AudioFlinger.cpp revision 030033342a6ea17003e6af38a56c7edc6d2ead01 
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    *status = lStatus;
541    return trackHandle;
542}
543
544uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
545{
546    Mutex::Autolock _l(mLock);
547    PlaybackThread *thread = checkPlaybackThread_l(output);
548    if (thread == NULL) {
549        ALOGW("sampleRate() unknown thread %d", output);
550        return 0;
551    }
552    return thread->sampleRate();
553}
554
555int AudioFlinger::channelCount(audio_io_handle_t output) const
556{
557    Mutex::Autolock _l(mLock);
558    PlaybackThread *thread = checkPlaybackThread_l(output);
559    if (thread == NULL) {
560        ALOGW("channelCount() unknown thread %d", output);
561        return 0;
562    }
563    return thread->channelCount();
564}
565
566audio_format_t AudioFlinger::format(audio_io_handle_t output) const
567{
568    Mutex::Autolock _l(mLock);
569    PlaybackThread *thread = checkPlaybackThread_l(output);
570    if (thread == NULL) {
571        ALOGW("format() unknown thread %d", output);
572        return AUDIO_FORMAT_INVALID;
573    }
574    return thread->format();
575}
576
577size_t AudioFlinger::frameCount(audio_io_handle_t output) const
578{
579    Mutex::Autolock _l(mLock);
580    PlaybackThread *thread = checkPlaybackThread_l(output);
581    if (thread == NULL) {
582        ALOGW("frameCount() unknown thread %d", output);
583        return 0;
584    }
585    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
586    //       should examine all callers and fix them to handle smaller counts
587    return thread->frameCount();
588}
589
590uint32_t AudioFlinger::latency(audio_io_handle_t output) const
591{
592    Mutex::Autolock _l(mLock);
593    PlaybackThread *thread = checkPlaybackThread_l(output);
594    if (thread == NULL) {
595        ALOGW("latency(): no playback thread found for output handle %d", output);
596        return 0;
597    }
598    return thread->latency();
599}
600
601status_t AudioFlinger::setMasterVolume(float value)
602{
603    status_t ret = initCheck();
604    if (ret != NO_ERROR) {
605        return ret;
606    }
607
608    // check calling permissions
609    if (!settingsAllowed()) {
610        return PERMISSION_DENIED;
611    }
612
613    Mutex::Autolock _l(mLock);
614    mMasterVolume = value;
615
616    // Set master volume in the HALs which support it.
617    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
618        AutoMutex lock(mHardwareLock);
619        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
620
621        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
622        if (dev->canSetMasterVolume()) {
623            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
624        }
625        mHardwareStatus = AUDIO_HW_IDLE;
626    }
627
628    // Now set the master volume in each playback thread.  Playback threads
629    // assigned to HALs which do not have master volume support will apply
630    // master volume during the mix operation.  Threads with HALs which do
631    // support master volume will simply ignore the setting.
632    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
633        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
634
635    return NO_ERROR;
636}
637
638status_t AudioFlinger::setMode(audio_mode_t mode)
639{
640    status_t ret = initCheck();
641    if (ret != NO_ERROR) {
642        return ret;
643    }
644
645    // check calling permissions
646    if (!settingsAllowed()) {
647        return PERMISSION_DENIED;
648    }
649    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
650        ALOGW("Illegal value: setMode(%d)", mode);
651        return BAD_VALUE;
652    }
653
654    { // scope for the lock
655        AutoMutex lock(mHardwareLock);
656        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
657        mHardwareStatus = AUDIO_HW_SET_MODE;
658        ret = dev->set_mode(dev, mode);
659        mHardwareStatus = AUDIO_HW_IDLE;
660    }
661
662    if (NO_ERROR == ret) {
663        Mutex::Autolock _l(mLock);
664        mMode = mode;
665        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
666            mPlaybackThreads.valueAt(i)->setMode(mode);
667    }
668
669    return ret;
670}
671
672status_t AudioFlinger::setMicMute(bool state)
673{
674    status_t ret = initCheck();
675    if (ret != NO_ERROR) {
676        return ret;
677    }
678
679    // check calling permissions
680    if (!settingsAllowed()) {
681        return PERMISSION_DENIED;
682    }
683
684    AutoMutex lock(mHardwareLock);
685    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
686    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
687    ret = dev->set_mic_mute(dev, state);
688    mHardwareStatus = AUDIO_HW_IDLE;
689    return ret;
690}
691
692bool AudioFlinger::getMicMute() const
693{
694    status_t ret = initCheck();
695    if (ret != NO_ERROR) {
696        return false;
697    }
698
699    bool state = AUDIO_MODE_INVALID;
700    AutoMutex lock(mHardwareLock);
701    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
702    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
703    dev->get_mic_mute(dev, &state);
704    mHardwareStatus = AUDIO_HW_IDLE;
705    return state;
706}
707
708status_t AudioFlinger::setMasterMute(bool muted)
709{
710    status_t ret = initCheck();
711    if (ret != NO_ERROR) {
712        return ret;
713    }
714
715    // check calling permissions
716    if (!settingsAllowed()) {
717        return PERMISSION_DENIED;
718    }
719
720    Mutex::Autolock _l(mLock);
721    mMasterMute = muted;
722
723    // Set master mute in the HALs which support it.
724    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
725        AutoMutex lock(mHardwareLock);
726        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
727
728        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
729        if (dev->canSetMasterMute()) {
730            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
731        }
732        mHardwareStatus = AUDIO_HW_IDLE;
733    }
734
735    // Now set the master mute in each playback thread.  Playback threads
736    // assigned to HALs which do not have master mute support will apply master
737    // mute during the mix operation.  Threads with HALs which do support master
738    // mute will simply ignore the setting.
739    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
740        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
741
742    return NO_ERROR;
743}
744
745float AudioFlinger::masterVolume() const
746{
747    Mutex::Autolock _l(mLock);
748    return masterVolume_l();
749}
750
751bool AudioFlinger::masterMute() const
752{
753    Mutex::Autolock _l(mLock);
754    return masterMute_l();
755}
756
757float AudioFlinger::masterVolume_l() const
758{
759    return mMasterVolume;
760}
761
762bool AudioFlinger::masterMute_l() const
763{
764    return mMasterMute;
765}
766
767status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
768        audio_io_handle_t output)
769{
770    // check calling permissions
771    if (!settingsAllowed()) {
772        return PERMISSION_DENIED;
773    }
774
775    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
776        ALOGE("setStreamVolume() invalid stream %d", stream);
777        return BAD_VALUE;
778    }
779
780    AutoMutex lock(mLock);
781    PlaybackThread *thread = NULL;
782    if (output) {
783        thread = checkPlaybackThread_l(output);
784        if (thread == NULL) {
785            return BAD_VALUE;
786        }
787    }
788
789    mStreamTypes[stream].volume = value;
790
791    if (thread == NULL) {
792        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
793            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
794        }
795    } else {
796        thread->setStreamVolume(stream, value);
797    }
798
799    return NO_ERROR;
800}
801
802status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
803{
804    // check calling permissions
805    if (!settingsAllowed()) {
806        return PERMISSION_DENIED;
807    }
808
809    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
810        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
811        ALOGE("setStreamMute() invalid stream %d", stream);
812        return BAD_VALUE;
813    }
814
815    AutoMutex lock(mLock);
816    mStreamTypes[stream].mute = muted;
817    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
818        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
819
820    return NO_ERROR;
821}
822
823float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
824{
825    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
826        return 0.0f;
827    }
828
829    AutoMutex lock(mLock);
830    float volume;
831    if (output) {
832        PlaybackThread *thread = checkPlaybackThread_l(output);
833        if (thread == NULL) {
834            return 0.0f;
835        }
836        volume = thread->streamVolume(stream);
837    } else {
838        volume = streamVolume_l(stream);
839    }
840
841    return volume;
842}
843
844bool AudioFlinger::streamMute(audio_stream_type_t stream) const
845{
846    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
847        return true;
848    }
849
850    AutoMutex lock(mLock);
851    return streamMute_l(stream);
852}
853
854status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
855{
856    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
857            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
858
859    // check calling permissions
860    if (!settingsAllowed()) {
861        return PERMISSION_DENIED;
862    }
863
864    // ioHandle == 0 means the parameters are global to the audio hardware interface
865    if (ioHandle == 0) {
866        Mutex::Autolock _l(mLock);
867        status_t final_result = NO_ERROR;
868        {
869            AutoMutex lock(mHardwareLock);
870            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
871            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
872                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
873                status_t result = dev->set_parameters(dev, keyValuePairs.string());
874                final_result = result ?: final_result;
875            }
876            mHardwareStatus = AUDIO_HW_IDLE;
877        }
878        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
879        AudioParameter param = AudioParameter(keyValuePairs);
880        String8 value;
881        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
882            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
883            if (mBtNrecIsOff != btNrecIsOff) {
884                for (size_t i = 0; i < mRecordThreads.size(); i++) {
885                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
886                    audio_devices_t device = thread->inDevice();
887                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
888                    // collect all of the thread's session IDs
889                    KeyedVector<int, bool> ids = thread->sessionIds();
890                    // suspend effects associated with those session IDs
891                    for (size_t j = 0; j < ids.size(); ++j) {
892                        int sessionId = ids.keyAt(j);
893                        thread->setEffectSuspended(FX_IID_AEC,
894                                                   suspend,
895                                                   sessionId);
896                        thread->setEffectSuspended(FX_IID_NS,
897                                                   suspend,
898                                                   sessionId);
899                    }
900                }
901                mBtNrecIsOff = btNrecIsOff;
902            }
903        }
904        String8 screenState;
905        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
906            bool isOff = screenState == "off";
907            if (isOff != (AudioFlinger::mScreenState & 1)) {
908                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
909            }
910        }
911        return final_result;
912    }
913
914    // hold a strong ref on thread in case closeOutput() or closeInput() is called
915    // and the thread is exited once the lock is released
916    sp<ThreadBase> thread;
917    {
918        Mutex::Autolock _l(mLock);
919        thread = checkPlaybackThread_l(ioHandle);
920        if (thread == 0) {
921            thread = checkRecordThread_l(ioHandle);
922        } else if (thread == primaryPlaybackThread_l()) {
923            // indicate output device change to all input threads for pre processing
924            AudioParameter param = AudioParameter(keyValuePairs);
925            int value;
926            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
927                    (value != 0)) {
928                for (size_t i = 0; i < mRecordThreads.size(); i++) {
929                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
930                }
931            }
932        }
933    }
934    if (thread != 0) {
935        return thread->setParameters(keyValuePairs);
936    }
937    return BAD_VALUE;
938}
939
940String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
941{
942    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
943            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
944
945    Mutex::Autolock _l(mLock);
946
947    if (ioHandle == 0) {
948        String8 out_s8;
949
950        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
951            char *s;
952            {
953            AutoMutex lock(mHardwareLock);
954            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
955            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
956            s = dev->get_parameters(dev, keys.string());
957            mHardwareStatus = AUDIO_HW_IDLE;
958            }
959            out_s8 += String8(s ? s : "");
960            free(s);
961        }
962        return out_s8;
963    }
964
965    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
966    if (playbackThread != NULL) {
967        return playbackThread->getParameters(keys);
968    }
969    RecordThread *recordThread = checkRecordThread_l(ioHandle);
970    if (recordThread != NULL) {
971        return recordThread->getParameters(keys);
972    }
973    return String8("");
974}
975
976size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
977        audio_channel_mask_t channelMask) const
978{
979    status_t ret = initCheck();
980    if (ret != NO_ERROR) {
981        return 0;
982    }
983
984    AutoMutex lock(mHardwareLock);
985    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
986    struct audio_config config;
987    memset(&config, 0, sizeof(config));
988    config.sample_rate = sampleRate;
989    config.channel_mask = channelMask;
990    config.format = format;
991
992    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
993    size_t size = dev->get_input_buffer_size(dev, &config);
994    mHardwareStatus = AUDIO_HW_IDLE;
995    return size;
996}
997
998unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
999{
1000    Mutex::Autolock _l(mLock);
1001
1002    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1003    if (recordThread != NULL) {
1004        return recordThread->getInputFramesLost();
1005    }
1006    return 0;
1007}
1008
1009status_t AudioFlinger::setVoiceVolume(float value)
1010{
1011    status_t ret = initCheck();
1012    if (ret != NO_ERROR) {
1013        return ret;
1014    }
1015
1016    // check calling permissions
1017    if (!settingsAllowed()) {
1018        return PERMISSION_DENIED;
1019    }
1020
1021    AutoMutex lock(mHardwareLock);
1022    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1023    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1024    ret = dev->set_voice_volume(dev, value);
1025    mHardwareStatus = AUDIO_HW_IDLE;
1026
1027    return ret;
1028}
1029
1030status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
1031        audio_io_handle_t output) const
1032{
1033    status_t status;
1034
1035    Mutex::Autolock _l(mLock);
1036
1037    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1038    if (playbackThread != NULL) {
1039        return playbackThread->getRenderPosition(halFrames, dspFrames);
1040    }
1041
1042    return BAD_VALUE;
1043}
1044
1045void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1046{
1047
1048    Mutex::Autolock _l(mLock);
1049
1050    pid_t pid = IPCThreadState::self()->getCallingPid();
1051    if (mNotificationClients.indexOfKey(pid) < 0) {
1052        sp<NotificationClient> notificationClient = new NotificationClient(this,
1053                                                                            client,
1054                                                                            pid);
1055        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1056
1057        mNotificationClients.add(pid, notificationClient);
1058
1059        sp<IBinder> binder = client->asBinder();
1060        binder->linkToDeath(notificationClient);
1061
1062        // the config change is always sent from playback or record threads to avoid deadlock
1063        // with AudioSystem::gLock
1064        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1065            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1066        }
1067
1068        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1069            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1070        }
1071    }
1072}
1073
1074void AudioFlinger::removeNotificationClient(pid_t pid)
1075{
1076    Mutex::Autolock _l(mLock);
1077
1078    mNotificationClients.removeItem(pid);
1079
1080    ALOGV("%d died, releasing its sessions", pid);
1081    size_t num = mAudioSessionRefs.size();
1082    bool removed = false;
1083    for (size_t i = 0; i< num; ) {
1084        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1085        ALOGV(" pid %d @ %d", ref->mPid, i);
1086        if (ref->mPid == pid) {
1087            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1088            mAudioSessionRefs.removeAt(i);
1089            delete ref;
1090            removed = true;
1091            num--;
1092        } else {
1093            i++;
1094        }
1095    }
1096    if (removed) {
1097        purgeStaleEffects_l();
1098    }
1099}
1100
1101// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1102void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1103{
1104    size_t size = mNotificationClients.size();
1105    for (size_t i = 0; i < size; i++) {
1106        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1107                                                                               param2);
1108    }
1109}
1110
1111// removeClient_l() must be called with AudioFlinger::mLock held
1112void AudioFlinger::removeClient_l(pid_t pid)
1113{
1114    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1115            IPCThreadState::self()->getCallingPid());
1116    mClients.removeItem(pid);
1117}
1118
1119// getEffectThread_l() must be called with AudioFlinger::mLock held
1120sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1121{
1122    sp<PlaybackThread> thread;
1123
1124    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1125        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1126            ALOG_ASSERT(thread == 0);
1127            thread = mPlaybackThreads.valueAt(i);
1128        }
1129    }
1130
1131    return thread;
1132}
1133
1134
1135
1136// ----------------------------------------------------------------------------
1137
1138AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1139    :   RefBase(),
1140        mAudioFlinger(audioFlinger),
1141        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1142        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1143        mPid(pid),
1144        mTimedTrackCount(0)
1145{
1146    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1147}
1148
1149// Client destructor must be called with AudioFlinger::mLock held
1150AudioFlinger::Client::~Client()
1151{
1152    mAudioFlinger->removeClient_l(mPid);
1153}
1154
1155sp<MemoryDealer> AudioFlinger::Client::heap() const
1156{
1157    return mMemoryDealer;
1158}
1159
1160// Reserve one of the limited slots for a timed audio track associated
1161// with this client
1162bool AudioFlinger::Client::reserveTimedTrack()
1163{
1164    const int kMaxTimedTracksPerClient = 4;
1165
1166    Mutex::Autolock _l(mTimedTrackLock);
1167
1168    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1169        ALOGW("can not create timed track - pid %d has exceeded the limit",
1170             mPid);
1171        return false;
1172    }
1173
1174    mTimedTrackCount++;
1175    return true;
1176}
1177
1178// Release a slot for a timed audio track
1179void AudioFlinger::Client::releaseTimedTrack()
1180{
1181    Mutex::Autolock _l(mTimedTrackLock);
1182    mTimedTrackCount--;
1183}
1184
1185// ----------------------------------------------------------------------------
1186
1187AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1188                                                     const sp<IAudioFlingerClient>& client,
1189                                                     pid_t pid)
1190    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1191{
1192}
1193
1194AudioFlinger::NotificationClient::~NotificationClient()
1195{
1196}
1197
1198void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
1199{
1200    sp<NotificationClient> keep(this);
1201    mAudioFlinger->removeNotificationClient(mPid);
1202}
1203
1204
1205// ----------------------------------------------------------------------------
1206
1207sp<IAudioRecord> AudioFlinger::openRecord(
1208        audio_io_handle_t input,
1209        uint32_t sampleRate,
1210        audio_format_t format,
1211        audio_channel_mask_t channelMask,
1212        size_t frameCount,
1213        IAudioFlinger::track_flags_t *flags,
1214        pid_t tid,
1215        int *sessionId,
1216        status_t *status)
1217{
1218    sp<RecordThread::RecordTrack> recordTrack;
1219    sp<RecordHandle> recordHandle;
1220    sp<Client> client;
1221    status_t lStatus;
1222    RecordThread *thread;
1223    size_t inFrameCount;
1224    int lSessionId;
1225
1226    // check calling permissions
1227    if (!recordingAllowed()) {
1228        lStatus = PERMISSION_DENIED;
1229        goto Exit;
1230    }
1231
1232    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1233        ALOGE("openRecord() invalid format %d", format);
1234        lStatus = BAD_VALUE;
1235        goto Exit;
1236    }
1237
1238    // add client to list
1239    { // scope for mLock
1240        Mutex::Autolock _l(mLock);
1241        thread = checkRecordThread_l(input);
1242        if (thread == NULL) {
1243            lStatus = BAD_VALUE;
1244            goto Exit;
1245        }
1246
1247        pid_t pid = IPCThreadState::self()->getCallingPid();
1248        client = registerPid_l(pid);
1249
1250        // If no audio session id is provided, create one here
1251        if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
1252            lSessionId = *sessionId;
1253        } else {
1254            lSessionId = nextUniqueId();
1255            if (sessionId != NULL) {
1256                *sessionId = lSessionId;
1257            }
1258        }
1259        // create new record track.
1260        // The record track uses one track in mHardwareMixerThread by convention.
1261        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1262                                                  frameCount, lSessionId, flags, tid, &lStatus);
1263    }
1264    if (lStatus != NO_ERROR) {
1265        // remove local strong reference to Client before deleting the RecordTrack so that the
1266        // Client destructor is called by the TrackBase destructor with mLock held
1267        client.clear();
1268        recordTrack.clear();
1269        goto Exit;
1270    }
1271
1272    // return to handle to client
1273    recordHandle = new RecordHandle(recordTrack);
1274
1275Exit:
1276    *status = lStatus;
1277    return recordHandle;
1278}
1279
1280
1281
1282// ----------------------------------------------------------------------------
1283
1284audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1285{
1286    if (!settingsAllowed()) {
1287        return 0;
1288    }
1289    Mutex::Autolock _l(mLock);
1290    return loadHwModule_l(name);
1291}
1292
1293// loadHwModule_l() must be called with AudioFlinger::mLock held
1294audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1295{
1296    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1297        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1298            ALOGW("loadHwModule() module %s already loaded", name);
1299            return mAudioHwDevs.keyAt(i);
1300        }
1301    }
1302
1303    audio_hw_device_t *dev;
1304
1305    int rc = load_audio_interface(name, &dev);
1306    if (rc) {
1307        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1308        return 0;
1309    }
1310
1311    mHardwareStatus = AUDIO_HW_INIT;
1312    rc = dev->init_check(dev);
1313    mHardwareStatus = AUDIO_HW_IDLE;
1314    if (rc) {
1315        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1316        return 0;
1317    }
1318
1319    // Check and cache this HAL's level of support for master mute and master
1320    // volume.  If this is the first HAL opened, and it supports the get
1321    // methods, use the initial values provided by the HAL as the current
1322    // master mute and volume settings.
1323
1324    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1325    {  // scope for auto-lock pattern
1326        AutoMutex lock(mHardwareLock);
1327
1328        if (0 == mAudioHwDevs.size()) {
1329            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1330            if (NULL != dev->get_master_volume) {
1331                float mv;
1332                if (OK == dev->get_master_volume(dev, &mv)) {
1333                    mMasterVolume = mv;
1334                }
1335            }
1336
1337            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1338            if (NULL != dev->get_master_mute) {
1339                bool mm;
1340                if (OK == dev->get_master_mute(dev, &mm)) {
1341                    mMasterMute = mm;
1342                }
1343            }
1344        }
1345
1346        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1347        if ((NULL != dev->set_master_volume) &&
1348            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1349            flags = static_cast<AudioHwDevice::Flags>(flags |
1350                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1351        }
1352
1353        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1354        if ((NULL != dev->set_master_mute) &&
1355            (OK == dev->set_master_mute(dev, mMasterMute))) {
1356            flags = static_cast<AudioHwDevice::Flags>(flags |
1357                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1358        }
1359
1360        mHardwareStatus = AUDIO_HW_IDLE;
1361    }
1362
1363    audio_module_handle_t handle = nextUniqueId();
1364    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1365
1366    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1367          name, dev->common.module->name, dev->common.module->id, handle);
1368
1369    return handle;
1370
1371}
1372
1373// ----------------------------------------------------------------------------
1374
1375uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1376{
1377    Mutex::Autolock _l(mLock);
1378    PlaybackThread *thread = primaryPlaybackThread_l();
1379    return thread != NULL ? thread->sampleRate() : 0;
1380}
1381
1382size_t AudioFlinger::getPrimaryOutputFrameCount()
1383{
1384    Mutex::Autolock _l(mLock);
1385    PlaybackThread *thread = primaryPlaybackThread_l();
1386    return thread != NULL ? thread->frameCountHAL() : 0;
1387}
1388
1389// ----------------------------------------------------------------------------
1390
1391status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1392{
1393    uid_t uid = IPCThreadState::self()->getCallingUid();
1394    if (uid != AID_SYSTEM) {
1395        return PERMISSION_DENIED;
1396    }
1397    Mutex::Autolock _l(mLock);
1398    if (mIsDeviceTypeKnown) {
1399        return INVALID_OPERATION;
1400    }
1401    mIsLowRamDevice = isLowRamDevice;
1402    mIsDeviceTypeKnown = true;
1403    return NO_ERROR;
1404}
1405
1406// ----------------------------------------------------------------------------
1407
1408audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1409                                           audio_devices_t *pDevices,
1410                                           uint32_t *pSamplingRate,
1411                                           audio_format_t *pFormat,
1412                                           audio_channel_mask_t *pChannelMask,
1413                                           uint32_t *pLatencyMs,
1414                                           audio_output_flags_t flags,
1415                                           const audio_offload_info_t *offloadInfo)
1416{
1417    PlaybackThread *thread = NULL;
1418    struct audio_config config;
1419    memset(&config, 0, sizeof(config));
1420    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1421    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1422    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1423    if (offloadInfo != NULL) {
1424        config.offload_info = *offloadInfo;
1425    }
1426
1427    audio_stream_out_t *outStream = NULL;
1428    AudioHwDevice *outHwDev;
1429
1430    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1431              module,
1432              (pDevices != NULL) ? *pDevices : 0,
1433              config.sample_rate,
1434              config.format,
1435              config.channel_mask,
1436              flags);
1437    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1438          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version );
1439
1440    if (pDevices == NULL || *pDevices == 0) {
1441        return 0;
1442    }
1443
1444    Mutex::Autolock _l(mLock);
1445
1446    outHwDev = findSuitableHwDev_l(module, *pDevices);
1447    if (outHwDev == NULL)
1448        return 0;
1449
1450    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1451    audio_io_handle_t id = nextUniqueId();
1452
1453    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1454
1455    status_t status = hwDevHal->open_output_stream(hwDevHal,
1456                                          id,
1457                                          *pDevices,
1458                                          (audio_output_flags_t)flags,
1459                                          &config,
1460                                          &outStream);
1461
1462    mHardwareStatus = AUDIO_HW_IDLE;
1463    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1464            "Channels %x, status %d",
1465            outStream,
1466            config.sample_rate,
1467            config.format,
1468            config.channel_mask,
1469            status);
1470
1471    if (status == NO_ERROR && outStream != NULL) {
1472        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1473
1474        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1475            thread = new OffloadThread(this, output, id, *pDevices);
1476            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1477        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1478            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1479            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1480            thread = new DirectOutputThread(this, output, id, *pDevices);
1481            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1482        } else {
1483            thread = new MixerThread(this, output, id, *pDevices);
1484            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1485        }
1486        mPlaybackThreads.add(id, thread);
1487
1488        if (pSamplingRate != NULL) {
1489            *pSamplingRate = config.sample_rate;
1490        }
1491        if (pFormat != NULL) {
1492            *pFormat = config.format;
1493        }
1494        if (pChannelMask != NULL) {
1495            *pChannelMask = config.channel_mask;
1496        }
1497        if (pLatencyMs != NULL) {
1498            *pLatencyMs = thread->latency();
1499        }
1500
1501        // notify client processes of the new output creation
1502        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1503
1504        // the first primary output opened designates the primary hw device
1505        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1506            ALOGI("Using module %d has the primary audio interface", module);
1507            mPrimaryHardwareDev = outHwDev;
1508
1509            AutoMutex lock(mHardwareLock);
1510            mHardwareStatus = AUDIO_HW_SET_MODE;
1511            hwDevHal->set_mode(hwDevHal, mMode);
1512            mHardwareStatus = AUDIO_HW_IDLE;
1513        }
1514        return id;
1515    }
1516
1517    return 0;
1518}
1519
1520audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1521        audio_io_handle_t output2)
1522{
1523    Mutex::Autolock _l(mLock);
1524    MixerThread *thread1 = checkMixerThread_l(output1);
1525    MixerThread *thread2 = checkMixerThread_l(output2);
1526
1527    if (thread1 == NULL || thread2 == NULL) {
1528        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1529                output2);
1530        return 0;
1531    }
1532
1533    audio_io_handle_t id = nextUniqueId();
1534    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1535    thread->addOutputTrack(thread2);
1536    mPlaybackThreads.add(id, thread);
1537    // notify client processes of the new output creation
1538    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1539    return id;
1540}
1541
1542status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1543{
1544    return closeOutput_nonvirtual(output);
1545}
1546
1547status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1548{
1549    // keep strong reference on the playback thread so that
1550    // it is not destroyed while exit() is executed
1551    sp<PlaybackThread> thread;
1552    {
1553        Mutex::Autolock _l(mLock);
1554        thread = checkPlaybackThread_l(output);
1555        if (thread == NULL) {
1556            return BAD_VALUE;
1557        }
1558
1559        ALOGV("closeOutput() %d", output);
1560
1561        if (thread->type() == ThreadBase::MIXER) {
1562            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1563                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1564                    DuplicatingThread *dupThread =
1565                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1566                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1567
1568                }
1569            }
1570        }
1571
1572
1573        mPlaybackThreads.removeItem(output);
1574        // save all effects to the default thread
1575        if (mPlaybackThreads.size()) {
1576            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1577            if (dstThread != NULL) {
1578                // audioflinger lock is held here so the acquisition order of thread locks does not
1579                // matter
1580                Mutex::Autolock _dl(dstThread->mLock);
1581                Mutex::Autolock _sl(thread->mLock);
1582                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1583                for (size_t i = 0; i < effectChains.size(); i ++) {
1584                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1585                }
1586            }
1587        }
1588        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1589    }
1590    thread->exit();
1591    // The thread entity (active unit of execution) is no longer running here,
1592    // but the ThreadBase container still exists.
1593
1594    if (thread->type() != ThreadBase::DUPLICATING) {
1595        AudioStreamOut *out = thread->clearOutput();
1596        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1597        // from now on thread->mOutput is NULL
1598        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1599        delete out;
1600    }
1601    return NO_ERROR;
1602}
1603
1604status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1605{
1606    Mutex::Autolock _l(mLock);
1607    PlaybackThread *thread = checkPlaybackThread_l(output);
1608
1609    if (thread == NULL) {
1610        return BAD_VALUE;
1611    }
1612
1613    ALOGV("suspendOutput() %d", output);
1614    thread->suspend();
1615
1616    return NO_ERROR;
1617}
1618
1619status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1620{
1621    Mutex::Autolock _l(mLock);
1622    PlaybackThread *thread = checkPlaybackThread_l(output);
1623
1624    if (thread == NULL) {
1625        return BAD_VALUE;
1626    }
1627
1628    ALOGV("restoreOutput() %d", output);
1629
1630    thread->restore();
1631
1632    return NO_ERROR;
1633}
1634
1635audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1636                                          audio_devices_t *pDevices,
1637                                          uint32_t *pSamplingRate,
1638                                          audio_format_t *pFormat,
1639                                          audio_channel_mask_t *pChannelMask)
1640{
1641    status_t status;
1642    RecordThread *thread = NULL;
1643    struct audio_config config;
1644    memset(&config, 0, sizeof(config));
1645    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1646    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1647    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1648
1649    uint32_t reqSamplingRate = config.sample_rate;
1650    audio_format_t reqFormat = config.format;
1651    audio_channel_mask_t reqChannelMask = config.channel_mask;
1652    audio_stream_in_t *inStream = NULL;
1653    AudioHwDevice *inHwDev;
1654
1655    if (pDevices == NULL || *pDevices == 0) {
1656        return 0;
1657    }
1658
1659    Mutex::Autolock _l(mLock);
1660
1661    inHwDev = findSuitableHwDev_l(module, *pDevices);
1662    if (inHwDev == NULL)
1663        return 0;
1664
1665    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1666    audio_io_handle_t id = nextUniqueId();
1667
1668    status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1669                                        &inStream);
1670    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
1671            "status %d",
1672            inStream,
1673            config.sample_rate,
1674            config.format,
1675            config.channel_mask,
1676            status);
1677
1678    // If the input could not be opened with the requested parameters and we can handle the
1679    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1680    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1681    if (status == BAD_VALUE &&
1682        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1683        (config.sample_rate <= 2 * reqSamplingRate) &&
1684        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1685        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1686        inStream = NULL;
1687        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1688    }
1689
1690    if (status == NO_ERROR && inStream != NULL) {
1691
1692#ifdef TEE_SINK
1693        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1694        // or (re-)create if current Pipe is idle and does not match the new format
1695        sp<NBAIO_Sink> teeSink;
1696        enum {
1697            TEE_SINK_NO,    // don't copy input
1698            TEE_SINK_NEW,   // copy input using a new pipe
1699            TEE_SINK_OLD,   // copy input using an existing pipe
1700        } kind;
1701        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1702                                        popcount(inStream->common.get_channels(&inStream->common)));
1703        if (!mTeeSinkInputEnabled) {
1704            kind = TEE_SINK_NO;
1705        } else if (format == Format_Invalid) {
1706            kind = TEE_SINK_NO;
1707        } else if (mRecordTeeSink == 0) {
1708            kind = TEE_SINK_NEW;
1709        } else if (mRecordTeeSink->getStrongCount() != 1) {
1710            kind = TEE_SINK_NO;
1711        } else if (format == mRecordTeeSink->format()) {
1712            kind = TEE_SINK_OLD;
1713        } else {
1714            kind = TEE_SINK_NEW;
1715        }
1716        switch (kind) {
1717        case TEE_SINK_NEW: {
1718            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1719            size_t numCounterOffers = 0;
1720            const NBAIO_Format offers[1] = {format};
1721            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1722            ALOG_ASSERT(index == 0);
1723            PipeReader *pipeReader = new PipeReader(*pipe);
1724            numCounterOffers = 0;
1725            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1726            ALOG_ASSERT(index == 0);
1727            mRecordTeeSink = pipe;
1728            mRecordTeeSource = pipeReader;
1729            teeSink = pipe;
1730            }
1731            break;
1732        case TEE_SINK_OLD:
1733            teeSink = mRecordTeeSink;
1734            break;
1735        case TEE_SINK_NO:
1736        default:
1737            break;
1738        }
1739#endif
1740
1741        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1742
1743        // Start record thread
1744        // RecordThread requires both input and output device indication to forward to audio
1745        // pre processing modules
1746        thread = new RecordThread(this,
1747                                  input,
1748                                  reqSamplingRate,
1749                                  reqChannelMask,
1750                                  id,
1751                                  primaryOutputDevice_l(),
1752                                  *pDevices
1753#ifdef TEE_SINK
1754                                  , teeSink
1755#endif
1756                                  );
1757        mRecordThreads.add(id, thread);
1758        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1759        if (pSamplingRate != NULL) {
1760            *pSamplingRate = reqSamplingRate;
1761        }
1762        if (pFormat != NULL) {
1763            *pFormat = config.format;
1764        }
1765        if (pChannelMask != NULL) {
1766            *pChannelMask = reqChannelMask;
1767        }
1768
1769        // notify client processes of the new input creation
1770        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1771        return id;
1772    }
1773
1774    return 0;
1775}
1776
1777status_t AudioFlinger::closeInput(audio_io_handle_t input)
1778{
1779    return closeInput_nonvirtual(input);
1780}
1781
1782status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1783{
1784    // keep strong reference on the record thread so that
1785    // it is not destroyed while exit() is executed
1786    sp<RecordThread> thread;
1787    {
1788        Mutex::Autolock _l(mLock);
1789        thread = checkRecordThread_l(input);
1790        if (thread == 0) {
1791            return BAD_VALUE;
1792        }
1793
1794        ALOGV("closeInput() %d", input);
1795        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1796        mRecordThreads.removeItem(input);
1797    }
1798    thread->exit();
1799    // The thread entity (active unit of execution) is no longer running here,
1800    // but the ThreadBase container still exists.
1801
1802    AudioStreamIn *in = thread->clearInput();
1803    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1804    // from now on thread->mInput is NULL
1805    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1806    delete in;
1807
1808    return NO_ERROR;
1809}
1810
1811status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
1812{
1813    Mutex::Autolock _l(mLock);
1814    ALOGV("setStreamOutput() stream %d to output %d", stream, output);
1815
1816    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1817        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1818        thread->invalidateTracks(stream);
1819    }
1820
1821    return NO_ERROR;
1822}
1823
1824
1825int AudioFlinger::newAudioSessionId()
1826{
1827    return nextUniqueId();
1828}
1829
1830void AudioFlinger::acquireAudioSessionId(int audioSession)
1831{
1832    Mutex::Autolock _l(mLock);
1833    pid_t caller = IPCThreadState::self()->getCallingPid();
1834    ALOGV("acquiring %d from %d", audioSession, caller);
1835    size_t num = mAudioSessionRefs.size();
1836    for (size_t i = 0; i< num; i++) {
1837        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1838        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1839            ref->mCnt++;
1840            ALOGV(" incremented refcount to %d", ref->mCnt);
1841            return;
1842        }
1843    }
1844    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1845    ALOGV(" added new entry for %d", audioSession);
1846}
1847
1848void AudioFlinger::releaseAudioSessionId(int audioSession)
1849{
1850    Mutex::Autolock _l(mLock);
1851    pid_t caller = IPCThreadState::self()->getCallingPid();
1852    ALOGV("releasing %d from %d", audioSession, caller);
1853    size_t num = mAudioSessionRefs.size();
1854    for (size_t i = 0; i< num; i++) {
1855        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1856        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1857            ref->mCnt--;
1858            ALOGV(" decremented refcount to %d", ref->mCnt);
1859            if (ref->mCnt == 0) {
1860                mAudioSessionRefs.removeAt(i);
1861                delete ref;
1862                purgeStaleEffects_l();
1863            }
1864            return;
1865        }
1866    }
1867    ALOGW("session id %d not found for pid %d", audioSession, caller);
1868}
1869
1870void AudioFlinger::purgeStaleEffects_l() {
1871
1872    ALOGV("purging stale effects");
1873
1874    Vector< sp<EffectChain> > chains;
1875
1876    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1877        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
1878        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1879            sp<EffectChain> ec = t->mEffectChains[j];
1880            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
1881                chains.push(ec);
1882            }
1883        }
1884    }
1885    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1886        sp<RecordThread> t = mRecordThreads.valueAt(i);
1887        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
1888            sp<EffectChain> ec = t->mEffectChains[j];
1889            chains.push(ec);
1890        }
1891    }
1892
1893    for (size_t i = 0; i < chains.size(); i++) {
1894        sp<EffectChain> ec = chains[i];
1895        int sessionid = ec->sessionId();
1896        sp<ThreadBase> t = ec->mThread.promote();
1897        if (t == 0) {
1898            continue;
1899        }
1900        size_t numsessionrefs = mAudioSessionRefs.size();
1901        bool found = false;
1902        for (size_t k = 0; k < numsessionrefs; k++) {
1903            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
1904            if (ref->mSessionid == sessionid) {
1905                ALOGV(" session %d still exists for %d with %d refs",
1906                    sessionid, ref->mPid, ref->mCnt);
1907                found = true;
1908                break;
1909            }
1910        }
1911        if (!found) {
1912            Mutex::Autolock _l (t->mLock);
1913            // remove all effects from the chain
1914            while (ec->mEffects.size()) {
1915                sp<EffectModule> effect = ec->mEffects[0];
1916                effect->unPin();
1917                t->removeEffect_l(effect);
1918                if (effect->purgeHandles()) {
1919                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
1920                }
1921                AudioSystem::unregisterEffect(effect->id());
1922            }
1923        }
1924    }
1925    return;
1926}
1927
1928// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
1929AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
1930{
1931    return mPlaybackThreads.valueFor(output).get();
1932}
1933
1934// checkMixerThread_l() must be called with AudioFlinger::mLock held
1935AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
1936{
1937    PlaybackThread *thread = checkPlaybackThread_l(output);
1938    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
1939}
1940
1941// checkRecordThread_l() must be called with AudioFlinger::mLock held
1942AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
1943{
1944    return mRecordThreads.valueFor(input).get();
1945}
1946
1947uint32_t AudioFlinger::nextUniqueId()
1948{
1949    return android_atomic_inc(&mNextUniqueId);
1950}
1951
1952AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
1953{
1954    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1955        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1956        AudioStreamOut *output = thread->getOutput();
1957        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
1958            return thread;
1959        }
1960    }
1961    return NULL;
1962}
1963
1964audio_devices_t AudioFlinger::primaryOutputDevice_l() const
1965{
1966    PlaybackThread *thread = primaryPlaybackThread_l();
1967
1968    if (thread == NULL) {
1969        return 0;
1970    }
1971
1972    return thread->outDevice();
1973}
1974
1975sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
1976                                    int triggerSession,
1977                                    int listenerSession,
1978                                    sync_event_callback_t callBack,
1979                                    void *cookie)
1980{
1981    Mutex::Autolock _l(mLock);
1982
1983    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
1984    status_t playStatus = NAME_NOT_FOUND;
1985    status_t recStatus = NAME_NOT_FOUND;
1986    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1987        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
1988        if (playStatus == NO_ERROR) {
1989            return event;
1990        }
1991    }
1992    for (size_t i = 0; i < mRecordThreads.size(); i++) {
1993        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
1994        if (recStatus == NO_ERROR) {
1995            return event;
1996        }
1997    }
1998    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
1999        mPendingSyncEvents.add(event);
2000    } else {
2001        ALOGV("createSyncEvent() invalid event %d", event->type());
2002        event.clear();
2003    }
2004    return event;
2005}
2006
2007// ----------------------------------------------------------------------------
2008//  Effect management
2009// ----------------------------------------------------------------------------
2010
2011
2012status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2013{
2014    Mutex::Autolock _l(mLock);
2015    return EffectQueryNumberEffects(numEffects);
2016}
2017
2018status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2019{
2020    Mutex::Autolock _l(mLock);
2021    return EffectQueryEffect(index, descriptor);
2022}
2023
2024status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2025        effect_descriptor_t *descriptor) const
2026{
2027    Mutex::Autolock _l(mLock);
2028    return EffectGetDescriptor(pUuid, descriptor);
2029}
2030
2031
2032sp<IEffect> AudioFlinger::createEffect(
2033        effect_descriptor_t *pDesc,
2034        const sp<IEffectClient>& effectClient,
2035        int32_t priority,
2036        audio_io_handle_t io,
2037        int sessionId,
2038        status_t *status,
2039        int *id,
2040        int *enabled)
2041{
2042    status_t lStatus = NO_ERROR;
2043    sp<EffectHandle> handle;
2044    effect_descriptor_t desc;
2045
2046    pid_t pid = IPCThreadState::self()->getCallingPid();
2047    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2048            pid, effectClient.get(), priority, sessionId, io);
2049
2050    if (pDesc == NULL) {
2051        lStatus = BAD_VALUE;
2052        goto Exit;
2053    }
2054
2055    // check audio settings permission for global effects
2056    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2057        lStatus = PERMISSION_DENIED;
2058        goto Exit;
2059    }
2060
2061    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2062    // that can only be created by audio policy manager (running in same process)
2063    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2064        lStatus = PERMISSION_DENIED;
2065        goto Exit;
2066    }
2067
2068    if (io == 0) {
2069        if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2070            // output must be specified by AudioPolicyManager when using session
2071            // AUDIO_SESSION_OUTPUT_STAGE
2072            lStatus = BAD_VALUE;
2073            goto Exit;
2074        } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2075            // if the output returned by getOutputForEffect() is removed before we lock the
2076            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2077            // and we will exit safely
2078            io = AudioSystem::getOutputForEffect(&desc);
2079        }
2080    }
2081
2082    {
2083        Mutex::Autolock _l(mLock);
2084
2085
2086        if (!EffectIsNullUuid(&pDesc->uuid)) {
2087            // if uuid is specified, request effect descriptor
2088            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2089            if (lStatus < 0) {
2090                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2091                goto Exit;
2092            }
2093        } else {
2094            // if uuid is not specified, look for an available implementation
2095            // of the required type in effect factory
2096            if (EffectIsNullUuid(&pDesc->type)) {
2097                ALOGW("createEffect() no effect type");
2098                lStatus = BAD_VALUE;
2099                goto Exit;
2100            }
2101            uint32_t numEffects = 0;
2102            effect_descriptor_t d;
2103            d.flags = 0; // prevent compiler warning
2104            bool found = false;
2105
2106            lStatus = EffectQueryNumberEffects(&numEffects);
2107            if (lStatus < 0) {
2108                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2109                goto Exit;
2110            }
2111            for (uint32_t i = 0; i < numEffects; i++) {
2112                lStatus = EffectQueryEffect(i, &desc);
2113                if (lStatus < 0) {
2114                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2115                    continue;
2116                }
2117                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2118                    // If matching type found save effect descriptor. If the session is
2119                    // 0 and the effect is not auxiliary, continue enumeration in case
2120                    // an auxiliary version of this effect type is available
2121                    found = true;
2122                    d = desc;
2123                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2124                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2125                        break;
2126                    }
2127                }
2128            }
2129            if (!found) {
2130                lStatus = BAD_VALUE;
2131                ALOGW("createEffect() effect not found");
2132                goto Exit;
2133            }
2134            // For same effect type, chose auxiliary version over insert version if
2135            // connect to output mix (Compliance to OpenSL ES)
2136            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2137                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2138                desc = d;
2139            }
2140        }
2141
2142        // Do not allow auxiliary effects on a session different from 0 (output mix)
2143        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2144             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2145            lStatus = INVALID_OPERATION;
2146            goto Exit;
2147        }
2148
2149        // check recording permission for visualizer
2150        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2151            !recordingAllowed()) {
2152            lStatus = PERMISSION_DENIED;
2153            goto Exit;
2154        }
2155
2156        // return effect descriptor
2157        *pDesc = desc;
2158
2159        // If output is not specified try to find a matching audio session ID in one of the
2160        // output threads.
2161        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2162        // because of code checking output when entering the function.
2163        // Note: io is never 0 when creating an effect on an input
2164        if (io == 0) {
2165            // look for the thread where the specified audio session is present
2166            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2167                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2168                    io = mPlaybackThreads.keyAt(i);
2169                    break;
2170                }
2171            }
2172            if (io == 0) {
2173                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2174                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2175                        io = mRecordThreads.keyAt(i);
2176                        break;
2177                    }
2178                }
2179            }
2180            // If no output thread contains the requested session ID, default to
2181            // first output. The effect chain will be moved to the correct output
2182            // thread when a track with the same session ID is created
2183            if (io == 0 && mPlaybackThreads.size()) {
2184                io = mPlaybackThreads.keyAt(0);
2185            }
2186            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2187        }
2188        ThreadBase *thread = checkRecordThread_l(io);
2189        if (thread == NULL) {
2190            thread = checkPlaybackThread_l(io);
2191            if (thread == NULL) {
2192                ALOGE("createEffect() unknown output thread");
2193                lStatus = BAD_VALUE;
2194                goto Exit;
2195            }
2196        }
2197
2198        sp<Client> client = registerPid_l(pid);
2199
2200        // create effect on selected output thread
2201        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2202                &desc, enabled, &lStatus);
2203        if (handle != 0 && id != NULL) {
2204            *id = handle->id();
2205        }
2206    }
2207
2208Exit:
2209    *status = lStatus;
2210    return handle;
2211}
2212
2213status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2214        audio_io_handle_t dstOutput)
2215{
2216    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2217            sessionId, srcOutput, dstOutput);
2218    Mutex::Autolock _l(mLock);
2219    if (srcOutput == dstOutput) {
2220        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2221        return NO_ERROR;
2222    }
2223    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2224    if (srcThread == NULL) {
2225        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2226        return BAD_VALUE;
2227    }
2228    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2229    if (dstThread == NULL) {
2230        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2231        return BAD_VALUE;
2232    }
2233
2234    Mutex::Autolock _dl(dstThread->mLock);
2235    Mutex::Autolock _sl(srcThread->mLock);
2236    moveEffectChain_l(sessionId, srcThread, dstThread, false);
2237
2238    return NO_ERROR;
2239}
2240
2241// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2242status_t AudioFlinger::moveEffectChain_l(int sessionId,
2243                                   AudioFlinger::PlaybackThread *srcThread,
2244                                   AudioFlinger::PlaybackThread *dstThread,
2245                                   bool reRegister)
2246{
2247    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2248            sessionId, srcThread, dstThread);
2249
2250    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2251    if (chain == 0) {
2252        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2253                sessionId, srcThread);
2254        return INVALID_OPERATION;
2255    }
2256
2257    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2258    // so that a new chain is created with correct parameters when first effect is added. This is
2259    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2260    // removed.
2261    srcThread->removeEffectChain_l(chain);
2262
2263    // transfer all effects one by one so that new effect chain is created on new thread with
2264    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2265    audio_io_handle_t dstOutput = dstThread->id();
2266    sp<EffectChain> dstChain;
2267    uint32_t strategy = 0; // prevent compiler warning
2268    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2269    while (effect != 0) {
2270        srcThread->removeEffect_l(effect);
2271        dstThread->addEffect_l(effect);
2272        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2273        if (effect->state() == EffectModule::ACTIVE ||
2274                effect->state() == EffectModule::STOPPING) {
2275            effect->start();
2276        }
2277        // if the move request is not received from audio policy manager, the effect must be
2278        // re-registered with the new strategy and output
2279        if (dstChain == 0) {
2280            dstChain = effect->chain().promote();
2281            if (dstChain == 0) {
2282                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2283                srcThread->addEffect_l(effect);
2284                return NO_INIT;
2285            }
2286            strategy = dstChain->strategy();
2287        }
2288        if (reRegister) {
2289            AudioSystem::unregisterEffect(effect->id());
2290            AudioSystem::registerEffect(&effect->desc(),
2291                                        dstOutput,
2292                                        strategy,
2293                                        sessionId,
2294                                        effect->id());
2295        }
2296        effect = chain->getEffectFromId_l(0);
2297    }
2298
2299    return NO_ERROR;
2300}
2301
2302struct Entry {
2303#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2304    char mName[MAX_NAME];
2305};
2306
2307int comparEntry(const void *p1, const void *p2)
2308{
2309    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2310}
2311
2312#ifdef TEE_SINK
2313void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2314{
2315    NBAIO_Source *teeSource = source.get();
2316    if (teeSource != NULL) {
2317        // .wav rotation
2318        // There is a benign race condition if 2 threads call this simultaneously.
2319        // They would both traverse the directory, but the result would simply be
2320        // failures at unlink() which are ignored.  It's also unlikely since
2321        // normally dumpsys is only done by bugreport or from the command line.
2322        char teePath[32+256];
2323        strcpy(teePath, "/data/misc/media");
2324        size_t teePathLen = strlen(teePath);
2325        DIR *dir = opendir(teePath);
2326        teePath[teePathLen++] = '/';
2327        if (dir != NULL) {
2328#define MAX_SORT 20 // number of entries to sort
2329#define MAX_KEEP 10 // number of entries to keep
2330            struct Entry entries[MAX_SORT];
2331            size_t entryCount = 0;
2332            while (entryCount < MAX_SORT) {
2333                struct dirent de;
2334                struct dirent *result = NULL;
2335                int rc = readdir_r(dir, &de, &result);
2336                if (rc != 0) {
2337                    ALOGW("readdir_r failed %d", rc);
2338                    break;
2339                }
2340                if (result == NULL) {
2341                    break;
2342                }
2343                if (result != &de) {
2344                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2345                    break;
2346                }
2347                // ignore non .wav file entries
2348                size_t nameLen = strlen(de.d_name);
2349                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2350                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2351                    continue;
2352                }
2353                strcpy(entries[entryCount++].mName, de.d_name);
2354            }
2355            (void) closedir(dir);
2356            if (entryCount > MAX_KEEP) {
2357                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2358                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2359                    strcpy(&teePath[teePathLen], entries[i].mName);
2360                    (void) unlink(teePath);
2361                }
2362            }
2363        } else {
2364            if (fd >= 0) {
2365                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2366            }
2367        }
2368        char teeTime[16];
2369        struct timeval tv;
2370        gettimeofday(&tv, NULL);
2371        struct tm tm;
2372        localtime_r(&tv.tv_sec, &tm);
2373        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2374        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2375        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2376        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2377        if (teeFd >= 0) {
2378            char wavHeader[44];
2379            memcpy(wavHeader,
2380                "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",
2381                sizeof(wavHeader));
2382            NBAIO_Format format = teeSource->format();
2383            unsigned channelCount = Format_channelCount(format);
2384            ALOG_ASSERT(channelCount <= FCC_2);
2385            uint32_t sampleRate = Format_sampleRate(format);
2386            wavHeader[22] = channelCount;       // number of channels
2387            wavHeader[24] = sampleRate;         // sample rate
2388            wavHeader[25] = sampleRate >> 8;
2389            wavHeader[32] = channelCount * 2;   // block alignment
2390            write(teeFd, wavHeader, sizeof(wavHeader));
2391            size_t total = 0;
2392            bool firstRead = true;
2393            for (;;) {
2394#define TEE_SINK_READ 1024
2395                short buffer[TEE_SINK_READ * FCC_2];
2396                size_t count = TEE_SINK_READ;
2397                ssize_t actual = teeSource->read(buffer, count,
2398                        AudioBufferProvider::kInvalidPTS);
2399                bool wasFirstRead = firstRead;
2400                firstRead = false;
2401                if (actual <= 0) {
2402                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2403                        continue;
2404                    }
2405                    break;
2406                }
2407                ALOG_ASSERT(actual <= (ssize_t)count);
2408                write(teeFd, buffer, actual * channelCount * sizeof(short));
2409                total += actual;
2410            }
2411            lseek(teeFd, (off_t) 4, SEEK_SET);
2412            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2413            write(teeFd, &temp, sizeof(temp));
2414            lseek(teeFd, (off_t) 40, SEEK_SET);
2415            temp =  total * channelCount * sizeof(short);
2416            write(teeFd, &temp, sizeof(temp));
2417            close(teeFd);
2418            if (fd >= 0) {
2419                fdprintf(fd, "tee copied to %s\n", teePath);
2420            }
2421        } else {
2422            if (fd >= 0) {
2423                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2424            }
2425        }
2426    }
2427}
2428#endif
2429
2430// ----------------------------------------------------------------------------
2431
2432status_t AudioFlinger::onTransact(
2433        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2434{
2435    return BnAudioFlinger::onTransact(code, data, reply, flags);
2436}
2437
2438}; // namespace android
2439