024
7025    for (size_t i = 0; i < size; i++) {
7026        // by convention, return first effect if id provided is 0 (0 is never a valid id)
7027        if (id == 0 || mEffects[i]->id() == id) {
7028            effect = mEffects[i];
7029            break;
7030        }
7031    }
7032    return effect;
7033}
7034
7035// getEffectFromType_l() must be called with ThreadBase::mLock held
7036sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
7037        const effect_uuid_t *type)
7038{
7039    sp<EffectModule> effect;
7040    size_t size = mEffects.size();
7041
7042    for (size_t i = 0; i < size; i++) {
7043        if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
7044            effect = mEffects[i];
7045            break;
7046        }
7047    }
7048    return effect;
7049}
7050
7051// Must be called with EffectChain::mLock locked
7052void AudioFlinger::EffectChain::process_l()
7053{
7054    sp<ThreadBase> thread = mThread.promote();
7055    if (thread == 0) {
7056        LOGW("process_l(): cannot promote mixer thread");
7057        return;
7058    }
7059    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
7060            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
7061    bool tracksOnSession = false;
7062    if (!isGlobalSession) {
7063        tracksOnSession = (trackCnt() != 0);
7064    }
7065
7066    // if no track is active, input buffer must be cleared here as the mixer process
7067    // will not do it
7068    if (tracksOnSession &&
7069            activeTrackCnt() == 0) {
7070        size_t numSamples = thread->frameCount() * thread->channelCount();
7071        memset(mInBuffer, 0, numSamples * sizeof(int16_t));
7072    }
7073
7074    size_t size = mEffects.size();
7075    // do not process effect if no track is present in same audio session
7076    if (isGlobalSession || tracksOnSession) {
7077        for (size_t i = 0; i < size; i++) {
7078            mEffects[i]->process();
7079        }
7080    }
7081    for (size_t i = 0; i < size; i++) {
7082        mEffects[i]->updateState();
7083    }
7084}
7085
7086// addEffect_l() must be called with PlaybackThread::mLock held
7087status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
7088{
7089    effect_descriptor_t desc = effect->desc();
7090    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
7091
7092    Mutex::Autolock _l(mLock);
7093    effect->setChain(this);
7094    sp<ThreadBase> thread = mThread.promote();
7095    if (thread == 0) {
7096        return NO_INIT;
7097    }
7098    effect->setThread(thread);
7099
7100    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
7101        // Auxiliary effects are inserted at the beginning of mEffects vector as
7102        // they are processed first and accumulated in chain input buffer
7103        mEffects.insertAt(effect, 0);
7104
7105        // the input buffer for auxiliary effect contains mono samples in
7106        // 32 bit format. This is to avoid saturation in AudoMixer
7107        // accumulation stage. Saturation is done in EffectModule::process() before
7108        // calling the process in effect engine
7109        size_t numSamples = thread->frameCount();
7110        int32_t *buffer = new int32_t[numSamples];
7111        memset(buffer, 0, numSamples * sizeof(int32_t));
7112        effect->setInBuffer((int16_t *)buffer);
7113        // auxiliary effects output samples to chain input buffer for further processing
7114        // by insert effects
7115        effect->setOutBuffer(mInBuffer);
7116    } else {
7117        // Insert effects are inserted at the end of mEffects vector as they are processed
7118        //  after track and auxiliary effects.
7119        // Insert effect order as a function of indicated preference:
7120        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
7121        //  another effect is present
7122        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
7123        //  last effect claiming first position
7124        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
7125        //  first effect claiming last position
7126        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
7127        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
7128        // already present
7129
7130        int size = (int)mEffects.size();
7131        int idx_insert = size;
7132        int idx_insert_first = -1;
7133        int idx_insert_last = -1;
7134
7135        for (int i = 0; i < size; i++) {
7136            effect_descriptor_t d = mEffects[i]->desc();
7137            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
7138            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
7139            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
7140                // check invalid effect chaining combinations
7141                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
7142                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
7143                    LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
7144                    return INVALID_OPERATION;
7145                }
7146                // remember position of first insert effect and by default
7147                // select this as insert position for new effect
7148                if (idx_insert == size) {
7149                    idx_insert = i;
7150                }
7151                // remember position of last insert effect claiming
7152                // first position
7153                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
7154                    idx_insert_first = i;
7155                }
7156                // remember position of first insert effect claiming
7157                // last position
7158                if (iPref == EFFECT_FLAG_INSERT_LAST &&
7159                    idx_insert_last == -1) {
7160                    idx_insert_last = i;
7161                }
7162            }
7163        }
7164
7165        // modify idx_insert from first position if needed
7166        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
7167            if (idx_insert_last != -1) {
7168                idx_insert = idx_insert_last;
7169            } else {
7170                idx_insert = size;
7171            }
7172        } else {
7173            if (idx_insert_first != -1) {
7174                idx_insert = idx_insert_first + 1;
7175            }
7176        }
7177
7178        // always read samples from chain input buffer
7179        effect->setInBuffer(mInBuffer);
7180
7181        // if last effect in the chain, output samples to chain
7182        // output buffer, otherwise to chain input buffer
7183        if (idx_insert == size) {
7184            if (idx_insert != 0) {
7185                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
7186                mEffects[idx_insert-1]->configure();
7187            }
7188            effect->setOutBuffer(mOutBuffer);
7189        } else {
7190            effect->setOutBuffer(mInBuffer);
7191        }
7192        mEffects.insertAt(effect, idx_insert);
7193
7194        LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
7195    }
7196    effect->configure();
7197    return NO_ERROR;
7198}
7199
7200// removeEffect_l() must be called with PlaybackThread::mLock held
7201size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
7202{
7203    Mutex::Autolock _l(mLock);
7204    int size = (int)mEffects.size();
7205    int i;
7206    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
7207
7208    for (i = 0; i < size; i++) {
7209        if (effect == mEffects[i]) {
7210            // calling stop here will remove pre-processing effect from the audio HAL.
7211            // This is safe as we hold the EffectChain mutex which guarantees that we are not in
7212            // the middle of a read from audio HAL
7213            mEffects[i]->stop();
7214            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
7215                delete[] effect->inBuffer();
7216            } else {
7217                if (i == size - 1 && i != 0) {
7218                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
7219                    mEffects[i - 1]->configure();
7220                }
7221            }
7222            mEffects.removeAt(i);
7223            LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
7224            break;
7225        }
7226    }
7227
7228    return mEffects.size();
7229}
7230
7231// setDevice_l() must be called with PlaybackThread::mLock held
7232void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
7233{
7234    size_t size = mEffects.size();
7235    for (size_t i = 0; i < size; i++) {
7236        mEffects[i]->setDevice(device);
7237    }
7238}
7239
7240// setMode_l() must be called with PlaybackThread::mLock held
7241void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
7242{
7243    size_t size = mEffects.size();
7244    for (size_t i = 0; i < size; i++) {
7245        mEffects[i]->setMode(mode);
7246    }
7247}
7248
7249// setVolume_l() must be called with PlaybackThread::mLock held
7250bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
7251{
7252    uint32_t newLeft = *left;
7253    uint32_t newRight = *right;
7254    bool hasControl = false;
7255    int ctrlIdx = -1;
7256    size_t size = mEffects.size();
7257
7258    // first update volume controller
7259    for (size_t i = size; i > 0; i--) {
7260        if (mEffects[i - 1]->isProcessEnabled() &&
7261            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
7262            ctrlIdx = i - 1;
7263            hasControl = true;
7264            break;
7265        }
7266    }
7267
7268    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
7269        if (hasControl) {
7270            *left = mNewLeftVolume;
7271            *right = mNewRightVolume;
7272        }
7273        return hasControl;
7274    }
7275
7276    mVolumeCtrlIdx = ctrlIdx;
7277    mLeftVolume = newLeft;
7278    mRightVolume = newRight;
7279
7280    // second get volume update from volume controller
7281    if (ctrlIdx >= 0) {
7282        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
7283        mNewLeftVolume = newLeft;
7284        mNewRightVolume = newRight;
7285    }
7286    // then indicate volume to all other effects in chain.
7287    // Pass altered volume to effects before volume controller
7288    // and requested volume to effects after controller
7289    uint32_t lVol = newLeft;
7290    uint32_t rVol = newRight;
7291
7292    for (size_t i = 0; i < size; i++) {
7293        if ((int)i == ctrlIdx) continue;
7294        // this also works for ctrlIdx == -1 when there is no volume controller
7295        if ((int)i > ctrlIdx) {
7296            lVol = *left;
7297            rVol = *right;
7298        }
7299        mEffects[i]->setVolume(&lVol, &rVol, false);
7300    }
7301    *left = newLeft;
7302    *right = newRight;
7303
7304    return hasControl;
7305}
7306
7307status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
7308{
7309    const size_t SIZE = 256;
7310    char buffer[SIZE];
7311    String8 result;
7312
7313    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
7314    result.append(buffer);
7315
7316    bool locked = tryLock(mLock);
7317    // failed to lock - AudioFlinger is probably deadlocked
7318    if (!locked) {
7319        result.append("\tCould not lock mutex:\n");
7320    }
7321
7322    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
7323    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
7324            mEffects.size(),
7325            (uint32_t)mInBuffer,
7326            (uint32_t)mOutBuffer,
7327            mActiveTrackCnt);
7328    result.append(buffer);
7329    write(fd, result.string(), result.size());
7330
7331    for (size_t i = 0; i < mEffects.size(); ++i) {
7332        sp<EffectModule> effect = mEffects[i];
7333        if (effect != 0) {
7334            effect->dump(fd, args);
7335        }
7336    }
7337
7338    if (locked) {
7339        mLock.unlock();
7340    }
7341
7342    return NO_ERROR;
7343}
7344
7345// must be called with ThreadBase::mLock held
7346void AudioFlinger::EffectChain::setEffectSuspended_l(
7347        const effect_uuid_t *type, bool suspend)
7348{
7349    sp<SuspendedEffectDesc> desc;
7350    // use effect type UUID timelow as key as there is no real risk of identical
7351    // timeLow fields among effect type UUIDs.
7352    int index = mSuspendedEffects.indexOfKey(type->timeLow);
7353    if (suspend) {
7354        if (index >= 0) {
7355            desc = mSuspendedEffects.valueAt(index);
7356        } else {
7357            desc = new SuspendedEffectDesc();
7358            memcpy(&desc->mType, type, sizeof(effect_uuid_t));
7359            mSuspendedEffects.add(type->timeLow, desc);
7360            LOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
7361        }
7362        if (desc->mRefCount++ == 0) {
7363            sp<EffectModule> effect = getEffectIfEnabled(type);
7364            if (effect != 0) {
7365                desc->mEffect = effect;
7366                effect->setSuspended(true);
7367                effect->setEnabled(false);
7368            }
7369        }
7370    } else {
7371        if (index < 0) {
7372            return;
7373        }
7374        desc = mSuspendedEffects.valueAt(index);
7375        if (desc->mRefCount <= 0) {
7376            LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
7377            desc->mRefCount = 1;
7378        }
7379        if (--desc->mRefCount == 0) {
7380            LOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7381            if (desc->mEffect != 0) {
7382                sp<EffectModule> effect = desc->mEffect.promote();
7383                if (effect != 0) {
7384                    effect->setSuspended(false);
7385                    sp<EffectHandle> handle = effect->controlHandle();
7386                    if (handle != 0) {
7387                        effect->setEnabled(handle->enabled());
7388                    }
7389                }
7390                desc->mEffect.clear();
7391            }
7392            mSuspendedEffects.removeItemsAt(index);
7393        }
7394    }
7395}
7396
7397// must be called with ThreadBase::mLock held
7398void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
7399{
7400    sp<SuspendedEffectDesc> desc;
7401
7402    int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7403    if (suspend) {
7404        if (index >= 0) {
7405            desc = mSuspendedEffects.valueAt(index);
7406        } else {
7407            desc = new SuspendedEffectDesc();
7408            mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
7409            LOGV("setEffectSuspendedAll_l() add entry for 0");
7410        }
7411        if (desc->mRefCount++ == 0) {
7412            Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
7413            for (size_t i = 0; i < effects.size(); i++) {
7414                setEffectSuspended_l(&effects[i]->desc().type, true);
7415            }
7416        }
7417    } else {
7418        if (index < 0) {
7419            return;
7420        }
7421        desc = mSuspendedEffects.valueAt(index);
7422        if (desc->mRefCount <= 0) {
7423            LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
7424            desc->mRefCount = 1;
7425        }
7426        if (--desc->mRefCount == 0) {
7427            Vector<const effect_uuid_t *> types;
7428            for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
7429                if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
7430                    continue;
7431                }
7432                types.add(&mSuspendedEffects.valueAt(i)->mType);
7433            }
7434            for (size_t i = 0; i < types.size(); i++) {
7435                setEffectSuspended_l(types[i], false);
7436            }
7437            LOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
7438            mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
7439        }
7440    }
7441}
7442
7443
7444// The volume effect is used for automated tests only
7445#ifndef OPENSL_ES_H_
7446static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
7447                                            { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
7448const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
7449#endif //OPENSL_ES_H_
7450
7451bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
7452{
7453    // auxiliary effects and visualizer are never suspended on output mix
7454    if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
7455        (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
7456         (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
7457         (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
7458        return false;
7459    }
7460    return true;
7461}
7462
7463Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
7464{
7465    Vector< sp<EffectModule> > effects;
7466    for (size_t i = 0; i < mEffects.size(); i++) {
7467        if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
7468            continue;
7469        }
7470        effects.add(mEffects[i]);
7471    }
7472    return effects;
7473}
7474
7475sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
7476                                                            const effect_uuid_t *type)
7477{
7478    sp<EffectModule> effect;
7479    effect = getEffectFromType_l(type);
7480    if (effect != 0 && !effect->isEnabled()) {
7481        effect.clear();
7482    }
7483    return effect;
7484}
7485
7486void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
7487                                                            bool enabled)
7488{
7489    int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7490    if (enabled) {
7491        if (index < 0) {
7492            // if the effect is not suspend check if all effects are suspended
7493            index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
7494            if (index < 0) {
7495                return;
7496            }
7497            if (!isEffectEligibleForSuspend(effect->desc())) {
7498                return;
7499            }
7500            setEffectSuspended_l(&effect->desc().type, enabled);
7501            index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
7502            if (index < 0) {
7503                LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
7504                return;
7505            }
7506        }
7507        LOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
7508             effect->desc().type.timeLow);
7509        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7510        // if effect is requested to suspended but was not yet enabled, supend it now.
7511        if (desc->mEffect == 0) {
7512            desc->mEffect = effect;
7513            effect->setEnabled(false);
7514            effect->setSuspended(true);
7515        }
7516    } else {
7517        if (index < 0) {
7518            return;
7519        }
7520        LOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
7521             effect->desc().type.timeLow);
7522        sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
7523        desc->mEffect.clear();
7524        effect->setSuspended(false);
7525    }
7526}
7527
7528#undef LOG_TAG
7529#define LOG_TAG "AudioFlinger"
7530
7531// ----------------------------------------------------------------------------
7532
7533status_t AudioFlinger::onTransact(
7534        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
7535{
7536    return BnAudioFlinger::onTransact(code, data, reply, flags);
7537}
7538
7539}; // namespace android
7540