AudioFlinger.cpp revision 1d2bff0e588afe183a1baaae731519b4e957bbdb
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
55#include <cpustats/ThreadCpuUsage.h>
56// #define DEBUG_CPU_USAGE 10  // log statistics every n wall clock seconds
57
58// ----------------------------------------------------------------------------
59
60
61namespace android {
62
63static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
64static const char* kHardwareLockedString = "Hardware lock is taken\n";
65
66//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
67static const float MAX_GAIN = 4096.0f;
68static const float MAX_GAIN_INT = 0x1000;
69
70// retry counts for buffer fill timeout
71// 50 * ~20msecs = 1 second
72static const int8_t kMaxTrackRetries = 50;
73static const int8_t kMaxTrackStartupRetries = 50;
74// allow less retry attempts on direct output thread.
75// direct outputs can be a scarce resource in audio hardware and should
76// be released as quickly as possible.
77static const int8_t kMaxTrackRetriesDirect = 2;
78
79static const int kDumpLockRetries = 50;
80static const int kDumpLockSleep = 20000;
81
82static const nsecs_t kWarningThrottle = seconds(5);
83
84// RecordThread loop sleep time upon application overrun or audio HAL read error
85static const int kRecordThreadSleepUs = 5000;
86
87// ----------------------------------------------------------------------------
88
89static bool recordingAllowed() {
90    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
91    bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
92    if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
93    return ok;
94}
95
96static bool settingsAllowed() {
97    if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
98    bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
99    if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
100    return ok;
101}
102
103// To collect the amplifier usage
104static void addBatteryData(uint32_t params) {
105    sp<IBinder> binder =
106        defaultServiceManager()->getService(String16("media.player"));
107    sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder);
108    if (service.get() == NULL) {
109        LOGW("Cannot connect to the MediaPlayerService for battery tracking");
110        return;
111    }
112
113    service->addBatteryData(params);
114}
115
116static int load_audio_interface(const char *if_name, const hw_module_t **mod,
117                                audio_hw_device_t **dev)
118{
119    int rc;
120
121    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
122    if (rc)
123        goto out;
124
125    rc = audio_hw_device_open(*mod, dev);
126    LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
127            AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
128    if (rc)
129        goto out;
130
131    return 0;
132
133out:
134    *mod = NULL;
135    *dev = NULL;
136    return rc;
137}
138
139static const char *audio_interfaces[] = {
140    "primary",
141    "a2dp",
142    "usb",
143};
144#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
145
146// ----------------------------------------------------------------------------
147
148AudioFlinger::AudioFlinger()
149    : BnAudioFlinger(),
150        mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1)
151{
152}
153
154void AudioFlinger::onFirstRef()
155{
156    int rc = 0;
157
158    Mutex::Autolock _l(mLock);
159
160    /* TODO: move all this work into an Init() function */
161    mHardwareStatus = AUDIO_HW_IDLE;
162
163    for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
164        const hw_module_t *mod;
165        audio_hw_device_t *dev;
166
167        rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
168        if (rc)
169            continue;
170
171        LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
172             mod->name, mod->id);
173        mAudioHwDevs.push(dev);
174
175        if (!mPrimaryHardwareDev) {
176            mPrimaryHardwareDev = dev;
177            LOGI("Using '%s' (%s.%s) as the primary audio interface",
178                 mod->name, mod->id, audio_interfaces[i]);
179        }
180    }
181
182    mHardwareStatus = AUDIO_HW_INIT;
183
184    if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
185        LOGE("Primary audio interface not found");
186        return;
187    }
188
189    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
190        audio_hw_device_t *dev = mAudioHwDevs[i];
191
192        mHardwareStatus = AUDIO_HW_INIT;
193        rc = dev->init_check(dev);
194        if (rc == 0) {
195            AutoMutex lock(mHardwareLock);
196
197            mMode = AUDIO_MODE_NORMAL;
198            mHardwareStatus = AUDIO_HW_SET_MODE;
199            dev->set_mode(dev, mMode);
200            mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
201            dev->set_master_volume(dev, 1.0f);
202            mHardwareStatus = AUDIO_HW_IDLE;
203        }
204    }
205}
206
207status_t AudioFlinger::initCheck() const
208{
209    Mutex::Autolock _l(mLock);
210    if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
211        return NO_INIT;
212    return NO_ERROR;
213}
214
215AudioFlinger::~AudioFlinger()
216{
217    int num_devs = mAudioHwDevs.size();
218
219    while (!mRecordThreads.isEmpty()) {
220        // closeInput() will remove first entry from mRecordThreads
221        closeInput(mRecordThreads.keyAt(0));
222    }
223    while (!mPlaybackThreads.isEmpty()) {
224        // closeOutput() will remove first entry from mPlaybackThreads
225        closeOutput(mPlaybackThreads.keyAt(0));
226    }
227
228    for (int i = 0; i < num_devs; i++) {
229        audio_hw_device_t *dev = mAudioHwDevs[i];
230        audio_hw_device_close(dev);
231    }
232    mAudioHwDevs.clear();
233}
234
235audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
236{
237    /* first matching HW device is returned */
238    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
239        audio_hw_device_t *dev = mAudioHwDevs[i];
240        if ((dev->get_supported_devices(dev) & devices) == devices)
241            return dev;
242    }
243    return NULL;
244}
245
246status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
247{
248    const size_t SIZE = 256;
249    char buffer[SIZE];
250    String8 result;
251
252    result.append("Clients:\n");
253    for (size_t i = 0; i < mClients.size(); ++i) {
254        wp<Client> wClient = mClients.valueAt(i);
255        if (wClient != 0) {
256            sp<Client> client = wClient.promote();
257            if (client != 0) {
258                snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
259                result.append(buffer);
260            }
261        }
262    }
263    write(fd, result.string(), result.size());
264    return NO_ERROR;
265}
266
267
268status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
269{
270    const size_t SIZE = 256;
271    char buffer[SIZE];
272    String8 result;
273    int hardwareStatus = mHardwareStatus;
274
275    snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
276    result.append(buffer);
277    write(fd, result.string(), result.size());
278    return NO_ERROR;
279}
280
281status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
282{
283    const size_t SIZE = 256;
284    char buffer[SIZE];
285    String8 result;
286    snprintf(buffer, SIZE, "Permission Denial: "
287            "can't dump AudioFlinger from pid=%d, uid=%d\n",
288            IPCThreadState::self()->getCallingPid(),
289            IPCThreadState::self()->getCallingUid());
290    result.append(buffer);
291    write(fd, result.string(), result.size());
292    return NO_ERROR;
293}
294
295static bool tryLock(Mutex& mutex)
296{
297    bool locked = false;
298    for (int i = 0; i < kDumpLockRetries; ++i) {
299        if (mutex.tryLock() == NO_ERROR) {
300            locked = true;
301            break;
302        }
303        usleep(kDumpLockSleep);
304    }
305    return locked;
306}
307
308status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
309{
310    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
311        dumpPermissionDenial(fd, args);
312    } else {
313        // get state of hardware lock
314        bool hardwareLocked = tryLock(mHardwareLock);
315        if (!hardwareLocked) {
316            String8 result(kHardwareLockedString);
317            write(fd, result.string(), result.size());
318        } else {
319            mHardwareLock.unlock();
320        }
321
322        bool locked = tryLock(mLock);
323
324        // failed to lock - AudioFlinger is probably deadlocked
325        if (!locked) {
326            String8 result(kDeadlockedString);
327            write(fd, result.string(), result.size());
328        }
329
330        dumpClients(fd, args);
331        dumpInternals(fd, args);
332
333        // dump playback threads
334        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
335            mPlaybackThreads.valueAt(i)->dump(fd, args);
336        }
337
338        // dump record threads
339        for (size_t i = 0; i < mRecordThreads.size(); i++) {
340            mRecordThreads.valueAt(i)->dump(fd, args);
341        }
342
343        // dump all hardware devs
344        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
345            audio_hw_device_t *dev = mAudioHwDevs[i];
346            dev->dump(dev, fd);
347        }
348        if (locked) mLock.unlock();
349    }
350    return NO_ERROR;
351}
352
353
354// IAudioFlinger interface
355
356
357sp<IAudioTrack> AudioFlinger::createTrack(
358        pid_t pid,
359        int streamType,
360        uint32_t sampleRate,
361        uint32_t format,
362        uint32_t channelMask,
363        int frameCount,
364        uint32_t flags,
365        const sp<IMemory>& sharedBuffer,
366        int output,
367        int *sessionId,
368        status_t *status)
369{
370    sp<PlaybackThread::Track> track;
371    sp<TrackHandle> trackHandle;
372    sp<Client> client;
373    wp<Client> wclient;
374    status_t lStatus;
375    int lSessionId;
376
377    if (streamType >= AUDIO_STREAM_CNT) {
378        LOGE("invalid stream type");
379        lStatus = BAD_VALUE;
380        goto Exit;
381    }
382
383    {
384        Mutex::Autolock _l(mLock);
385        PlaybackThread *thread = checkPlaybackThread_l(output);
386        PlaybackThread *effectThread = NULL;
387        if (thread == NULL) {
388            LOGE("unknown output thread");
389            lStatus = BAD_VALUE;
390            goto Exit;
391        }
392
393        wclient = mClients.valueFor(pid);
394
395        if (wclient != NULL) {
396            client = wclient.promote();
397        } else {
398            client = new Client(this, pid);
399            mClients.add(pid, client);
400        }
401
402        LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
403        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
404            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
405                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
406                if (mPlaybackThreads.keyAt(i) != output) {
407                    // prevent same audio session on different output threads
408                    uint32_t sessions = t->hasAudioSession(*sessionId);
409                    if (sessions & PlaybackThread::TRACK_SESSION) {
410                        lStatus = BAD_VALUE;
411                        goto Exit;
412                    }
413                    // check if an effect with same session ID is waiting for a track to be created
414                    if (sessions & PlaybackThread::EFFECT_SESSION) {
415                        effectThread = t.get();
416                    }
417                }
418            }
419            lSessionId = *sessionId;
420        } else {
421            // if no audio session id is provided, create one here
422            lSessionId = nextUniqueId();
423            if (sessionId != NULL) {
424                *sessionId = lSessionId;
425            }
426        }
427        LOGV("createTrack() lSessionId: %d", lSessionId);
428
429        track = thread->createTrack_l(client, streamType, sampleRate, format,
430                channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
431
432        // move effect chain to this output thread if an effect on same session was waiting
433        // for a track to be created
434        if (lStatus == NO_ERROR && effectThread != NULL) {
435            Mutex::Autolock _dl(thread->mLock);
436            Mutex::Autolock _sl(effectThread->mLock);
437            moveEffectChain_l(lSessionId, effectThread, thread, true);
438        }
439    }
440    if (lStatus == NO_ERROR) {
441        trackHandle = new TrackHandle(track);
442    } else {
443        // remove local strong reference to Client before deleting the Track so that the Client
444        // destructor is called by the TrackBase destructor with mLock held
445        client.clear();
446        track.clear();
447    }
448
449Exit:
450    if(status) {
451        *status = lStatus;
452    }
453    return trackHandle;
454}
455
456uint32_t AudioFlinger::sampleRate(int output) const
457{
458    Mutex::Autolock _l(mLock);
459    PlaybackThread *thread = checkPlaybackThread_l(output);
460    if (thread == NULL) {
461        LOGW("sampleRate() unknown thread %d", output);
462        return 0;
463    }
464    return thread->sampleRate();
465}
466
467int AudioFlinger::channelCount(int output) const
468{
469    Mutex::Autolock _l(mLock);
470    PlaybackThread *thread = checkPlaybackThread_l(output);
471    if (thread == NULL) {
472        LOGW("channelCount() unknown thread %d", output);
473        return 0;
474    }
475    return thread->channelCount();
476}
477
478uint32_t AudioFlinger::format(int output) const
479{
480    Mutex::Autolock _l(mLock);
481    PlaybackThread *thread = checkPlaybackThread_l(output);
482    if (thread == NULL) {
483        LOGW("format() unknown thread %d", output);
484        return 0;
485    }
486    return thread->format();
487}
488
489size_t AudioFlinger::frameCount(int output) const
490{
491    Mutex::Autolock _l(mLock);
492    PlaybackThread *thread = checkPlaybackThread_l(output);
493    if (thread == NULL) {
494        LOGW("frameCount() unknown thread %d", output);
495        return 0;
496    }
497    return thread->frameCount();
498}
499
500uint32_t AudioFlinger::latency(int output) const
501{
502    Mutex::Autolock _l(mLock);
503    PlaybackThread *thread = checkPlaybackThread_l(output);
504    if (thread == NULL) {
505        LOGW("latency() unknown thread %d", output);
506        return 0;
507    }
508    return thread->latency();
509}
510
511status_t AudioFlinger::setMasterVolume(float value)
512{
513    // check calling permissions
514    if (!settingsAllowed()) {
515        return PERMISSION_DENIED;
516    }
517
518    // when hw supports master volume, don't scale in sw mixer
519    { // scope for the lock
520        AutoMutex lock(mHardwareLock);
521        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
522        if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
523            value = 1.0f;
524        }
525        mHardwareStatus = AUDIO_HW_IDLE;
526    }
527
528    Mutex::Autolock _l(mLock);
529    mMasterVolume = value;
530    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
531       mPlaybackThreads.valueAt(i)->setMasterVolume(value);
532
533    return NO_ERROR;
534}
535
536status_t AudioFlinger::setMode(int mode)
537{
538    status_t ret;
539
540    // check calling permissions
541    if (!settingsAllowed()) {
542        return PERMISSION_DENIED;
543    }
544    if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) {
545        LOGW("Illegal value: setMode(%d)", mode);
546        return BAD_VALUE;
547    }
548
549    { // scope for the lock
550        AutoMutex lock(mHardwareLock);
551        mHardwareStatus = AUDIO_HW_SET_MODE;
552        ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
553        mHardwareStatus = AUDIO_HW_IDLE;
554    }
555
556    if (NO_ERROR == ret) {
557        Mutex::Autolock _l(mLock);
558        mMode = mode;
559        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
560           mPlaybackThreads.valueAt(i)->setMode(mode);
561    }
562
563    return ret;
564}
565
566status_t AudioFlinger::setMicMute(bool state)
567{
568    // check calling permissions
569    if (!settingsAllowed()) {
570        return PERMISSION_DENIED;
571    }
572
573    AutoMutex lock(mHardwareLock);
574    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
575    status_t ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
576    mHardwareStatus = AUDIO_HW_IDLE;
577    return ret;
578}
579
580bool AudioFlinger::getMicMute() const
581{
582    bool state = AUDIO_MODE_INVALID;
583    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
584    mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
585    mHardwareStatus = AUDIO_HW_IDLE;
586    return state;
587}
588
589status_t AudioFlinger::setMasterMute(bool muted)
590{
591    // check calling permissions
592    if (!settingsAllowed()) {
593        return PERMISSION_DENIED;
594    }
595
596    Mutex::Autolock _l(mLock);
597    mMasterMute = muted;
598    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
599       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
600
601    return NO_ERROR;
602}
603
604float AudioFlinger::masterVolume() const
605{
606    return mMasterVolume;
607}
608
609bool AudioFlinger::masterMute() const
610{
611    return mMasterMute;
612}
613
614status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
615{
616    // check calling permissions
617    if (!settingsAllowed()) {
618        return PERMISSION_DENIED;
619    }
620
621    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
622        return BAD_VALUE;
623    }
624
625    AutoMutex lock(mLock);
626    PlaybackThread *thread = NULL;
627    if (output) {
628        thread = checkPlaybackThread_l(output);
629        if (thread == NULL) {
630            return BAD_VALUE;
631        }
632    }
633
634    mStreamTypes[stream].volume = value;
635
636    if (thread == NULL) {
637        for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
638           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
639        }
640    } else {
641        thread->setStreamVolume(stream, value);
642    }
643
644    return NO_ERROR;
645}
646
647status_t AudioFlinger::setStreamMute(int stream, bool muted)
648{
649    // check calling permissions
650    if (!settingsAllowed()) {
651        return PERMISSION_DENIED;
652    }
653
654    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
655        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
656        return BAD_VALUE;
657    }
658
659    AutoMutex lock(mLock);
660    mStreamTypes[stream].mute = muted;
661    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
662       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
663
664    return NO_ERROR;
665}
666
667float AudioFlinger::streamVolume(int stream, int output) const
668{
669    if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
670        return 0.0f;
671    }
672
673    AutoMutex lock(mLock);
674    float volume;
675    if (output) {
676        PlaybackThread *thread = checkPlaybackThread_l(output);
677        if (thread == NULL) {
678            return 0.0f;
679        }
680        volume = thread->streamVolume(stream);
681    } else {
682        volume = mStreamTypes[stream].volume;
683    }
684
685    return volume;
686}
687
688bool AudioFlinger::streamMute(int stream) const
689{
690    if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
691        return true;
692    }
693
694    return mStreamTypes[stream].mute;
695}
696
697status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
698{
699    status_t result;
700
701    LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
702            ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
703    // check calling permissions
704    if (!settingsAllowed()) {
705        return PERMISSION_DENIED;
706    }
707
708    // ioHandle == 0 means the parameters are global to the audio hardware interface
709    if (ioHandle == 0) {
710        AutoMutex lock(mHardwareLock);
711        mHardwareStatus = AUDIO_SET_PARAMETER;
712        status_t final_result = NO_ERROR;
713        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
714            audio_hw_device_t *dev = mAudioHwDevs[i];
715            result = dev->set_parameters(dev, keyValuePairs.string());
716            final_result = result ?: final_result;
717        }
718        mHardwareStatus = AUDIO_HW_IDLE;
719        return final_result;
720    }
721
722    // hold a strong ref on thread in case closeOutput() or closeInput() is called
723    // and the thread is exited once the lock is released
724    sp<ThreadBase> thread;
725    {
726        Mutex::Autolock _l(mLock);
727        thread = checkPlaybackThread_l(ioHandle);
728        if (thread == NULL) {
729            thread = checkRecordThread_l(ioHandle);
730        } else if (thread.get() == primaryPlaybackThread_l()) {
731            // indicate output device change to all input threads for pre processing
732            AudioParameter param = AudioParameter(keyValuePairs);
733            int value;
734            if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
735                for (size_t i = 0; i < mRecordThreads.size(); i++) {
736                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
737                }
738            }
739        }
740    }
741    if (thread != NULL) {
742        result = thread->setParameters(keyValuePairs);
743        return result;
744    }
745    return BAD_VALUE;
746}
747
748String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
749{
750//    LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
751//            ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
752
753    if (ioHandle == 0) {
754        String8 out_s8;
755
756        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
757            audio_hw_device_t *dev = mAudioHwDevs[i];
758            char *s = dev->get_parameters(dev, keys.string());
759            out_s8 += String8(s);
760            free(s);
761        }
762        return out_s8;
763    }
764
765    Mutex::Autolock _l(mLock);
766
767    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
768    if (playbackThread != NULL) {
769        return playbackThread->getParameters(keys);
770    }
771    RecordThread *recordThread = checkRecordThread_l(ioHandle);
772    if (recordThread != NULL) {
773        return recordThread->getParameters(keys);
774    }
775    return String8("");
776}
777
778size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
779{
780    return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
781}
782
783unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
784{
785    if (ioHandle == 0) {
786        return 0;
787    }
788
789    Mutex::Autolock _l(mLock);
790
791    RecordThread *recordThread = checkRecordThread_l(ioHandle);
792    if (recordThread != NULL) {
793        return recordThread->getInputFramesLost();
794    }
795    return 0;
796}
797
798status_t AudioFlinger::setVoiceVolume(float value)
799{
800    // check calling permissions
801    if (!settingsAllowed()) {
802        return PERMISSION_DENIED;
803    }
804
805    AutoMutex lock(mHardwareLock);
806    mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
807    status_t ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
808    mHardwareStatus = AUDIO_HW_IDLE;
809
810    return ret;
811}
812
813status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
814{
815    status_t status;
816
817    Mutex::Autolock _l(mLock);
818
819    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
820    if (playbackThread != NULL) {
821        return playbackThread->getRenderPosition(halFrames, dspFrames);
822    }
823
824    return BAD_VALUE;
825}
826
827void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
828{
829
830    Mutex::Autolock _l(mLock);
831
832    int pid = IPCThreadState::self()->getCallingPid();
833    if (mNotificationClients.indexOfKey(pid) < 0) {
834        sp<NotificationClient> notificationClient = new NotificationClient(this,
835                                                                            client,
836                                                                            pid);
837        LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
838
839        mNotificationClients.add(pid, notificationClient);
840
841        sp<IBinder> binder = client->asBinder();
842        binder->linkToDeath(notificationClient);
843
844        // the config change is always sent from playback or record threads to avoid deadlock
845        // with AudioSystem::gLock
846        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
847            mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
848        }
849
850        for (size_t i = 0; i < mRecordThreads.size(); i++) {
851            mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
852        }
853    }
854}
855
856void AudioFlinger::removeNotificationClient(pid_t pid)
857{
858    Mutex::Autolock _l(mLock);
859
860    int index = mNotificationClients.indexOfKey(pid);
861    if (index >= 0) {
862        sp <NotificationClient> client = mNotificationClients.valueFor(pid);
863        LOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
864        mNotificationClients.removeItem(pid);
865    }
866}
867
868// audioConfigChanged_l() must be called with AudioFlinger::mLock held
869void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
870{
871    size_t size = mNotificationClients.size();
872    for (size_t i = 0; i < size; i++) {
873        mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
874    }
875}
876
877// removeClient_l() must be called with AudioFlinger::mLock held
878void AudioFlinger::removeClient_l(pid_t pid)
879{
880    LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
881    mClients.removeItem(pid);
882}
883
884
885// ----------------------------------------------------------------------------
886
887AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
888    :   Thread(false),
889        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
890        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), mDevice(device)
891{
892}
893
894AudioFlinger::ThreadBase::~ThreadBase()
895{
896    mParamCond.broadcast();
897    mNewParameters.clear();
898}
899
900void AudioFlinger::ThreadBase::exit()
901{
902    // keep a strong ref on ourself so that we wont get
903    // destroyed in the middle of requestExitAndWait()
904    sp <ThreadBase> strongMe = this;
905
906    LOGV("ThreadBase::exit");
907    {
908        AutoMutex lock(&mLock);
909        mExiting = true;
910        requestExit();
911        mWaitWorkCV.signal();
912    }
913    requestExitAndWait();
914}
915
916uint32_t AudioFlinger::ThreadBase::sampleRate() const
917{
918    return mSampleRate;
919}
920
921int AudioFlinger::ThreadBase::channelCount() const
922{
923    return (int)mChannelCount;
924}
925
926uint32_t AudioFlinger::ThreadBase::format() const
927{
928    return mFormat;
929}
930
931size_t AudioFlinger::ThreadBase::frameCount() const
932{
933    return mFrameCount;
934}
935
936status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
937{
938    status_t status;
939
940    LOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
941    Mutex::Autolock _l(mLock);
942
943    mNewParameters.add(keyValuePairs);
944    mWaitWorkCV.signal();
945    // wait condition with timeout in case the thread loop has exited
946    // before the request could be processed
947    if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) {
948        status = mParamStatus;
949        mWaitWorkCV.signal();
950    } else {
951        status = TIMED_OUT;
952    }
953    return status;
954}
955
956void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
957{
958    Mutex::Autolock _l(mLock);
959    sendConfigEvent_l(event, param);
960}
961
962// sendConfigEvent_l() must be called with ThreadBase::mLock held
963void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
964{
965    ConfigEvent *configEvent = new ConfigEvent();
966    configEvent->mEvent = event;
967    configEvent->mParam = param;
968    mConfigEvents.add(configEvent);
969    LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
970    mWaitWorkCV.signal();
971}
972
973void AudioFlinger::ThreadBase::processConfigEvents()
974{
975    mLock.lock();
976    while(!mConfigEvents.isEmpty()) {
977        LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
978        ConfigEvent *configEvent = mConfigEvents[0];
979        mConfigEvents.removeAt(0);
980        // release mLock before locking AudioFlinger mLock: lock order is always
981        // AudioFlinger then ThreadBase to avoid cross deadlock
982        mLock.unlock();
983        mAudioFlinger->mLock.lock();
984        audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
985        mAudioFlinger->mLock.unlock();
986        delete configEvent;
987        mLock.lock();
988    }
989    mLock.unlock();
990}
991
992status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
993{
994    const size_t SIZE = 256;
995    char buffer[SIZE];
996    String8 result;
997
998    bool locked = tryLock(mLock);
999    if (!locked) {
1000        snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
1001        write(fd, buffer, strlen(buffer));
1002    }
1003
1004    snprintf(buffer, SIZE, "standby: %d\n", mStandby);
1005    result.append(buffer);
1006    snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
1007    result.append(buffer);
1008    snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
1009    result.append(buffer);
1010    snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
1011    result.append(buffer);
1012    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
1013    result.append(buffer);
1014    snprintf(buffer, SIZE, "Format: %d\n", mFormat);
1015    result.append(buffer);
1016    snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
1017    result.append(buffer);
1018
1019    snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
1020    result.append(buffer);
1021    result.append(" Index Command");
1022    for (size_t i = 0; i < mNewParameters.size(); ++i) {
1023        snprintf(buffer, SIZE, "\n %02d    ", i);
1024        result.append(buffer);
1025        result.append(mNewParameters[i]);
1026    }
1027
1028    snprintf(buffer, SIZE, "\n\nPending config events: \n");
1029    result.append(buffer);
1030    snprintf(buffer, SIZE, " Index event param\n");
1031    result.append(buffer);
1032    for (size_t i = 0; i < mConfigEvents.size(); i++) {
1033        snprintf(buffer, SIZE, " %02d    %02d    %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
1034        result.append(buffer);
1035    }
1036    result.append("\n");
1037
1038    write(fd, result.string(), result.size());
1039
1040    if (locked) {
1041        mLock.unlock();
1042    }
1043    return NO_ERROR;
1044}
1045
1046status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
1047{
1048    const size_t SIZE = 256;
1049    char buffer[SIZE];
1050    String8 result;
1051
1052    snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
1053    write(fd, buffer, strlen(buffer));
1054
1055    for (size_t i = 0; i < mEffectChains.size(); ++i) {
1056        sp<EffectChain> chain = mEffectChains[i];
1057        if (chain != 0) {
1058            chain->dump(fd, args);
1059        }
1060    }
1061    return NO_ERROR;
1062}
1063
1064
1065// ----------------------------------------------------------------------------
1066
1067AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1068                                             AudioStreamOut* output,
1069                                             int id,
1070                                             uint32_t device)
1071    :   ThreadBase(audioFlinger, id, device),
1072        mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
1073        mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
1074{
1075    readOutputParameters();
1076
1077    mMasterVolume = mAudioFlinger->masterVolume();
1078    mMasterMute = mAudioFlinger->masterMute();
1079
1080    for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
1081        mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
1082        mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
1083    }
1084}
1085
1086AudioFlinger::PlaybackThread::~PlaybackThread()
1087{
1088    delete [] mMixBuffer;
1089}
1090
1091status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1092{
1093    dumpInternals(fd, args);
1094    dumpTracks(fd, args);
1095    dumpEffectChains(fd, args);
1096    return NO_ERROR;
1097}
1098
1099status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
1100{
1101    const size_t SIZE = 256;
1102    char buffer[SIZE];
1103    String8 result;
1104
1105    snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1106    result.append(buffer);
1107    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1108    for (size_t i = 0; i < mTracks.size(); ++i) {
1109        sp<Track> track = mTracks[i];
1110        if (track != 0) {
1111            track->dump(buffer, SIZE);
1112            result.append(buffer);
1113        }
1114    }
1115
1116    snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1117    result.append(buffer);
1118    result.append("   Name  Clien Typ Fmt Chn mask   Session Buf  S M F SRate LeftV RighV  Serv       User       Main buf   Aux Buf\n");
1119    for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1120        wp<Track> wTrack = mActiveTracks[i];
1121        if (wTrack != 0) {
1122            sp<Track> track = wTrack.promote();
1123            if (track != 0) {
1124                track->dump(buffer, SIZE);
1125                result.append(buffer);
1126            }
1127        }
1128    }
1129    write(fd, result.string(), result.size());
1130    return NO_ERROR;
1131}
1132
1133status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1134{
1135    const size_t SIZE = 256;
1136    char buffer[SIZE];
1137    String8 result;
1138
1139    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1140    result.append(buffer);
1141    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
1142    result.append(buffer);
1143    snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1144    result.append(buffer);
1145    snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1146    result.append(buffer);
1147    snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1148    result.append(buffer);
1149    snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1150    result.append(buffer);
1151    snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1152    result.append(buffer);
1153    write(fd, result.string(), result.size());
1154
1155    dumpBase(fd, args);
1156
1157    return NO_ERROR;
1158}
1159
1160// Thread virtuals
1161status_t AudioFlinger::PlaybackThread::readyToRun()
1162{
1163    if (mSampleRate == 0) {
1164        LOGE("No working audio driver found.");
1165        return NO_INIT;
1166    }
1167    LOGI("AudioFlinger's thread %p ready to run", this);
1168    return NO_ERROR;
1169}
1170
1171void AudioFlinger::PlaybackThread::onFirstRef()
1172{
1173    const size_t SIZE = 256;
1174    char buffer[SIZE];
1175
1176    snprintf(buffer, SIZE, "Playback Thread %p", this);
1177
1178    run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
1179}
1180
1181// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1182sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
1183        const sp<AudioFlinger::Client>& client,
1184        int streamType,
1185        uint32_t sampleRate,
1186        uint32_t format,
1187        uint32_t channelMask,
1188        int frameCount,
1189        const sp<IMemory>& sharedBuffer,
1190        int sessionId,
1191        status_t *status)
1192{
1193    sp<Track> track;
1194    status_t lStatus;
1195
1196    if (mType == DIRECT) {
1197        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1198            if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1199                LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
1200                        "for output %p with format %d",
1201                        sampleRate, format, channelMask, mOutput, mFormat);
1202                lStatus = BAD_VALUE;
1203                goto Exit;
1204            }
1205        }
1206    } else {
1207        // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1208        if (sampleRate > mSampleRate*2) {
1209            LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
1210            lStatus = BAD_VALUE;
1211            goto Exit;
1212        }
1213    }
1214
1215    lStatus = initCheck();
1216    if (lStatus != NO_ERROR) {
1217        LOGE("Audio driver not initialized.");
1218        goto Exit;
1219    }
1220
1221    { // scope for mLock
1222        Mutex::Autolock _l(mLock);
1223
1224        // all tracks in same audio session must share the same routing strategy otherwise
1225        // conflicts will happen when tracks are moved from one output to another by audio policy
1226        // manager
1227        uint32_t strategy =
1228                AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
1229        for (size_t i = 0; i < mTracks.size(); ++i) {
1230            sp<Track> t = mTracks[i];
1231            if (t != 0) {
1232                if (sessionId == t->sessionId() &&
1233                        strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
1234                    lStatus = BAD_VALUE;
1235                    goto Exit;
1236                }
1237            }
1238        }
1239
1240        track = new Track(this, client, streamType, sampleRate, format,
1241                channelMask, frameCount, sharedBuffer, sessionId);
1242        if (track->getCblk() == NULL || track->name() < 0) {
1243            lStatus = NO_MEMORY;
1244            goto Exit;
1245        }
1246        mTracks.add(track);
1247
1248        sp<EffectChain> chain = getEffectChain_l(sessionId);
1249        if (chain != 0) {
1250            LOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1251            track->setMainBuffer(chain->inBuffer());
1252            chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
1253            chain->incTrackCnt();
1254        }
1255    }
1256    lStatus = NO_ERROR;
1257
1258Exit:
1259    if(status) {
1260        *status = lStatus;
1261    }
1262    return track;
1263}
1264
1265uint32_t AudioFlinger::PlaybackThread::latency() const
1266{
1267    if (mOutput) {
1268        return mOutput->stream->get_latency(mOutput->stream);
1269    }
1270    else {
1271        return 0;
1272    }
1273}
1274
1275status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
1276{
1277    mMasterVolume = value;
1278    return NO_ERROR;
1279}
1280
1281status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1282{
1283    mMasterMute = muted;
1284    return NO_ERROR;
1285}
1286
1287float AudioFlinger::PlaybackThread::masterVolume() const
1288{
1289    return mMasterVolume;
1290}
1291
1292bool AudioFlinger::PlaybackThread::masterMute() const
1293{
1294    return mMasterMute;
1295}
1296
1297status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
1298{
1299    mStreamTypes[stream].volume = value;
1300    return NO_ERROR;
1301}
1302
1303status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
1304{
1305    mStreamTypes[stream].mute = muted;
1306    return NO_ERROR;
1307}
1308
1309float AudioFlinger::PlaybackThread::streamVolume(int stream) const
1310{
1311    return mStreamTypes[stream].volume;
1312}
1313
1314bool AudioFlinger::PlaybackThread::streamMute(int stream) const
1315{
1316    return mStreamTypes[stream].mute;
1317}
1318
1319// addTrack_l() must be called with ThreadBase::mLock held
1320status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1321{
1322    status_t status = ALREADY_EXISTS;
1323
1324    // set retry count for buffer fill
1325    track->mRetryCount = kMaxTrackStartupRetries;
1326    if (mActiveTracks.indexOf(track) < 0) {
1327        // the track is newly added, make sure it fills up all its
1328        // buffers before playing. This is to ensure the client will
1329        // effectively get the latency it requested.
1330        track->mFillingUpStatus = Track::FS_FILLING;
1331        track->mResetDone = false;
1332        mActiveTracks.add(track);
1333        if (track->mainBuffer() != mMixBuffer) {
1334            sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1335            if (chain != 0) {
1336                LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
1337                chain->incActiveTrackCnt();
1338            }
1339        }
1340
1341        status = NO_ERROR;
1342    }
1343
1344    LOGV("mWaitWorkCV.broadcast");
1345    mWaitWorkCV.broadcast();
1346
1347    return status;
1348}
1349
1350// destroyTrack_l() must be called with ThreadBase::mLock held
1351void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1352{
1353    track->mState = TrackBase::TERMINATED;
1354    if (mActiveTracks.indexOf(track) < 0) {
1355        removeTrack_l(track);
1356    }
1357}
1358
1359void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1360{
1361    mTracks.remove(track);
1362    deleteTrackName_l(track->name());
1363    sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1364    if (chain != 0) {
1365        chain->decTrackCnt();
1366    }
1367}
1368
1369String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1370{
1371    String8 out_s8;
1372    char *s;
1373
1374    s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1375    out_s8 = String8(s);
1376    free(s);
1377    return out_s8;
1378}
1379
1380// destroyTrack_l() must be called with AudioFlinger::mLock held
1381void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1382    AudioSystem::OutputDescriptor desc;
1383    void *param2 = 0;
1384
1385    LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
1386
1387    switch (event) {
1388    case AudioSystem::OUTPUT_OPENED:
1389    case AudioSystem::OUTPUT_CONFIG_CHANGED:
1390        desc.channels = mChannelMask;
1391        desc.samplingRate = mSampleRate;
1392        desc.format = mFormat;
1393        desc.frameCount = mFrameCount;
1394        desc.latency = latency();
1395        param2 = &desc;
1396        break;
1397
1398    case AudioSystem::STREAM_CONFIG_CHANGED:
1399        param2 = &param;
1400    case AudioSystem::OUTPUT_CLOSED:
1401    default:
1402        break;
1403    }
1404    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1405}
1406
1407void AudioFlinger::PlaybackThread::readOutputParameters()
1408{
1409    mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1410    mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1411    mChannelCount = (uint16_t)popcount(mChannelMask);
1412    mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1413    mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
1414    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1415
1416    // FIXME - Current mixer implementation only supports stereo output: Always
1417    // Allocate a stereo buffer even if HW output is mono.
1418    if (mMixBuffer != NULL) delete[] mMixBuffer;
1419    mMixBuffer = new int16_t[mFrameCount * 2];
1420    memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
1421
1422    // force reconfiguration of effect chains and engines to take new buffer size and audio
1423    // parameters into account
1424    // Note that mLock is not held when readOutputParameters() is called from the constructor
1425    // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1426    // matter.
1427    // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1428    Vector< sp<EffectChain> > effectChains = mEffectChains;
1429    for (size_t i = 0; i < effectChains.size(); i ++) {
1430        mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1431    }
1432}
1433
1434status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
1435{
1436    if (halFrames == 0 || dspFrames == 0) {
1437        return BAD_VALUE;
1438    }
1439    if (initCheck() != NO_ERROR) {
1440        return INVALID_OPERATION;
1441    }
1442    *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
1443
1444    return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1445}
1446
1447uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
1448{
1449    Mutex::Autolock _l(mLock);
1450    uint32_t result = 0;
1451    if (getEffectChain_l(sessionId) != 0) {
1452        result = EFFECT_SESSION;
1453    }
1454
1455    for (size_t i = 0; i < mTracks.size(); ++i) {
1456        sp<Track> track = mTracks[i];
1457        if (sessionId == track->sessionId() &&
1458                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1459            result |= TRACK_SESSION;
1460            break;
1461        }
1462    }
1463
1464    return result;
1465}
1466
1467uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1468{
1469    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1470    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1471    if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1472        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1473    }
1474    for (size_t i = 0; i < mTracks.size(); i++) {
1475        sp<Track> track = mTracks[i];
1476        if (sessionId == track->sessionId() &&
1477                !(track->mCblk->flags & CBLK_INVALID_MSK)) {
1478            return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
1479        }
1480    }
1481    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1482}
1483
1484
1485// ----------------------------------------------------------------------------
1486
1487AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
1488    :   PlaybackThread(audioFlinger, output, id, device),
1489        mAudioMixer(0)
1490{
1491    mType = ThreadBase::MIXER;
1492    mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
1493
1494    // FIXME - Current mixer implementation only supports stereo output
1495    if (mChannelCount == 1) {
1496        LOGE("Invalid audio hardware channel count");
1497    }
1498}
1499
1500AudioFlinger::MixerThread::~MixerThread()
1501{
1502    delete mAudioMixer;
1503}
1504
1505bool AudioFlinger::MixerThread::threadLoop()
1506{
1507    Vector< sp<Track> > tracksToRemove;
1508    uint32_t mixerStatus = MIXER_IDLE;
1509    nsecs_t standbyTime = systemTime();
1510    size_t mixBufferSize = mFrameCount * mFrameSize;
1511    // FIXME: Relaxed timing because of a certain device that can't meet latency
1512    // Should be reduced to 2x after the vendor fixes the driver issue
1513    nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
1514    nsecs_t lastWarning = 0;
1515    bool longStandbyExit = false;
1516    uint32_t activeSleepTime = activeSleepTimeUs();
1517    uint32_t idleSleepTime = idleSleepTimeUs();
1518    uint32_t sleepTime = idleSleepTime;
1519    Vector< sp<EffectChain> > effectChains;
1520#ifdef DEBUG_CPU_USAGE
1521    ThreadCpuUsage cpu;
1522    const CentralTendencyStatistics& stats = cpu.statistics();
1523#endif
1524
1525    while (!exitPending())
1526    {
1527#ifdef DEBUG_CPU_USAGE
1528        cpu.sampleAndEnable();
1529        unsigned n = stats.n();
1530        // cpu.elapsed() is expensive, so don't call it every loop
1531        if ((n & 127) == 1) {
1532            long long elapsed = cpu.elapsed();
1533            if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
1534                double perLoop = elapsed / (double) n;
1535                double perLoop100 = perLoop * 0.01;
1536                double mean = stats.mean();
1537                double stddev = stats.stddev();
1538                double minimum = stats.minimum();
1539                double maximum = stats.maximum();
1540                cpu.resetStatistics();
1541                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",
1542                        elapsed * .000000001, n, perLoop * .000001,
1543                        mean * .001,
1544                        stddev * .001,
1545                        minimum * .001,
1546                        maximum * .001,
1547                        mean / perLoop100,
1548                        stddev / perLoop100,
1549                        minimum / perLoop100,
1550                        maximum / perLoop100);
1551            }
1552        }
1553#endif
1554        processConfigEvents();
1555
1556        mixerStatus = MIXER_IDLE;
1557        { // scope for mLock
1558
1559            Mutex::Autolock _l(mLock);
1560
1561            if (checkForNewParameters_l()) {
1562                mixBufferSize = mFrameCount * mFrameSize;
1563                // FIXME: Relaxed timing because of a certain device that can't meet latency
1564                // Should be reduced to 2x after the vendor fixes the driver issue
1565                maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
1566                activeSleepTime = activeSleepTimeUs();
1567                idleSleepTime = idleSleepTimeUs();
1568            }
1569
1570            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
1571
1572            // put audio hardware into standby after short delay
1573            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
1574                        mSuspended) {
1575                if (!mStandby) {
1576                    LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
1577                    mOutput->stream->common.standby(&mOutput->stream->common);
1578                    mStandby = true;
1579                    mBytesWritten = 0;
1580                }
1581
1582                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
1583                    // we're about to wait, flush the binder command buffer
1584                    IPCThreadState::self()->flushCommands();
1585
1586                    if (exitPending()) break;
1587
1588                    // wait until we have something to do...
1589                    LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
1590                    mWaitWorkCV.wait(mLock);
1591                    LOGV("MixerThread %p TID %d waking up\n", this, gettid());
1592
1593                    if (mMasterMute == false) {
1594                        char value[PROPERTY_VALUE_MAX];
1595                        property_get("ro.audio.silent", value, "0");
1596                        if (atoi(value)) {
1597                            LOGD("Silence is golden");
1598                            setMasterMute(true);
1599                        }
1600                    }
1601
1602                    standbyTime = systemTime() + kStandbyTimeInNsecs;
1603                    sleepTime = idleSleepTime;
1604                    continue;
1605                }
1606            }
1607
1608            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
1609
1610            // prevent any changes in effect chain list and in each effect chain
1611            // during mixing and effect process as the audio buffers could be deleted
1612            // or modified if an effect is created or deleted
1613            lockEffectChains_l(effectChains);
1614       }
1615
1616        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
1617            // mix buffers...
1618            mAudioMixer->process();
1619            sleepTime = 0;
1620            standbyTime = systemTime() + kStandbyTimeInNsecs;
1621            //TODO: delay standby when effects have a tail
1622        } else {
1623            // If no tracks are ready, sleep once for the duration of an output
1624            // buffer size, then write 0s to the output
1625            if (sleepTime == 0) {
1626                if (mixerStatus == MIXER_TRACKS_ENABLED) {
1627                    sleepTime = activeSleepTime;
1628                } else {
1629                    sleepTime = idleSleepTime;
1630                }
1631            } else if (mBytesWritten != 0 ||
1632                       (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
1633                memset (mMixBuffer, 0, mixBufferSize);
1634                sleepTime = 0;
1635                LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
1636            }
1637            // TODO add standby time extension fct of effect tail
1638        }
1639
1640        if (mSuspended) {
1641            sleepTime = suspendSleepTimeUs();
1642        }
1643        // sleepTime == 0 means we must write to audio hardware
1644        if (sleepTime == 0) {
1645             for (size_t i = 0; i < effectChains.size(); i ++) {
1646                 effectChains[i]->process_l();
1647             }
1648             // enable changes in effect chain
1649             unlockEffectChains(effectChains);
1650            mLastWriteTime = systemTime();
1651            mInWrite = true;
1652            mBytesWritten += mixBufferSize;
1653
1654            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
1655            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
1656            mNumWrites++;
1657            mInWrite = false;
1658            nsecs_t now = systemTime();
1659            nsecs_t delta = now - mLastWriteTime;
1660            if (delta > maxPeriod) {
1661                mNumDelayedWrites++;
1662                if ((now - lastWarning) > kWarningThrottle) {
1663                    LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
1664                            ns2ms(delta), mNumDelayedWrites, this);
1665                    lastWarning = now;
1666                }
1667                if (mStandby) {
1668                    longStandbyExit = true;
1669                }
1670            }
1671            mStandby = false;
1672        } else {
1673            // enable changes in effect chain
1674            unlockEffectChains(effectChains);
1675            usleep(sleepTime);
1676        }
1677
1678        // finally let go of all our tracks, without the lock held
1679        // since we can't guarantee the destructors won't acquire that
1680        // same lock.
1681        tracksToRemove.clear();
1682
1683        // Effect chains will be actually deleted here if they were removed from
1684        // mEffectChains list during mixing or effects processing
1685        effectChains.clear();
1686    }
1687
1688    if (!mStandby) {
1689        mOutput->stream->common.standby(&mOutput->stream->common);
1690    }
1691
1692    LOGV("MixerThread %p exiting", this);
1693    return false;
1694}
1695
1696// prepareTracks_l() must be called with ThreadBase::mLock held
1697uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
1698{
1699
1700    uint32_t mixerStatus = MIXER_IDLE;
1701    // find out which tracks need to be processed
1702    size_t count = activeTracks.size();
1703    size_t mixedTracks = 0;
1704    size_t tracksWithEffect = 0;
1705
1706    float masterVolume = mMasterVolume;
1707    bool  masterMute = mMasterMute;
1708
1709    if (masterMute) {
1710        masterVolume = 0;
1711    }
1712    // Delegate master volume control to effect in output mix effect chain if needed
1713    sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
1714    if (chain != 0) {
1715        uint32_t v = (uint32_t)(masterVolume * (1 << 24));
1716        chain->setVolume_l(&v, &v);
1717        masterVolume = (float)((v + (1 << 23)) >> 24);
1718        chain.clear();
1719    }
1720
1721    for (size_t i=0 ; i<count ; i++) {
1722        sp<Track> t = activeTracks[i].promote();
1723        if (t == 0) continue;
1724
1725        Track* const track = t.get();
1726        audio_track_cblk_t* cblk = track->cblk();
1727
1728        // The first time a track is added we wait
1729        // for all its buffers to be filled before processing it
1730        mAudioMixer->setActiveTrack(track->name());
1731        if (cblk->framesReady() && track->isReady() &&
1732                !track->isPaused() && !track->isTerminated())
1733        {
1734            //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
1735
1736            mixedTracks++;
1737
1738            // track->mainBuffer() != mMixBuffer means there is an effect chain
1739            // connected to the track
1740            chain.clear();
1741            if (track->mainBuffer() != mMixBuffer) {
1742                chain = getEffectChain_l(track->sessionId());
1743                // Delegate volume control to effect in track effect chain if needed
1744                if (chain != 0) {
1745                    tracksWithEffect++;
1746                } else {
1747                    LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
1748                            track->name(), track->sessionId());
1749                }
1750            }
1751
1752
1753            int param = AudioMixer::VOLUME;
1754            if (track->mFillingUpStatus == Track::FS_FILLED) {
1755                // no ramp for the first volume setting
1756                track->mFillingUpStatus = Track::FS_ACTIVE;
1757                if (track->mState == TrackBase::RESUMING) {
1758                    track->mState = TrackBase::ACTIVE;
1759                    param = AudioMixer::RAMP_VOLUME;
1760                }
1761                mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
1762            } else if (cblk->server != 0) {
1763                // If the track is stopped before the first frame was mixed,
1764                // do not apply ramp
1765                param = AudioMixer::RAMP_VOLUME;
1766            }
1767
1768            // compute volume for this track
1769            uint32_t vl, vr, va;
1770            if (track->isMuted() || track->isPausing() ||
1771                mStreamTypes[track->type()].mute) {
1772                vl = vr = va = 0;
1773                if (track->isPausing()) {
1774                    track->setPaused();
1775                }
1776            } else {
1777
1778                // read original volumes with volume control
1779                float typeVolume = mStreamTypes[track->type()].volume;
1780                float v = masterVolume * typeVolume;
1781                vl = (uint32_t)(v * cblk->volume[0]) << 12;
1782                vr = (uint32_t)(v * cblk->volume[1]) << 12;
1783
1784                va = (uint32_t)(v * cblk->sendLevel);
1785            }
1786            // Delegate volume control to effect in track effect chain if needed
1787            if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
1788                // Do not ramp volume if volume is controlled by effect
1789                param = AudioMixer::VOLUME;
1790                track->mHasVolumeController = true;
1791            } else {
1792                // force no volume ramp when volume controller was just disabled or removed
1793                // from effect chain to avoid volume spike
1794                if (track->mHasVolumeController) {
1795                    param = AudioMixer::VOLUME;
1796                }
1797                track->mHasVolumeController = false;
1798            }
1799
1800            // Convert volumes from 8.24 to 4.12 format
1801            int16_t left, right, aux;
1802            uint32_t v_clamped = (vl + (1 << 11)) >> 12;
1803            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
1804            left = int16_t(v_clamped);
1805            v_clamped = (vr + (1 << 11)) >> 12;
1806            if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
1807            right = int16_t(v_clamped);
1808
1809            if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
1810            aux = int16_t(va);
1811
1812            // XXX: these things DON'T need to be done each time
1813            mAudioMixer->setBufferProvider(track);
1814            mAudioMixer->enable(AudioMixer::MIXING);
1815
1816            mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
1817            mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
1818            mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
1819            mAudioMixer->setParameter(
1820                AudioMixer::TRACK,
1821                AudioMixer::FORMAT, (void *)track->format());
1822            mAudioMixer->setParameter(
1823                AudioMixer::TRACK,
1824                AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
1825            mAudioMixer->setParameter(
1826                AudioMixer::RESAMPLE,
1827                AudioMixer::SAMPLE_RATE,
1828                (void *)(cblk->sampleRate));
1829            mAudioMixer->setParameter(
1830                AudioMixer::TRACK,
1831                AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
1832            mAudioMixer->setParameter(
1833                AudioMixer::TRACK,
1834                AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
1835
1836            // reset retry count
1837            track->mRetryCount = kMaxTrackRetries;
1838            mixerStatus = MIXER_TRACKS_READY;
1839        } else {
1840            //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
1841            if (track->isStopped()) {
1842                track->reset();
1843            }
1844            if (track->isTerminated() || track->isStopped() || track->isPaused()) {
1845                // We have consumed all the buffers of this track.
1846                // Remove it from the list of active tracks.
1847                tracksToRemove->add(track);
1848            } else {
1849                // No buffers for this track. Give it a few chances to
1850                // fill a buffer, then remove it from active list.
1851                if (--(track->mRetryCount) <= 0) {
1852                    LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
1853                    tracksToRemove->add(track);
1854                    // indicate to client process that the track was disabled because of underrun
1855                    android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
1856                } else if (mixerStatus != MIXER_TRACKS_READY) {
1857                    mixerStatus = MIXER_TRACKS_ENABLED;
1858                }
1859            }
1860            mAudioMixer->disable(AudioMixer::MIXING);
1861        }
1862    }
1863
1864    // remove all the tracks that need to be...
1865    count = tracksToRemove->size();
1866    if (UNLIKELY(count)) {
1867        for (size_t i=0 ; i<count ; i++) {
1868            const sp<Track>& track = tracksToRemove->itemAt(i);
1869            mActiveTracks.remove(track);
1870            if (track->mainBuffer() != mMixBuffer) {
1871                chain = getEffectChain_l(track->sessionId());
1872                if (chain != 0) {
1873                    LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
1874                    chain->decActiveTrackCnt();
1875                }
1876            }
1877            if (track->isTerminated()) {
1878                removeTrack_l(track);
1879            }
1880        }
1881    }
1882
1883    // mix buffer must be cleared if all tracks are connected to an
1884    // effect chain as in this case the mixer will not write to
1885    // mix buffer and track effects will accumulate into it
1886    if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
1887        memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
1888    }
1889
1890    return mixerStatus;
1891}
1892
1893void AudioFlinger::MixerThread::invalidateTracks(int streamType)
1894{
1895    LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
1896            this,  streamType, mTracks.size());
1897    Mutex::Autolock _l(mLock);
1898
1899    size_t size = mTracks.size();
1900    for (size_t i = 0; i < size; i++) {
1901        sp<Track> t = mTracks[i];
1902        if (t->type() == streamType) {
1903            android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
1904            t->mCblk->cv.signal();
1905        }
1906    }
1907}
1908
1909
1910// getTrackName_l() must be called with ThreadBase::mLock held
1911int AudioFlinger::MixerThread::getTrackName_l()
1912{
1913    return mAudioMixer->getTrackName();
1914}
1915
1916// deleteTrackName_l() must be called with ThreadBase::mLock held
1917void AudioFlinger::MixerThread::deleteTrackName_l(int name)
1918{
1919    LOGV("remove track (%d) and delete from mixer", name);
1920    mAudioMixer->deleteTrackName(name);
1921}
1922
1923// checkForNewParameters_l() must be called with ThreadBase::mLock held
1924bool AudioFlinger::MixerThread::checkForNewParameters_l()
1925{
1926    bool reconfig = false;
1927
1928    while (!mNewParameters.isEmpty()) {
1929        status_t status = NO_ERROR;
1930        String8 keyValuePair = mNewParameters[0];
1931        AudioParameter param = AudioParameter(keyValuePair);
1932        int value;
1933
1934        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
1935            reconfig = true;
1936        }
1937        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
1938            if (value != AUDIO_FORMAT_PCM_16_BIT) {
1939                status = BAD_VALUE;
1940            } else {
1941                reconfig = true;
1942            }
1943        }
1944        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
1945            if (value != AUDIO_CHANNEL_OUT_STEREO) {
1946                status = BAD_VALUE;
1947            } else {
1948                reconfig = true;
1949            }
1950        }
1951        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
1952            // do not accept frame count changes if tracks are open as the track buffer
1953            // size depends on frame count and correct behavior would not be garantied
1954            // if frame count is changed after track creation
1955            if (!mTracks.isEmpty()) {
1956                status = INVALID_OPERATION;
1957            } else {
1958                reconfig = true;
1959            }
1960        }
1961        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
1962            // when changing the audio output device, call addBatteryData to notify
1963            // the change
1964            if ((int)mDevice != value) {
1965                uint32_t params = 0;
1966                // check whether speaker is on
1967                if (value & AUDIO_DEVICE_OUT_SPEAKER) {
1968                    params |= IMediaPlayerService::kBatteryDataSpeakerOn;
1969                }
1970
1971                int deviceWithoutSpeaker
1972                    = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
1973                // check if any other device (except speaker) is on
1974                if (value & deviceWithoutSpeaker ) {
1975                    params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
1976                }
1977
1978                if (params != 0) {
1979                    addBatteryData(params);
1980                }
1981            }
1982
1983            // forward device change to effects that have requested to be
1984            // aware of attached audio device.
1985            mDevice = (uint32_t)value;
1986            for (size_t i = 0; i < mEffectChains.size(); i++) {
1987                mEffectChains[i]->setDevice_l(mDevice);
1988            }
1989        }
1990
1991        if (status == NO_ERROR) {
1992            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
1993                                                    keyValuePair.string());
1994            if (!mStandby && status == INVALID_OPERATION) {
1995               mOutput->stream->common.standby(&mOutput->stream->common);
1996               mStandby = true;
1997               mBytesWritten = 0;
1998               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
1999                                                       keyValuePair.string());
2000            }
2001            if (status == NO_ERROR && reconfig) {
2002                delete mAudioMixer;
2003                readOutputParameters();
2004                mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
2005                for (size_t i = 0; i < mTracks.size() ; i++) {
2006                    int name = getTrackName_l();
2007                    if (name < 0) break;
2008                    mTracks[i]->mName = name;
2009                    // limit track sample rate to 2 x new output sample rate
2010                    if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
2011                        mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
2012                    }
2013                }
2014                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2015            }
2016        }
2017
2018        mNewParameters.removeAt(0);
2019
2020        mParamStatus = status;
2021        mParamCond.signal();
2022        mWaitWorkCV.wait(mLock);
2023    }
2024    return reconfig;
2025}
2026
2027status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
2028{
2029    const size_t SIZE = 256;
2030    char buffer[SIZE];
2031    String8 result;
2032
2033    PlaybackThread::dumpInternals(fd, args);
2034
2035    snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
2036    result.append(buffer);
2037    write(fd, result.string(), result.size());
2038    return NO_ERROR;
2039}
2040
2041uint32_t AudioFlinger::MixerThread::activeSleepTimeUs()
2042{
2043    return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2044}
2045
2046uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
2047{
2048    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2049}
2050
2051uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
2052{
2053    return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2054}
2055
2056// ----------------------------------------------------------------------------
2057AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
2058    :   PlaybackThread(audioFlinger, output, id, device)
2059{
2060    mType = ThreadBase::DIRECT;
2061}
2062
2063AudioFlinger::DirectOutputThread::~DirectOutputThread()
2064{
2065}
2066
2067
2068static inline int16_t clamp16(int32_t sample)
2069{
2070    if ((sample>>15) ^ (sample>>31))
2071        sample = 0x7FFF ^ (sample>>31);
2072    return sample;
2073}
2074
2075static inline
2076int32_t mul(int16_t in, int16_t v)
2077{
2078#if defined(__arm__) && !defined(__thumb__)
2079    int32_t out;
2080    asm( "smulbb %[out], %[in], %[v] \n"
2081         : [out]"=r"(out)
2082         : [in]"%r"(in), [v]"r"(v)
2083         : );
2084    return out;
2085#else
2086    return in * int32_t(v);
2087#endif
2088}
2089
2090void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
2091{
2092    // Do not apply volume on compressed audio
2093    if (!audio_is_linear_pcm(mFormat)) {
2094        return;
2095    }
2096
2097    // convert to signed 16 bit before volume calculation
2098    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2099        size_t count = mFrameCount * mChannelCount;
2100        uint8_t *src = (uint8_t *)mMixBuffer + count-1;
2101        int16_t *dst = mMixBuffer + count-1;
2102        while(count--) {
2103            *dst-- = (int16_t)(*src--^0x80) << 8;
2104        }
2105    }
2106
2107    size_t frameCount = mFrameCount;
2108    int16_t *out = mMixBuffer;
2109    if (ramp) {
2110        if (mChannelCount == 1) {
2111            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2112            int32_t vlInc = d / (int32_t)frameCount;
2113            int32_t vl = ((int32_t)mLeftVolShort << 16);
2114            do {
2115                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2116                out++;
2117                vl += vlInc;
2118            } while (--frameCount);
2119
2120        } else {
2121            int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
2122            int32_t vlInc = d / (int32_t)frameCount;
2123            d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
2124            int32_t vrInc = d / (int32_t)frameCount;
2125            int32_t vl = ((int32_t)mLeftVolShort << 16);
2126            int32_t vr = ((int32_t)mRightVolShort << 16);
2127            do {
2128                out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
2129                out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
2130                out += 2;
2131                vl += vlInc;
2132                vr += vrInc;
2133            } while (--frameCount);
2134        }
2135    } else {
2136        if (mChannelCount == 1) {
2137            do {
2138                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2139                out++;
2140            } while (--frameCount);
2141        } else {
2142            do {
2143                out[0] = clamp16(mul(out[0], leftVol) >> 12);
2144                out[1] = clamp16(mul(out[1], rightVol) >> 12);
2145                out += 2;
2146            } while (--frameCount);
2147        }
2148    }
2149
2150    // convert back to unsigned 8 bit after volume calculation
2151    if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
2152        size_t count = mFrameCount * mChannelCount;
2153        int16_t *src = mMixBuffer;
2154        uint8_t *dst = (uint8_t *)mMixBuffer;
2155        while(count--) {
2156            *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
2157        }
2158    }
2159
2160    mLeftVolShort = leftVol;
2161    mRightVolShort = rightVol;
2162}
2163
2164bool AudioFlinger::DirectOutputThread::threadLoop()
2165{
2166    uint32_t mixerStatus = MIXER_IDLE;
2167    sp<Track> trackToRemove;
2168    sp<Track> activeTrack;
2169    nsecs_t standbyTime = systemTime();
2170    int8_t *curBuf;
2171    size_t mixBufferSize = mFrameCount*mFrameSize;
2172    uint32_t activeSleepTime = activeSleepTimeUs();
2173    uint32_t idleSleepTime = idleSleepTimeUs();
2174    uint32_t sleepTime = idleSleepTime;
2175    // use shorter standby delay as on normal output to release
2176    // hardware resources as soon as possible
2177    nsecs_t standbyDelay = microseconds(activeSleepTime*2);
2178
2179    while (!exitPending())
2180    {
2181        bool rampVolume;
2182        uint16_t leftVol;
2183        uint16_t rightVol;
2184        Vector< sp<EffectChain> > effectChains;
2185
2186        processConfigEvents();
2187
2188        mixerStatus = MIXER_IDLE;
2189
2190        { // scope for the mLock
2191
2192            Mutex::Autolock _l(mLock);
2193
2194            if (checkForNewParameters_l()) {
2195                mixBufferSize = mFrameCount*mFrameSize;
2196                activeSleepTime = activeSleepTimeUs();
2197                idleSleepTime = idleSleepTimeUs();
2198                standbyDelay = microseconds(activeSleepTime*2);
2199            }
2200
2201            // put audio hardware into standby after short delay
2202            if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
2203                        mSuspended) {
2204                // wait until we have something to do...
2205                if (!mStandby) {
2206                    LOGV("Audio hardware entering standby, mixer %p\n", this);
2207                    mOutput->stream->common.standby(&mOutput->stream->common);
2208                    mStandby = true;
2209                    mBytesWritten = 0;
2210                }
2211
2212                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2213                    // we're about to wait, flush the binder command buffer
2214                    IPCThreadState::self()->flushCommands();
2215
2216                    if (exitPending()) break;
2217
2218                    LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
2219                    mWaitWorkCV.wait(mLock);
2220                    LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
2221
2222                    if (mMasterMute == false) {
2223                        char value[PROPERTY_VALUE_MAX];
2224                        property_get("ro.audio.silent", value, "0");
2225                        if (atoi(value)) {
2226                            LOGD("Silence is golden");
2227                            setMasterMute(true);
2228                        }
2229                    }
2230
2231                    standbyTime = systemTime() + standbyDelay;
2232                    sleepTime = idleSleepTime;
2233                    continue;
2234                }
2235            }
2236
2237            effectChains = mEffectChains;
2238
2239            // find out which tracks need to be processed
2240            if (mActiveTracks.size() != 0) {
2241                sp<Track> t = mActiveTracks[0].promote();
2242                if (t == 0) continue;
2243
2244                Track* const track = t.get();
2245                audio_track_cblk_t* cblk = track->cblk();
2246
2247                // The first time a track is added we wait
2248                // for all its buffers to be filled before processing it
2249                if (cblk->framesReady() && track->isReady() &&
2250                        !track->isPaused() && !track->isTerminated())
2251                {
2252                    //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
2253
2254                    if (track->mFillingUpStatus == Track::FS_FILLED) {
2255                        track->mFillingUpStatus = Track::FS_ACTIVE;
2256                        mLeftVolFloat = mRightVolFloat = 0;
2257                        mLeftVolShort = mRightVolShort = 0;
2258                        if (track->mState == TrackBase::RESUMING) {
2259                            track->mState = TrackBase::ACTIVE;
2260                            rampVolume = true;
2261                        }
2262                    } else if (cblk->server != 0) {
2263                        // If the track is stopped before the first frame was mixed,
2264                        // do not apply ramp
2265                        rampVolume = true;
2266                    }
2267                    // compute volume for this track
2268                    float left, right;
2269                    if (track->isMuted() || mMasterMute || track->isPausing() ||
2270                        mStreamTypes[track->type()].mute) {
2271                        left = right = 0;
2272                        if (track->isPausing()) {
2273                            track->setPaused();
2274                        }
2275                    } else {
2276                        float typeVolume = mStreamTypes[track->type()].volume;
2277                        float v = mMasterVolume * typeVolume;
2278                        float v_clamped = v * cblk->volume[0];
2279                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2280                        left = v_clamped/MAX_GAIN;
2281                        v_clamped = v * cblk->volume[1];
2282                        if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
2283                        right = v_clamped/MAX_GAIN;
2284                    }
2285
2286                    if (left != mLeftVolFloat || right != mRightVolFloat) {
2287                        mLeftVolFloat = left;
2288                        mRightVolFloat = right;
2289
2290                        // If audio HAL implements volume control,
2291                        // force software volume to nominal value
2292                        if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
2293                            left = 1.0f;
2294                            right = 1.0f;
2295                        }
2296
2297                        // Convert volumes from float to 8.24
2298                        uint32_t vl = (uint32_t)(left * (1 << 24));
2299                        uint32_t vr = (uint32_t)(right * (1 << 24));
2300
2301                        // Delegate volume control to effect in track effect chain if needed
2302                        // only one effect chain can be present on DirectOutputThread, so if
2303                        // there is one, the track is connected to it
2304                        if (!effectChains.isEmpty()) {
2305                            // Do not ramp volume if volume is controlled by effect
2306                            if(effectChains[0]->setVolume_l(&vl, &vr)) {
2307                                rampVolume = false;
2308                            }
2309                        }
2310
2311                        // Convert volumes from 8.24 to 4.12 format
2312                        uint32_t v_clamped = (vl + (1 << 11)) >> 12;
2313                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2314                        leftVol = (uint16_t)v_clamped;
2315                        v_clamped = (vr + (1 << 11)) >> 12;
2316                        if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
2317                        rightVol = (uint16_t)v_clamped;
2318                    } else {
2319                        leftVol = mLeftVolShort;
2320                        rightVol = mRightVolShort;
2321                        rampVolume = false;
2322                    }
2323
2324                    // reset retry count
2325                    track->mRetryCount = kMaxTrackRetriesDirect;
2326                    activeTrack = t;
2327                    mixerStatus = MIXER_TRACKS_READY;
2328                } else {
2329                    //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
2330                    if (track->isStopped()) {
2331                        track->reset();
2332                    }
2333                    if (track->isTerminated() || track->isStopped() || track->isPaused()) {
2334                        // We have consumed all the buffers of this track.
2335                        // Remove it from the list of active tracks.
2336                        trackToRemove = track;
2337                    } else {
2338                        // No buffers for this track. Give it a few chances to
2339                        // fill a buffer, then remove it from active list.
2340                        if (--(track->mRetryCount) <= 0) {
2341                            LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
2342                            trackToRemove = track;
2343                        } else {
2344                            mixerStatus = MIXER_TRACKS_ENABLED;
2345                        }
2346                    }
2347                }
2348            }
2349
2350            // remove all the tracks that need to be...
2351            if (UNLIKELY(trackToRemove != 0)) {
2352                mActiveTracks.remove(trackToRemove);
2353                if (!effectChains.isEmpty()) {
2354                    LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
2355                            trackToRemove->sessionId());
2356                    effectChains[0]->decActiveTrackCnt();
2357                }
2358                if (trackToRemove->isTerminated()) {
2359                    removeTrack_l(trackToRemove);
2360                }
2361            }
2362
2363            lockEffectChains_l(effectChains);
2364       }
2365
2366        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2367            AudioBufferProvider::Buffer buffer;
2368            size_t frameCount = mFrameCount;
2369            curBuf = (int8_t *)mMixBuffer;
2370            // output audio to hardware
2371            while (frameCount) {
2372                buffer.frameCount = frameCount;
2373                activeTrack->getNextBuffer(&buffer);
2374                if (UNLIKELY(buffer.raw == 0)) {
2375                    memset(curBuf, 0, frameCount * mFrameSize);
2376                    break;
2377                }
2378                memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
2379                frameCount -= buffer.frameCount;
2380                curBuf += buffer.frameCount * mFrameSize;
2381                activeTrack->releaseBuffer(&buffer);
2382            }
2383            sleepTime = 0;
2384            standbyTime = systemTime() + standbyDelay;
2385        } else {
2386            if (sleepTime == 0) {
2387                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2388                    sleepTime = activeSleepTime;
2389                } else {
2390                    sleepTime = idleSleepTime;
2391                }
2392            } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
2393                memset (mMixBuffer, 0, mFrameCount * mFrameSize);
2394                sleepTime = 0;
2395            }
2396        }
2397
2398        if (mSuspended) {
2399            sleepTime = suspendSleepTimeUs();
2400        }
2401        // sleepTime == 0 means we must write to audio hardware
2402        if (sleepTime == 0) {
2403            if (mixerStatus == MIXER_TRACKS_READY) {
2404                applyVolume(leftVol, rightVol, rampVolume);
2405            }
2406            for (size_t i = 0; i < effectChains.size(); i ++) {
2407                effectChains[i]->process_l();
2408            }
2409            unlockEffectChains(effectChains);
2410
2411            mLastWriteTime = systemTime();
2412            mInWrite = true;
2413            mBytesWritten += mixBufferSize;
2414            int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
2415            if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
2416            mNumWrites++;
2417            mInWrite = false;
2418            mStandby = false;
2419        } else {
2420            unlockEffectChains(effectChains);
2421            usleep(sleepTime);
2422        }
2423
2424        // finally let go of removed track, without the lock held
2425        // since we can't guarantee the destructors won't acquire that
2426        // same lock.
2427        trackToRemove.clear();
2428        activeTrack.clear();
2429
2430        // Effect chains will be actually deleted here if they were removed from
2431        // mEffectChains list during mixing or effects processing
2432        effectChains.clear();
2433    }
2434
2435    if (!mStandby) {
2436        mOutput->stream->common.standby(&mOutput->stream->common);
2437    }
2438
2439    LOGV("DirectOutputThread %p exiting", this);
2440    return false;
2441}
2442
2443// getTrackName_l() must be called with ThreadBase::mLock held
2444int AudioFlinger::DirectOutputThread::getTrackName_l()
2445{
2446    return 0;
2447}
2448
2449// deleteTrackName_l() must be called with ThreadBase::mLock held
2450void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
2451{
2452}
2453
2454// checkForNewParameters_l() must be called with ThreadBase::mLock held
2455bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
2456{
2457    bool reconfig = false;
2458
2459    while (!mNewParameters.isEmpty()) {
2460        status_t status = NO_ERROR;
2461        String8 keyValuePair = mNewParameters[0];
2462        AudioParameter param = AudioParameter(keyValuePair);
2463        int value;
2464
2465        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2466            // do not accept frame count changes if tracks are open as the track buffer
2467            // size depends on frame count and correct behavior would not be garantied
2468            // if frame count is changed after track creation
2469            if (!mTracks.isEmpty()) {
2470                status = INVALID_OPERATION;
2471            } else {
2472                reconfig = true;
2473            }
2474        }
2475        if (status == NO_ERROR) {
2476            status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2477                                                    keyValuePair.string());
2478            if (!mStandby && status == INVALID_OPERATION) {
2479               mOutput->stream->common.standby(&mOutput->stream->common);
2480               mStandby = true;
2481               mBytesWritten = 0;
2482               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
2483                                                       keyValuePair.string());
2484            }
2485            if (status == NO_ERROR && reconfig) {
2486                readOutputParameters();
2487                sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
2488            }
2489        }
2490
2491        mNewParameters.removeAt(0);
2492
2493        mParamStatus = status;
2494        mParamCond.signal();
2495        mWaitWorkCV.wait(mLock);
2496    }
2497    return reconfig;
2498}
2499
2500uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
2501{
2502    uint32_t time;
2503    if (audio_is_linear_pcm(mFormat)) {
2504        time = (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
2505    } else {
2506        time = 10000;
2507    }
2508    return time;
2509}
2510
2511uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
2512{
2513    uint32_t time;
2514    if (audio_is_linear_pcm(mFormat)) {
2515        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
2516    } else {
2517        time = 10000;
2518    }
2519    return time;
2520}
2521
2522uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
2523{
2524    uint32_t time;
2525    if (audio_is_linear_pcm(mFormat)) {
2526        time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
2527    } else {
2528        time = 10000;
2529    }
2530    return time;
2531}
2532
2533
2534// ----------------------------------------------------------------------------
2535
2536AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
2537    :   MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
2538{
2539    mType = ThreadBase::DUPLICATING;
2540    addOutputTrack(mainThread);
2541}
2542
2543AudioFlinger::DuplicatingThread::~DuplicatingThread()
2544{
2545    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2546        mOutputTracks[i]->destroy();
2547    }
2548    mOutputTracks.clear();
2549}
2550
2551bool AudioFlinger::DuplicatingThread::threadLoop()
2552{
2553    Vector< sp<Track> > tracksToRemove;
2554    uint32_t mixerStatus = MIXER_IDLE;
2555    nsecs_t standbyTime = systemTime();
2556    size_t mixBufferSize = mFrameCount*mFrameSize;
2557    SortedVector< sp<OutputTrack> > outputTracks;
2558    uint32_t writeFrames = 0;
2559    uint32_t activeSleepTime = activeSleepTimeUs();
2560    uint32_t idleSleepTime = idleSleepTimeUs();
2561    uint32_t sleepTime = idleSleepTime;
2562    Vector< sp<EffectChain> > effectChains;
2563
2564    while (!exitPending())
2565    {
2566        processConfigEvents();
2567
2568        mixerStatus = MIXER_IDLE;
2569        { // scope for the mLock
2570
2571            Mutex::Autolock _l(mLock);
2572
2573            if (checkForNewParameters_l()) {
2574                mixBufferSize = mFrameCount*mFrameSize;
2575                updateWaitTime();
2576                activeSleepTime = activeSleepTimeUs();
2577                idleSleepTime = idleSleepTimeUs();
2578            }
2579
2580            const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
2581
2582            for (size_t i = 0; i < mOutputTracks.size(); i++) {
2583                outputTracks.add(mOutputTracks[i]);
2584            }
2585
2586            // put audio hardware into standby after short delay
2587            if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
2588                         mSuspended) {
2589                if (!mStandby) {
2590                    for (size_t i = 0; i < outputTracks.size(); i++) {
2591                        outputTracks[i]->stop();
2592                    }
2593                    mStandby = true;
2594                    mBytesWritten = 0;
2595                }
2596
2597                if (!activeTracks.size() && mConfigEvents.isEmpty()) {
2598                    // we're about to wait, flush the binder command buffer
2599                    IPCThreadState::self()->flushCommands();
2600                    outputTracks.clear();
2601
2602                    if (exitPending()) break;
2603
2604                    LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
2605                    mWaitWorkCV.wait(mLock);
2606                    LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
2607                    if (mMasterMute == false) {
2608                        char value[PROPERTY_VALUE_MAX];
2609                        property_get("ro.audio.silent", value, "0");
2610                        if (atoi(value)) {
2611                            LOGD("Silence is golden");
2612                            setMasterMute(true);
2613                        }
2614                    }
2615
2616                    standbyTime = systemTime() + kStandbyTimeInNsecs;
2617                    sleepTime = idleSleepTime;
2618                    continue;
2619                }
2620            }
2621
2622            mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
2623
2624            // prevent any changes in effect chain list and in each effect chain
2625            // during mixing and effect process as the audio buffers could be deleted
2626            // or modified if an effect is created or deleted
2627            lockEffectChains_l(effectChains);
2628        }
2629
2630        if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
2631            // mix buffers...
2632            if (outputsReady(outputTracks)) {
2633                mAudioMixer->process();
2634            } else {
2635                memset(mMixBuffer, 0, mixBufferSize);
2636            }
2637            sleepTime = 0;
2638            writeFrames = mFrameCount;
2639        } else {
2640            if (sleepTime == 0) {
2641                if (mixerStatus == MIXER_TRACKS_ENABLED) {
2642                    sleepTime = activeSleepTime;
2643                } else {
2644                    sleepTime = idleSleepTime;
2645                }
2646            } else if (mBytesWritten != 0) {
2647                // flush remaining overflow buffers in output tracks
2648                for (size_t i = 0; i < outputTracks.size(); i++) {
2649                    if (outputTracks[i]->isActive()) {
2650                        sleepTime = 0;
2651                        writeFrames = 0;
2652                        memset(mMixBuffer, 0, mixBufferSize);
2653                        break;
2654                    }
2655                }
2656            }
2657        }
2658
2659        if (mSuspended) {
2660            sleepTime = suspendSleepTimeUs();
2661        }
2662        // sleepTime == 0 means we must write to audio hardware
2663        if (sleepTime == 0) {
2664            for (size_t i = 0; i < effectChains.size(); i ++) {
2665                effectChains[i]->process_l();
2666            }
2667            // enable changes in effect chain
2668            unlockEffectChains(effectChains);
2669
2670            standbyTime = systemTime() + kStandbyTimeInNsecs;
2671            for (size_t i = 0; i < outputTracks.size(); i++) {
2672                outputTracks[i]->write(mMixBuffer, writeFrames);
2673            }
2674            mStandby = false;
2675            mBytesWritten += mixBufferSize;
2676        } else {
2677            // enable changes in effect chain
2678            unlockEffectChains(effectChains);
2679            usleep(sleepTime);
2680        }
2681
2682        // finally let go of all our tracks, without the lock held
2683        // since we can't guarantee the destructors won't acquire that
2684        // same lock.
2685        tracksToRemove.clear();
2686        outputTracks.clear();
2687
2688        // Effect chains will be actually deleted here if they were removed from
2689        // mEffectChains list during mixing or effects processing
2690        effectChains.clear();
2691    }
2692
2693    return false;
2694}
2695
2696void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
2697{
2698    int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
2699    OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
2700                                            this,
2701                                            mSampleRate,
2702                                            mFormat,
2703                                            mChannelMask,
2704                                            frameCount);
2705    if (outputTrack->cblk() != NULL) {
2706        thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
2707        mOutputTracks.add(outputTrack);
2708        LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
2709        updateWaitTime();
2710    }
2711}
2712
2713void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
2714{
2715    Mutex::Autolock _l(mLock);
2716    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2717        if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
2718            mOutputTracks[i]->destroy();
2719            mOutputTracks.removeAt(i);
2720            updateWaitTime();
2721            return;
2722        }
2723    }
2724    LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
2725}
2726
2727void AudioFlinger::DuplicatingThread::updateWaitTime()
2728{
2729    mWaitTimeMs = UINT_MAX;
2730    for (size_t i = 0; i < mOutputTracks.size(); i++) {
2731        sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
2732        if (strong != NULL) {
2733            uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
2734            if (waitTimeMs < mWaitTimeMs) {
2735                mWaitTimeMs = waitTimeMs;
2736            }
2737        }
2738    }
2739}
2740
2741
2742bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
2743{
2744    for (size_t i = 0; i < outputTracks.size(); i++) {
2745        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
2746        if (thread == 0) {
2747            LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
2748            return false;
2749        }
2750        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
2751        if (playbackThread->standby() && !playbackThread->isSuspended()) {
2752            LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
2753            return false;
2754        }
2755    }
2756    return true;
2757}
2758
2759uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
2760{
2761    return (mWaitTimeMs * 1000) / 2;
2762}
2763
2764// ----------------------------------------------------------------------------
2765
2766// TrackBase constructor must be called with AudioFlinger::mLock held
2767AudioFlinger::ThreadBase::TrackBase::TrackBase(
2768            const wp<ThreadBase>& thread,
2769            const sp<Client>& client,
2770            uint32_t sampleRate,
2771            uint32_t format,
2772            uint32_t channelMask,
2773            int frameCount,
2774            uint32_t flags,
2775            const sp<IMemory>& sharedBuffer,
2776            int sessionId)
2777    :   RefBase(),
2778        mThread(thread),
2779        mClient(client),
2780        mCblk(0),
2781        mFrameCount(0),
2782        mState(IDLE),
2783        mClientTid(-1),
2784        mFormat(format),
2785        mFlags(flags & ~SYSTEM_FLAGS_MASK),
2786        mSessionId(sessionId)
2787{
2788    LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
2789
2790    // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
2791   size_t size = sizeof(audio_track_cblk_t);
2792   uint8_t channelCount = popcount(channelMask);
2793   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
2794   if (sharedBuffer == 0) {
2795       size += bufferSize;
2796   }
2797
2798   if (client != NULL) {
2799        mCblkMemory = client->heap()->allocate(size);
2800        if (mCblkMemory != 0) {
2801            mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
2802            if (mCblk) { // construct the shared structure in-place.
2803                new(mCblk) audio_track_cblk_t();
2804                // clear all buffers
2805                mCblk->frameCount = frameCount;
2806                mCblk->sampleRate = sampleRate;
2807                mChannelCount = channelCount;
2808                mChannelMask = channelMask;
2809                if (sharedBuffer == 0) {
2810                    mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
2811                    memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
2812                    // Force underrun condition to avoid false underrun callback until first data is
2813                    // written to buffer (other flags are cleared)
2814                    mCblk->flags = CBLK_UNDERRUN_ON;
2815                } else {
2816                    mBuffer = sharedBuffer->pointer();
2817                }
2818                mBufferEnd = (uint8_t *)mBuffer + bufferSize;
2819            }
2820        } else {
2821            LOGE("not enough memory for AudioTrack size=%u", size);
2822            client->heap()->dump("AudioTrack");
2823            return;
2824        }
2825   } else {
2826       mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
2827       if (mCblk) { // construct the shared structure in-place.
2828           new(mCblk) audio_track_cblk_t();
2829           // clear all buffers
2830           mCblk->frameCount = frameCount;
2831           mCblk->sampleRate = sampleRate;
2832           mChannelCount = channelCount;
2833           mChannelMask = channelMask;
2834           mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
2835           memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
2836           // Force underrun condition to avoid false underrun callback until first data is
2837           // written to buffer (other flags are cleared)
2838           mCblk->flags = CBLK_UNDERRUN_ON;
2839           mBufferEnd = (uint8_t *)mBuffer + bufferSize;
2840       }
2841   }
2842}
2843
2844AudioFlinger::ThreadBase::TrackBase::~TrackBase()
2845{
2846    if (mCblk) {
2847        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
2848        if (mClient == NULL) {
2849            delete mCblk;
2850        }
2851    }
2852    mCblkMemory.clear();            // and free the shared memory
2853    if (mClient != NULL) {
2854        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
2855        mClient.clear();
2856    }
2857}
2858
2859void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
2860{
2861    buffer->raw = 0;
2862    mFrameCount = buffer->frameCount;
2863    step();
2864    buffer->frameCount = 0;
2865}
2866
2867bool AudioFlinger::ThreadBase::TrackBase::step() {
2868    bool result;
2869    audio_track_cblk_t* cblk = this->cblk();
2870
2871    result = cblk->stepServer(mFrameCount);
2872    if (!result) {
2873        LOGV("stepServer failed acquiring cblk mutex");
2874        mFlags |= STEPSERVER_FAILED;
2875    }
2876    return result;
2877}
2878
2879void AudioFlinger::ThreadBase::TrackBase::reset() {
2880    audio_track_cblk_t* cblk = this->cblk();
2881
2882    cblk->user = 0;
2883    cblk->server = 0;
2884    cblk->userBase = 0;
2885    cblk->serverBase = 0;
2886    mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
2887    LOGV("TrackBase::reset");
2888}
2889
2890sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
2891{
2892    return mCblkMemory;
2893}
2894
2895int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
2896    return (int)mCblk->sampleRate;
2897}
2898
2899int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
2900    return (const int)mChannelCount;
2901}
2902
2903uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
2904    return mChannelMask;
2905}
2906
2907void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
2908    audio_track_cblk_t* cblk = this->cblk();
2909    int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
2910    int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
2911
2912    // Check validity of returned pointer in case the track control block would have been corrupted.
2913    if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
2914        ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
2915        LOGE("TrackBase::getBuffer buffer out of range:\n    start: %p, end %p , mBuffer %p mBufferEnd %p\n    \
2916                server %d, serverBase %d, user %d, userBase %d",
2917                bufferStart, bufferEnd, mBuffer, mBufferEnd,
2918                cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
2919        return 0;
2920    }
2921
2922    return bufferStart;
2923}
2924
2925// ----------------------------------------------------------------------------
2926
2927// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
2928AudioFlinger::PlaybackThread::Track::Track(
2929            const wp<ThreadBase>& thread,
2930            const sp<Client>& client,
2931            int streamType,
2932            uint32_t sampleRate,
2933            uint32_t format,
2934            uint32_t channelMask,
2935            int frameCount,
2936            const sp<IMemory>& sharedBuffer,
2937            int sessionId)
2938    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
2939    mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
2940    mAuxEffectId(0), mHasVolumeController(false)
2941{
2942    if (mCblk != NULL) {
2943        sp<ThreadBase> baseThread = thread.promote();
2944        if (baseThread != 0) {
2945            PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
2946            mName = playbackThread->getTrackName_l();
2947            mMainBuffer = playbackThread->mixBuffer();
2948        }
2949        LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
2950        if (mName < 0) {
2951            LOGE("no more track names available");
2952        }
2953        mVolume[0] = 1.0f;
2954        mVolume[1] = 1.0f;
2955        mStreamType = streamType;
2956        // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
2957        // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
2958        mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
2959    }
2960}
2961
2962AudioFlinger::PlaybackThread::Track::~Track()
2963{
2964    LOGV("PlaybackThread::Track destructor");
2965    sp<ThreadBase> thread = mThread.promote();
2966    if (thread != 0) {
2967        Mutex::Autolock _l(thread->mLock);
2968        mState = TERMINATED;
2969    }
2970}
2971
2972void AudioFlinger::PlaybackThread::Track::destroy()
2973{
2974    // NOTE: destroyTrack_l() can remove a strong reference to this Track
2975    // by removing it from mTracks vector, so there is a risk that this Tracks's
2976    // desctructor is called. As the destructor needs to lock mLock,
2977    // we must acquire a strong reference on this Track before locking mLock
2978    // here so that the destructor is called only when exiting this function.
2979    // On the other hand, as long as Track::destroy() is only called by
2980    // TrackHandle destructor, the TrackHandle still holds a strong ref on
2981    // this Track with its member mTrack.
2982    sp<Track> keep(this);
2983    { // scope for mLock
2984        sp<ThreadBase> thread = mThread.promote();
2985        if (thread != 0) {
2986            if (!isOutputTrack()) {
2987                if (mState == ACTIVE || mState == RESUMING) {
2988                    AudioSystem::stopOutput(thread->id(),
2989                                            (audio_stream_type_t)mStreamType,
2990                                            mSessionId);
2991
2992                    // to track the speaker usage
2993                    addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
2994                }
2995                AudioSystem::releaseOutput(thread->id());
2996            }
2997            Mutex::Autolock _l(thread->mLock);
2998            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
2999            playbackThread->destroyTrack_l(this);
3000        }
3001    }
3002}
3003
3004void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
3005{
3006    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",
3007            mName - AudioMixer::TRACK0,
3008            (mClient == NULL) ? getpid() : mClient->pid(),
3009            mStreamType,
3010            mFormat,
3011            mChannelMask,
3012            mSessionId,
3013            mFrameCount,
3014            mState,
3015            mMute,
3016            mFillingUpStatus,
3017            mCblk->sampleRate,
3018            mCblk->volume[0],
3019            mCblk->volume[1],
3020            mCblk->server,
3021            mCblk->user,
3022            (int)mMainBuffer,
3023            (int)mAuxBuffer);
3024}
3025
3026status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3027{
3028     audio_track_cblk_t* cblk = this->cblk();
3029     uint32_t framesReady;
3030     uint32_t framesReq = buffer->frameCount;
3031
3032     // Check if last stepServer failed, try to step now
3033     if (mFlags & TrackBase::STEPSERVER_FAILED) {
3034         if (!step())  goto getNextBuffer_exit;
3035         LOGV("stepServer recovered");
3036         mFlags &= ~TrackBase::STEPSERVER_FAILED;
3037     }
3038
3039     framesReady = cblk->framesReady();
3040
3041     if (LIKELY(framesReady)) {
3042        uint32_t s = cblk->server;
3043        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3044
3045        bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
3046        if (framesReq > framesReady) {
3047            framesReq = framesReady;
3048        }
3049        if (s + framesReq > bufferEnd) {
3050            framesReq = bufferEnd - s;
3051        }
3052
3053         buffer->raw = getBuffer(s, framesReq);
3054         if (buffer->raw == 0) goto getNextBuffer_exit;
3055
3056         buffer->frameCount = framesReq;
3057        return NO_ERROR;
3058     }
3059
3060getNextBuffer_exit:
3061     buffer->raw = 0;
3062     buffer->frameCount = 0;
3063     LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
3064     return NOT_ENOUGH_DATA;
3065}
3066
3067bool AudioFlinger::PlaybackThread::Track::isReady() const {
3068    if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
3069
3070    if (mCblk->framesReady() >= mCblk->frameCount ||
3071            (mCblk->flags & CBLK_FORCEREADY_MSK)) {
3072        mFillingUpStatus = FS_FILLED;
3073        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3074        return true;
3075    }
3076    return false;
3077}
3078
3079status_t AudioFlinger::PlaybackThread::Track::start()
3080{
3081    status_t status = NO_ERROR;
3082    LOGV("start(%d), calling thread %d session %d",
3083            mName, IPCThreadState::self()->getCallingPid(), mSessionId);
3084    sp<ThreadBase> thread = mThread.promote();
3085    if (thread != 0) {
3086        Mutex::Autolock _l(thread->mLock);
3087        int state = mState;
3088        // here the track could be either new, or restarted
3089        // in both cases "unstop" the track
3090        if (mState == PAUSED) {
3091            mState = TrackBase::RESUMING;
3092            LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
3093        } else {
3094            mState = TrackBase::ACTIVE;
3095            LOGV("? => ACTIVE (%d) on thread %p", mName, this);
3096        }
3097
3098        if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
3099            thread->mLock.unlock();
3100            status = AudioSystem::startOutput(thread->id(),
3101                                              (audio_stream_type_t)mStreamType,
3102                                              mSessionId);
3103            thread->mLock.lock();
3104
3105            // to track the speaker usage
3106            if (status == NO_ERROR) {
3107                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
3108            }
3109        }
3110        if (status == NO_ERROR) {
3111            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3112            playbackThread->addTrack_l(this);
3113        } else {
3114            mState = state;
3115        }
3116    } else {
3117        status = BAD_VALUE;
3118    }
3119    return status;
3120}
3121
3122void AudioFlinger::PlaybackThread::Track::stop()
3123{
3124    LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3125    sp<ThreadBase> thread = mThread.promote();
3126    if (thread != 0) {
3127        Mutex::Autolock _l(thread->mLock);
3128        int state = mState;
3129        if (mState > STOPPED) {
3130            mState = STOPPED;
3131            // If the track is not active (PAUSED and buffers full), flush buffers
3132            PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3133            if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3134                reset();
3135            }
3136            LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
3137        }
3138        if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
3139            thread->mLock.unlock();
3140            AudioSystem::stopOutput(thread->id(),
3141                                    (audio_stream_type_t)mStreamType,
3142                                    mSessionId);
3143            thread->mLock.lock();
3144
3145            // to track the speaker usage
3146            addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3147        }
3148    }
3149}
3150
3151void AudioFlinger::PlaybackThread::Track::pause()
3152{
3153    LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
3154    sp<ThreadBase> thread = mThread.promote();
3155    if (thread != 0) {
3156        Mutex::Autolock _l(thread->mLock);
3157        if (mState == ACTIVE || mState == RESUMING) {
3158            mState = PAUSING;
3159            LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
3160            if (!isOutputTrack()) {
3161                thread->mLock.unlock();
3162                AudioSystem::stopOutput(thread->id(),
3163                                        (audio_stream_type_t)mStreamType,
3164                                        mSessionId);
3165                thread->mLock.lock();
3166
3167                // to track the speaker usage
3168                addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
3169            }
3170        }
3171    }
3172}
3173
3174void AudioFlinger::PlaybackThread::Track::flush()
3175{
3176    LOGV("flush(%d)", mName);
3177    sp<ThreadBase> thread = mThread.promote();
3178    if (thread != 0) {
3179        Mutex::Autolock _l(thread->mLock);
3180        if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
3181            return;
3182        }
3183        // No point remaining in PAUSED state after a flush => go to
3184        // STOPPED state
3185        mState = STOPPED;
3186
3187        // do not reset the track if it is still in the process of being stopped or paused.
3188        // this will be done by prepareTracks_l() when the track is stopped.
3189        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3190        if (playbackThread->mActiveTracks.indexOf(this) < 0) {
3191            reset();
3192        }
3193    }
3194}
3195
3196void AudioFlinger::PlaybackThread::Track::reset()
3197{
3198    // Do not reset twice to avoid discarding data written just after a flush and before
3199    // the audioflinger thread detects the track is stopped.
3200    if (!mResetDone) {
3201        TrackBase::reset();
3202        // Force underrun condition to avoid false underrun callback until first data is
3203        // written to buffer
3204        android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
3205        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3206        mFillingUpStatus = FS_FILLING;
3207        mResetDone = true;
3208    }
3209}
3210
3211void AudioFlinger::PlaybackThread::Track::mute(bool muted)
3212{
3213    mMute = muted;
3214}
3215
3216void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
3217{
3218    mVolume[0] = left;
3219    mVolume[1] = right;
3220}
3221
3222status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
3223{
3224    status_t status = DEAD_OBJECT;
3225    sp<ThreadBase> thread = mThread.promote();
3226    if (thread != 0) {
3227       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3228       status = playbackThread->attachAuxEffect(this, EffectId);
3229    }
3230    return status;
3231}
3232
3233void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
3234{
3235    mAuxEffectId = EffectId;
3236    mAuxBuffer = buffer;
3237}
3238
3239// ----------------------------------------------------------------------------
3240
3241// RecordTrack constructor must be called with AudioFlinger::mLock held
3242AudioFlinger::RecordThread::RecordTrack::RecordTrack(
3243            const wp<ThreadBase>& thread,
3244            const sp<Client>& client,
3245            uint32_t sampleRate,
3246            uint32_t format,
3247            uint32_t channelMask,
3248            int frameCount,
3249            uint32_t flags,
3250            int sessionId)
3251    :   TrackBase(thread, client, sampleRate, format,
3252                  channelMask, frameCount, flags, 0, sessionId),
3253        mOverflow(false)
3254{
3255    if (mCblk != NULL) {
3256       LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
3257       if (format == AUDIO_FORMAT_PCM_16_BIT) {
3258           mCblk->frameSize = mChannelCount * sizeof(int16_t);
3259       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
3260           mCblk->frameSize = mChannelCount * sizeof(int8_t);
3261       } else {
3262           mCblk->frameSize = sizeof(int8_t);
3263       }
3264    }
3265}
3266
3267AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
3268{
3269    sp<ThreadBase> thread = mThread.promote();
3270    if (thread != 0) {
3271        AudioSystem::releaseInput(thread->id());
3272    }
3273}
3274
3275status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
3276{
3277    audio_track_cblk_t* cblk = this->cblk();
3278    uint32_t framesAvail;
3279    uint32_t framesReq = buffer->frameCount;
3280
3281     // Check if last stepServer failed, try to step now
3282    if (mFlags & TrackBase::STEPSERVER_FAILED) {
3283        if (!step()) goto getNextBuffer_exit;
3284        LOGV("stepServer recovered");
3285        mFlags &= ~TrackBase::STEPSERVER_FAILED;
3286    }
3287
3288    framesAvail = cblk->framesAvailable_l();
3289
3290    if (LIKELY(framesAvail)) {
3291        uint32_t s = cblk->server;
3292        uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
3293
3294        if (framesReq > framesAvail) {
3295            framesReq = framesAvail;
3296        }
3297        if (s + framesReq > bufferEnd) {
3298            framesReq = bufferEnd - s;
3299        }
3300
3301        buffer->raw = getBuffer(s, framesReq);
3302        if (buffer->raw == 0) goto getNextBuffer_exit;
3303
3304        buffer->frameCount = framesReq;
3305        return NO_ERROR;
3306    }
3307
3308getNextBuffer_exit:
3309    buffer->raw = 0;
3310    buffer->frameCount = 0;
3311    return NOT_ENOUGH_DATA;
3312}
3313
3314status_t AudioFlinger::RecordThread::RecordTrack::start()
3315{
3316    sp<ThreadBase> thread = mThread.promote();
3317    if (thread != 0) {
3318        RecordThread *recordThread = (RecordThread *)thread.get();
3319        return recordThread->start(this);
3320    } else {
3321        return BAD_VALUE;
3322    }
3323}
3324
3325void AudioFlinger::RecordThread::RecordTrack::stop()
3326{
3327    sp<ThreadBase> thread = mThread.promote();
3328    if (thread != 0) {
3329        RecordThread *recordThread = (RecordThread *)thread.get();
3330        recordThread->stop(this);
3331        TrackBase::reset();
3332        // Force overerrun condition to avoid false overrun callback until first data is
3333        // read from buffer
3334        android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
3335    }
3336}
3337
3338void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
3339{
3340    snprintf(buffer, size, "   %05d %03u 0x%08x %05d   %04u %01d %05u  %08x %08x\n",
3341            (mClient == NULL) ? getpid() : mClient->pid(),
3342            mFormat,
3343            mChannelMask,
3344            mSessionId,
3345            mFrameCount,
3346            mState,
3347            mCblk->sampleRate,
3348            mCblk->server,
3349            mCblk->user);
3350}
3351
3352
3353// ----------------------------------------------------------------------------
3354
3355AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
3356            const wp<ThreadBase>& thread,
3357            DuplicatingThread *sourceThread,
3358            uint32_t sampleRate,
3359            uint32_t format,
3360            uint32_t channelMask,
3361            int frameCount)
3362    :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
3363    mActive(false), mSourceThread(sourceThread)
3364{
3365
3366    PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
3367    if (mCblk != NULL) {
3368        mCblk->flags |= CBLK_DIRECTION_OUT;
3369        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
3370        mCblk->volume[0] = mCblk->volume[1] = 0x1000;
3371        mOutBuffer.frameCount = 0;
3372        playbackThread->mTracks.add(this);
3373        LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
3374                "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
3375                mCblk, mBuffer, mCblk->buffers,
3376                mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
3377    } else {
3378        LOGW("Error creating output track on thread %p", playbackThread);
3379    }
3380}
3381
3382AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
3383{
3384    clearBufferQueue();
3385}
3386
3387status_t AudioFlinger::PlaybackThread::OutputTrack::start()
3388{
3389    status_t status = Track::start();
3390    if (status != NO_ERROR) {
3391        return status;
3392    }
3393
3394    mActive = true;
3395    mRetryCount = 127;
3396    return status;
3397}
3398
3399void AudioFlinger::PlaybackThread::OutputTrack::stop()
3400{
3401    Track::stop();
3402    clearBufferQueue();
3403    mOutBuffer.frameCount = 0;
3404    mActive = false;
3405}
3406
3407bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
3408{
3409    Buffer *pInBuffer;
3410    Buffer inBuffer;
3411    uint32_t channelCount = mChannelCount;
3412    bool outputBufferFull = false;
3413    inBuffer.frameCount = frames;
3414    inBuffer.i16 = data;
3415
3416    uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
3417
3418    if (!mActive && frames != 0) {
3419        start();
3420        sp<ThreadBase> thread = mThread.promote();
3421        if (thread != 0) {
3422            MixerThread *mixerThread = (MixerThread *)thread.get();
3423            if (mCblk->frameCount > frames){
3424                if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3425                    uint32_t startFrames = (mCblk->frameCount - frames);
3426                    pInBuffer = new Buffer;
3427                    pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
3428                    pInBuffer->frameCount = startFrames;
3429                    pInBuffer->i16 = pInBuffer->mBuffer;
3430                    memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
3431                    mBufferQueue.add(pInBuffer);
3432                } else {
3433                    LOGW ("OutputTrack::write() %p no more buffers in queue", this);
3434                }
3435            }
3436        }
3437    }
3438
3439    while (waitTimeLeftMs) {
3440        // First write pending buffers, then new data
3441        if (mBufferQueue.size()) {
3442            pInBuffer = mBufferQueue.itemAt(0);
3443        } else {
3444            pInBuffer = &inBuffer;
3445        }
3446
3447        if (pInBuffer->frameCount == 0) {
3448            break;
3449        }
3450
3451        if (mOutBuffer.frameCount == 0) {
3452            mOutBuffer.frameCount = pInBuffer->frameCount;
3453            nsecs_t startTime = systemTime();
3454            if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
3455                LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
3456                outputBufferFull = true;
3457                break;
3458            }
3459            uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
3460            if (waitTimeLeftMs >= waitTimeMs) {
3461                waitTimeLeftMs -= waitTimeMs;
3462            } else {
3463                waitTimeLeftMs = 0;
3464            }
3465        }
3466
3467        uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
3468        memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
3469        mCblk->stepUser(outFrames);
3470        pInBuffer->frameCount -= outFrames;
3471        pInBuffer->i16 += outFrames * channelCount;
3472        mOutBuffer.frameCount -= outFrames;
3473        mOutBuffer.i16 += outFrames * channelCount;
3474
3475        if (pInBuffer->frameCount == 0) {
3476            if (mBufferQueue.size()) {
3477                mBufferQueue.removeAt(0);
3478                delete [] pInBuffer->mBuffer;
3479                delete pInBuffer;
3480                LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3481            } else {
3482                break;
3483            }
3484        }
3485    }
3486
3487    // If we could not write all frames, allocate a buffer and queue it for next time.
3488    if (inBuffer.frameCount) {
3489        sp<ThreadBase> thread = mThread.promote();
3490        if (thread != 0 && !thread->standby()) {
3491            if (mBufferQueue.size() < kMaxOverFlowBuffers) {
3492                pInBuffer = new Buffer;
3493                pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
3494                pInBuffer->frameCount = inBuffer.frameCount;
3495                pInBuffer->i16 = pInBuffer->mBuffer;
3496                memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
3497                mBufferQueue.add(pInBuffer);
3498                LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
3499            } else {
3500                LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
3501            }
3502        }
3503    }
3504
3505    // Calling write() with a 0 length buffer, means that no more data will be written:
3506    // If no more buffers are pending, fill output track buffer to make sure it is started
3507    // by output mixer.
3508    if (frames == 0 && mBufferQueue.size() == 0) {
3509        if (mCblk->user < mCblk->frameCount) {
3510            frames = mCblk->frameCount - mCblk->user;
3511            pInBuffer = new Buffer;
3512            pInBuffer->mBuffer = new int16_t[frames * channelCount];
3513            pInBuffer->frameCount = frames;
3514            pInBuffer->i16 = pInBuffer->mBuffer;
3515            memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
3516            mBufferQueue.add(pInBuffer);
3517        } else if (mActive) {
3518            stop();
3519        }
3520    }
3521
3522    return outputBufferFull;
3523}
3524
3525status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
3526{
3527    int active;
3528    status_t result;
3529    audio_track_cblk_t* cblk = mCblk;
3530    uint32_t framesReq = buffer->frameCount;
3531
3532//    LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
3533    buffer->frameCount  = 0;
3534
3535    uint32_t framesAvail = cblk->framesAvailable();
3536
3537
3538    if (framesAvail == 0) {
3539        Mutex::Autolock _l(cblk->lock);
3540        goto start_loop_here;
3541        while (framesAvail == 0) {
3542            active = mActive;
3543            if (UNLIKELY(!active)) {
3544                LOGV("Not active and NO_MORE_BUFFERS");
3545                return AudioTrack::NO_MORE_BUFFERS;
3546            }
3547            result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
3548            if (result != NO_ERROR) {
3549                return AudioTrack::NO_MORE_BUFFERS;
3550            }
3551            // read the server count again
3552        start_loop_here:
3553            framesAvail = cblk->framesAvailable_l();
3554        }
3555    }
3556
3557//    if (framesAvail < framesReq) {
3558//        return AudioTrack::NO_MORE_BUFFERS;
3559//    }
3560
3561    if (framesReq > framesAvail) {
3562        framesReq = framesAvail;
3563    }
3564
3565    uint32_t u = cblk->user;
3566    uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
3567
3568    if (u + framesReq > bufferEnd) {
3569        framesReq = bufferEnd - u;
3570    }
3571
3572    buffer->frameCount  = framesReq;
3573    buffer->raw         = (void *)cblk->buffer(u);
3574    return NO_ERROR;
3575}
3576
3577
3578void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
3579{
3580    size_t size = mBufferQueue.size();
3581    Buffer *pBuffer;
3582
3583    for (size_t i = 0; i < size; i++) {
3584        pBuffer = mBufferQueue.itemAt(i);
3585        delete [] pBuffer->mBuffer;
3586        delete pBuffer;
3587    }
3588    mBufferQueue.clear();
3589}
3590
3591// ----------------------------------------------------------------------------
3592
3593AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
3594    :   RefBase(),
3595        mAudioFlinger(audioFlinger),
3596        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
3597        mPid(pid)
3598{
3599    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
3600}
3601
3602// Client destructor must be called with AudioFlinger::mLock held
3603AudioFlinger::Client::~Client()
3604{
3605    mAudioFlinger->removeClient_l(mPid);
3606}
3607
3608const sp<MemoryDealer>& AudioFlinger::Client::heap() const
3609{
3610    return mMemoryDealer;
3611}
3612
3613// ----------------------------------------------------------------------------
3614
3615AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
3616                                                     const sp<IAudioFlingerClient>& client,
3617                                                     pid_t pid)
3618    : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
3619{
3620}
3621
3622AudioFlinger::NotificationClient::~NotificationClient()
3623{
3624    mClient.clear();
3625}
3626
3627void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
3628{
3629    sp<NotificationClient> keep(this);
3630    {
3631        mAudioFlinger->removeNotificationClient(mPid);
3632    }
3633}
3634
3635// ----------------------------------------------------------------------------
3636
3637AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
3638    : BnAudioTrack(),
3639      mTrack(track)
3640{
3641}
3642
3643AudioFlinger::TrackHandle::~TrackHandle() {
3644    // just stop the track on deletion, associated resources
3645    // will be freed from the main thread once all pending buffers have
3646    // been played. Unless it's not in the active track list, in which
3647    // case we free everything now...
3648    mTrack->destroy();
3649}
3650
3651status_t AudioFlinger::TrackHandle::start() {
3652    return mTrack->start();
3653}
3654
3655void AudioFlinger::TrackHandle::stop() {
3656    mTrack->stop();
3657}
3658
3659void AudioFlinger::TrackHandle::flush() {
3660    mTrack->flush();
3661}
3662
3663void AudioFlinger::TrackHandle::mute(bool e) {
3664    mTrack->mute(e);
3665}
3666
3667void AudioFlinger::TrackHandle::pause() {
3668    mTrack->pause();
3669}
3670
3671void AudioFlinger::TrackHandle::setVolume(float left, float right) {
3672    mTrack->setVolume(left, right);
3673}
3674
3675sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
3676    return mTrack->getCblk();
3677}
3678
3679status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
3680{
3681    return mTrack->attachAuxEffect(EffectId);
3682}
3683
3684status_t AudioFlinger::TrackHandle::onTransact(
3685    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
3686{
3687    return BnAudioTrack::onTransact(code, data, reply, flags);
3688}
3689
3690// ----------------------------------------------------------------------------
3691
3692sp<IAudioRecord> AudioFlinger::openRecord(
3693        pid_t pid,
3694        int input,
3695        uint32_t sampleRate,
3696        uint32_t format,
3697        uint32_t channelMask,
3698        int frameCount,
3699        uint32_t flags,
3700        int *sessionId,
3701        status_t *status)
3702{
3703    sp<RecordThread::RecordTrack> recordTrack;
3704    sp<RecordHandle> recordHandle;
3705    sp<Client> client;
3706    wp<Client> wclient;
3707    status_t lStatus;
3708    RecordThread *thread;
3709    size_t inFrameCount;
3710    int lSessionId;
3711
3712    // check calling permissions
3713    if (!recordingAllowed()) {
3714        lStatus = PERMISSION_DENIED;
3715        goto Exit;
3716    }
3717
3718    // add client to list
3719    { // scope for mLock
3720        Mutex::Autolock _l(mLock);
3721        thread = checkRecordThread_l(input);
3722        if (thread == NULL) {
3723            lStatus = BAD_VALUE;
3724            goto Exit;
3725        }
3726
3727        wclient = mClients.valueFor(pid);
3728        if (wclient != NULL) {
3729            client = wclient.promote();
3730        } else {
3731            client = new Client(this, pid);
3732            mClients.add(pid, client);
3733        }
3734
3735        // If no audio session id is provided, create one here
3736        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
3737            lSessionId = *sessionId;
3738        } else {
3739            lSessionId = nextUniqueId();
3740            if (sessionId != NULL) {
3741                *sessionId = lSessionId;
3742            }
3743        }
3744        // create new record track. The record track uses one track in mHardwareMixerThread by convention.
3745        recordTrack = thread->createRecordTrack_l(client,
3746                                                sampleRate,
3747                                                format,
3748                                                channelMask,
3749                                                frameCount,
3750                                                flags,
3751                                                lSessionId,
3752                                                &lStatus);
3753    }
3754    if (lStatus != NO_ERROR) {
3755        // remove local strong reference to Client before deleting the RecordTrack so that the Client
3756        // destructor is called by the TrackBase destructor with mLock held
3757        client.clear();
3758        recordTrack.clear();
3759        goto Exit;
3760    }
3761
3762    // return to handle to client
3763    recordHandle = new RecordHandle(recordTrack);
3764    lStatus = NO_ERROR;
3765
3766Exit:
3767    if (status) {
3768        *status = lStatus;
3769    }
3770    return recordHandle;
3771}
3772
3773// ----------------------------------------------------------------------------
3774
3775AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
3776    : BnAudioRecord(),
3777    mRecordTrack(recordTrack)
3778{
3779}
3780
3781AudioFlinger::RecordHandle::~RecordHandle() {
3782    stop();
3783}
3784
3785status_t AudioFlinger::RecordHandle::start() {
3786    LOGV("RecordHandle::start()");
3787    return mRecordTrack->start();
3788}
3789
3790void AudioFlinger::RecordHandle::stop() {
3791    LOGV("RecordHandle::stop()");
3792    mRecordTrack->stop();
3793}
3794
3795sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
3796    return mRecordTrack->getCblk();
3797}
3798
3799status_t AudioFlinger::RecordHandle::onTransact(
3800    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
3801{
3802    return BnAudioRecord::onTransact(code, data, reply, flags);
3803}
3804
3805// ----------------------------------------------------------------------------
3806
3807AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
3808                                         AudioStreamIn *input,
3809                                         uint32_t sampleRate,
3810                                         uint32_t channels,
3811                                         int id,
3812                                         uint32_t device) :
3813    ThreadBase(audioFlinger, id, device),
3814    mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
3815{
3816    mType = ThreadBase::RECORD;
3817    mReqChannelCount = popcount(channels);
3818    mReqSampleRate = sampleRate;
3819    readInputParameters();
3820}
3821
3822
3823AudioFlinger::RecordThread::~RecordThread()
3824{
3825    delete[] mRsmpInBuffer;
3826    if (mResampler != 0) {
3827        delete mResampler;
3828        delete[] mRsmpOutBuffer;
3829    }
3830}
3831
3832void AudioFlinger::RecordThread::onFirstRef()
3833{
3834    const size_t SIZE = 256;
3835    char buffer[SIZE];
3836
3837    snprintf(buffer, SIZE, "Record Thread %p", this);
3838
3839    run(buffer, PRIORITY_URGENT_AUDIO);
3840}
3841
3842bool AudioFlinger::RecordThread::threadLoop()
3843{
3844    AudioBufferProvider::Buffer buffer;
3845    sp<RecordTrack> activeTrack;
3846    Vector< sp<EffectChain> > effectChains;
3847
3848    nsecs_t lastWarning = 0;
3849
3850    // start recording
3851    while (!exitPending()) {
3852
3853        processConfigEvents();
3854
3855        { // scope for mLock
3856            Mutex::Autolock _l(mLock);
3857            checkForNewParameters_l();
3858            if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
3859                if (!mStandby) {
3860                    mInput->stream->common.standby(&mInput->stream->common);
3861                    mStandby = true;
3862                }
3863
3864                if (exitPending()) break;
3865
3866                LOGV("RecordThread: loop stopping");
3867                // go to sleep
3868                mWaitWorkCV.wait(mLock);
3869                LOGV("RecordThread: loop starting");
3870                continue;
3871            }
3872            if (mActiveTrack != 0) {
3873                if (mActiveTrack->mState == TrackBase::PAUSING) {
3874                    if (!mStandby) {
3875                        mInput->stream->common.standby(&mInput->stream->common);
3876                        mStandby = true;
3877                    }
3878                    mActiveTrack.clear();
3879                    mStartStopCond.broadcast();
3880                } else if (mActiveTrack->mState == TrackBase::RESUMING) {
3881                    if (mReqChannelCount != mActiveTrack->channelCount()) {
3882                        mActiveTrack.clear();
3883                        mStartStopCond.broadcast();
3884                    } else if (mBytesRead != 0) {
3885                        // record start succeeds only if first read from audio input
3886                        // succeeds
3887                        if (mBytesRead > 0) {
3888                            mActiveTrack->mState = TrackBase::ACTIVE;
3889                        } else {
3890                            mActiveTrack.clear();
3891                        }
3892                        mStartStopCond.broadcast();
3893                    }
3894                    mStandby = false;
3895                }
3896            }
3897            lockEffectChains_l(effectChains);
3898        }
3899
3900        if (mActiveTrack != 0) {
3901            if (mActiveTrack->mState != TrackBase::ACTIVE &&
3902                mActiveTrack->mState != TrackBase::RESUMING) {
3903                unlockEffectChains(effectChains);
3904                usleep(kRecordThreadSleepUs);
3905                continue;
3906            }
3907            for (size_t i = 0; i < effectChains.size(); i ++) {
3908                effectChains[i]->process_l();
3909            }
3910            // enable changes in effect chain
3911            unlockEffectChains(effectChains);
3912
3913            buffer.frameCount = mFrameCount;
3914            if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
3915                size_t framesOut = buffer.frameCount;
3916                if (mResampler == 0) {
3917                    // no resampling
3918                    while (framesOut) {
3919                        size_t framesIn = mFrameCount - mRsmpInIndex;
3920                        if (framesIn) {
3921                            int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
3922                            int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
3923                            if (framesIn > framesOut)
3924                                framesIn = framesOut;
3925                            mRsmpInIndex += framesIn;
3926                            framesOut -= framesIn;
3927                            if ((int)mChannelCount == mReqChannelCount ||
3928                                mFormat != AUDIO_FORMAT_PCM_16_BIT) {
3929                                memcpy(dst, src, framesIn * mFrameSize);
3930                            } else {
3931                                int16_t *src16 = (int16_t *)src;
3932                                int16_t *dst16 = (int16_t *)dst;
3933                                if (mChannelCount == 1) {
3934                                    while (framesIn--) {
3935                                        *dst16++ = *src16;
3936                                        *dst16++ = *src16++;
3937                                    }
3938                                } else {
3939                                    while (framesIn--) {
3940                                        *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
3941                                        src16 += 2;
3942                                    }
3943                                }
3944                            }
3945                        }
3946                        if (framesOut && mFrameCount == mRsmpInIndex) {
3947                            if (framesOut == mFrameCount &&
3948                                ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
3949                                mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
3950                                framesOut = 0;
3951                            } else {
3952                                mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
3953                                mRsmpInIndex = 0;
3954                            }
3955                            if (mBytesRead < 0) {
3956                                LOGE("Error reading audio input");
3957                                if (mActiveTrack->mState == TrackBase::ACTIVE) {
3958                                    // Force input into standby so that it tries to
3959                                    // recover at next read attempt
3960                                    mInput->stream->common.standby(&mInput->stream->common);
3961                                    usleep(kRecordThreadSleepUs);
3962                                }
3963                                mRsmpInIndex = mFrameCount;
3964                                framesOut = 0;
3965                                buffer.frameCount = 0;
3966                            }
3967                        }
3968                    }
3969                } else {
3970                    // resampling
3971
3972                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
3973                    // alter output frame count as if we were expecting stereo samples
3974                    if (mChannelCount == 1 && mReqChannelCount == 1) {
3975                        framesOut >>= 1;
3976                    }
3977                    mResampler->resample(mRsmpOutBuffer, framesOut, this);
3978                    // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
3979                    // are 32 bit aligned which should be always true.
3980                    if (mChannelCount == 2 && mReqChannelCount == 1) {
3981                        AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
3982                        // the resampler always outputs stereo samples: do post stereo to mono conversion
3983                        int16_t *src = (int16_t *)mRsmpOutBuffer;
3984                        int16_t *dst = buffer.i16;
3985                        while (framesOut--) {
3986                            *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
3987                            src += 2;
3988                        }
3989                    } else {
3990                        AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
3991                    }
3992
3993                }
3994                mActiveTrack->releaseBuffer(&buffer);
3995                mActiveTrack->overflow();
3996            }
3997            // client isn't retrieving buffers fast enough
3998            else {
3999                if (!mActiveTrack->setOverflow()) {
4000                    nsecs_t now = systemTime();
4001                    if ((now - lastWarning) > kWarningThrottle) {
4002                        LOGW("RecordThread: buffer overflow");
4003                        lastWarning = now;
4004                    }
4005                }
4006                // Release the processor for a while before asking for a new buffer.
4007                // This will give the application more chance to read from the buffer and
4008                // clear the overflow.
4009                usleep(kRecordThreadSleepUs);
4010            }
4011        } else {
4012            unlockEffectChains(effectChains);
4013        }
4014        effectChains.clear();
4015    }
4016
4017    if (!mStandby) {
4018        mInput->stream->common.standby(&mInput->stream->common);
4019    }
4020    mActiveTrack.clear();
4021
4022    mStartStopCond.broadcast();
4023
4024    LOGV("RecordThread %p exiting", this);
4025    return false;
4026}
4027
4028
4029sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
4030        const sp<AudioFlinger::Client>& client,
4031        uint32_t sampleRate,
4032        int format,
4033        int channelMask,
4034        int frameCount,
4035        uint32_t flags,
4036        int sessionId,
4037        status_t *status)
4038{
4039    sp<RecordTrack> track;
4040    status_t lStatus;
4041
4042    lStatus = initCheck();
4043    if (lStatus != NO_ERROR) {
4044        LOGE("Audio driver not initialized.");
4045        goto Exit;
4046    }
4047
4048    { // scope for mLock
4049        Mutex::Autolock _l(mLock);
4050
4051        track = new RecordTrack(this, client, sampleRate,
4052                      format, channelMask, frameCount, flags, sessionId);
4053
4054        if (track->getCblk() == NULL) {
4055            lStatus = NO_MEMORY;
4056            goto Exit;
4057        }
4058
4059        mTrack = track.get();
4060
4061    }
4062    lStatus = NO_ERROR;
4063
4064Exit:
4065    if (status) {
4066        *status = lStatus;
4067    }
4068    return track;
4069}
4070
4071status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
4072{
4073    LOGV("RecordThread::start");
4074    sp <ThreadBase> strongMe = this;
4075    status_t status = NO_ERROR;
4076    {
4077        AutoMutex lock(&mLock);
4078        if (mActiveTrack != 0) {
4079            if (recordTrack != mActiveTrack.get()) {
4080                status = -EBUSY;
4081            } else if (mActiveTrack->mState == TrackBase::PAUSING) {
4082                mActiveTrack->mState = TrackBase::ACTIVE;
4083            }
4084            return status;
4085        }
4086
4087        recordTrack->mState = TrackBase::IDLE;
4088        mActiveTrack = recordTrack;
4089        mLock.unlock();
4090        status_t status = AudioSystem::startInput(mId);
4091        mLock.lock();
4092        if (status != NO_ERROR) {
4093            mActiveTrack.clear();
4094            return status;
4095        }
4096        mRsmpInIndex = mFrameCount;
4097        mBytesRead = 0;
4098        if (mResampler != NULL) {
4099            mResampler->reset();
4100        }
4101        mActiveTrack->mState = TrackBase::RESUMING;
4102        // signal thread to start
4103        LOGV("Signal record thread");
4104        mWaitWorkCV.signal();
4105        // do not wait for mStartStopCond if exiting
4106        if (mExiting) {
4107            mActiveTrack.clear();
4108            status = INVALID_OPERATION;
4109            goto startError;
4110        }
4111        mStartStopCond.wait(mLock);
4112        if (mActiveTrack == 0) {
4113            LOGV("Record failed to start");
4114            status = BAD_VALUE;
4115            goto startError;
4116        }
4117        LOGV("Record started OK");
4118        return status;
4119    }
4120startError:
4121    AudioSystem::stopInput(mId);
4122    return status;
4123}
4124
4125void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
4126    LOGV("RecordThread::stop");
4127    sp <ThreadBase> strongMe = this;
4128    {
4129        AutoMutex lock(&mLock);
4130        if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
4131            mActiveTrack->mState = TrackBase::PAUSING;
4132            // do not wait for mStartStopCond if exiting
4133            if (mExiting) {
4134                return;
4135            }
4136            mStartStopCond.wait(mLock);
4137            // if we have been restarted, recordTrack == mActiveTrack.get() here
4138            if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
4139                mLock.unlock();
4140                AudioSystem::stopInput(mId);
4141                mLock.lock();
4142                LOGV("Record stopped OK");
4143            }
4144        }
4145    }
4146}
4147
4148status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4149{
4150    const size_t SIZE = 256;
4151    char buffer[SIZE];
4152    String8 result;
4153    pid_t pid = 0;
4154
4155    snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4156    result.append(buffer);
4157
4158    if (mActiveTrack != 0) {
4159        result.append("Active Track:\n");
4160        result.append("   Clien Fmt Chn mask   Session Buf  S SRate  Serv     User\n");
4161        mActiveTrack->dump(buffer, SIZE);
4162        result.append(buffer);
4163
4164        snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4165        result.append(buffer);
4166        snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4167        result.append(buffer);
4168        snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
4169        result.append(buffer);
4170        snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
4171        result.append(buffer);
4172        snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
4173        result.append(buffer);
4174
4175
4176    } else {
4177        result.append("No record client\n");
4178    }
4179    write(fd, result.string(), result.size());
4180
4181    dumpBase(fd, args);
4182    dumpEffectChains(fd, args);
4183
4184    return NO_ERROR;
4185}
4186
4187status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
4188{
4189    size_t framesReq = buffer->frameCount;
4190    size_t framesReady = mFrameCount - mRsmpInIndex;
4191    int channelCount;
4192
4193    if (framesReady == 0) {
4194        mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4195        if (mBytesRead < 0) {
4196            LOGE("RecordThread::getNextBuffer() Error reading audio input");
4197            if (mActiveTrack->mState == TrackBase::ACTIVE) {
4198                // Force input into standby so that it tries to
4199                // recover at next read attempt
4200                mInput->stream->common.standby(&mInput->stream->common);
4201                usleep(kRecordThreadSleepUs);
4202            }
4203            buffer->raw = 0;
4204            buffer->frameCount = 0;
4205            return NOT_ENOUGH_DATA;
4206        }
4207        mRsmpInIndex = 0;
4208        framesReady = mFrameCount;
4209    }
4210
4211    if (framesReq > framesReady) {
4212        framesReq = framesReady;
4213    }
4214
4215    if (mChannelCount == 1 && mReqChannelCount == 2) {
4216        channelCount = 1;
4217    } else {
4218        channelCount = 2;
4219    }
4220    buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4221    buffer->frameCount = framesReq;
4222    return NO_ERROR;
4223}
4224
4225void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4226{
4227    mRsmpInIndex += buffer->frameCount;
4228    buffer->frameCount = 0;
4229}
4230
4231bool AudioFlinger::RecordThread::checkForNewParameters_l()
4232{
4233    bool reconfig = false;
4234
4235    while (!mNewParameters.isEmpty()) {
4236        status_t status = NO_ERROR;
4237        String8 keyValuePair = mNewParameters[0];
4238        AudioParameter param = AudioParameter(keyValuePair);
4239        int value;
4240        int reqFormat = mFormat;
4241        int reqSamplingRate = mReqSampleRate;
4242        int reqChannelCount = mReqChannelCount;
4243
4244        if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4245            reqSamplingRate = value;
4246            reconfig = true;
4247        }
4248        if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4249            reqFormat = value;
4250            reconfig = true;
4251        }
4252        if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4253            reqChannelCount = popcount(value);
4254            reconfig = true;
4255        }
4256        if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4257            // do not accept frame count changes if tracks are open as the track buffer
4258            // size depends on frame count and correct behavior would not be garantied
4259            // if frame count is changed after track creation
4260            if (mActiveTrack != 0) {
4261                status = INVALID_OPERATION;
4262            } else {
4263                reconfig = true;
4264            }
4265        }
4266        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4267            // forward device change to effects that have requested to be
4268            // aware of attached audio device.
4269            for (size_t i = 0; i < mEffectChains.size(); i++) {
4270                mEffectChains[i]->setDevice_l(value);
4271            }
4272            // store input device and output device but do not forward output device to audio HAL.
4273            // Note that status is ignored by the caller for output device
4274            // (see AudioFlinger::setParameters()
4275            if (value & AUDIO_DEVICE_OUT_ALL) {
4276                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
4277                status = BAD_VALUE;
4278            } else {
4279                mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
4280            }
4281            mDevice |= (uint32_t)value;
4282        }
4283        if (status == NO_ERROR) {
4284            status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4285            if (status == INVALID_OPERATION) {
4286               mInput->stream->common.standby(&mInput->stream->common);
4287               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
4288            }
4289            if (reconfig) {
4290                if (status == BAD_VALUE &&
4291                    reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4292                    reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
4293                    ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
4294                    (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
4295                    (reqChannelCount < 3)) {
4296                    status = NO_ERROR;
4297                }
4298                if (status == NO_ERROR) {
4299                    readInputParameters();
4300                    sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4301                }
4302            }
4303        }
4304
4305        mNewParameters.removeAt(0);
4306
4307        mParamStatus = status;
4308        mParamCond.signal();
4309        mWaitWorkCV.wait(mLock);
4310    }
4311    return reconfig;
4312}
4313
4314String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4315{
4316    char *s;
4317    String8 out_s8;
4318
4319    s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4320    out_s8 = String8(s);
4321    free(s);
4322    return out_s8;
4323}
4324
4325void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4326    AudioSystem::OutputDescriptor desc;
4327    void *param2 = 0;
4328
4329    switch (event) {
4330    case AudioSystem::INPUT_OPENED:
4331    case AudioSystem::INPUT_CONFIG_CHANGED:
4332        desc.channels = mChannelMask;
4333        desc.samplingRate = mSampleRate;
4334        desc.format = mFormat;
4335        desc.frameCount = mFrameCount;
4336        desc.latency = 0;
4337        param2 = &desc;
4338        break;
4339
4340    case AudioSystem::INPUT_CLOSED:
4341    default:
4342        break;
4343    }
4344    mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4345}
4346
4347void AudioFlinger::RecordThread::readInputParameters()
4348{
4349    if (mRsmpInBuffer) delete mRsmpInBuffer;
4350    if (mRsmpOutBuffer) delete mRsmpOutBuffer;
4351    if (mResampler) delete mResampler;
4352    mResampler = 0;
4353
4354    mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4355    mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4356    mChannelCount = (uint16_t)popcount(mChannelMask);
4357    mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4358    mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
4359    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4360    mFrameCount = mInputBytes / mFrameSize;
4361    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4362
4363    if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
4364    {
4365        int channelCount;
4366         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4367         // stereo to mono post process as the resampler always outputs stereo.
4368        if (mChannelCount == 1 && mReqChannelCount == 2) {
4369            channelCount = 1;
4370        } else {
4371            channelCount = 2;
4372        }
4373        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4374        mResampler->setSampleRate(mSampleRate);
4375        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4376        mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4377
4378        // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
4379        if (mChannelCount == 1 && mReqChannelCount == 1) {
4380            mFrameCount >>= 1;
4381        }
4382
4383    }
4384    mRsmpInIndex = mFrameCount;
4385}
4386
4387unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4388{
4389    return mInput->stream->get_input_frames_lost(mInput->stream);
4390}
4391
4392uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
4393{
4394    Mutex::Autolock _l(mLock);
4395    uint32_t result = 0;
4396    if (getEffectChain_l(sessionId) != 0) {
4397        result = EFFECT_SESSION;
4398    }
4399
4400    if (mTrack != NULL && sessionId == mTrack->sessionId()) {
4401        result |= TRACK_SESSION;
4402    }
4403
4404    return result;
4405}
4406
4407// ----------------------------------------------------------------------------
4408
4409int AudioFlinger::openOutput(uint32_t *pDevices,
4410                                uint32_t *pSamplingRate,
4411                                uint32_t *pFormat,
4412                                uint32_t *pChannels,
4413                                uint32_t *pLatencyMs,
4414                                uint32_t flags)
4415{
4416    status_t status;
4417    PlaybackThread *thread = NULL;
4418    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
4419    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4420    uint32_t format = pFormat ? *pFormat : 0;
4421    uint32_t channels = pChannels ? *pChannels : 0;
4422    uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
4423    audio_stream_out_t *outStream;
4424    audio_hw_device_t *outHwDev;
4425
4426    LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
4427            pDevices ? *pDevices : 0,
4428            samplingRate,
4429            format,
4430            channels,
4431            flags);
4432
4433    if (pDevices == NULL || *pDevices == 0) {
4434        return 0;
4435    }
4436
4437    Mutex::Autolock _l(mLock);
4438
4439    outHwDev = findSuitableHwDev_l(*pDevices);
4440    if (outHwDev == NULL)
4441        return 0;
4442
4443    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
4444                                          &channels, &samplingRate, &outStream);
4445    LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
4446            outStream,
4447            samplingRate,
4448            format,
4449            channels,
4450            status);
4451
4452    mHardwareStatus = AUDIO_HW_IDLE;
4453    if (outStream != NULL) {
4454        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
4455        int id = nextUniqueId();
4456
4457        if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
4458            (format != AUDIO_FORMAT_PCM_16_BIT) ||
4459            (channels != AUDIO_CHANNEL_OUT_STEREO)) {
4460            thread = new DirectOutputThread(this, output, id, *pDevices);
4461            LOGV("openOutput() created direct output: ID %d thread %p", id, thread);
4462        } else {
4463            thread = new MixerThread(this, output, id, *pDevices);
4464            LOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
4465        }
4466        mPlaybackThreads.add(id, thread);
4467
4468        if (pSamplingRate) *pSamplingRate = samplingRate;
4469        if (pFormat) *pFormat = format;
4470        if (pChannels) *pChannels = channels;
4471        if (pLatencyMs) *pLatencyMs = thread->latency();
4472
4473        // notify client processes of the new output creation
4474        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4475        return id;
4476    }
4477
4478    return 0;
4479}
4480
4481int AudioFlinger::openDuplicateOutput(int output1, int output2)
4482{
4483    Mutex::Autolock _l(mLock);
4484    MixerThread *thread1 = checkMixerThread_l(output1);
4485    MixerThread *thread2 = checkMixerThread_l(output2);
4486
4487    if (thread1 == NULL || thread2 == NULL) {
4488        LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
4489        return 0;
4490    }
4491
4492    int id = nextUniqueId();
4493    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
4494    thread->addOutputTrack(thread2);
4495    mPlaybackThreads.add(id, thread);
4496    // notify client processes of the new output creation
4497    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
4498    return id;
4499}
4500
4501status_t AudioFlinger::closeOutput(int output)
4502{
4503    // keep strong reference on the playback thread so that
4504    // it is not destroyed while exit() is executed
4505    sp <PlaybackThread> thread;
4506    {
4507        Mutex::Autolock _l(mLock);
4508        thread = checkPlaybackThread_l(output);
4509        if (thread == NULL) {
4510            return BAD_VALUE;
4511        }
4512
4513        LOGV("closeOutput() %d", output);
4514
4515        if (thread->type() == ThreadBase::MIXER) {
4516            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4517                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
4518                    DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
4519                    dupThread->removeOutputTrack((MixerThread *)thread.get());
4520                }
4521            }
4522        }
4523        void *param2 = 0;
4524        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
4525        mPlaybackThreads.removeItem(output);
4526    }
4527    thread->exit();
4528
4529    if (thread->type() != ThreadBase::DUPLICATING) {
4530        AudioStreamOut *out = thread->getOutput();
4531        out->hwDev->close_output_stream(out->hwDev, out->stream);
4532        delete out;
4533    }
4534    return NO_ERROR;
4535}
4536
4537status_t AudioFlinger::suspendOutput(int output)
4538{
4539    Mutex::Autolock _l(mLock);
4540    PlaybackThread *thread = checkPlaybackThread_l(output);
4541
4542    if (thread == NULL) {
4543        return BAD_VALUE;
4544    }
4545
4546    LOGV("suspendOutput() %d", output);
4547    thread->suspend();
4548
4549    return NO_ERROR;
4550}
4551
4552status_t AudioFlinger::restoreOutput(int output)
4553{
4554    Mutex::Autolock _l(mLock);
4555    PlaybackThread *thread = checkPlaybackThread_l(output);
4556
4557    if (thread == NULL) {
4558        return BAD_VALUE;
4559    }
4560
4561    LOGV("restoreOutput() %d", output);
4562
4563    thread->restore();
4564
4565    return NO_ERROR;
4566}
4567
4568int AudioFlinger::openInput(uint32_t *pDevices,
4569                                uint32_t *pSamplingRate,
4570                                uint32_t *pFormat,
4571                                uint32_t *pChannels,
4572                                uint32_t acoustics)
4573{
4574    status_t status;
4575    RecordThread *thread = NULL;
4576    uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
4577    uint32_t format = pFormat ? *pFormat : 0;
4578    uint32_t channels = pChannels ? *pChannels : 0;
4579    uint32_t reqSamplingRate = samplingRate;
4580    uint32_t reqFormat = format;
4581    uint32_t reqChannels = channels;
4582    audio_stream_in_t *inStream;
4583    audio_hw_device_t *inHwDev;
4584
4585    if (pDevices == NULL || *pDevices == 0) {
4586        return 0;
4587    }
4588
4589    Mutex::Autolock _l(mLock);
4590
4591    inHwDev = findSuitableHwDev_l(*pDevices);
4592    if (inHwDev == NULL)
4593        return 0;
4594
4595    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
4596                                        &channels, &samplingRate,
4597                                        (audio_in_acoustics_t)acoustics,
4598                                        &inStream);
4599    LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
4600            inStream,
4601            samplingRate,
4602            format,
4603            channels,
4604            acoustics,
4605            status);
4606
4607    // If the input could not be opened with the requested parameters and we can handle the conversion internally,
4608    // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
4609    // or stereo to mono conversions on 16 bit PCM inputs.
4610    if (inStream == NULL && status == BAD_VALUE &&
4611        reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
4612        (samplingRate <= 2 * reqSamplingRate) &&
4613        (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
4614        LOGV("openInput() reopening with proposed sampling rate and channels");
4615        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
4616                                            &channels, &samplingRate,
4617                                            (audio_in_acoustics_t)acoustics,
4618                                            &inStream);
4619    }
4620
4621    if (inStream != NULL) {
4622        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
4623
4624        int id = nextUniqueId();
4625        // Start record thread
4626        // RecorThread require both input and output device indication to forward to audio
4627        // pre processing modules
4628        uint32_t device = (*pDevices) | primaryOutputDevice_l();
4629        thread = new RecordThread(this,
4630                                  input,
4631                                  reqSamplingRate,
4632                                  reqChannels,
4633                                  id,
4634                                  device);
4635        mRecordThreads.add(id, thread);
4636        LOGV("openInput() created record thread: ID %d thread %p", id, thread);
4637        if (pSamplingRate) *pSamplingRate = reqSamplingRate;
4638        if (pFormat) *pFormat = format;
4639        if (pChannels) *pChannels = reqChannels;
4640
4641        input->stream->common.standby(&input->stream->common);
4642
4643        // notify client processes of the new input creation
4644        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
4645        return id;
4646    }
4647
4648    return 0;
4649}
4650
4651status_t AudioFlinger::closeInput(int input)
4652{
4653    // keep strong reference on the record thread so that
4654    // it is not destroyed while exit() is executed
4655    sp <RecordThread> thread;
4656    {
4657        Mutex::Autolock _l(mLock);
4658        thread = checkRecordThread_l(input);
4659        if (thread == NULL) {
4660            return BAD_VALUE;
4661        }
4662
4663        LOGV("closeInput() %d", input);
4664        void *param2 = 0;
4665        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
4666        mRecordThreads.removeItem(input);
4667    }
4668    thread->exit();
4669
4670    AudioStreamIn *in = thread->getInput();
4671    in->hwDev->close_input_stream(in->hwDev, in->stream);
4672    delete in;
4673
4674    return NO_ERROR;
4675}
4676
4677status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
4678{
4679    Mutex::Autolock _l(mLock);
4680    MixerThread *dstThread = checkMixerThread_l(output);
4681    if (dstThread == NULL) {
4682        LOGW("setStreamOutput() bad output id %d", output);
4683        return BAD_VALUE;
4684    }
4685
4686    LOGV("setStreamOutput() stream %d to output %d", stream, output);
4687    audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
4688
4689    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4690        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
4691        if (thread != dstThread &&
4692            thread->type() != ThreadBase::DIRECT) {
4693            MixerThread *srcThread = (MixerThread *)thread;
4694            srcThread->invalidateTracks(stream);
4695        }
4696    }
4697
4698    return NO_ERROR;
4699}
4700
4701
4702int AudioFlinger::newAudioSessionId()
4703{
4704    return nextUniqueId();
4705}
4706
4707// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
4708AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
4709{
4710    PlaybackThread *thread = NULL;
4711    if (mPlaybackThreads.indexOfKey(output) >= 0) {
4712        thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
4713    }
4714    return thread;
4715}
4716
4717// checkMixerThread_l() must be called with AudioFlinger::mLock held
4718AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
4719{
4720    PlaybackThread *thread = checkPlaybackThread_l(output);
4721    if (thread != NULL) {
4722        if (thread->type() == ThreadBase::DIRECT) {
4723            thread = NULL;
4724        }
4725    }
4726    return (MixerThread *)thread;
4727}
4728
4729// checkRecordThread_l() must be called with AudioFlinger::mLock held
4730AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
4731{
4732    RecordThread *thread = NULL;
4733    if (mRecordThreads.indexOfKey(input) >= 0) {
4734        thread = (RecordThread *)mRecordThreads.valueFor(input).get();
4735    }
4736    return thread;
4737}
4738
4739uint32_t AudioFlinger::nextUniqueId()
4740{
4741    return android_atomic_inc(&mNextUniqueId);
4742}
4743
4744AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
4745{
4746    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4747        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
4748        if (thread->getOutput()->hwDev == mPrimaryHardwareDev) {
4749            return thread;
4750        }
4751    }
4752    return NULL;
4753}
4754
4755uint32_t AudioFlinger::primaryOutputDevice_l()
4756{
4757    PlaybackThread *thread = primaryPlaybackThread_l();
4758
4759    if (thread == NULL) {
4760        return 0;
4761    }
4762
4763    return thread->device();
4764}
4765
4766
4767// ----------------------------------------------------------------------------
4768//  Effect management
4769// ----------------------------------------------------------------------------
4770
4771
4772status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
4773{
4774    Mutex::Autolock _l(mLock);
4775    return EffectQueryNumberEffects(numEffects);
4776}
4777
4778status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
4779{
4780    Mutex::Autolock _l(mLock);
4781    return EffectQueryEffect(index, descriptor);
4782}
4783
4784status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
4785{
4786    Mutex::Autolock _l(mLock);
4787    return EffectGetDescriptor(pUuid, descriptor);
4788}
4789
4790
4791// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp
4792static const effect_uuid_t VISUALIZATION_UUID_ =
4793    {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
4794
4795sp<IEffect> AudioFlinger::createEffect(pid_t pid,
4796        effect_descriptor_t *pDesc,
4797        const sp<IEffectClient>& effectClient,
4798        int32_t priority,
4799        int io,
4800        int sessionId,
4801        status_t *status,
4802        int *id,
4803        int *enabled)
4804{
4805    status_t lStatus = NO_ERROR;
4806    sp<EffectHandle> handle;
4807    effect_descriptor_t desc;
4808    sp<Client> client;
4809    wp<Client> wclient;
4810
4811    LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
4812            pid, effectClient.get(), priority, sessionId, io);
4813
4814    if (pDesc == NULL) {
4815        lStatus = BAD_VALUE;
4816        goto Exit;
4817    }
4818
4819    // check audio settings permission for global effects
4820    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
4821        lStatus = PERMISSION_DENIED;
4822        goto Exit;
4823    }
4824
4825    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
4826    // that can only be created by audio policy manager (running in same process)
4827    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
4828        lStatus = PERMISSION_DENIED;
4829        goto Exit;
4830    }
4831
4832    // check recording permission for visualizer
4833    if ((memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 ||
4834         memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) &&
4835        !recordingAllowed()) {
4836        lStatus = PERMISSION_DENIED;
4837        goto Exit;
4838    }
4839
4840    if (io == 0) {
4841        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
4842            // output must be specified by AudioPolicyManager when using session
4843            // AUDIO_SESSION_OUTPUT_STAGE
4844            lStatus = BAD_VALUE;
4845            goto Exit;
4846        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
4847            // if the output returned by getOutputForEffect() is removed before we lock the
4848            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
4849            // and we will exit safely
4850            io = AudioSystem::getOutputForEffect(&desc);
4851        }
4852    }
4853
4854    {
4855        Mutex::Autolock _l(mLock);
4856
4857
4858        if (!EffectIsNullUuid(&pDesc->uuid)) {
4859            // if uuid is specified, request effect descriptor
4860            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
4861            if (lStatus < 0) {
4862                LOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
4863                goto Exit;
4864            }
4865        } else {
4866            // if uuid is not specified, look for an available implementation
4867            // of the required type in effect factory
4868            if (EffectIsNullUuid(&pDesc->type)) {
4869                LOGW("createEffect() no effect type");
4870                lStatus = BAD_VALUE;
4871                goto Exit;
4872            }
4873            uint32_t numEffects = 0;
4874            effect_descriptor_t d;
4875            bool found = false;
4876
4877            lStatus = EffectQueryNumberEffects(&numEffects);
4878            if (lStatus < 0) {
4879                LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
4880                goto Exit;
4881            }
4882            for (uint32_t i = 0; i < numEffects; i++) {
4883                lStatus = EffectQueryEffect(i, &desc);
4884                if (lStatus < 0) {
4885                    LOGW("createEffect() error %d from EffectQueryEffect", lStatus);
4886                    continue;
4887                }
4888                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
4889                    // If matching type found save effect descriptor. If the session is
4890                    // 0 and the effect is not auxiliary, continue enumeration in case
4891                    // an auxiliary version of this effect type is available
4892                    found = true;
4893                    memcpy(&d, &desc, sizeof(effect_descriptor_t));
4894                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
4895                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
4896                        break;
4897                    }
4898                }
4899            }
4900            if (!found) {
4901                lStatus = BAD_VALUE;
4902                LOGW("createEffect() effect not found");
4903                goto Exit;
4904            }
4905            // For same effect type, chose auxiliary version over insert version if
4906            // connect to output mix (Compliance to OpenSL ES)
4907            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
4908                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
4909                memcpy(&desc, &d, sizeof(effect_descriptor_t));
4910            }
4911        }
4912
4913        // Do not allow auxiliary effects on a session different from 0 (output mix)
4914        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
4915             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
4916            lStatus = INVALID_OPERATION;
4917            goto Exit;
4918        }
4919
4920        // return effect descriptor
4921        memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
4922
4923        // If output is not specified try to find a matching audio session ID in one of the
4924        // output threads.
4925        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
4926        // because of code checking output when entering the function.
4927        // Note: io is never 0 when creating an effect on an input
4928        if (io == 0) {
4929             // look for the thread where the specified audio session is present
4930            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
4931                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
4932                    io = mPlaybackThreads.keyAt(i);
4933                    break;
4934                }
4935            }
4936            if (io == 0) {
4937               for (size_t i = 0; i < mRecordThreads.size(); i++) {
4938                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
4939                       io = mRecordThreads.keyAt(i);
4940                       break;
4941                   }
4942               }
4943            }
4944            // If no output thread contains the requested session ID, default to
4945            // first output. The effect chain will be moved to the correct output
4946            // thread when a track with the same session ID is created
4947            if (io == 0 && mPlaybackThreads.size()) {
4948                io = mPlaybackThreads.keyAt(0);
4949            }
4950            LOGV("createEffect() got io %d for effect %s", io, desc.name);
4951        }
4952        ThreadBase *thread = checkRecordThread_l(io);
4953        if (thread == NULL) {
4954            thread = checkPlaybackThread_l(io);
4955            if (thread == NULL) {
4956                LOGE("createEffect() unknown output thread");
4957                lStatus = BAD_VALUE;
4958                goto Exit;
4959            }
4960        }
4961
4962        wclient = mClients.valueFor(pid);
4963
4964        if (wclient != NULL) {
4965            client = wclient.promote();
4966        } else {
4967            client = new Client(this, pid);
4968            mClients.add(pid, client);
4969        }
4970
4971        // create effect on selected output trhead
4972        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
4973                &desc, enabled, &lStatus);
4974        if (handle != 0 && id != NULL) {
4975            *id = handle->id();
4976        }
4977    }
4978
4979Exit:
4980    if(status) {
4981        *status = lStatus;
4982    }
4983    return handle;
4984}
4985
4986status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput)
4987{
4988    LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
4989            session, srcOutput, dstOutput);
4990    Mutex::Autolock _l(mLock);
4991    if (srcOutput == dstOutput) {
4992        LOGW("moveEffects() same dst and src outputs %d", dstOutput);
4993        return NO_ERROR;
4994    }
4995    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
4996    if (srcThread == NULL) {
4997        LOGW("moveEffects() bad srcOutput %d", srcOutput);
4998        return BAD_VALUE;
4999    }
5000    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
5001    if (dstThread == NULL) {
5002        LOGW("moveEffects() bad dstOutput %d", dstOutput);
5003        return BAD_VALUE;
5004    }
5005
5006    Mutex::Autolock _dl(dstThread->mLock);
5007    Mutex::Autolock _sl(srcThread->mLock);
5008    moveEffectChain_l(session, srcThread, dstThread, false);
5009
5010    return NO_ERROR;
5011}
5012
5013// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held
5014status_t AudioFlinger::moveEffectChain_l(int session,
5015                                   AudioFlinger::PlaybackThread *srcThread,
5016                                   AudioFlinger::PlaybackThread *dstThread,
5017                                   bool reRegister)
5018{
5019    LOGV("moveEffectChain_l() session %d from thread %p to thread %p",
5020            session, srcThread, dstThread);
5021
5022    sp<EffectChain> chain = srcThread->getEffectChain_l(session);
5023    if (chain == 0) {
5024        LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
5025                session, srcThread);
5026        return INVALID_OPERATION;
5027    }
5028
5029    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
5030    // so that a new chain is created with correct parameters when first effect is added. This is
5031    // otherwise unecessary as removeEffect_l() will remove the chain when last effect is
5032    // removed.
5033    srcThread->removeEffectChain_l(chain);
5034
5035    // transfer all effects one by one so that new effect chain is created on new thread with
5036    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
5037    int dstOutput = dstThread->id();
5038    sp<EffectChain> dstChain;
5039    uint32_t strategy;
5040    sp<EffectModule> effect = chain->getEffectFromId_l(0);
5041    while (effect != 0) {
5042        srcThread->removeEffect_l(effect);
5043        dstThread->addEffect_l(effect);
5044        // if the move request is not received from audio policy manager, the effect must be
5045        // re-registered with the new strategy and output
5046        if (dstChain == 0) {
5047            dstChain = effect->chain().promote();
5048            if (dstChain == 0) {
5049                LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
5050                srcThread->addEffect_l(effect);
5051                return NO_INIT;
5052            }
5053            strategy = dstChain->strategy();
5054        }
5055        if (reRegister) {
5056            AudioSystem::unregisterEffect(effect->id());
5057            AudioSystem::registerEffect(&effect->desc(),
5058                                        dstOutput,
5059                                        strategy,
5060                                        session,
5061                                        effect->id());
5062        }
5063        effect = chain->getEffectFromId_l(0);
5064    }
5065
5066    return NO_ERROR;
5067}
5068
5069
5070// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
5071sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
5072        const sp<AudioFlinger::Client>& client,
5073        const sp<IEffectClient>& effectClient,
5074        int32_t priority,
5075        int sessionId,
5076        effect_descriptor_t *desc,
5077        int *enabled,
5078        status_t *status
5079        )
5080{
5081    sp<EffectModule> effect;
5082    sp<EffectHandle> handle;
5083    status_t lStatus;
5084    sp<EffectChain> chain;
5085    bool chainCreated = false;
5086    bool effectCreated = false;
5087    bool effectRegistered = false;
5088
5089    lStatus = initCheck();
5090    if (lStatus != NO_ERROR) {
5091        LOGW("createEffect_l() Audio driver not initialized.");
5092        goto Exit;
5093    }
5094
5095    // Do not allow effects with session ID 0 on direct output or duplicating threads
5096    // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
5097    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
5098        LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
5099                desc->name, sessionId);
5100        lStatus = BAD_VALUE;
5101        goto Exit;
5102    }
5103    // Only Pre processor effects are allowed on input threads and only on input threads
5104    if ((mType == RECORD &&
5105            (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
5106            (mType != RECORD &&
5107                    (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
5108        LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
5109                desc->name, desc->flags, mType);
5110        lStatus = BAD_VALUE;
5111        goto Exit;
5112    }
5113
5114    LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
5115
5116    { // scope for mLock
5117        Mutex::Autolock _l(mLock);
5118
5119        // check for existing effect chain with the requested audio session
5120        chain = getEffectChain_l(sessionId);
5121        if (chain == 0) {
5122            // create a new chain for this session
5123            LOGV("createEffect_l() new effect chain for session %d", sessionId);
5124            chain = new EffectChain(this, sessionId);
5125            addEffectChain_l(chain);
5126            chain->setStrategy(getStrategyForSession_l(sessionId));
5127            chainCreated = true;
5128        } else {
5129            effect = chain->getEffectFromDesc_l(desc);
5130        }
5131
5132        LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
5133
5134        if (effect == 0) {
5135            int id = mAudioFlinger->nextUniqueId();
5136            // Check CPU and memory usage
5137            lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
5138            if (lStatus != NO_ERROR) {
5139                goto Exit;
5140            }
5141            effectRegistered = true;
5142            // create a new effect module if none present in the chain
5143            effect = new EffectModule(this, chain, desc, id, sessionId);
5144            lStatus = effect->status();
5145            if (lStatus != NO_ERROR) {
5146                goto Exit;
5147            }
5148            lStatus = chain->addEffect_l(effect);
5149            if (lStatus != NO_ERROR) {
5150                goto Exit;
5151            }
5152            effectCreated = true;
5153
5154            effect->setDevice(mDevice);
5155            effect->setMode(mAudioFlinger->getMode());
5156        }
5157        // create effect handle and connect it to effect module
5158        handle = new EffectHandle(effect, client, effectClient, priority);
5159        lStatus = effect->addHandle(handle);
5160        if (enabled) {
5161            *enabled = (int)effect->isEnabled();
5162        }
5163    }
5164
5165Exit:
5166    if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
5167        Mutex::Autolock _l(mLock);
5168        if (effectCreated) {
5169            chain->removeEffect_l(effect);
5170        }
5171        if (effectRegistered) {
5172            AudioSystem::unregisterEffect(effect->id());
5173        }
5174        if (chainCreated) {
5175            removeEffectChain_l(chain);
5176        }
5177        handle.clear();
5178    }
5179
5180    if(status) {
5181        *status = lStatus;
5182    }
5183    return handle;
5184}
5185
5186sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
5187{
5188    sp<EffectModule> effect;
5189
5190    sp<EffectChain> chain = getEffectChain_l(sessionId);
5191    if (chain != 0) {
5192        effect = chain->getEffectFromId_l(effectId);
5193    }
5194    return effect;
5195}
5196
5197// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
5198// PlaybackThread::mLock held
5199status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
5200{
5201    // check for existing effect chain with the requested audio session
5202    int sessionId = effect->sessionId();
5203    sp<EffectChain> chain = getEffectChain_l(sessionId);
5204    bool chainCreated = false;
5205
5206    if (chain == 0) {
5207        // create a new chain for this session
5208        LOGV("addEffect_l() new effect chain for session %d", sessionId);
5209        chain = new EffectChain(this, sessionId);
5210        addEffectChain_l(chain);
5211        chain->setStrategy(getStrategyForSession_l(sessionId));
5212        chainCreated = true;
5213    }
5214    LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
5215
5216    if (chain->getEffectFromId_l(effect->id()) != 0) {
5217        LOGW("addEffect_l() %p effect %s already present in chain %p",
5218                this, effect->desc().name, chain.get());
5219        return BAD_VALUE;
5220    }
5221
5222    status_t status = chain->addEffect_l(effect);
5223    if (status != NO_ERROR) {
5224        if (chainCreated) {
5225            removeEffectChain_l(chain);
5226        }
5227        return status;
5228    }
5229
5230    effect->setDevice(mDevice);
5231    effect->setMode(mAudioFlinger->getMode());
5232    return NO_ERROR;
5233}
5234
5235void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
5236
5237    LOGV("removeEffect_l() %p effect %p", this, effect.get());
5238    effect_descriptor_t desc = effect->desc();
5239    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5240        detachAuxEffect_l(effect->id());
5241    }
5242
5243    sp<EffectChain> chain = effect->chain().promote();
5244    if (chain != 0) {
5245        // remove effect chain if removing last effect
5246        if (chain->removeEffect_l(effect) == 0) {
5247            removeEffectChain_l(chain);
5248        }
5249    } else {
5250        LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
5251    }
5252}
5253
5254void AudioFlinger::ThreadBase::lockEffectChains_l(
5255        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5256{
5257    effectChains = mEffectChains;
5258    for (size_t i = 0; i < mEffectChains.size(); i++) {
5259        mEffectChains[i]->lock();
5260    }
5261}
5262
5263void AudioFlinger::ThreadBase::unlockEffectChains(
5264        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
5265{
5266    for (size_t i = 0; i < effectChains.size(); i++) {
5267        effectChains[i]->unlock();
5268    }
5269}
5270
5271sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
5272{
5273    Mutex::Autolock _l(mLock);
5274    return getEffectChain_l(sessionId);
5275}
5276
5277sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
5278{
5279    sp<EffectChain> chain;
5280
5281    size_t size = mEffectChains.size();
5282    for (size_t i = 0; i < size; i++) {
5283        if (mEffectChains[i]->sessionId() == sessionId) {
5284            chain = mEffectChains[i];
5285            break;
5286        }
5287    }
5288    return chain;
5289}
5290
5291void AudioFlinger::ThreadBase::setMode(uint32_t mode)
5292{
5293    Mutex::Autolock _l(mLock);
5294    size_t size = mEffectChains.size();
5295    for (size_t i = 0; i < size; i++) {
5296        mEffectChains[i]->setMode_l(mode);
5297    }
5298}
5299
5300void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
5301                                                    const wp<EffectHandle>& handle) {
5302    Mutex::Autolock _l(mLock);
5303    LOGV("disconnectEffect() %p effect %p", this, effect.get());
5304    // delete the effect module if removing last handle on it
5305    if (effect->removeHandle(handle) == 0) {
5306        removeEffect_l(effect);
5307        AudioSystem::unregisterEffect(effect->id());
5308    }
5309}
5310
5311status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
5312{
5313    int session = chain->sessionId();
5314    int16_t *buffer = mMixBuffer;
5315    bool ownsBuffer = false;
5316
5317    LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
5318    if (session > 0) {
5319        // Only one effect chain can be present in direct output thread and it uses
5320        // the mix buffer as input
5321        if (mType != DIRECT) {
5322            size_t numSamples = mFrameCount * mChannelCount;
5323            buffer = new int16_t[numSamples];
5324            memset(buffer, 0, numSamples * sizeof(int16_t));
5325            LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
5326            ownsBuffer = true;
5327        }
5328
5329        // Attach all tracks with same session ID to this chain.
5330        for (size_t i = 0; i < mTracks.size(); ++i) {
5331            sp<Track> track = mTracks[i];
5332            if (session == track->sessionId()) {
5333                LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
5334                track->setMainBuffer(buffer);
5335                chain->incTrackCnt();
5336            }
5337        }
5338
5339        // indicate all active tracks in the chain
5340        for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5341            sp<Track> track = mActiveTracks[i].promote();
5342            if (track == 0) continue;
5343            if (session == track->sessionId()) {
5344                LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
5345                chain->incActiveTrackCnt();
5346            }
5347        }
5348    }
5349
5350    chain->setInBuffer(buffer, ownsBuffer);
5351    chain->setOutBuffer(mMixBuffer);
5352    // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
5353    // chains list in order to be processed last as it contains output stage effects
5354    // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
5355    // session AUDIO_SESSION_OUTPUT_STAGE to be processed
5356    // after track specific effects and before output stage
5357    // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
5358    // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
5359    // Effect chain for other sessions are inserted at beginning of effect
5360    // chains list to be processed before output mix effects. Relative order between other
5361    // sessions is not important
5362    size_t size = mEffectChains.size();
5363    size_t i = 0;
5364    for (i = 0; i < size; i++) {
5365        if (mEffectChains[i]->sessionId() < session) break;
5366    }
5367    mEffectChains.insertAt(chain, i);
5368
5369    return NO_ERROR;
5370}
5371
5372size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
5373{
5374    int session = chain->sessionId();
5375
5376    LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
5377
5378    for (size_t i = 0; i < mEffectChains.size(); i++) {
5379        if (chain == mEffectChains[i]) {
5380            mEffectChains.removeAt(i);
5381            // detach all active tracks from the chain
5382            for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
5383                sp<Track> track = mActiveTracks[i].promote();
5384                if (track == 0) continue;
5385                if (session == track->sessionId()) {
5386                    LOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
5387                            chain.get(), session);
5388                    chain->decActiveTrackCnt();
5389                }
5390            }
5391
5392            // detach all tracks with same session ID from this chain
5393            for (size_t i = 0; i < mTracks.size(); ++i) {
5394                sp<Track> track = mTracks[i];
5395                if (session == track->sessionId()) {
5396                    track->setMainBuffer(mMixBuffer);
5397                    chain->decTrackCnt();
5398                }
5399            }
5400            break;
5401        }
5402    }
5403    return mEffectChains.size();
5404}
5405
5406status_t AudioFlinger::PlaybackThread::attachAuxEffect(
5407        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
5408{
5409    Mutex::Autolock _l(mLock);
5410    return attachAuxEffect_l(track, EffectId);
5411}
5412
5413status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
5414        const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
5415{
5416    status_t status = NO_ERROR;
5417
5418    if (EffectId == 0) {
5419        track->setAuxBuffer(0, NULL);
5420    } else {
5421        // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
5422        sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
5423        if (effect != 0) {
5424            if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5425                track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
5426            } else {
5427                status = INVALID_OPERATION;
5428            }
5429        } else {
5430            status = BAD_VALUE;
5431        }
5432    }
5433    return status;
5434}
5435
5436void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
5437{
5438     for (size_t i = 0; i < mTracks.size(); ++i) {
5439        sp<Track> track = mTracks[i];
5440        if (track->auxEffectId() == effectId) {
5441            attachAuxEffect_l(track, 0);
5442        }
5443    }
5444}
5445
5446status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
5447{
5448    // only one chain per input thread
5449    if (mEffectChains.size() != 0) {
5450        return INVALID_OPERATION;
5451    }
5452    LOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
5453
5454    chain->setInBuffer(NULL);
5455    chain->setOutBuffer(NULL);
5456
5457    mEffectChains.add(chain);
5458
5459    return NO_ERROR;
5460}
5461
5462size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
5463{
5464    LOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
5465    LOGW_IF(mEffectChains.size() != 1,
5466            "removeEffectChain_l() %p invalid chain size %d on thread %p",
5467            chain.get(), mEffectChains.size(), this);
5468    if (mEffectChains.size() == 1) {
5469        mEffectChains.removeAt(0);
5470    }
5471    return 0;
5472}
5473
5474// ----------------------------------------------------------------------------
5475//  EffectModule implementation
5476// ----------------------------------------------------------------------------
5477
5478#undef LOG_TAG
5479#define LOG_TAG "AudioFlinger::EffectModule"
5480
5481AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
5482                                        const wp<AudioFlinger::EffectChain>& chain,
5483                                        effect_descriptor_t *desc,
5484                                        int id,
5485                                        int sessionId)
5486    : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
5487      mStatus(NO_INIT), mState(IDLE)
5488{
5489    LOGV("Constructor %p", this);
5490    int lStatus;
5491    sp<ThreadBase> thread = mThread.promote();
5492    if (thread == 0) {
5493        return;
5494    }
5495
5496    memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
5497
5498    // create effect engine from effect factory
5499    mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
5500
5501    if (mStatus != NO_ERROR) {
5502        return;
5503    }
5504    lStatus = init();
5505    if (lStatus < 0) {
5506        mStatus = lStatus;
5507        goto Error;
5508    }
5509
5510    LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
5511    return;
5512Error:
5513    EffectRelease(mEffectInterface);
5514    mEffectInterface = NULL;
5515    LOGV("Constructor Error %d", mStatus);
5516}
5517
5518AudioFlinger::EffectModule::~EffectModule()
5519{
5520    LOGV("Destructor %p", this);
5521    if (mEffectInterface != NULL) {
5522        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
5523                (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
5524            sp<ThreadBase> thread = mThread.promote();
5525            if (thread != 0) {
5526                thread->stream()->remove_audio_effect(thread->stream(), mEffectInterface);
5527            }
5528        }
5529        // release effect engine
5530        EffectRelease(mEffectInterface);
5531    }
5532}
5533
5534status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
5535{
5536    status_t status;
5537
5538    Mutex::Autolock _l(mLock);
5539    // First handle in mHandles has highest priority and controls the effect module
5540    int priority = handle->priority();
5541    size_t size = mHandles.size();
5542    sp<EffectHandle> h;
5543    size_t i;
5544    for (i = 0; i < size; i++) {
5545        h = mHandles[i].promote();
5546        if (h == 0) continue;
5547        if (h->priority() <= priority) break;
5548    }
5549    // if inserted in first place, move effect control from previous owner to this handle
5550    if (i == 0) {
5551        if (h != 0) {
5552            h->setControl(false, true);
5553        }
5554        handle->setControl(true, false);
5555        status = NO_ERROR;
5556    } else {
5557        status = ALREADY_EXISTS;
5558    }
5559    mHandles.insertAt(handle, i);
5560    return status;
5561}
5562
5563size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
5564{
5565    Mutex::Autolock _l(mLock);
5566    size_t size = mHandles.size();
5567    size_t i;
5568    for (i = 0; i < size; i++) {
5569        if (mHandles[i] == handle) break;
5570    }
5571    if (i == size) {
5572        return size;
5573    }
5574    mHandles.removeAt(i);
5575    size = mHandles.size();
5576    // if removed from first place, move effect control from this handle to next in line
5577    if (i == 0 && size != 0) {
5578        sp<EffectHandle> h = mHandles[0].promote();
5579        if (h != 0) {
5580            h->setControl(true, true);
5581        }
5582    }
5583
5584    // Release effect engine here so that it is done immediately. Otherwise it will be released
5585    // by the destructor when the last strong reference on the this object is released which can
5586    // happen after next process is called on this effect.
5587    if (size == 0 && mEffectInterface != NULL) {
5588        // release effect engine
5589        EffectRelease(mEffectInterface);
5590        mEffectInterface = NULL;
5591    }
5592
5593    return size;
5594}
5595
5596void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle)
5597{
5598    // keep a strong reference on this EffectModule to avoid calling the
5599    // destructor before we exit
5600    sp<EffectModule> keep(this);
5601    {
5602        sp<ThreadBase> thread = mThread.promote();
5603        if (thread != 0) {
5604            thread->disconnectEffect(keep, handle);
5605        }
5606    }
5607}
5608
5609void AudioFlinger::EffectModule::updateState() {
5610    Mutex::Autolock _l(mLock);
5611
5612    switch (mState) {
5613    case RESTART:
5614        reset_l();
5615        // FALL THROUGH
5616
5617    case STARTING:
5618        // clear auxiliary effect input buffer for next accumulation
5619        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5620            memset(mConfig.inputCfg.buffer.raw,
5621                   0,
5622                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
5623        }
5624        start_l();
5625        mState = ACTIVE;
5626        break;
5627    case STOPPING:
5628        stop_l();
5629        mDisableWaitCnt = mMaxDisableWaitCnt;
5630        mState = STOPPED;
5631        break;
5632    case STOPPED:
5633        // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
5634        // turn off sequence.
5635        if (--mDisableWaitCnt == 0) {
5636            reset_l();
5637            mState = IDLE;
5638        }
5639        break;
5640    default: //IDLE , ACTIVE
5641        break;
5642    }
5643}
5644
5645void AudioFlinger::EffectModule::process()
5646{
5647    Mutex::Autolock _l(mLock);
5648
5649    if (mEffectInterface == NULL ||
5650            mConfig.inputCfg.buffer.raw == NULL ||
5651            mConfig.outputCfg.buffer.raw == NULL) {
5652        return;
5653    }
5654
5655    if (isProcessEnabled()) {
5656        // do 32 bit to 16 bit conversion for auxiliary effect input buffer
5657        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5658            AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
5659                                        mConfig.inputCfg.buffer.s32,
5660                                        mConfig.inputCfg.buffer.frameCount/2);
5661        }
5662
5663        // do the actual processing in the effect engine
5664        int ret = (*mEffectInterface)->process(mEffectInterface,
5665                                               &mConfig.inputCfg.buffer,
5666                                               &mConfig.outputCfg.buffer);
5667
5668        // force transition to IDLE state when engine is ready
5669        if (mState == STOPPED && ret == -ENODATA) {
5670            mDisableWaitCnt = 1;
5671        }
5672
5673        // clear auxiliary effect input buffer for next accumulation
5674        if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5675            memset(mConfig.inputCfg.buffer.raw, 0,
5676                   mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
5677        }
5678    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
5679                mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
5680        // If an insert effect is idle and input buffer is different from output buffer,
5681        // accumulate input onto output
5682        sp<EffectChain> chain = mChain.promote();
5683        if (chain != 0 && chain->activeTrackCnt() != 0) {
5684            size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2;  //always stereo here
5685            int16_t *in = mConfig.inputCfg.buffer.s16;
5686            int16_t *out = mConfig.outputCfg.buffer.s16;
5687            for (size_t i = 0; i < frameCnt; i++) {
5688                out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
5689            }
5690        }
5691    }
5692}
5693
5694void AudioFlinger::EffectModule::reset_l()
5695{
5696    if (mEffectInterface == NULL) {
5697        return;
5698    }
5699    (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
5700}
5701
5702status_t AudioFlinger::EffectModule::configure()
5703{
5704    uint32_t channels;
5705    if (mEffectInterface == NULL) {
5706        return NO_INIT;
5707    }
5708
5709    sp<ThreadBase> thread = mThread.promote();
5710    if (thread == 0) {
5711        return DEAD_OBJECT;
5712    }
5713
5714    // TODO: handle configuration of effects replacing track process
5715    if (thread->channelCount() == 1) {
5716        channels = AUDIO_CHANNEL_OUT_MONO;
5717    } else {
5718        channels = AUDIO_CHANNEL_OUT_STEREO;
5719    }
5720
5721    if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
5722        mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
5723    } else {
5724        mConfig.inputCfg.channels = channels;
5725    }
5726    mConfig.outputCfg.channels = channels;
5727    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
5728    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
5729    mConfig.inputCfg.samplingRate = thread->sampleRate();
5730    mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
5731    mConfig.inputCfg.bufferProvider.cookie = NULL;
5732    mConfig.inputCfg.bufferProvider.getBuffer = NULL;
5733    mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
5734    mConfig.outputCfg.bufferProvider.cookie = NULL;
5735    mConfig.outputCfg.bufferProvider.getBuffer = NULL;
5736    mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
5737    mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
5738    // Insert effect:
5739    // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
5740    // always overwrites output buffer: input buffer == output buffer
5741    // - in other sessions:
5742    //      last effect in the chain accumulates in output buffer: input buffer != output buffer
5743    //      other effect: overwrites output buffer: input buffer == output buffer
5744    // Auxiliary effect:
5745    //      accumulates in output buffer: input buffer != output buffer
5746    // Therefore: accumulate <=> input buffer != output buffer
5747    if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
5748        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
5749    } else {
5750        mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
5751    }
5752    mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
5753    mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
5754    mConfig.inputCfg.buffer.frameCount = thread->frameCount();
5755    mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
5756
5757    LOGV("configure() %p thread %p buffer %p framecount %d",
5758            this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
5759
5760    status_t cmdStatus;
5761    uint32_t size = sizeof(int);
5762    status_t status = (*mEffectInterface)->command(mEffectInterface,
5763                                                   EFFECT_CMD_CONFIGURE,
5764                                                   sizeof(effect_config_t),
5765                                                   &mConfig,
5766                                                   &size,
5767                                                   &cmdStatus);
5768    if (status == 0) {
5769        status = cmdStatus;
5770    }
5771
5772    mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
5773            (1000 * mConfig.outputCfg.buffer.frameCount);
5774
5775    return status;
5776}
5777
5778status_t AudioFlinger::EffectModule::init()
5779{
5780    Mutex::Autolock _l(mLock);
5781    if (mEffectInterface == NULL) {
5782        return NO_INIT;
5783    }
5784    status_t cmdStatus;
5785    uint32_t size = sizeof(status_t);
5786    status_t status = (*mEffectInterface)->command(mEffectInterface,
5787                                                   EFFECT_CMD_INIT,
5788                                                   0,
5789                                                   NULL,
5790                                                   &size,
5791                                                   &cmdStatus);
5792    if (status == 0) {
5793        status = cmdStatus;
5794    }
5795    return status;
5796}
5797
5798status_t AudioFlinger::EffectModule::start_l()
5799{
5800    if (mEffectInterface == NULL) {
5801        return NO_INIT;
5802    }
5803    status_t cmdStatus;
5804    uint32_t size = sizeof(status_t);
5805    status_t status = (*mEffectInterface)->command(mEffectInterface,
5806                                                   EFFECT_CMD_ENABLE,
5807                                                   0,
5808                                                   NULL,
5809                                                   &size,
5810                                                   &cmdStatus);
5811    if (status == 0) {
5812        status = cmdStatus;
5813    }
5814    if (status == 0 &&
5815            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
5816             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
5817        sp<ThreadBase> thread = mThread.promote();
5818        if (thread != 0) {
5819            thread->stream()->add_audio_effect(thread->stream(), mEffectInterface);
5820        }
5821    }
5822    return status;
5823}
5824
5825status_t AudioFlinger::EffectModule::stop_l()
5826{
5827    if (mEffectInterface == NULL) {
5828        return NO_INIT;
5829    }
5830    status_t cmdStatus;
5831    uint32_t size = sizeof(status_t);
5832    status_t status = (*mEffectInterface)->command(mEffectInterface,
5833                                                   EFFECT_CMD_DISABLE,
5834                                                   0,
5835                                                   NULL,
5836                                                   &size,
5837                                                   &cmdStatus);
5838    if (status == 0) {
5839        status = cmdStatus;
5840    }
5841    if (status == 0 &&
5842            ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
5843             (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
5844        sp<ThreadBase> thread = mThread.promote();
5845        if (thread != 0) {
5846            thread->stream()->remove_audio_effect(thread->stream(), mEffectInterface);
5847        }
5848    }
5849    return status;
5850}
5851
5852status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
5853                                             uint32_t cmdSize,
5854                                             void *pCmdData,
5855                                             uint32_t *replySize,
5856                                             void *pReplyData)
5857{
5858    Mutex::Autolock _l(mLock);
5859//    LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
5860
5861    if (mEffectInterface == NULL) {
5862        return NO_INIT;
5863    }
5864    status_t status = (*mEffectInterface)->command(mEffectInterface,
5865                                                   cmdCode,
5866                                                   cmdSize,
5867                                                   pCmdData,
5868                                                   replySize,
5869                                                   pReplyData);
5870    if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
5871        uint32_t size = (replySize == NULL) ? 0 : *replySize;
5872        for (size_t i = 1; i < mHandles.size(); i++) {
5873            sp<EffectHandle> h = mHandles[i].promote();
5874            if (h != 0) {
5875                h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
5876            }
5877        }
5878    }
5879    return status;
5880}
5881
5882status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
5883{
5884    Mutex::Autolock _l(mLock);
5885    LOGV("setEnabled %p enabled %d", this, enabled);
5886
5887    if (enabled != isEnabled()) {
5888        switch (mState) {
5889        // going from disabled to enabled
5890        case IDLE:
5891            mState = STARTING;
5892            break;
5893        case STOPPED:
5894            mState = RESTART;
5895            break;
5896        case STOPPING:
5897            mState = ACTIVE;
5898            break;
5899
5900        // going from enabled to disabled
5901        case RESTART:
5902            mState = STOPPED;
5903            break;
5904        case STARTING:
5905            mState = IDLE;
5906            break;
5907        case ACTIVE:
5908            mState = STOPPING;
5909            break;
5910        }
5911        for (size_t i = 1; i < mHandles.size(); i++) {
5912            sp<EffectHandle> h = mHandles[i].promote();
5913            if (h != 0) {
5914                h->setEnabled(enabled);
5915            }
5916        }
5917    }
5918    return NO_ERROR;
5919}
5920
5921bool AudioFlinger::EffectModule::isEnabled()
5922{
5923    switch (mState) {
5924    case RESTART:
5925    case STARTING:
5926    case ACTIVE:
5927        return true;
5928    case IDLE:
5929    case STOPPING:
5930    case STOPPED:
5931    default:
5932        return false;
5933    }
5934}
5935
5936bool AudioFlinger::EffectModule::isProcessEnabled()
5937{
5938    switch (mState) {
5939    case RESTART:
5940    case ACTIVE:
5941    case STOPPING:
5942    case STOPPED:
5943        return true;
5944    case IDLE:
5945    case STARTING:
5946    default:
5947        return false;
5948    }
5949}
5950
5951status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
5952{
5953    Mutex::Autolock _l(mLock);
5954    status_t status = NO_ERROR;
5955
5956    // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
5957    // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
5958    if (isProcessEnabled() &&
5959            ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
5960            (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
5961        status_t cmdStatus;
5962        uint32_t volume[2];
5963        uint32_t *pVolume = NULL;
5964        uint32_t size = sizeof(volume);
5965        volume[0] = *left;
5966        volume[1] = *right;
5967        if (controller) {
5968            pVolume = volume;
5969        }
5970        status = (*mEffectInterface)->command(mEffectInterface,
5971                                              EFFECT_CMD_SET_VOLUME,
5972                                              size,
5973                                              volume,
5974                                              &size,
5975                                              pVolume);
5976        if (controller && status == NO_ERROR && size == sizeof(volume)) {
5977            *left = volume[0];
5978            *right = volume[1];
5979        }
5980    }
5981    return status;
5982}
5983
5984status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
5985{
5986    Mutex::Autolock _l(mLock);
5987    status_t status = NO_ERROR;
5988    if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
5989        // audio pre processing modules on RecordThread can receive both output and
5990        // input device indication in the same call
5991        uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
5992        if (dev) {
5993            status_t cmdStatus;
5994            uint32_t size = sizeof(status_t);
5995
5996            status = (*mEffectInterface)->command(mEffectInterface,
5997                                                  EFFECT_CMD_SET_DEVICE,
5998                                                  sizeof(uint32_t),
5999                                                  &dev,
6000                                                  &size,
6001                                                  &cmdStatus);
6002            if (status == NO_ERROR) {
6003                status = cmdStatus;
6004            }
6005        }
6006        dev = device & AUDIO_DEVICE_IN_ALL;
6007        if (dev) {
6008            status_t cmdStatus;
6009            uint32_t size = sizeof(status_t);
6010
6011            status_t status2 = (*mEffectInterface)->command(mEffectInterface,
6012                                                  EFFECT_CMD_SET_INPUT_DEVICE,
6013                                                  sizeof(uint32_t),
6014                                                  &dev,
6015                                                  &size,
6016                                                  &cmdStatus);
6017            if (status2 == NO_ERROR) {
6018                status2 = cmdStatus;
6019            }
6020            if (status == NO_ERROR) {
6021                status = status2;
6022            }
6023        }
6024    }
6025    return status;
6026}
6027
6028status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
6029{
6030    Mutex::Autolock _l(mLock);
6031    status_t status = NO_ERROR;
6032    if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
6033        status_t cmdStatus;
6034        uint32_t size = sizeof(status_t);
6035        status = (*mEffectInterface)->command(mEffectInterface,
6036                                              EFFECT_CMD_SET_AUDIO_MODE,
6037                                              sizeof(int),
6038                                              &mode,
6039                                              &size,
6040                                              &cmdStatus);
6041        if (status == NO_ERROR) {
6042            status = cmdStatus;
6043        }
6044    }
6045    return status;
6046}
6047
6048status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
6049{
6050    const size_t SIZE = 256;
6051    char buffer[SIZE];
6052    String8 result;
6053
6054    snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
6055    result.append(buffer);
6056
6057    bool locked = tryLock(mLock);
6058    // failed to lock - AudioFlinger is probably deadlocked
6059    if (!locked) {
6060        result.append("\t\tCould not lock Fx mutex:\n");
6061    }
6062
6063    result.append("\t\tSession Status State Engine:\n");
6064    snprintf(buffer, SIZE, "\t\t%05d   %03d    %03d   0x%08x\n",
6065            mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
6066    result.append(buffer);
6067
6068    result.append("\t\tDescriptor:\n");
6069    snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6070            mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
6071            mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
6072            mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
6073    result.append(buffer);
6074    snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
6075                mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
6076                mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
6077                mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
6078    result.append(buffer);
6079    snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
6080            mDescriptor.apiVersion,
6081            mDescriptor.flags);
6082    result.append(buffer);
6083    snprintf(buffer, SIZE, "\t\t- name: %s\n",
6084            mDescriptor.name);
6085    result.append(buffer);
6086    snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
6087            mDescriptor.implementor);
6088    result.append(buffer);
6089
6090    result.append("\t\t- Input configuration:\n");
6091    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6092    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6093            (uint32_t)mConfig.inputCfg.buffer.raw,
6094            mConfig.inputCfg.buffer.frameCount,
6095            mConfig.inputCfg.samplingRate,
6096            mConfig.inputCfg.channels,
6097            mConfig.inputCfg.format);
6098    result.append(buffer);
6099
6100    result.append("\t\t- Output configuration:\n");
6101    result.append("\t\t\tBuffer     Frames  Smp rate Channels Format\n");
6102    snprintf(buffer, SIZE, "\t\t\t0x%08x %05d   %05d    %08x %d\n",
6103            (uint32_t)mConfig.outputCfg.buffer.raw,
6104            mConfig.outputCfg.buffer.frameCount,
6105            mConfig.outputCfg.samplingRate,
6106            mConfig.outputCfg.channels,
6107            mConfig.outputCfg.format);
6108    result.append(buffer);
6109
6110    snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
6111    result.append(buffer);
6112    result.append("\t\t\tPid   Priority Ctrl Locked client server\n");
6113    for (size_t i = 0; i < mHandles.size(); ++i) {
6114        sp<EffectHandle> handle = mHandles[i].promote();
6115        if (handle != 0) {
6116            handle->dump(buffer, SIZE);
6117            result.append(buffer);
6118        }
6119    }
6120
6121    result.append("\n");
6122
6123    write(fd, result.string(), result.length());
6124
6125    if (locked) {
6126        mLock.unlock();
6127    }
6128
6129    return NO_ERROR;
6130}
6131
6132// ----------------------------------------------------------------------------
6133//  EffectHandle implementation
6134// ----------------------------------------------------------------------------
6135
6136#undef LOG_TAG
6137#define LOG_TAG "AudioFlinger::EffectHandle"
6138
6139AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
6140                                        const sp<AudioFlinger::Client>& client,
6141                                        const sp<IEffectClient>& effectClient,
6142                                        int32_t priority)
6143    : BnEffect(),
6144    mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false)
6145{
6146    LOGV("constructor %p", this);
6147
6148    int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
6149    mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
6150    if (mCblkMemory != 0) {
6151        mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
6152
6153        if (mCblk) {
6154            new(mCblk) effect_param_cblk_t();
6155            mBuffer = (uint8_t *)mCblk + bufOffset;
6156         }
6157    } else {
6158        LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
6159        return;
6160    }
6161}
6162
6163AudioFlinger::EffectHandle::~EffectHandle()
6164{
6165    LOGV("Destructor %p", this);
6166    disconnect();
6167}
6168
6169status_t AudioFlinger::EffectHandle::enable()
6170{
6171    if (!mHasControl) return INVALID_OPERATION;
6172    if (mEffect == 0) return DEAD_OBJECT;
6173
6174    return mEffect->setEnabled(true);
6175}
6176
6177status_t AudioFlinger::EffectHandle::disable()
6178{
6179    if (!mHasControl) return INVALID_OPERATION;
6180    if (mEffect == NULL) return DEAD_OBJECT;
6181
6182    return mEffect->setEnabled(false);
6183}
6184
6185void AudioFlinger::EffectHandle::disconnect()
6186{
6187    if (mEffect == 0) {
6188        return;
6189    }
6190    mEffect->disconnect(this);
6191    // release sp on module => module destructor can be called now
6192    mEffect.clear();
6193    if (mCblk) {
6194        mCblk->~effect_param_cblk_t();   // destroy our shared-structure.
6195    }
6196    mCblkMemory.clear();            // and free the shared memory
6197    if (mClient != 0) {
6198        Mutex::Autolock _l(mClient->audioFlinger()->mLock);
6199        mClient.clear();
6200    }
6201}
6202
6203status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
6204                                             uint32_t cmdSize,
6205                                             void *pCmdData,
6206                                             uint32_t *replySize,
6207                                             void *pReplyData)
6208{
6209//    LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
6210//              cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
6211
6212    // only get parameter command is permitted for applications not controlling the effect
6213    if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
6214        return INVALID_OPERATION;
6215    }
6216    if (mEffect == 0) return DEAD_OBJECT;
6217
6218    // handle commands that are not forwarded transparently to effect engine
6219    if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
6220        // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
6221        // no risk to block the whole media server process or mixer threads is we are stuck here
6222        Mutex::Autolock _l(mCblk->lock);
6223        if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
6224            mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
6225            mCblk->serverIndex = 0;
6226            mCblk->clientIndex = 0;
6227            return BAD_VALUE;
6228        }
6229        status_t status = NO_ERROR;
6230        while (mCblk->serverIndex < mCblk->clientIndex) {
6231            int reply;
6232            uint32_t rsize = sizeof(int);
6233            int *p = (int *)(mBuffer + mCblk->serverIndex);
6234            int size = *p++;
6235            if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
6236                LOGW("command(): invalid parameter block size");
6237                break;
6238            }
6239            effect_param_t *param = (effect_param_t *)p;
6240            if (param->psize == 0 || param->vsize == 0) {
6241                LOGW("command(): null parameter or value size");
6242                mCblk->serverIndex += size;
6243                continue;
6244            }
6245            uint32_t psize = sizeof(effect_param_t) +
6246                             ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
6247                             param->vsize;
6248            status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
6249                                            psize,
6250                                            p,
6251                                            &rsize,
6252                                            &reply);
6253            // stop at first error encountered
6254            if (ret != NO_ERROR) {
6255                status = ret;
6256                *(int *)pReplyData = reply;
6257                break;
6258            } else if (reply != NO_ERROR) {
6259                *(int *)pReplyData = reply;
6260                break;
6261            }
6262            mCblk->serverIndex += size;
6263        }
6264        mCblk->serverIndex = 0;
6265        mCblk->clientIndex = 0;
6266        return status;
6267    } else if (cmdCode == EFFECT_CMD_ENABLE) {
6268        *(int *)pReplyData = NO_ERROR;
6269        return enable();
6270    } else if (cmdCode == EFFECT_CMD_DISABLE) {
6271        *(int *)pReplyData = NO_ERROR;
6272        return disable();
6273    }
6274
6275    return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
6276}
6277
6278sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
6279    return mCblkMemory;
6280}
6281
6282void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal)
6283{
6284    LOGV("setControl %p control %d", this, hasControl);
6285
6286    mHasControl = hasControl;
6287    if (signal && mEffectClient != 0) {
6288        mEffectClient->controlStatusChanged(hasControl);
6289    }
6290}
6291
6292void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
6293                                                 uint32_t cmdSize,
6294                                                 void *pCmdData,
6295                                                 uint32_t replySize,
6296                                                 void *pReplyData)
6297{
6298    if (mEffectClient != 0) {
6299        mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
6300    }
6301}
6302
6303
6304
6305void AudioFlinger::EffectHandle::setEnabled(bool enabled)
6306{
6307    if (mEffectClient != 0) {
6308        mEffectClient->enableStatusChanged(enabled);
6309    }
6310}
6311
6312status_t AudioFlinger::EffectHandle::onTransact(
6313    uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
6314{
6315    return BnEffect::onTransact(code, data, reply, flags);
6316}
6317
6318
6319void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
6320{
6321    bool locked = tryLock(mCblk->lock);
6322
6323    snprintf(buffer, size, "\t\t\t%05d %05d    %01u    %01u      %05u  %05u\n",
6324            (mClient == NULL) ? getpid() : mClient->pid(),
6325            mPriority,
6326            mHasControl,
6327            !locked,
6328            mCblk->clientIndex,
6329            mCblk->serverIndex
6330            );
6331
6332    if (locked) {
6333        mCblk->lock.unlock();
6334    }
6335}
6336
6337#undef LOG_TAG
6338#define LOG_TAG "AudioFlinger::EffectChain"
6339
6340AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
6341                                        int sessionId)
6342    : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0),
6343      mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
6344      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
6345{
6346    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
6347}
6348
6349AudioFlinger::EffectChain::~EffectChain()
6350{
6351    if (mOwnInBuffer) {
6352        delete mInBuffer;
6353    }
6354
6355}
6356
6357// getEffectFromDesc_l() must be called with PlaybackThread::mLock held
6358sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
6359{
6360    sp<EffectModule> effect;
6361    size_t size = mEffects.size();
6362
6363    for (size_t i = 0; i < size; i++) {
6364        if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
6365            effect = mEffects[i];
6366            break;
6367        }
6368    }
6369    return effect;
6370}
6371
6372// getEffectFromId_l() must be called with PlaybackThread::mLock held
6373sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
6374{
6375    sp<EffectModule> effect;
6376    size_t size = mEffects.size();
6377
6378    for (size_t i = 0; i < size; i++) {
6379        // by convention, return first effect if id provided is 0 (0 is never a valid id)
6380        if (id == 0 || mEffects[i]->id() == id) {
6381            effect = mEffects[i];
6382            break;
6383        }
6384    }
6385    return effect;
6386}
6387
6388// Must be called with EffectChain::mLock locked
6389void AudioFlinger::EffectChain::process_l()
6390{
6391    sp<ThreadBase> thread = mThread.promote();
6392    if (thread == 0) {
6393        LOGW("process_l(): cannot promote mixer thread");
6394        return;
6395    }
6396    bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
6397            (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
6398    bool tracksOnSession = false;
6399    if (!isGlobalSession) {
6400        tracksOnSession = (trackCnt() != 0);
6401    }
6402
6403    // if no track is active, input buffer must be cleared here as the mixer process
6404    // will not do it
6405    if (tracksOnSession &&
6406            activeTrackCnt() == 0) {
6407        size_t numSamples = thread->frameCount() * thread->channelCount();
6408        memset(mInBuffer, 0, numSamples * sizeof(int16_t));
6409    }
6410
6411    size_t size = mEffects.size();
6412    // do not process effect if no track is present in same audio session
6413    if (isGlobalSession || tracksOnSession) {
6414        for (size_t i = 0; i < size; i++) {
6415            mEffects[i]->process();
6416        }
6417    }
6418    for (size_t i = 0; i < size; i++) {
6419        mEffects[i]->updateState();
6420    }
6421}
6422
6423// addEffect_l() must be called with PlaybackThread::mLock held
6424status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
6425{
6426    effect_descriptor_t desc = effect->desc();
6427    uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
6428
6429    Mutex::Autolock _l(mLock);
6430    effect->setChain(this);
6431    sp<ThreadBase> thread = mThread.promote();
6432    if (thread == 0) {
6433        return NO_INIT;
6434    }
6435    effect->setThread(thread);
6436
6437    if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
6438        // Auxiliary effects are inserted at the beginning of mEffects vector as
6439        // they are processed first and accumulated in chain input buffer
6440        mEffects.insertAt(effect, 0);
6441
6442        // the input buffer for auxiliary effect contains mono samples in
6443        // 32 bit format. This is to avoid saturation in AudoMixer
6444        // accumulation stage. Saturation is done in EffectModule::process() before
6445        // calling the process in effect engine
6446        size_t numSamples = thread->frameCount();
6447        int32_t *buffer = new int32_t[numSamples];
6448        memset(buffer, 0, numSamples * sizeof(int32_t));
6449        effect->setInBuffer((int16_t *)buffer);
6450        // auxiliary effects output samples to chain input buffer for further processing
6451        // by insert effects
6452        effect->setOutBuffer(mInBuffer);
6453    } else {
6454        // Insert effects are inserted at the end of mEffects vector as they are processed
6455        //  after track and auxiliary effects.
6456        // Insert effect order as a function of indicated preference:
6457        //  if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
6458        //  another effect is present
6459        //  else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
6460        //  last effect claiming first position
6461        //  else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
6462        //  first effect claiming last position
6463        //  else if EFFECT_FLAG_INSERT_ANY insert after first or before last
6464        // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
6465        // already present
6466
6467        int size = (int)mEffects.size();
6468        int idx_insert = size;
6469        int idx_insert_first = -1;
6470        int idx_insert_last = -1;
6471
6472        for (int i = 0; i < size; i++) {
6473            effect_descriptor_t d = mEffects[i]->desc();
6474            uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
6475            uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
6476            if (iMode == EFFECT_FLAG_TYPE_INSERT) {
6477                // check invalid effect chaining combinations
6478                if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
6479                    iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
6480                    LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
6481                    return INVALID_OPERATION;
6482                }
6483                // remember position of first insert effect and by default
6484                // select this as insert position for new effect
6485                if (idx_insert == size) {
6486                    idx_insert = i;
6487                }
6488                // remember position of last insert effect claiming
6489                // first position
6490                if (iPref == EFFECT_FLAG_INSERT_FIRST) {
6491                    idx_insert_first = i;
6492                }
6493                // remember position of first insert effect claiming
6494                // last position
6495                if (iPref == EFFECT_FLAG_INSERT_LAST &&
6496                    idx_insert_last == -1) {
6497                    idx_insert_last = i;
6498                }
6499            }
6500        }
6501
6502        // modify idx_insert from first position if needed
6503        if (insertPref == EFFECT_FLAG_INSERT_LAST) {
6504            if (idx_insert_last != -1) {
6505                idx_insert = idx_insert_last;
6506            } else {
6507                idx_insert = size;
6508            }
6509        } else {
6510            if (idx_insert_first != -1) {
6511                idx_insert = idx_insert_first + 1;
6512            }
6513        }
6514
6515        // always read samples from chain input buffer
6516        effect->setInBuffer(mInBuffer);
6517
6518        // if last effect in the chain, output samples to chain
6519        // output buffer, otherwise to chain input buffer
6520        if (idx_insert == size) {
6521            if (idx_insert != 0) {
6522                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
6523                mEffects[idx_insert-1]->configure();
6524            }
6525            effect->setOutBuffer(mOutBuffer);
6526        } else {
6527            effect->setOutBuffer(mInBuffer);
6528        }
6529        mEffects.insertAt(effect, idx_insert);
6530
6531        LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
6532    }
6533    effect->configure();
6534    return NO_ERROR;
6535}
6536
6537// removeEffect_l() must be called with PlaybackThread::mLock held
6538size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
6539{
6540    Mutex::Autolock _l(mLock);
6541    int size = (int)mEffects.size();
6542    int i;
6543    uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
6544
6545    for (i = 0; i < size; i++) {
6546        if (effect == mEffects[i]) {
6547            if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
6548                delete[] effect->inBuffer();
6549            } else {
6550                if (i == size - 1 && i != 0) {
6551                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
6552                    mEffects[i - 1]->configure();
6553                }
6554            }
6555            mEffects.removeAt(i);
6556            LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
6557            break;
6558        }
6559    }
6560
6561    return mEffects.size();
6562}
6563
6564// setDevice_l() must be called with PlaybackThread::mLock held
6565void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
6566{
6567    size_t size = mEffects.size();
6568    for (size_t i = 0; i < size; i++) {
6569        mEffects[i]->setDevice(device);
6570    }
6571}
6572
6573// setMode_l() must be called with PlaybackThread::mLock held
6574void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
6575{
6576    size_t size = mEffects.size();
6577    for (size_t i = 0; i < size; i++) {
6578        mEffects[i]->setMode(mode);
6579    }
6580}
6581
6582// setVolume_l() must be called with PlaybackThread::mLock held
6583bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
6584{
6585    uint32_t newLeft = *left;
6586    uint32_t newRight = *right;
6587    bool hasControl = false;
6588    int ctrlIdx = -1;
6589    size_t size = mEffects.size();
6590
6591    // first update volume controller
6592    for (size_t i = size; i > 0; i--) {
6593        if (mEffects[i - 1]->isProcessEnabled() &&
6594            (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
6595            ctrlIdx = i - 1;
6596            hasControl = true;
6597            break;
6598        }
6599    }
6600
6601    if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
6602        if (hasControl) {
6603            *left = mNewLeftVolume;
6604            *right = mNewRightVolume;
6605        }
6606        return hasControl;
6607    }
6608
6609    mVolumeCtrlIdx = ctrlIdx;
6610    mLeftVolume = newLeft;
6611    mRightVolume = newRight;
6612
6613    // second get volume update from volume controller
6614    if (ctrlIdx >= 0) {
6615        mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
6616        mNewLeftVolume = newLeft;
6617        mNewRightVolume = newRight;
6618    }
6619    // then indicate volume to all other effects in chain.
6620    // Pass altered volume to effects before volume controller
6621    // and requested volume to effects after controller
6622    uint32_t lVol = newLeft;
6623    uint32_t rVol = newRight;
6624
6625    for (size_t i = 0; i < size; i++) {
6626        if ((int)i == ctrlIdx) continue;
6627        // this also works for ctrlIdx == -1 when there is no volume controller
6628        if ((int)i > ctrlIdx) {
6629            lVol = *left;
6630            rVol = *right;
6631        }
6632        mEffects[i]->setVolume(&lVol, &rVol, false);
6633    }
6634    *left = newLeft;
6635    *right = newRight;
6636
6637    return hasControl;
6638}
6639
6640status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
6641{
6642    const size_t SIZE = 256;
6643    char buffer[SIZE];
6644    String8 result;
6645
6646    snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
6647    result.append(buffer);
6648
6649    bool locked = tryLock(mLock);
6650    // failed to lock - AudioFlinger is probably deadlocked
6651    if (!locked) {
6652        result.append("\tCould not lock mutex:\n");
6653    }
6654
6655    result.append("\tNum fx In buffer   Out buffer   Active tracks:\n");
6656    snprintf(buffer, SIZE, "\t%02d     0x%08x  0x%08x   %d\n",
6657            mEffects.size(),
6658            (uint32_t)mInBuffer,
6659            (uint32_t)mOutBuffer,
6660            mActiveTrackCnt);
6661    result.append(buffer);
6662    write(fd, result.string(), result.size());
6663
6664    for (size_t i = 0; i < mEffects.size(); ++i) {
6665        sp<EffectModule> effect = mEffects[i];
6666        if (effect != 0) {
6667            effect->dump(fd, args);
6668        }
6669    }
6670
6671    if (locked) {
6672        mLock.unlock();
6673    }
6674
6675    return NO_ERROR;
6676}
6677
6678#undef LOG_TAG
6679#define LOG_TAG "AudioFlinger"
6680
6681// ----------------------------------------------------------------------------
6682
6683status_t AudioFlinger::onTransact(
6684        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
6685{
6686    return BnAudioFlinger::onTransact(code, data, reply, flags);
6687}
6688
6689}; // namespace android
6690