AudioFlinger.cpp revision 893a5642871114fca3b2a00c6ff8e5699ce3e3ed 
1/*
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 "Configuration.h"
23#include <dirent.h>
24#include <math.h>
25#include <signal.h>
26#include <sys/time.h>
27#include <sys/resource.h>
28
29#include <binder/IPCThreadState.h>
30#include <binder/IServiceManager.h>
31#include <utils/Log.h>
32#include <utils/Trace.h>
33#include <binder/Parcel.h>
34#include <utils/String16.h>
35#include <utils/threads.h>
36#include <utils/Atomic.h>
37
38#include <cutils/bitops.h>
39#include <cutils/properties.h>
40
41#include <system/audio.h>
42#include <hardware/audio.h>
43
44#include "AudioMixer.h"
45#include "AudioFlinger.h"
46#include "ServiceUtilities.h"
47
48#include <media/EffectsFactoryApi.h>
49#include <audio_effects/effect_visualizer.h>
50#include <audio_effects/effect_ns.h>
51#include <audio_effects/effect_aec.h>
52
53#include <audio_utils/primitives.h>
54
55#include <powermanager/PowerManager.h>
56
57#include <common_time/cc_helper.h>
58
59#include <media/IMediaLogService.h>
60
61#include <media/nbaio/Pipe.h>
62#include <media/nbaio/PipeReader.h>
63#include <media/AudioParameter.h>
64#include <private/android_filesystem_config.h>
65
66// ----------------------------------------------------------------------------
67
68// Note: the following macro is used for extremely verbose logging message.  In
69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
70// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
71// are so verbose that we want to suppress them even when we have ALOG_ASSERT
72// turned on.  Do not uncomment the #def below unless you really know what you
73// are doing and want to see all of the extremely verbose messages.
74//#define VERY_VERY_VERBOSE_LOGGING
75#ifdef VERY_VERY_VERBOSE_LOGGING
76#define ALOGVV ALOGV
77#else
78#define ALOGVV(a...) do { } while(0)
79#endif
80
81namespace android {
82
83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
84static const char kHardwareLockedString[] = "Hardware lock is taken\n";
85
86
87nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
88
89uint32_t AudioFlinger::mScreenState;
90
91#ifdef TEE_SINK
92bool AudioFlinger::mTeeSinkInputEnabled = false;
93bool AudioFlinger::mTeeSinkOutputEnabled = false;
94bool AudioFlinger::mTeeSinkTrackEnabled = false;
95
96size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
97size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
98size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
99#endif
100
101// ----------------------------------------------------------------------------
102
103static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
104{
105    const hw_module_t *mod;
106    int rc;
107
108    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
109    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
110                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
111    if (rc) {
112        goto out;
113    }
114    rc = audio_hw_device_open(mod, dev);
115    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
116                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
117    if (rc) {
118        goto out;
119    }
120    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
121        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
122        rc = BAD_VALUE;
123        goto out;
124    }
125    return 0;
126
127out:
128    *dev = NULL;
129    return rc;
130}
131
132// ----------------------------------------------------------------------------
133
134AudioFlinger::AudioFlinger()
135    : BnAudioFlinger(),
136      mPrimaryHardwareDev(NULL),
137      mHardwareStatus(AUDIO_HW_IDLE),
138      mMasterVolume(1.0f),
139      mMasterMute(false),
140      mNextUniqueId(1),
141      mMode(AUDIO_MODE_INVALID),
142      mBtNrecIsOff(false),
143      mIsLowRamDevice(true),
144      mIsDeviceTypeKnown(false)
145{
146    getpid_cached = getpid();
147    char value[PROPERTY_VALUE_MAX];
148    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
149    if (doLog) {
150        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
151    }
152#ifdef TEE_SINK
153    (void) property_get("ro.debuggable", value, "0");
154    int debuggable = atoi(value);
155    int teeEnabled = 0;
156    if (debuggable) {
157        (void) property_get("af.tee", value, "0");
158        teeEnabled = atoi(value);
159    }
160    if (teeEnabled & 1)
161        mTeeSinkInputEnabled = true;
162    if (teeEnabled & 2)
163        mTeeSinkOutputEnabled = true;
164    if (teeEnabled & 4)
165        mTeeSinkTrackEnabled = true;
166#endif
167}
168
169void AudioFlinger::onFirstRef()
170{
171    int rc = 0;
172
173    Mutex::Autolock _l(mLock);
174
175    /* TODO: move all this work into an Init() function */
176    char val_str[PROPERTY_VALUE_MAX] = { 0 };
177    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
178        uint32_t int_val;
179        if (1 == sscanf(val_str, "%u", &int_val)) {
180            mStandbyTimeInNsecs = milliseconds(int_val);
181            ALOGI("Using %u mSec as standby time.", int_val);
182        } else {
183            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
184            ALOGI("Using default %u mSec as standby time.",
185                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
186        }
187    }
188
189    mMode = AUDIO_MODE_NORMAL;
190}
191
192AudioFlinger::~AudioFlinger()
193{
194    while (!mRecordThreads.isEmpty()) {
195        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
196        closeInput_nonvirtual(mRecordThreads.keyAt(0));
197    }
198    while (!mPlaybackThreads.isEmpty()) {
199        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
200        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
201    }
202
203    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
204        // no mHardwareLock needed, as there are no other references to this
205        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
206        delete mAudioHwDevs.valueAt(i);
207    }
208}
209
210static const char * const audio_interfaces[] = {
211    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
212    AUDIO_HARDWARE_MODULE_ID_A2DP,
213    AUDIO_HARDWARE_MODULE_ID_USB,
214};
215#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
216
217AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
218        audio_module_handle_t module,
219        audio_devices_t devices)
220{
221    // if module is 0, the request comes from an old policy manager and we should load
222    // well known modules
223    if (module == 0) {
224        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
225        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
226            loadHwModule_l(audio_interfaces[i]);
227        }
228        // then try to find a module supporting the requested device.
229        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
230            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
231            audio_hw_device_t *dev = audioHwDevice->hwDevice();
232            if ((dev->get_supported_devices != NULL) &&
233                    (dev->get_supported_devices(dev) & devices) == devices)
234                return audioHwDevice;
235        }
236    } else {
237        // check a match for the requested module handle
238        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
239        if (audioHwDevice != NULL) {
240            return audioHwDevice;
241        }
242    }
243
244    return NULL;
245}
246
247void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
248{
249    const size_t SIZE = 256;
250    char buffer[SIZE];
251    String8 result;
252
253    result.append("Clients:\n");
254    for (size_t i = 0; i < mClients.size(); ++i) {
255        sp<Client> client = mClients.valueAt(i).promote();
256        if (client != 0) {
257            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
258            result.append(buffer);
259        }
260    }
261
262    result.append("Global session refs:\n");
263    result.append(" session pid count\n");
264    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
265        AudioSessionRef *r = mAudioSessionRefs[i];
266        snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
267        result.append(buffer);
268    }
269    write(fd, result.string(), result.size());
270}
271
272
273void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
274{
275    const size_t SIZE = 256;
276    char buffer[SIZE];
277    String8 result;
278    hardware_call_state hardwareStatus = mHardwareStatus;
279
280    snprintf(buffer, SIZE, "Hardware status: %d\n"
281                           "Standby Time mSec: %u\n",
282                            hardwareStatus,
283                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
284    result.append(buffer);
285    write(fd, result.string(), result.size());
286}
287
288void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
289{
290    const size_t SIZE = 256;
291    char buffer[SIZE];
292    String8 result;
293    snprintf(buffer, SIZE, "Permission Denial: "
294            "can't dump AudioFlinger from pid=%d, uid=%d\n",
295            IPCThreadState::self()->getCallingPid(),
296            IPCThreadState::self()->getCallingUid());
297    result.append(buffer);
298    write(fd, result.string(), result.size());
299}
300
301bool AudioFlinger::dumpTryLock(Mutex& mutex)
302{
303    bool locked = false;
304    for (int i = 0; i < kDumpLockRetries; ++i) {
305        if (mutex.tryLock() == NO_ERROR) {
306            locked = true;
307            break;
308        }
309        usleep(kDumpLockSleepUs);
310    }
311    return locked;
312}
313
314status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
315{
316    if (!dumpAllowed()) {
317        dumpPermissionDenial(fd, args);
318    } else {
319        // get state of hardware lock
320        bool hardwareLocked = dumpTryLock(mHardwareLock);
321        if (!hardwareLocked) {
322            String8 result(kHardwareLockedString);
323            write(fd, result.string(), result.size());
324        } else {
325            mHardwareLock.unlock();
326        }
327
328        bool locked = dumpTryLock(mLock);
329
330        // failed to lock - AudioFlinger is probably deadlocked
331        if (!locked) {
332            String8 result(kDeadlockedString);
333            write(fd, result.string(), result.size());
334        }
335
336        dumpClients(fd, args);
337        dumpInternals(fd, args);
338
339        // dump playback threads
340        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
341            mPlaybackThreads.valueAt(i)->dump(fd, args);
342        }
343
344        // dump record threads
345        for (size_t i = 0; i < mRecordThreads.size(); i++) {
346            mRecordThreads.valueAt(i)->dump(fd, args);
347        }
348
349        // dump all hardware devs
350        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
351            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
352            dev->dump(dev, fd);
353        }
354
355#ifdef TEE_SINK
356        // dump the serially shared record tee sink
357        if (mRecordTeeSource != 0) {
358            dumpTee(fd, mRecordTeeSource);
359        }
360#endif
361
362        if (locked) {
363            mLock.unlock();
364        }
365
366        // append a copy of media.log here by forwarding fd to it, but don't attempt
367        // to lookup the service if it's not running, as it will block for a second
368        if (mLogMemoryDealer != 0) {
369            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
370            if (binder != 0) {
371                fdprintf(fd, "\nmedia.log:\n");
372                Vector<String16> args;
373                binder->dump(fd, args);
374            }
375        }
376    }
377    return NO_ERROR;
378}
379
380sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
381{
382    // If pid is already in the mClients wp<> map, then use that entry
383    // (for which promote() is always != 0), otherwise create a new entry and Client.
384    sp<Client> client = mClients.valueFor(pid).promote();
385    if (client == 0) {
386        client = new Client(this, pid);
387        mClients.add(pid, client);
388    }
389
390    return client;
391}
392
393sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
394{
395    if (mLogMemoryDealer == 0) {
396        return new NBLog::Writer();
397    }
398    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
399    sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
400    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
401    if (binder != 0) {
402        interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
403    }
404    return writer;
405}
406
407void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
408{
409    if (writer == 0) {
410        return;
411    }
412    sp<IMemory> iMemory(writer->getIMemory());
413    if (iMemory == 0) {
414        return;
415    }
416    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
417    if (binder != 0) {
418        interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
419        // Now the media.log remote reference to IMemory is gone.
420        // When our last local reference to IMemory also drops to zero,
421        // the IMemory destructor will deallocate the region from mMemoryDealer.
422    }
423}
424
425// IAudioFlinger interface
426
427
428sp<IAudioTrack> AudioFlinger::createTrack(
429        audio_stream_type_t streamType,
430        uint32_t sampleRate,
431        audio_format_t format,
432        audio_channel_mask_t channelMask,
433        size_t frameCount,
434        IAudioFlinger::track_flags_t *flags,
435        const sp<IMemory>& sharedBuffer,
436        audio_io_handle_t output,
437        pid_t tid,
438        int *sessionId,
439        String8& name,
440        status_t *status)
441{
442    sp<PlaybackThread::Track> track;
443    sp<TrackHandle> trackHandle;
444    sp<Client> client;
445    status_t lStatus;
446    int lSessionId;
447
448    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
449    // but if someone uses binder directly they could bypass that and cause us to crash
450    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
451        ALOGE("createTrack() invalid stream type %d", streamType);
452        lStatus = BAD_VALUE;
453        goto Exit;
454    }
455
456    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
457    // and we don't yet support 8.24 or 32-bit PCM
458    if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
459        ALOGE("createTrack() invalid format %d", format);
460        lStatus = BAD_VALUE;
461        goto Exit;
462    }
463
464    {
465        Mutex::Autolock _l(mLock);
466        PlaybackThread *thread = checkPlaybackThread_l(output);
467        PlaybackThread *effectThread = NULL;
468        if (thread == NULL) {
469            ALOGE("no playback thread found for output handle %d", output);
470            lStatus = BAD_VALUE;
471            goto Exit;
472        }
473
474        pid_t pid = IPCThreadState::self()->getCallingPid();
475        client = registerPid_l(pid);
476
477        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
478        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
479            // check if an effect chain with the same session ID is present on another
480            // output thread and move it here.
481            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
482                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
483                if (mPlaybackThreads.keyAt(i) != output) {
484                    uint32_t sessions = t->hasAudioSession(*sessionId);
485                    if (sessions & PlaybackThread::EFFECT_SESSION) {
486                        effectThread = t.get();
487                        break;
488                    }
489                }
490            }
491            lSessionId = *sessionId;
492        } else {
493            // if no audio session id is provided, create one here
494            lSessionId = nextUniqueId();
495            if (sessionId != NULL) {
496                *sessionId = lSessionId;
497            }
498        }
499        ALOGV("createTrack() lSessionId: %d", lSessionId);
500
501        track = thread->createTrack_l(client, streamType, sampleRate, format,
502                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus);
503
504        // move effect chain to this output thread if an effect on same session was waiting
505        // for a track to be created
506        if (lStatus == NO_ERROR && effectThread != NULL) {
507            Mutex::Autolock _dl(thread->mLock);
508            Mutex::Autolock _sl(effectThread->mLock);
509            moveEffectChain_l(lSessionId, effectThread, thread, true);
510        }
511
512        // Look for sync events awaiting for a session to be used.
513        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
514            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
515                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
516                    if (lStatus == NO_ERROR) {
517                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
518                    } else {
519                        mPendingSyncEvents[i]->cancel();
520                    }
521                    mPendingSyncEvents.removeAt(i);
522                    i--;
523                }
524            }
525        }
526    }
527    if (lStatus == NO_ERROR) {
528        // s for server's pid, n for normal mixer name, f for fast index
529        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
530                track->fastIndex());
531        trackHandle = new TrackHandle(track);
532    } else {
533        // remove local strong reference to Client before deleting the Track so that the Client
534        // destructor is called by the TrackBase destructor with mLock held
535        client.clear();
536        track.clear();
537    }
538
539Exit:
540    if (status != NULL) {
541        *status = lStatus;
542    }
543    return trackHandle;
544}
545
546uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
547{
548    Mutex::Autolock _l(mLock);
549    PlaybackThread *thread = checkPlaybackThread_l(output);
550    if (thread == NULL) {
551        ALOGW("sampleRate() unknown thread %d", output);
552        return 0;
553    }
554    return thread->sampleRate();
555}
556
557int AudioFlinger::channelCount(audio_io_handle_t output) const
558{
559    Mutex::Autolock _l(mLock);
560    PlaybackThread *thread = checkPlaybackThread_l(output);
561    if (thread == NULL) {
562        ALOGW("channelCount() unknown thread %d", output);
563        return 0;
564    }
565    return thread->channelCount();
566}
567
568audio_format_t AudioFlinger::format(audio_io_handle_t output) const
569{
570    Mutex::Autolock _l(mLock);
571    PlaybackThread *thread = checkPlaybackThread_l(output);
572    if (thread == NULL) {
573        ALOGW("format() unknown thread %d", output);
574        return AUDIO_FORMAT_INVALID;
575    }
576    return thread->format();
577}
578
579size_t AudioFlinger::frameCount(audio_io_handle_t output) const
580{
581    Mutex::Autolock _l(mLock);
582    PlaybackThread *thread = checkPlaybackThread_l(output);
583    if (thread == NULL) {
584        ALOGW("frameCount() unknown thread %d", output);
585        return 0;
586    }
587    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
588    //       should examine all callers and fix them to handle smaller counts
589    return thread->frameCount();
590}
591
592uint32_t AudioFlinger::latency(audio_io_handle_t output) const
593{
594    Mutex::Autolock _l(mLock);
595    PlaybackThread *thread = checkPlaybackThread_l(output);
596    if (thread == NULL) {
597        ALOGW("latency(): no playback thread found for output handle %d", output);
598        return 0;
599    }
600    return thread->latency();
601}
602
603status_t AudioFlinger::setMasterVolume(float value)
604{
605    status_t ret = initCheck();
606    if (ret != NO_ERROR) {
607        return ret;
608    }
609
610    // check calling permissions
611    if (!settingsAllowed()) {
612        return PERMISSION_DENIED;
613    }
614
615    Mutex::Autolock _l(mLock);
616    mMasterVolume = value;
617
618    // Set master volume in the HALs which support it.
619    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
620        AutoMutex lock(mHardwareLock);
621        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
622
623        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
624        if (dev->canSetMasterVolume()) {
625            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
626        }
627        mHardwareStatus = AUDIO_HW_IDLE;
628    }
629
630    // Now set the master volume in each playback thread.  Playback threads
631    // assigned to HALs which do not have master volume support will apply
632    // master volume during the mix operation.  Threads with HALs which do
633    // support master volume will simply ignore the setting.
634    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
635        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
636
637    return NO_ERROR;
638}
639
640status_t AudioFlinger::setMode(audio_mode_t mode)
641{
642    status_t ret = initCheck();
643    if (ret != NO_ERROR) {
644        return ret;
645    }
646
647    // check calling permissions
648    if (!settingsAllowed()) {
649        return PERMISSION_DENIED;
650    }
651    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
652        ALOGW("Illegal value: setMode(%d)", mode);
653        return BAD_VALUE;
654    }
655
656    { // scope for the lock
657        AutoMutex lock(mHardwareLock);
658        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
659        mHardwareStatus = AUDIO_HW_SET_MODE;
660        ret = dev->set_mode(dev, mode);
661        mHardwareStatus = AUDIO_HW_IDLE;
662    }
663
664    if (NO_ERROR == ret) {
665        Mutex::Autolock _l(mLock);
666        mMode = mode;
667        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
668            mPlaybackThreads.valueAt(i)->setMode(mode);
669    }
670
671    return ret;
672}
673
674status_t AudioFlinger::setMicMute(bool state)
675{
676    status_t ret = initCheck();
677    if (ret != NO_ERROR) {
678        return ret;
679    }
680
681    // check calling permissions
682    if (!settingsAllowed()) {
683        return PERMISSION_DENIED;
684    }
685
686    AutoMutex lock(mHardwareLock);
687    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
688    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
689    ret = dev->set_mic_mute(dev, state);
690    mHardwareStatus = AUDIO_HW_IDLE;
691    return ret;
692}
693
694bool AudioFlinger::getMicMute() const
695{
696    status_t ret = initCheck();
697    if (ret != NO_ERROR) {
698        return false;
699    }
700
701    bool state = AUDIO_MODE_INVALID;
702    AutoMutex lock(mHardwareLock);
703    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
704    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
705    dev->get_mic_mute(dev, &state);
706    mHardwareStatus = AUDIO_HW_IDLE;
707    return state;
708}
709
710status_t AudioFlinger::setMasterMute(bool muted)
711{
712    status_t ret = initCheck();
713    if (ret != NO_ERROR) {
714        return ret;
715    }
716
717    // check calling permissions
718    if (!settingsAllowed()) {
719        return PERMISSION_DENIED;
720    }
721
722    Mutex::Autolock _l(mLock);
723    mMasterMute = muted;
724
725    // Set master mute in the HALs which support it.
726    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
727        AutoMutex lock(mHardwareLock);
728        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
729
730        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
731        if (dev->canSetMasterMute()) {
732            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
733        }
734        mHardwareStatus = AUDIO_HW_IDLE;
735    }
736
737    // Now set the master mute in each playback thread.  Playback threads
738    // assigned to HALs which do not have master mute support will apply master
739    // mute during the mix operation.  Threads with HALs which do support master
740    // mute will simply ignore the setting.
741    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
742        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
743
744    return NO_ERROR;
745}
746
747float AudioFlinger::masterVolume() const
748{
749    Mutex::Autolock _l(mLock);
750    return masterVolume_l();
751}
752
753bool AudioFlinger::masterMute() const
754{
755    Mutex::Autolock _l(mLock);
756    return masterMute_l();
757}
758
759float AudioFlinger::masterVolume_l() const
760{
761    return mMasterVolume;
762}
763
764bool AudioFlinger::masterMute_l() const
765{
766    return mMasterMute;
767}
768
769status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
770        audio_io_handle_t output)
771{
772    // check calling permissions
773    if (!settingsAllowed()) {
774        return PERMISSION_DENIED;
775    }
776
777    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
778        ALOGE("setStreamVolume() invalid stream %d", stream);
779        return BAD_VALUE;
780    }
781
782    AutoMutex lock(mLock);
783    PlaybackThread *thread = NULL;
784    if (output) {
785        thread = checkPlaybackThread_l(output);
786        if (thread == NULL) {
787            return BAD_VALUE;
788        }
789    }
790
791    mStreamTypes[stream].volume = value;
792
793    if (thread == NULL) {
794        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
795            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
796        }
797    } else {
798        thread->setStreamVolume(stream, value);
799    }
800
801    return NO_ERROR;
802}
803
804status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
805{
806    // check calling permissions
807    if (!settingsAllowed()) {
808        return PERMISSION_DENIED;
809    }
810
811    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
812        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
813        ALOGE("setStreamMute() invalid stream %d", stream);
814        return BAD_VALUE;
815    }
816
817    AutoMutex lock(mLock);
818    mStreamTypes[stream].mute = muted;
819    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
820        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
821
822    return NO_ERROR;
823}
824
825float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
826{
827    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
828        return 0.0f;
829    }
830
831    AutoMutex lock(mLock);
832    float volume;
833    if (output) {
834        PlaybackThread *thread = checkPlaybackThread_l(output);
835        if (thread == NULL) {
836            return 0.0f;
837        }
838        volume = thread->streamVolume(stream);
839    } else {
840        volume = streamVolume_l(stream);
841    }
842
843    return volume;
844}
845
846bool AudioFlinger::streamMute(audio_stream_type_t stream) const
847{
848    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
849        return true;
850    }
851
852    AutoMutex lock(mLock);
853    return streamMute_l(stream);
854}
855
856status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
857{
858    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
859            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
860
861    // check calling permissions
862    if (!settingsAllowed()) {
863        return PERMISSION_DENIED;
864    }
865
866    // ioHandle == 0 means the parameters are global to the audio hardware interface
867    if (ioHandle == 0) {
868        Mutex::Autolock _l(mLock);
869        status_t final_result = NO_ERROR;
870        {
871            AutoMutex lock(mHardwareLock);
872            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
873            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
874                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
875                status_t result = dev->set_parameters(dev, keyValuePairs.string());
876                final_result = result ?: final_result;
877            }
878            mHardwareStatus = AUDIO_HW_IDLE;
879        }
880        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
881        AudioParameter param = AudioParameter(keyValuePairs);
882        String8 value;
883        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
884            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
885            if (mBtNrecIsOff != btNrecIsOff) {
886                for (size_t i = 0; i < mRecordThreads.size(); i++) {
887                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
888                    audio_devices_t device = thread->inDevice();
889                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
890                    // collect all of the thread's session IDs
891                    KeyedVector<int, bool> ids = thread->sessionIds();
892                    // suspend effects associated with those session IDs
893                    for (size_t j = 0; j < ids.size(); ++j) {
894                        int sessionId = ids.keyAt(j);
895                        thread->setEffectSuspended(FX_IID_AEC,
896                                                   suspend,
897                                                   sessionId);
898                        thread->setEffectSuspended(FX_IID_NS,
899                                                   suspend,
900                                                   sessionId);
901                    }
902                }
903                mBtNrecIsOff = btNrecIsOff;
904            }
905        }
906        String8 screenState;
907        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
908            bool isOff = screenState == "off";
909            if (isOff != (AudioFlinger::mScreenState & 1)) {
910                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
911            }
912        }
913        return final_result;
914    }
915
916    // hold a strong ref on thread in case closeOutput() or closeInput() is called
917    // and the thread is exited once the lock is released
918    sp<ThreadBase> thread;
919    {
920        Mutex::Autolock _l(mLock);
921        thread = checkPlaybackThread_l(ioHandle);
922        if (thread == 0) {
923            thread = checkRecordThread_l(ioHandle);
924        } else if (thread == primaryPlaybackThread_l()) {
925            // indicate output device change to all input threads for pre processing
926            AudioParameter param = AudioParameter(keyValuePairs);
927            int value;
928            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
929                    (value != 0)) {
930                for (size_t i = 0; i < mRecordThreads.size(); i++) {
931                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
932                }
933            }
934        }
935    }
936    if (thread != 0) {
937        return thread->setParameters(keyValuePairs);
938    }
939    return BAD_VALUE;
940}
941
942String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
943{
944    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
945            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
946
947    Mutex::Autolock _l(mLock);
948
949    if (ioHandle == 0) {
950        String8 out_s8;
951
952        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
953            char *s;
954            {
955            AutoMutex lock(mHardwareLock);
956            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
957            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
958            s = dev->get_parameters(dev, keys.string());
959            mHardwareStatus = AUDIO_HW_IDLE;
960            }
961            out_s8 += String8(s ? s : "");
962            free(s);
963        }
964        return out_s8;
965    }
966
967    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
968    if (playbackThread != NULL) {
969        return playbackThread->getParameters(keys);
970    }
971    RecordThread *recordThread = checkRecordThread_l(ioHandle);
972    if (recordThread != NULL) {
973        return recordThread->getParameters(keys);
974    }
975    return String8("");
976}
977
978size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
979        audio_channel_mask_t channelMask) const
980{
981    status_t ret = initCheck();
982    if (ret != NO_ERROR) {
983        return 0;
984    }
985
986    AutoMutex lock(mHardwareLock);
987    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
988    struct audio_config config;
989    memset(&config, 0, sizeof(config));
990    config.sample_rate = sampleRate;
991    config.channel_mask = channelMask;
992    config.format = format;
993
994    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
995    size_t size = dev->get_input_buffer_size(dev, &config);
996    mHardwareStatus = AUDIO_HW_IDLE;
997    return size;
998}
999
1000unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1001{
1002    Mutex::Autolock _l(mLock);
1003
1004    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1005    if (recordThread != NULL) {
1006        return recordThread->getInputFramesLost();
1007    }
1008    return 0;
1009}
1010
1011status_t AudioFlinger::setVoiceVolume(float value)
1012{
1013    status_t ret = initCheck();
1014    if (ret != NO_ERROR) {
1015        return ret;
1016    }
1017
1018    // check calling permissions
1019    if (!settingsAllowed()) {
1020        return PERMISSION_DENIED;
1021    }
1022
1023    AutoMutex lock(mHardwareLock);
1024    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1025    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1026    ret = dev->set_voice_volume(dev, value);
1027    mHardwareStatus = AUDIO_HW_IDLE;
1028
1029    return ret;
1030}
1031
1032status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1033        audio_io_handle_t output) const
1034{
1035    status_t status;
1036
1037    Mutex::Autolock _l(mLock);
1038
1039    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1040    if (playbackThread != NULL) {
1041        return playbackThread->getRenderPosition(halFrames, dspFrames);
1042    }
1043
1044    return BAD_VALUE;
1045}
1046
1047void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1048{
1049
1050    Mutex::Autolock _l(mLock);
1051
1052    pid_t pid = IPCThreadState::self()->getCallingPid();
1053    if (mNotificationClients.indexOfKey(pid) < 0) {
1054        sp<NotificationClient> notificationClient = new NotificationClient(this,
1055                                                                            client,
1056                                                                            pid);
1057        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1058
1059        mNotificationClients.add(pid, notificationClient);
1060
1061        sp<IBinder> binder = client->asBinder();
1062        binder->linkToDeath(notificationClient);
1063
1064        // the config change is always sent from playback or record threads to avoid deadlock
1065        // with AudioSystem::gLock
1066        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1067            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1068        }
1069
1070        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1071            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1072        }
1073    }
1074}
1075
1076void AudioFlinger::removeNotificationClient(pid_t pid)
1077{
1078    Mutex::Autolock _l(mLock);
1079
1080    mNotificationClients.removeItem(pid);
1081
1082    ALOGV("%d died, releasing its sessions", pid);
1083    size_t num = mAudioSessionRefs.size();
1084    bool removed = false;
1085    for (size_t i = 0; i< num; ) {
1086        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1087        ALOGV(" pid %d @ %d", ref->mPid, i);
1088        if (ref->mPid == pid) {
1089            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1090            mAudioSessionRefs.removeAt(i);
1091            delete ref;
1092            removed = true;
1093            num--;
1094        } else {
1095            i++;
1096        }
1097    }
1098    if (removed) {
1099        purgeStaleEffects_l();
1100    }
1101}
1102
1103// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1104void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1105{
1106    size_t size = mNotificationClients.size();
1107    for (size_t i = 0; i < size; i++) {
1108        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1109                                                                               param2);
1110    }
1111}
1112
1113// removeClient_l() must be called with AudioFlinger::mLock held
1114void AudioFlinger::removeClient_l(pid_t pid)
1115{
1116    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1117            IPCThreadState::self()->getCallingPid());
1118    mClients.removeItem(pid);
1119}
1120
1121// getEffectThread_l() must be called with AudioFlinger::mLock held
1122sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1123{
1124    sp<PlaybackThread> thread;
1125
1126    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1127        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1128            ALOG_ASSERT(thread == 0);
1129            thread = mPlaybackThreads.valueAt(i);
1130        }
1131    }
1132
1133    return thread;
1134}
1135
1136
1137
1138// ----------------------------------------------------------------------------
1139
1140AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1141    :   RefBase(),
1142        mAudioFlinger(audioFlinger),
1143        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1144        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1145        mPid(pid),
1146        mTimedTrackCount(0)
1147{
1148    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1149}
1150
1151// Client destructor must be called with AudioFlinger::mLock held
1152AudioFlinger::Client::~Client()
1153{
1154    mAudioFlinger->removeClient_l(mPid);
1155}
1156
1157sp<MemoryDealer> AudioFlinger::Client::heap() const
1158{
1159    return mMemoryDealer;
1160}
1161
1162// Reserve one of the limited slots for a timed audio track associated
1163// with this client
1164bool AudioFlinger::Client::reserveTimedTrack()
1165{
1166    const int kMaxTimedTracksPerClient = 4;
1167
1168    Mutex::Autolock _l(mTimedTrackLock);
1169
1170    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1171        ALOGW("can not create timed track - pid %d has exceeded the limit",
1172             mPid);
1173        return false;
1174    }
1175
1176    mTimedTrackCount++;
1177    return true;
1178}
1179
1180// Release a slot for a timed audio track
1181void AudioFlinger::Client::releaseTimedTrack()
1182{
1183    Mutex::Autolock _l(mTimedTrackLock);
1184    mTimedTrackCount--;
1185}
1186
1187// ----------------------------------------------------------------------------
1188
1189AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1190                                                     const sp<IAudioFlingerClient>& client,
1191                                                     pid_t pid)
1192    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1193{
1194}
1195
1196AudioFlinger::NotificationClient::~NotificationClient()
1197{
1198}
1199
1200void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1201{
1202    sp<NotificationClient> keep(this);
1203    mAudioFlinger->removeNotificationClient(mPid);
1204}
1205
1206
1207// ----------------------------------------------------------------------------
1208
1209static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1210    return audio_is_remote_submix_device(inDevice);
1211}
1212
1213sp<IAudioRecord> AudioFlinger::openRecord(
1214        audio_io_handle_t input,
1215        uint32_t sampleRate,
1216        audio_format_t format,
1217        audio_channel_mask_t channelMask,
1218        size_t frameCount,
1219        IAudioFlinger::track_flags_t *flags,
1220        pid_t tid,
1221        int *sessionId,
1222        status_t *status)
1223{
1224    sp<RecordThread::RecordTrack> recordTrack;
1225    sp<RecordHandle> recordHandle;
1226    sp<Client> client;
1227    status_t lStatus;
1228    RecordThread *thread;
1229    size_t inFrameCount;
1230    int lSessionId;
1231
1232    // check calling permissions
1233    if (!recordingAllowed()) {
1234        lStatus = PERMISSION_DENIED;
1235        goto Exit;
1236    }
1237
1238    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1239        ALOGE("openRecord() invalid format %d", format);
1240        lStatus = BAD_VALUE;
1241        goto Exit;
1242    }
1243
1244    // add client to list
1245    { // scope for mLock
1246        Mutex::Autolock _l(mLock);
1247        thread = checkRecordThread_l(input);
1248        if (thread == NULL) {
1249            lStatus = BAD_VALUE;
1250            goto Exit;
1251        }
1252
1253        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1254                && !captureAudioOutputAllowed()) {
1255            lStatus = PERMISSION_DENIED;
1256            goto Exit;
1257        }
1258
1259        pid_t pid = IPCThreadState::self()->getCallingPid();
1260        client = registerPid_l(pid);
1261
1262        // If no audio session id is provided, create one here
1263        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1264            lSessionId = *sessionId;
1265        } else {
1266            lSessionId = nextUniqueId();
1267            if (sessionId != NULL) {
1268                *sessionId = lSessionId;
1269            }
1270        }
1271        // create new record track.
1272        // The record track uses one track in mHardwareMixerThread by convention.
1273        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1274                                                  frameCount, lSessionId, flags, tid, &lStatus);
1275    }
1276    if (lStatus != NO_ERROR) {
1277        // remove local strong reference to Client before deleting the RecordTrack so that the
1278        // Client destructor is called by the TrackBase destructor with mLock held
1279        client.clear();
1280        recordTrack.clear();
1281        goto Exit;
1282    }
1283
1284    // return to handle to client
1285    recordHandle = new RecordHandle(recordTrack);
1286    lStatus = NO_ERROR;
1287
1288Exit:
1289    if (status) {
1290        *status = lStatus;
1291    }
1292    return recordHandle;
1293}
1294
1295
1296
1297// ----------------------------------------------------------------------------
1298
1299audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1300{
1301    if (!settingsAllowed()) {
1302        return 0;
1303    }
1304    Mutex::Autolock _l(mLock);
1305    return loadHwModule_l(name);
1306}
1307
1308// loadHwModule_l() must be called with AudioFlinger::mLock held
1309audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1310{
1311    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1312        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1313            ALOGW("loadHwModule() module %s already loaded", name);
1314            return mAudioHwDevs.keyAt(i);
1315        }
1316    }
1317
1318    audio_hw_device_t *dev;
1319
1320    int rc = load_audio_interface(name, &dev);
1321    if (rc) {
1322        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1323        return 0;
1324    }
1325
1326    mHardwareStatus = AUDIO_HW_INIT;
1327    rc = dev->init_check(dev);
1328    mHardwareStatus = AUDIO_HW_IDLE;
1329    if (rc) {
1330        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1331        return 0;
1332    }
1333
1334    // Check and cache this HAL's level of support for master mute and master
1335    // volume.  If this is the first HAL opened, and it supports the get
1336    // methods, use the initial values provided by the HAL as the current
1337    // master mute and volume settings.
1338
1339    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1340    {  // scope for auto-lock pattern
1341        AutoMutex lock(mHardwareLock);
1342
1343        if (0 == mAudioHwDevs.size()) {
1344            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1345            if (NULL != dev->get_master_volume) {
1346                float mv;
1347                if (OK == dev->get_master_volume(dev, &mv)) {
1348                    mMasterVolume = mv;
1349                }
1350            }
1351
1352            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1353            if (NULL != dev->get_master_mute) {
1354                bool mm;
1355                if (OK == dev->get_master_mute(dev, &mm)) {
1356                    mMasterMute = mm;
1357                }
1358            }
1359        }
1360
1361        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1362        if ((NULL != dev->set_master_volume) &&
1363            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1364            flags = static_cast<AudioHwDevice::Flags>(flags |
1365                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1366        }
1367
1368        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1369        if ((NULL != dev->set_master_mute) &&
1370            (OK == dev->set_master_mute(dev, mMasterMute))) {
1371            flags = static_cast<AudioHwDevice::Flags>(flags |
1372                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1373        }
1374
1375        mHardwareStatus = AUDIO_HW_IDLE;
1376    }
1377
1378    audio_module_handle_t handle = nextUniqueId();
1379    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1380
1381    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1382          name, dev->common.module->name, dev->common.module->id, handle);
1383
1384    return handle;
1385
1386}
1387
1388// ----------------------------------------------------------------------------
1389
1390uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1391{
1392    Mutex::Autolock _l(mLock);
1393    PlaybackThread *thread = primaryPlaybackThread_l();
1394    return thread != NULL ? thread->sampleRate() : 0;
1395}
1396
1397size_t AudioFlinger::getPrimaryOutputFrameCount()
1398{
1399    Mutex::Autolock _l(mLock);
1400    PlaybackThread *thread = primaryPlaybackThread_l();
1401    return thread != NULL ? thread->frameCountHAL() : 0;
1402}
1403
1404// ----------------------------------------------------------------------------
1405
1406status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1407{
1408    uid_t uid = IPCThreadState::self()->getCallingUid();
1409    if (uid != AID_SYSTEM) {
1410        return PERMISSION_DENIED;
1411    }
1412    Mutex::Autolock _l(mLock);
1413    if (mIsDeviceTypeKnown) {
1414        return INVALID_OPERATION;
1415    }
1416    mIsLowRamDevice = isLowRamDevice;
1417    mIsDeviceTypeKnown = true;
1418    return NO_ERROR;
1419}
1420
1421// ----------------------------------------------------------------------------
1422
1423audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1424                                           audio_devices_t *pDevices,
1425                                           uint32_t *pSamplingRate,
1426                                           audio_format_t *pFormat,
1427                                           audio_channel_mask_t *pChannelMask,
1428                                           uint32_t *pLatencyMs,
1429                                           audio_output_flags_t flags,
1430                                           const audio_offload_info_t *offloadInfo)
1431{
1432    PlaybackThread *thread = NULL;
1433    struct audio_config config;
1434    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1435    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1436    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1437    if (offloadInfo) {
1438        config.offload_info = *offloadInfo;
1439    }
1440
1441    audio_stream_out_t *outStream = NULL;
1442    AudioHwDevice *outHwDev;
1443
1444    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1445              module,
1446              (pDevices != NULL) ? *pDevices : 0,
1447              config.sample_rate,
1448              config.format,
1449              config.channel_mask,
1450              flags);
1451    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1452          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version );
1453
1454    if (pDevices == NULL || *pDevices == 0) {
1455        return 0;
1456    }
1457
1458    Mutex::Autolock _l(mLock);
1459
1460    outHwDev = findSuitableHwDev_l(module, *pDevices);
1461    if (outHwDev == NULL)
1462        return 0;
1463
1464    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1465    audio_io_handle_t id = nextUniqueId();
1466
1467    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1468
1469    status_t status = hwDevHal->open_output_stream(hwDevHal,
1470                                          id,
1471                                          *pDevices,
1472                                          (audio_output_flags_t)flags,
1473                                          &config,
1474                                          &outStream);
1475
1476    mHardwareStatus = AUDIO_HW_IDLE;
1477    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1478            "Channels %x, status %d",
1479            outStream,
1480            config.sample_rate,
1481            config.format,
1482            config.channel_mask,
1483            status);
1484
1485    if (status == NO_ERROR && outStream != NULL) {
1486        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1487
1488        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1489            thread = new OffloadThread(this, output, id, *pDevices);
1490            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1491        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1492            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1493            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1494            thread = new DirectOutputThread(this, output, id, *pDevices);
1495            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1496        } else {
1497            thread = new MixerThread(this, output, id, *pDevices);
1498            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1499        }
1500        mPlaybackThreads.add(id, thread);
1501
1502        if (pSamplingRate != NULL) {
1503            *pSamplingRate = config.sample_rate;
1504        }
1505        if (pFormat != NULL) {
1506            *pFormat = config.format;
1507        }
1508        if (pChannelMask != NULL) {
1509            *pChannelMask = config.channel_mask;
1510        }
1511        if (pLatencyMs != NULL) {
1512            *pLatencyMs = thread->latency();
1513        }
1514
1515        // notify client processes of the new output creation
1516        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1517
1518        // the first primary output opened designates the primary hw device
1519        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1520            ALOGI("Using module %d has the primary audio interface", module);
1521            mPrimaryHardwareDev = outHwDev;
1522
1523            AutoMutex lock(mHardwareLock);
1524            mHardwareStatus = AUDIO_HW_SET_MODE;
1525            hwDevHal->set_mode(hwDevHal, mMode);
1526            mHardwareStatus = AUDIO_HW_IDLE;
1527        }
1528        return id;
1529    }
1530
1531    return 0;
1532}
1533
1534audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1535        audio_io_handle_t output2)
1536{
1537    Mutex::Autolock _l(mLock);
1538    MixerThread *thread1 = checkMixerThread_l(output1);
1539    MixerThread *thread2 = checkMixerThread_l(output2);
1540
1541    if (thread1 == NULL || thread2 == NULL) {
1542        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1543                output2);
1544        return 0;
1545    }
1546
1547    audio_io_handle_t id = nextUniqueId();
1548    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1549    thread->addOutputTrack(thread2);
1550    mPlaybackThreads.add(id, thread);
1551    // notify client processes of the new output creation
1552    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1553    return id;
1554}
1555
1556status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1557{
1558    return closeOutput_nonvirtual(output);
1559}
1560
1561status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1562{
1563    // keep strong reference on the playback thread so that
1564    // it is not destroyed while exit() is executed
1565    sp<PlaybackThread> thread;
1566    {
1567        Mutex::Autolock _l(mLock);
1568        thread = checkPlaybackThread_l(output);
1569        if (thread == NULL) {
1570            return BAD_VALUE;
1571        }
1572
1573        ALOGV("closeOutput() %d", output);
1574
1575        if (thread->type() == ThreadBase::MIXER) {
1576            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1577                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1578                    DuplicatingThread *dupThread =
1579                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1580                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1581
1582                }
1583            }
1584        }
1585
1586
1587        mPlaybackThreads.removeItem(output);
1588        // save all effects to the default thread
1589        if (mPlaybackThreads.size()) {
1590            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1591            if (dstThread != NULL) {
1592                // audioflinger lock is held here so the acquisition order of thread locks does not
1593                // matter
1594                Mutex::Autolock _dl(dstThread->mLock);
1595                Mutex::Autolock _sl(thread->mLock);
1596                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1597                for (size_t i = 0; i < effectChains.size(); i ++) {
1598                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1599                }
1600            }
1601        }
1602        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1603    }
1604    thread->exit();
1605    // The thread entity (active unit of execution) is no longer running here,
1606    // but the ThreadBase container still exists.
1607
1608    if (thread->type() != ThreadBase::DUPLICATING) {
1609        AudioStreamOut *out = thread->clearOutput();
1610        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1611        // from now on thread->mOutput is NULL
1612        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1613        delete out;
1614    }
1615    return NO_ERROR;
1616}
1617
1618status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1619{
1620    Mutex::Autolock _l(mLock);
1621    PlaybackThread *thread = checkPlaybackThread_l(output);
1622
1623    if (thread == NULL) {
1624        return BAD_VALUE;
1625    }
1626
1627    ALOGV("suspendOutput() %d", output);
1628    thread->suspend();
1629
1630    return NO_ERROR;
1631}
1632
1633status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1634{
1635    Mutex::Autolock _l(mLock);
1636    PlaybackThread *thread = checkPlaybackThread_l(output);
1637
1638    if (thread == NULL) {
1639        return BAD_VALUE;
1640    }
1641
1642    ALOGV("restoreOutput() %d", output);
1643
1644    thread->restore();
1645
1646    return NO_ERROR;
1647}
1648
1649audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1650                                          audio_devices_t *pDevices,
1651                                          uint32_t *pSamplingRate,
1652                                          audio_format_t *pFormat,
1653                                          audio_channel_mask_t *pChannelMask)
1654{
1655    status_t status;
1656    RecordThread *thread = NULL;
1657    struct audio_config config;
1658    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1659    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1660    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1661
1662    uint32_t reqSamplingRate = config.sample_rate;
1663    audio_format_t reqFormat = config.format;
1664    audio_channel_mask_t reqChannels = config.channel_mask;
1665    audio_stream_in_t *inStream = NULL;
1666    AudioHwDevice *inHwDev;
1667
1668    if (pDevices == NULL || *pDevices == 0) {
1669        return 0;
1670    }
1671
1672    Mutex::Autolock _l(mLock);
1673
1674    inHwDev = findSuitableHwDev_l(module, *pDevices);
1675    if (inHwDev == NULL)
1676        return 0;
1677
1678    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1679    audio_io_handle_t id = nextUniqueId();
1680
1681    status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1682                                        &inStream);
1683    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1684            "status %d",
1685            inStream,
1686            config.sample_rate,
1687            config.format,
1688            config.channel_mask,
1689            status);
1690
1691    // If the input could not be opened with the requested parameters and we can handle the
1692    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1693    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1694    if (status == BAD_VALUE &&
1695        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1696        (config.sample_rate <= 2 * reqSamplingRate) &&
1697        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
1698        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1699        inStream = NULL;
1700        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1701    }
1702
1703    if (status == NO_ERROR && inStream != NULL) {
1704
1705#ifdef TEE_SINK
1706        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1707        // or (re-)create if current Pipe is idle and does not match the new format
1708        sp<NBAIO_Sink> teeSink;
1709        enum {
1710            TEE_SINK_NO,    // don't copy input
1711            TEE_SINK_NEW,   // copy input using a new pipe
1712            TEE_SINK_OLD,   // copy input using an existing pipe
1713        } kind;
1714        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1715                                        popcount(inStream->common.get_channels(&inStream->common)));
1716        if (!mTeeSinkInputEnabled) {
1717            kind = TEE_SINK_NO;
1718        } else if (format == Format_Invalid) {
1719            kind = TEE_SINK_NO;
1720        } else if (mRecordTeeSink == 0) {
1721            kind = TEE_SINK_NEW;
1722        } else if (mRecordTeeSink->getStrongCount() != 1) {
1723            kind = TEE_SINK_NO;
1724        } else if (format == mRecordTeeSink->format()) {
1725            kind = TEE_SINK_OLD;
1726        } else {
1727            kind = TEE_SINK_NEW;
1728        }
1729        switch (kind) {
1730        case TEE_SINK_NEW: {
1731            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1732            size_t numCounterOffers = 0;
1733            const NBAIO_Format offers[1] = {format};
1734            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1735            ALOG_ASSERT(index == 0);
1736            PipeReader *pipeReader = new PipeReader(*pipe);
1737            numCounterOffers = 0;
1738            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1739            ALOG_ASSERT(index == 0);
1740            mRecordTeeSink = pipe;
1741            mRecordTeeSource = pipeReader;
1742            teeSink = pipe;
1743            }
1744            break;
1745        case TEE_SINK_OLD:
1746            teeSink = mRecordTeeSink;
1747            break;
1748        case TEE_SINK_NO:
1749        default:
1750            break;
1751        }
1752#endif
1753
1754        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1755
1756        // Start record thread
1757        // RecordThread requires both input and output device indication to forward to audio
1758        // pre processing modules
1759        thread = new RecordThread(this,
1760                                  input,
1761                                  reqSamplingRate,
1762                                  reqChannels,
1763                                  id,
1764                                  primaryOutputDevice_l(),
1765                                  *pDevices
1766#ifdef TEE_SINK
1767                                  , teeSink
1768#endif
1769                                  );
1770        mRecordThreads.add(id, thread);
1771        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1772        if (pSamplingRate != NULL) {
1773            *pSamplingRate = reqSamplingRate;
1774        }
1775        if (pFormat != NULL) {
1776            *pFormat = config.format;
1777        }
1778        if (pChannelMask != NULL) {
1779            *pChannelMask = reqChannels;
1780        }
1781
1782        // notify client processes of the new input creation
1783        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1784        return id;
1785    }
1786
1787    return 0;
1788}
1789
1790status_t AudioFlinger::closeInput(audio_io_handle_t input)
1791{
1792    return closeInput_nonvirtual(input);
1793}
1794
1795status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1796{
1797    // keep strong reference on the record thread so that
1798    // it is not destroyed while exit() is executed
1799    sp<RecordThread> thread;
1800    {
1801        Mutex::Autolock _l(mLock);
1802        thread = checkRecordThread_l(input);
1803        if (thread == 0) {
1804            return BAD_VALUE;
1805        }
1806
1807        ALOGV("closeInput() %d", input);
1808        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1809        mRecordThreads.removeItem(input);
1810    }
1811    thread->exit();
1812    // The thread entity (active unit of execution) is no longer running here,
1813    // but the ThreadBase container still exists.
1814
1815    AudioStreamIn *in = thread->clearInput();
1816    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1817    // from now on thread->mInput is NULL
1818    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1819    delete in;
1820
1821    return NO_ERROR;
1822}
1823
1824status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1825{
1826    Mutex::Autolock _l(mLock);
1827    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1828
1829    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1830        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1831        thread->invalidateTracks(stream);
1832    }
1833
1834    return NO_ERROR;
1835}
1836
1837
1838int AudioFlinger::newAudioSessionId()
1839{
1840    return nextUniqueId();
1841}
1842
1843void AudioFlinger::acquireAudioSessionId(int audioSession)
1844{
1845    Mutex::Autolock _l(mLock);
1846    pid_t caller = IPCThreadState::self()->getCallingPid();
1847    ALOGV("acquiring %d from %d", audioSession, caller);
1848    size_t num = mAudioSessionRefs.size();
1849    for (size_t i = 0; i< num; i++) {
1850        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1851        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1852            ref->mCnt++;
1853            ALOGV(" incremented refcount to %d", ref->mCnt);
1854            return;
1855        }
1856    }
1857    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1858    ALOGV(" added new entry for %d", audioSession);
1859}
1860
1861void AudioFlinger::releaseAudioSessionId(int audioSession)
1862{
1863    Mutex::Autolock _l(mLock);
1864    pid_t caller = IPCThreadState::self()->getCallingPid();
1865    ALOGV("releasing %d from %d", audioSession, caller);
1866    size_t num = mAudioSessionRefs.size();
1867    for (size_t i = 0; i< num; i++) {
1868        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1869        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1870            ref->mCnt--;
1871            ALOGV(" decremented refcount to %d", ref->mCnt);
1872            if (ref->mCnt == 0) {
1873                mAudioSessionRefs.removeAt(i);
1874                delete ref;
1875                purgeStaleEffects_l();
1876            }
1877            return;
1878        }
1879    }
1880    ALOGW("session id %d not found for pid %d", audioSession, caller);
1881}
1882
1883void AudioFlinger::purgeStaleEffects_l() {
1884
1885    ALOGV("purging stale effects");
1886
1887    Vector< sp<EffectChain> > chains;
1888
1889    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1890        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1891        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1892            sp<EffectChain> ec = t->mEffectChains[j];
1893            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1894                chains.push(ec);
1895            }
1896        }
1897    }
1898    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1899        sp<RecordThread> t = mRecordThreads.valueAt(i);
1900        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1901            sp<EffectChain> ec = t->mEffectChains[j];
1902            chains.push(ec);
1903        }
1904    }
1905
1906    for (size_t i = 0; i < chains.size(); i++) {
1907        sp<EffectChain> ec = chains[i];
1908        int sessionid = ec->sessionId();
1909        sp<ThreadBase> t = ec->mThread.promote();
1910        if (t == 0) {
1911            continue;
1912        }
1913        size_t numsessionrefs = mAudioSessionRefs.size();
1914        bool found = false;
1915        for (size_t k = 0; k < numsessionrefs; k++) {
1916            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1917            if (ref->mSessionid == sessionid) {
1918                ALOGV(" session %d still exists for %d with %d refs",
1919                    sessionid, ref->mPid, ref->mCnt);
1920                found = true;
1921                break;
1922            }
1923        }
1924        if (!found) {
1925            Mutex::Autolock _l (t->mLock);
1926            // remove all effects from the chain
1927            while (ec->mEffects.size()) {
1928                sp<EffectModule> effect = ec->mEffects[0];
1929                effect->unPin();
1930                t->removeEffect_l(effect);
1931                if (effect->purgeHandles()) {
1932                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
1933                }
1934                AudioSystem::unregisterEffect(effect->id());
1935            }
1936        }
1937    }
1938    return;
1939}
1940
1941// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
1942AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
1943{
1944    return mPlaybackThreads.valueFor(output).get();
1945}
1946
1947// checkMixerThread_l() must be called with AudioFlinger::mLock held
1948AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
1949{
1950    PlaybackThread *thread = checkPlaybackThread_l(output);
1951    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
1952}
1953
1954// checkRecordThread_l() must be called with AudioFlinger::mLock held
1955AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
1956{
1957    return mRecordThreads.valueFor(input).get();
1958}
1959
1960uint32_t AudioFlinger::nextUniqueId()
1961{
1962    return android_atomic_inc(&mNextUniqueId);
1963}
1964
1965AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
1966{
1967    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1968        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1969        AudioStreamOut *output = thread->getOutput();
1970        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
1971            return thread;
1972        }
1973    }
1974    return NULL;
1975}
1976
1977audio_devices_t AudioFlinger::primaryOutputDevice_l() const
1978{
1979    PlaybackThread *thread = primaryPlaybackThread_l();
1980
1981    if (thread == NULL) {
1982        return 0;
1983    }
1984
1985    return thread->outDevice();
1986}
1987
1988sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
1989                                    int triggerSession,
1990                                    int listenerSession,
1991                                    sync_event_callback_t callBack,
1992                                    void *cookie)
1993{
1994    Mutex::Autolock _l(mLock);
1995
1996    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
1997    status_t playStatus = NAME_NOT_FOUND;
1998    status_t recStatus = NAME_NOT_FOUND;
1999    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2000        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2001        if (playStatus == NO_ERROR) {
2002            return event;
2003        }
2004    }
2005    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2006        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2007        if (recStatus == NO_ERROR) {
2008            return event;
2009        }
2010    }
2011    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2012        mPendingSyncEvents.add(event);
2013    } else {
2014        ALOGV("createSyncEvent() invalid event %d", event->type());
2015        event.clear();
2016    }
2017    return event;
2018}
2019
2020// ----------------------------------------------------------------------------
2021//  Effect management
2022// ----------------------------------------------------------------------------
2023
2024
2025status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2026{
2027    Mutex::Autolock _l(mLock);
2028    return EffectQueryNumberEffects(numEffects);
2029}
2030
2031status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2032{
2033    Mutex::Autolock _l(mLock);
2034    return EffectQueryEffect(index, descriptor);
2035}
2036
2037status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2038        effect_descriptor_t *descriptor) const
2039{
2040    Mutex::Autolock _l(mLock);
2041    return EffectGetDescriptor(pUuid, descriptor);
2042}
2043
2044
2045sp<IEffect> AudioFlinger::createEffect(
2046        effect_descriptor_t *pDesc,
2047        const sp<IEffectClient>& effectClient,
2048        int32_t priority,
2049        audio_io_handle_t io,
2050        int sessionId,
2051        status_t *status,
2052        int *id,
2053        int *enabled)
2054{
2055    status_t lStatus = NO_ERROR;
2056    sp<EffectHandle> handle;
2057    effect_descriptor_t desc;
2058
2059    pid_t pid = IPCThreadState::self()->getCallingPid();
2060    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2061            pid, effectClient.get(), priority, sessionId, io);
2062
2063    if (pDesc == NULL) {
2064        lStatus = BAD_VALUE;
2065        goto Exit;
2066    }
2067
2068    // check audio settings permission for global effects
2069    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2070        lStatus = PERMISSION_DENIED;
2071        goto Exit;
2072    }
2073
2074    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2075    // that can only be created by audio policy manager (running in same process)
2076    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2077        lStatus = PERMISSION_DENIED;
2078        goto Exit;
2079    }
2080
2081    if (io == 0) {
2082        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2083            // output must be specified by AudioPolicyManager when using session
2084            // AUDIO_SESSION_OUTPUT_STAGE
2085            lStatus = BAD_VALUE;
2086            goto Exit;
2087        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2088            // if the output returned by getOutputForEffect() is removed before we lock the
2089            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2090            // and we will exit safely
2091            io = AudioSystem::getOutputForEffect(&desc);
2092        }
2093    }
2094
2095    {
2096        Mutex::Autolock _l(mLock);
2097
2098
2099        if (!EffectIsNullUuid(&pDesc->uuid)) {
2100            // if uuid is specified, request effect descriptor
2101            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2102            if (lStatus < 0) {
2103                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2104                goto Exit;
2105            }
2106        } else {
2107            // if uuid is not specified, look for an available implementation
2108            // of the required type in effect factory
2109            if (EffectIsNullUuid(&pDesc->type)) {
2110                ALOGW("createEffect() no effect type");
2111                lStatus = BAD_VALUE;
2112                goto Exit;
2113            }
2114            uint32_t numEffects = 0;
2115            effect_descriptor_t d;
2116            d.flags = 0; // prevent compiler warning
2117            bool found = false;
2118
2119            lStatus = EffectQueryNumberEffects(&numEffects);
2120            if (lStatus < 0) {
2121                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2122                goto Exit;
2123            }
2124            for (uint32_t i = 0; i < numEffects; i++) {
2125                lStatus = EffectQueryEffect(i, &desc);
2126                if (lStatus < 0) {
2127                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2128                    continue;
2129                }
2130                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2131                    // If matching type found save effect descriptor. If the session is
2132                    // 0 and the effect is not auxiliary, continue enumeration in case
2133                    // an auxiliary version of this effect type is available
2134                    found = true;
2135                    d = desc;
2136                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2137                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2138                        break;
2139                    }
2140                }
2141            }
2142            if (!found) {
2143                lStatus = BAD_VALUE;
2144                ALOGW("createEffect() effect not found");
2145                goto Exit;
2146            }
2147            // For same effect type, chose auxiliary version over insert version if
2148            // connect to output mix (Compliance to OpenSL ES)
2149            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2150                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2151                desc = d;
2152            }
2153        }
2154
2155        // Do not allow auxiliary effects on a session different from 0 (output mix)
2156        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2157             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2158            lStatus = INVALID_OPERATION;
2159            goto Exit;
2160        }
2161
2162        // check recording permission for visualizer
2163        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2164            !recordingAllowed()) {
2165            lStatus = PERMISSION_DENIED;
2166            goto Exit;
2167        }
2168
2169        // return effect descriptor
2170        *pDesc = desc;
2171
2172        // If output is not specified try to find a matching audio session ID in one of the
2173        // output threads.
2174        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2175        // because of code checking output when entering the function.
2176        // Note: io is never 0 when creating an effect on an input
2177        if (io == 0) {
2178            // look for the thread where the specified audio session is present
2179            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2180                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2181                    io = mPlaybackThreads.keyAt(i);
2182                    break;
2183                }
2184            }
2185            if (io == 0) {
2186                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2187                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2188                        io = mRecordThreads.keyAt(i);
2189                        break;
2190                    }
2191                }
2192            }
2193            // If no output thread contains the requested session ID, default to
2194            // first output. The effect chain will be moved to the correct output
2195            // thread when a track with the same session ID is created
2196            if (io == 0 && mPlaybackThreads.size()) {
2197                io = mPlaybackThreads.keyAt(0);
2198            }
2199            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2200        }
2201        ThreadBase *thread = checkRecordThread_l(io);
2202        if (thread == NULL) {
2203            thread = checkPlaybackThread_l(io);
2204            if (thread == NULL) {
2205                ALOGE("createEffect() unknown output thread");
2206                lStatus = BAD_VALUE;
2207                goto Exit;
2208            }
2209        }
2210
2211        sp<Client> client = registerPid_l(pid);
2212
2213        // create effect on selected output thread
2214        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2215                &desc, enabled, &lStatus);
2216        if (handle != 0 && id != NULL) {
2217            *id = handle->id();
2218        }
2219    }
2220
2221Exit:
2222    if (status != NULL) {
2223        *status = lStatus;
2224    }
2225    return handle;
2226}
2227
2228status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2229        audio_io_handle_t dstOutput)
2230{
2231    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2232            sessionId, srcOutput, dstOutput);
2233    Mutex::Autolock _l(mLock);
2234    if (srcOutput == dstOutput) {
2235        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2236        return NO_ERROR;
2237    }
2238    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2239    if (srcThread == NULL) {
2240        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2241        return BAD_VALUE;
2242    }
2243    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2244    if (dstThread == NULL) {
2245        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2246        return BAD_VALUE;
2247    }
2248
2249    Mutex::Autolock _dl(dstThread->mLock);
2250    Mutex::Autolock _sl(srcThread->mLock);
2251    moveEffectChain_l(sessionId, srcThread, dstThread, false);
2252
2253    return NO_ERROR;
2254}
2255
2256// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2257status_t AudioFlinger::moveEffectChain_l(int sessionId,
2258                                   AudioFlinger::PlaybackThread *srcThread,
2259                                   AudioFlinger::PlaybackThread *dstThread,
2260                                   bool reRegister)
2261{
2262    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2263            sessionId, srcThread, dstThread);
2264
2265    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2266    if (chain == 0) {
2267        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2268                sessionId, srcThread);
2269        return INVALID_OPERATION;
2270    }
2271
2272    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2273    // so that a new chain is created with correct parameters when first effect is added. This is
2274    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2275    // removed.
2276    srcThread->removeEffectChain_l(chain);
2277
2278    // transfer all effects one by one so that new effect chain is created on new thread with
2279    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2280    audio_io_handle_t dstOutput = dstThread->id();
2281    sp<EffectChain> dstChain;
2282    uint32_t strategy = 0; // prevent compiler warning
2283    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2284    while (effect != 0) {
2285        srcThread->removeEffect_l(effect);
2286        dstThread->addEffect_l(effect);
2287        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2288        if (effect->state() == EffectModule::ACTIVE ||
2289                effect->state() == EffectModule::STOPPING) {
2290            effect->start();
2291        }
2292        // if the move request is not received from audio policy manager, the effect must be
2293        // re-registered with the new strategy and output
2294        if (dstChain == 0) {
2295            dstChain = effect->chain().promote();
2296            if (dstChain == 0) {
2297                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2298                srcThread->addEffect_l(effect);
2299                return NO_INIT;
2300            }
2301            strategy = dstChain->strategy();
2302        }
2303        if (reRegister) {
2304            AudioSystem::unregisterEffect(effect->id());
2305            AudioSystem::registerEffect(&effect->desc(),
2306                                        dstOutput,
2307                                        strategy,
2308                                        sessionId,
2309                                        effect->id());
2310        }
2311        effect = chain->getEffectFromId_l(0);
2312    }
2313
2314    return NO_ERROR;
2315}
2316
2317struct Entry {
2318#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2319    char mName[MAX_NAME];
2320};
2321
2322int comparEntry(const void *p1, const void *p2)
2323{
2324    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2325}
2326
2327#ifdef TEE_SINK
2328void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2329{
2330    NBAIO_Source *teeSource = source.get();
2331    if (teeSource != NULL) {
2332        // .wav rotation
2333        // There is a benign race condition if 2 threads call this simultaneously.
2334        // They would both traverse the directory, but the result would simply be
2335        // failures at unlink() which are ignored.  It's also unlikely since
2336        // normally dumpsys is only done by bugreport or from the command line.
2337        char teePath[32+256];
2338        strcpy(teePath, "/data/misc/media");
2339        size_t teePathLen = strlen(teePath);
2340        DIR *dir = opendir(teePath);
2341        teePath[teePathLen++] = '/';
2342        if (dir != NULL) {
2343#define MAX_SORT 20 // number of entries to sort
2344#define MAX_KEEP 10 // number of entries to keep
2345            struct Entry entries[MAX_SORT];
2346            size_t entryCount = 0;
2347            while (entryCount < MAX_SORT) {
2348                struct dirent de;
2349                struct dirent *result = NULL;
2350                int rc = readdir_r(dir, &de, &result);
2351                if (rc != 0) {
2352                    ALOGW("readdir_r failed %d", rc);
2353                    break;
2354                }
2355                if (result == NULL) {
2356                    break;
2357                }
2358                if (result != &de) {
2359                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2360                    break;
2361                }
2362                // ignore non .wav file entries
2363                size_t nameLen = strlen(de.d_name);
2364                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2365                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2366                    continue;
2367                }
2368                strcpy(entries[entryCount++].mName, de.d_name);
2369            }
2370            (void) closedir(dir);
2371            if (entryCount > MAX_KEEP) {
2372                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2373                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2374                    strcpy(&teePath[teePathLen], entries[i].mName);
2375                    (void) unlink(teePath);
2376                }
2377            }
2378        } else {
2379            if (fd >= 0) {
2380                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2381            }
2382        }
2383        char teeTime[16];
2384        struct timeval tv;
2385        gettimeofday(&tv, NULL);
2386        struct tm tm;
2387        localtime_r(&tv.tv_sec, &tm);
2388        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2389        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2390        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2391        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2392        if (teeFd >= 0) {
2393            char wavHeader[44];
2394            memcpy(wavHeader,
2395                "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0",
2396                sizeof(wavHeader));
2397            NBAIO_Format format = teeSource->format();
2398            unsigned channelCount = Format_channelCount(format);
2399            ALOG_ASSERT(channelCount <= FCC_2);
2400            uint32_t sampleRate = Format_sampleRate(format);
2401            wavHeader[22] = channelCount;       // number of channels
2402            wavHeader[24] = sampleRate;         // sample rate
2403            wavHeader[25] = sampleRate >> 8;
2404            wavHeader[32] = channelCount * 2;   // block alignment
2405            write(teeFd, wavHeader, sizeof(wavHeader));
2406            size_t total = 0;
2407            bool firstRead = true;
2408            for (;;) {
2409#define TEE_SINK_READ 1024
2410                short buffer[TEE_SINK_READ * FCC_2];
2411                size_t count = TEE_SINK_READ;
2412                ssize_t actual = teeSource->read(buffer, count,
2413                        AudioBufferProvider::kInvalidPTS);
2414                bool wasFirstRead = firstRead;
2415                firstRead = false;
2416                if (actual <= 0) {
2417                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2418                        continue;
2419                    }
2420                    break;
2421                }
2422                ALOG_ASSERT(actual <= (ssize_t)count);
2423                write(teeFd, buffer, actual * channelCount * sizeof(short));
2424                total += actual;
2425            }
2426            lseek(teeFd, (off_t) 4, SEEK_SET);
2427            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2428            write(teeFd, &temp, sizeof(temp));
2429            lseek(teeFd, (off_t) 40, SEEK_SET);
2430            temp =  total * channelCount * sizeof(short);
2431            write(teeFd, &temp, sizeof(temp));
2432            close(teeFd);
2433            if (fd >= 0) {
2434                fdprintf(fd, "tee copied to %s\n", teePath);
2435            }
2436        } else {
2437            if (fd >= 0) {
2438                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2439            }
2440        }
2441    }
2442}
2443#endif
2444
2445// ----------------------------------------------------------------------------
2446
2447status_t AudioFlinger::onTransact(
2448        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2449{
2450    return BnAudioFlinger::onTransact(code, data, reply, flags);
2451}
2452
2453}; // namespace android
2454