AudioFlinger.cpp revision 5806b3533437e3d83208e8e9d6bd74ed304e51ec 
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";
85static const char kClientLockedString[] = "Client lock is taken\n";
86
87
88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
89
90uint32_t AudioFlinger::mScreenState;
91
92#ifdef TEE_SINK
93bool AudioFlinger::mTeeSinkInputEnabled = false;
94bool AudioFlinger::mTeeSinkOutputEnabled = false;
95bool AudioFlinger::mTeeSinkTrackEnabled = false;
96
97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
100#endif
101
102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
103// we define a minimum time during which a global effect is considered enabled.
104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
105
106// ----------------------------------------------------------------------------
107
108const char *formatToString(audio_format_t format) {
109    switch(format) {
110    case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8";
111    case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16";
112    case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32";
113    case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24";
114    case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24";
115    case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat";
116    case AUDIO_FORMAT_MP3: return "mp3";
117    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
118    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
119    case AUDIO_FORMAT_AAC: return "aac";
120    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
121    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
122    case AUDIO_FORMAT_VORBIS: return "vorbis";
123    default:
124        break;
125    }
126    return "unknown";
127}
128
129static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
130{
131    const hw_module_t *mod;
132    int rc;
133
134    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
135    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
136                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
137    if (rc) {
138        goto out;
139    }
140    rc = audio_hw_device_open(mod, dev);
141    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
142                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
143    if (rc) {
144        goto out;
145    }
146    if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
147        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
148        rc = BAD_VALUE;
149        goto out;
150    }
151    return 0;
152
153out:
154    *dev = NULL;
155    return rc;
156}
157
158// ----------------------------------------------------------------------------
159
160AudioFlinger::AudioFlinger()
161    : BnAudioFlinger(),
162      mPrimaryHardwareDev(NULL),
163      mAudioHwDevs(NULL),
164      mHardwareStatus(AUDIO_HW_IDLE),
165      mMasterVolume(1.0f),
166      mMasterMute(false),
167      mNextUniqueId(1),
168      mMode(AUDIO_MODE_INVALID),
169      mBtNrecIsOff(false),
170      mIsLowRamDevice(true),
171      mIsDeviceTypeKnown(false),
172      mGlobalEffectEnableTime(0)
173{
174    getpid_cached = getpid();
175    char value[PROPERTY_VALUE_MAX];
176    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
177    if (doLog) {
178        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY);
179    }
180#ifdef TEE_SINK
181    (void) property_get("ro.debuggable", value, "0");
182    int debuggable = atoi(value);
183    int teeEnabled = 0;
184    if (debuggable) {
185        (void) property_get("af.tee", value, "0");
186        teeEnabled = atoi(value);
187    }
188    // FIXME symbolic constants here
189    if (teeEnabled & 1) {
190        mTeeSinkInputEnabled = true;
191    }
192    if (teeEnabled & 2) {
193        mTeeSinkOutputEnabled = true;
194    }
195    if (teeEnabled & 4) {
196        mTeeSinkTrackEnabled = true;
197    }
198#endif
199}
200
201void AudioFlinger::onFirstRef()
202{
203    int rc = 0;
204
205    Mutex::Autolock _l(mLock);
206
207    /* TODO: move all this work into an Init() function */
208    char val_str[PROPERTY_VALUE_MAX] = { 0 };
209    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
210        uint32_t int_val;
211        if (1 == sscanf(val_str, "%u", &int_val)) {
212            mStandbyTimeInNsecs = milliseconds(int_val);
213            ALOGI("Using %u mSec as standby time.", int_val);
214        } else {
215            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
216            ALOGI("Using default %u mSec as standby time.",
217                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
218        }
219    }
220
221    mMode = AUDIO_MODE_NORMAL;
222}
223
224AudioFlinger::~AudioFlinger()
225{
226    while (!mRecordThreads.isEmpty()) {
227        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
228        closeInput_nonvirtual(mRecordThreads.keyAt(0));
229    }
230    while (!mPlaybackThreads.isEmpty()) {
231        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
232        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
233    }
234
235    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
236        // no mHardwareLock needed, as there are no other references to this
237        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
238        delete mAudioHwDevs.valueAt(i);
239    }
240
241    // Tell media.log service about any old writers that still need to be unregistered
242    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
243    if (binder != 0) {
244        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
245        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
246            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
247            mUnregisteredWriters.pop();
248            mediaLogService->unregisterWriter(iMemory);
249        }
250    }
251
252}
253
254static const char * const audio_interfaces[] = {
255    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
256    AUDIO_HARDWARE_MODULE_ID_A2DP,
257    AUDIO_HARDWARE_MODULE_ID_USB,
258};
259#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
260
261AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
262        audio_module_handle_t module,
263        audio_devices_t devices)
264{
265    // if module is 0, the request comes from an old policy manager and we should load
266    // well known modules
267    if (module == 0) {
268        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
269        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
270            loadHwModule_l(audio_interfaces[i]);
271        }
272        // then try to find a module supporting the requested device.
273        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
274            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
275            audio_hw_device_t *dev = audioHwDevice->hwDevice();
276            if ((dev->get_supported_devices != NULL) &&
277                    (dev->get_supported_devices(dev) & devices) == devices)
278                return audioHwDevice;
279        }
280    } else {
281        // check a match for the requested module handle
282        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
283        if (audioHwDevice != NULL) {
284            return audioHwDevice;
285        }
286    }
287
288    return NULL;
289}
290
291void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
292{
293    const size_t SIZE = 256;
294    char buffer[SIZE];
295    String8 result;
296
297    result.append("Clients:\n");
298    for (size_t i = 0; i < mClients.size(); ++i) {
299        sp<Client> client = mClients.valueAt(i).promote();
300        if (client != 0) {
301            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
302            result.append(buffer);
303        }
304    }
305
306    result.append("Notification Clients:\n");
307    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
308        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
309        result.append(buffer);
310    }
311
312    result.append("Global session refs:\n");
313    result.append("  session   pid count\n");
314    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
315        AudioSessionRef *r = mAudioSessionRefs[i];
316        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
317        result.append(buffer);
318    }
319    write(fd, result.string(), result.size());
320}
321
322
323void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
324{
325    const size_t SIZE = 256;
326    char buffer[SIZE];
327    String8 result;
328    hardware_call_state hardwareStatus = mHardwareStatus;
329
330    snprintf(buffer, SIZE, "Hardware status: %d\n"
331                           "Standby Time mSec: %u\n",
332                            hardwareStatus,
333                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
334    result.append(buffer);
335    write(fd, result.string(), result.size());
336}
337
338void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
339{
340    const size_t SIZE = 256;
341    char buffer[SIZE];
342    String8 result;
343    snprintf(buffer, SIZE, "Permission Denial: "
344            "can't dump AudioFlinger from pid=%d, uid=%d\n",
345            IPCThreadState::self()->getCallingPid(),
346            IPCThreadState::self()->getCallingUid());
347    result.append(buffer);
348    write(fd, result.string(), result.size());
349}
350
351bool AudioFlinger::dumpTryLock(Mutex& mutex)
352{
353    bool locked = false;
354    for (int i = 0; i < kDumpLockRetries; ++i) {
355        if (mutex.tryLock() == NO_ERROR) {
356            locked = true;
357            break;
358        }
359        usleep(kDumpLockSleepUs);
360    }
361    return locked;
362}
363
364status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
365{
366    if (!dumpAllowed()) {
367        dumpPermissionDenial(fd, args);
368    } else {
369        // get state of hardware lock
370        bool hardwareLocked = dumpTryLock(mHardwareLock);
371        if (!hardwareLocked) {
372            String8 result(kHardwareLockedString);
373            write(fd, result.string(), result.size());
374        } else {
375            mHardwareLock.unlock();
376        }
377
378        bool locked = dumpTryLock(mLock);
379
380        // failed to lock - AudioFlinger is probably deadlocked
381        if (!locked) {
382            String8 result(kDeadlockedString);
383            write(fd, result.string(), result.size());
384        }
385
386        bool clientLocked = dumpTryLock(mClientLock);
387        if (!clientLocked) {
388            String8 result(kClientLockedString);
389            write(fd, result.string(), result.size());
390        }
391        dumpClients(fd, args);
392        if (clientLocked) {
393            mClientLock.unlock();
394        }
395
396        dumpInternals(fd, args);
397
398        // dump playback threads
399        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
400            mPlaybackThreads.valueAt(i)->dump(fd, args);
401        }
402
403        // dump record threads
404        for (size_t i = 0; i < mRecordThreads.size(); i++) {
405            mRecordThreads.valueAt(i)->dump(fd, args);
406        }
407
408        // dump all hardware devs
409        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
410            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
411            dev->dump(dev, fd);
412        }
413
414#ifdef TEE_SINK
415        // dump the serially shared record tee sink
416        if (mRecordTeeSource != 0) {
417            dumpTee(fd, mRecordTeeSource);
418        }
419#endif
420
421        if (locked) {
422            mLock.unlock();
423        }
424
425        // append a copy of media.log here by forwarding fd to it, but don't attempt
426        // to lookup the service if it's not running, as it will block for a second
427        if (mLogMemoryDealer != 0) {
428            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
429            if (binder != 0) {
430                dprintf(fd, "\nmedia.log:\n");
431                Vector<String16> args;
432                binder->dump(fd, args);
433            }
434        }
435    }
436    return NO_ERROR;
437}
438
439sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
440{
441    Mutex::Autolock _cl(mClientLock);
442    // If pid is already in the mClients wp<> map, then use that entry
443    // (for which promote() is always != 0), otherwise create a new entry and Client.
444    sp<Client> client = mClients.valueFor(pid).promote();
445    if (client == 0) {
446        client = new Client(this, pid);
447        mClients.add(pid, client);
448    }
449
450    return client;
451}
452
453sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
454{
455    // If there is no memory allocated for logs, return a dummy writer that does nothing
456    if (mLogMemoryDealer == 0) {
457        return new NBLog::Writer();
458    }
459    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
460    // Similarly if we can't contact the media.log service, also return a dummy writer
461    if (binder == 0) {
462        return new NBLog::Writer();
463    }
464    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
465    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
466    // If allocation fails, consult the vector of previously unregistered writers
467    // and garbage-collect one or more them until an allocation succeeds
468    if (shared == 0) {
469        Mutex::Autolock _l(mUnregisteredWritersLock);
470        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
471            {
472                // Pick the oldest stale writer to garbage-collect
473                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
474                mUnregisteredWriters.removeAt(0);
475                mediaLogService->unregisterWriter(iMemory);
476                // Now the media.log remote reference to IMemory is gone.  When our last local
477                // reference to IMemory also drops to zero at end of this block,
478                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
479            }
480            // Re-attempt the allocation
481            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
482            if (shared != 0) {
483                goto success;
484            }
485        }
486        // Even after garbage-collecting all old writers, there is still not enough memory,
487        // so return a dummy writer
488        return new NBLog::Writer();
489    }
490success:
491    mediaLogService->registerWriter(shared, size, name);
492    return new NBLog::Writer(size, shared);
493}
494
495void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
496{
497    if (writer == 0) {
498        return;
499    }
500    sp<IMemory> iMemory(writer->getIMemory());
501    if (iMemory == 0) {
502        return;
503    }
504    // Rather than removing the writer immediately, append it to a queue of old writers to
505    // be garbage-collected later.  This allows us to continue to view old logs for a while.
506    Mutex::Autolock _l(mUnregisteredWritersLock);
507    mUnregisteredWriters.push(writer);
508}
509
510// IAudioFlinger interface
511
512
513sp<IAudioTrack> AudioFlinger::createTrack(
514        audio_stream_type_t streamType,
515        uint32_t sampleRate,
516        audio_format_t format,
517        audio_channel_mask_t channelMask,
518        size_t *frameCount,
519        IAudioFlinger::track_flags_t *flags,
520        const sp<IMemory>& sharedBuffer,
521        audio_io_handle_t output,
522        pid_t tid,
523        int *sessionId,
524        int clientUid,
525        status_t *status)
526{
527    sp<PlaybackThread::Track> track;
528    sp<TrackHandle> trackHandle;
529    sp<Client> client;
530    status_t lStatus;
531    int lSessionId;
532
533    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
534    // but if someone uses binder directly they could bypass that and cause us to crash
535    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
536        ALOGE("createTrack() invalid stream type %d", streamType);
537        lStatus = BAD_VALUE;
538        goto Exit;
539    }
540
541    // further sample rate checks are performed by createTrack_l() depending on the thread type
542    if (sampleRate == 0) {
543        ALOGE("createTrack() invalid sample rate %u", sampleRate);
544        lStatus = BAD_VALUE;
545        goto Exit;
546    }
547
548    // further channel mask checks are performed by createTrack_l() depending on the thread type
549    if (!audio_is_output_channel(channelMask)) {
550        ALOGE("createTrack() invalid channel mask %#x", channelMask);
551        lStatus = BAD_VALUE;
552        goto Exit;
553    }
554
555    // further format checks are performed by createTrack_l() depending on the thread type
556    if (!audio_is_valid_format(format)) {
557        ALOGE("createTrack() invalid format %#x", format);
558        lStatus = BAD_VALUE;
559        goto Exit;
560    }
561
562    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
563        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
564        lStatus = BAD_VALUE;
565        goto Exit;
566    }
567
568    {
569        Mutex::Autolock _l(mLock);
570        PlaybackThread *thread = checkPlaybackThread_l(output);
571        if (thread == NULL) {
572            ALOGE("no playback thread found for output handle %d", output);
573            lStatus = BAD_VALUE;
574            goto Exit;
575        }
576
577        pid_t pid = IPCThreadState::self()->getCallingPid();
578        client = registerPid(pid);
579
580        PlaybackThread *effectThread = NULL;
581        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
582            lSessionId = *sessionId;
583            // check if an effect chain with the same session ID is present on another
584            // output thread and move it here.
585            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
586                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
587                if (mPlaybackThreads.keyAt(i) != output) {
588                    uint32_t sessions = t->hasAudioSession(lSessionId);
589                    if (sessions & PlaybackThread::EFFECT_SESSION) {
590                        effectThread = t.get();
591                        break;
592                    }
593                }
594            }
595        } else {
596            // if no audio session id is provided, create one here
597            lSessionId = nextUniqueId();
598            if (sessionId != NULL) {
599                *sessionId = lSessionId;
600            }
601        }
602        ALOGV("createTrack() lSessionId: %d", lSessionId);
603
604        track = thread->createTrack_l(client, streamType, sampleRate, format,
605                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
606        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
607        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
608
609        // move effect chain to this output thread if an effect on same session was waiting
610        // for a track to be created
611        if (lStatus == NO_ERROR && effectThread != NULL) {
612            // no risk of deadlock because AudioFlinger::mLock is held
613            Mutex::Autolock _dl(thread->mLock);
614            Mutex::Autolock _sl(effectThread->mLock);
615            moveEffectChain_l(lSessionId, effectThread, thread, true);
616        }
617
618        // Look for sync events awaiting for a session to be used.
619        for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
620            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
621                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
622                    if (lStatus == NO_ERROR) {
623                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
624                    } else {
625                        mPendingSyncEvents[i]->cancel();
626                    }
627                    mPendingSyncEvents.removeAt(i);
628                    i--;
629                }
630            }
631        }
632
633    }
634
635    if (lStatus != NO_ERROR) {
636        // remove local strong reference to Client before deleting the Track so that the
637        // Client destructor is called by the TrackBase destructor with mClientLock held
638        Mutex::Autolock _cl(mClientLock);
639        client.clear();
640        track.clear();
641        goto Exit;
642    }
643
644    // return handle to client
645    trackHandle = new TrackHandle(track);
646
647Exit:
648    *status = lStatus;
649    return trackHandle;
650}
651
652uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
653{
654    Mutex::Autolock _l(mLock);
655    PlaybackThread *thread = checkPlaybackThread_l(output);
656    if (thread == NULL) {
657        ALOGW("sampleRate() unknown thread %d", output);
658        return 0;
659    }
660    return thread->sampleRate();
661}
662
663int AudioFlinger::channelCount(audio_io_handle_t output) const
664{
665    Mutex::Autolock _l(mLock);
666    PlaybackThread *thread = checkPlaybackThread_l(output);
667    if (thread == NULL) {
668        ALOGW("channelCount() unknown thread %d", output);
669        return 0;
670    }
671    return thread->channelCount();
672}
673
674audio_format_t AudioFlinger::format(audio_io_handle_t output) const
675{
676    Mutex::Autolock _l(mLock);
677    PlaybackThread *thread = checkPlaybackThread_l(output);
678    if (thread == NULL) {
679        ALOGW("format() unknown thread %d", output);
680        return AUDIO_FORMAT_INVALID;
681    }
682    return thread->format();
683}
684
685size_t AudioFlinger::frameCount(audio_io_handle_t output) const
686{
687    Mutex::Autolock _l(mLock);
688    PlaybackThread *thread = checkPlaybackThread_l(output);
689    if (thread == NULL) {
690        ALOGW("frameCount() unknown thread %d", output);
691        return 0;
692    }
693    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
694    //       should examine all callers and fix them to handle smaller counts
695    return thread->frameCount();
696}
697
698uint32_t AudioFlinger::latency(audio_io_handle_t output) const
699{
700    Mutex::Autolock _l(mLock);
701    PlaybackThread *thread = checkPlaybackThread_l(output);
702    if (thread == NULL) {
703        ALOGW("latency(): no playback thread found for output handle %d", output);
704        return 0;
705    }
706    return thread->latency();
707}
708
709status_t AudioFlinger::setMasterVolume(float value)
710{
711    status_t ret = initCheck();
712    if (ret != NO_ERROR) {
713        return ret;
714    }
715
716    // check calling permissions
717    if (!settingsAllowed()) {
718        return PERMISSION_DENIED;
719    }
720
721    Mutex::Autolock _l(mLock);
722    mMasterVolume = value;
723
724    // Set master volume in the HALs which support it.
725    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
726        AutoMutex lock(mHardwareLock);
727        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
728
729        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
730        if (dev->canSetMasterVolume()) {
731            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
732        }
733        mHardwareStatus = AUDIO_HW_IDLE;
734    }
735
736    // Now set the master volume in each playback thread.  Playback threads
737    // assigned to HALs which do not have master volume support will apply
738    // master volume during the mix operation.  Threads with HALs which do
739    // support master volume will simply ignore the setting.
740    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
741        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
742
743    return NO_ERROR;
744}
745
746status_t AudioFlinger::setMode(audio_mode_t mode)
747{
748    status_t ret = initCheck();
749    if (ret != NO_ERROR) {
750        return ret;
751    }
752
753    // check calling permissions
754    if (!settingsAllowed()) {
755        return PERMISSION_DENIED;
756    }
757    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
758        ALOGW("Illegal value: setMode(%d)", mode);
759        return BAD_VALUE;
760    }
761
762    { // scope for the lock
763        AutoMutex lock(mHardwareLock);
764        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
765        mHardwareStatus = AUDIO_HW_SET_MODE;
766        ret = dev->set_mode(dev, mode);
767        mHardwareStatus = AUDIO_HW_IDLE;
768    }
769
770    if (NO_ERROR == ret) {
771        Mutex::Autolock _l(mLock);
772        mMode = mode;
773        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
774            mPlaybackThreads.valueAt(i)->setMode(mode);
775    }
776
777    return ret;
778}
779
780status_t AudioFlinger::setMicMute(bool state)
781{
782    status_t ret = initCheck();
783    if (ret != NO_ERROR) {
784        return ret;
785    }
786
787    // check calling permissions
788    if (!settingsAllowed()) {
789        return PERMISSION_DENIED;
790    }
791
792    AutoMutex lock(mHardwareLock);
793    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
794    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
795    ret = dev->set_mic_mute(dev, state);
796    mHardwareStatus = AUDIO_HW_IDLE;
797    return ret;
798}
799
800bool AudioFlinger::getMicMute() const
801{
802    status_t ret = initCheck();
803    if (ret != NO_ERROR) {
804        return false;
805    }
806
807    bool state = AUDIO_MODE_INVALID;
808    AutoMutex lock(mHardwareLock);
809    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
810    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
811    dev->get_mic_mute(dev, &state);
812    mHardwareStatus = AUDIO_HW_IDLE;
813    return state;
814}
815
816status_t AudioFlinger::setMasterMute(bool muted)
817{
818    status_t ret = initCheck();
819    if (ret != NO_ERROR) {
820        return ret;
821    }
822
823    // check calling permissions
824    if (!settingsAllowed()) {
825        return PERMISSION_DENIED;
826    }
827
828    Mutex::Autolock _l(mLock);
829    mMasterMute = muted;
830
831    // Set master mute in the HALs which support it.
832    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
833        AutoMutex lock(mHardwareLock);
834        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
835
836        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
837        if (dev->canSetMasterMute()) {
838            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
839        }
840        mHardwareStatus = AUDIO_HW_IDLE;
841    }
842
843    // Now set the master mute in each playback thread.  Playback threads
844    // assigned to HALs which do not have master mute support will apply master
845    // mute during the mix operation.  Threads with HALs which do support master
846    // mute will simply ignore the setting.
847    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
848        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
849
850    return NO_ERROR;
851}
852
853float AudioFlinger::masterVolume() const
854{
855    Mutex::Autolock _l(mLock);
856    return masterVolume_l();
857}
858
859bool AudioFlinger::masterMute() const
860{
861    Mutex::Autolock _l(mLock);
862    return masterMute_l();
863}
864
865float AudioFlinger::masterVolume_l() const
866{
867    return mMasterVolume;
868}
869
870bool AudioFlinger::masterMute_l() const
871{
872    return mMasterMute;
873}
874
875status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
876        audio_io_handle_t output)
877{
878    // check calling permissions
879    if (!settingsAllowed()) {
880        return PERMISSION_DENIED;
881    }
882
883    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
884        ALOGE("setStreamVolume() invalid stream %d", stream);
885        return BAD_VALUE;
886    }
887
888    AutoMutex lock(mLock);
889    PlaybackThread *thread = NULL;
890    if (output != AUDIO_IO_HANDLE_NONE) {
891        thread = checkPlaybackThread_l(output);
892        if (thread == NULL) {
893            return BAD_VALUE;
894        }
895    }
896
897    mStreamTypes[stream].volume = value;
898
899    if (thread == NULL) {
900        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
901            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
902        }
903    } else {
904        thread->setStreamVolume(stream, value);
905    }
906
907    return NO_ERROR;
908}
909
910status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
911{
912    // check calling permissions
913    if (!settingsAllowed()) {
914        return PERMISSION_DENIED;
915    }
916
917    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
918        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
919        ALOGE("setStreamMute() invalid stream %d", stream);
920        return BAD_VALUE;
921    }
922
923    AutoMutex lock(mLock);
924    mStreamTypes[stream].mute = muted;
925    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
926        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
927
928    return NO_ERROR;
929}
930
931float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
932{
933    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
934        return 0.0f;
935    }
936
937    AutoMutex lock(mLock);
938    float volume;
939    if (output != AUDIO_IO_HANDLE_NONE) {
940        PlaybackThread *thread = checkPlaybackThread_l(output);
941        if (thread == NULL) {
942            return 0.0f;
943        }
944        volume = thread->streamVolume(stream);
945    } else {
946        volume = streamVolume_l(stream);
947    }
948
949    return volume;
950}
951
952bool AudioFlinger::streamMute(audio_stream_type_t stream) const
953{
954    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
955        return true;
956    }
957
958    AutoMutex lock(mLock);
959    return streamMute_l(stream);
960}
961
962status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
963{
964    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
965            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
966
967    // check calling permissions
968    if (!settingsAllowed()) {
969        return PERMISSION_DENIED;
970    }
971
972    // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
973    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
974        Mutex::Autolock _l(mLock);
975        status_t final_result = NO_ERROR;
976        {
977            AutoMutex lock(mHardwareLock);
978            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
979            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
980                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
981                status_t result = dev->set_parameters(dev, keyValuePairs.string());
982                final_result = result ?: final_result;
983            }
984            mHardwareStatus = AUDIO_HW_IDLE;
985        }
986        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
987        AudioParameter param = AudioParameter(keyValuePairs);
988        String8 value;
989        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
990            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
991            if (mBtNrecIsOff != btNrecIsOff) {
992                for (size_t i = 0; i < mRecordThreads.size(); i++) {
993                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
994                    audio_devices_t device = thread->inDevice();
995                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
996                    // collect all of the thread's session IDs
997                    KeyedVector<int, bool> ids = thread->sessionIds();
998                    // suspend effects associated with those session IDs
999                    for (size_t j = 0; j < ids.size(); ++j) {
1000                        int sessionId = ids.keyAt(j);
1001                        thread->setEffectSuspended(FX_IID_AEC,
1002                                                   suspend,
1003                                                   sessionId);
1004                        thread->setEffectSuspended(FX_IID_NS,
1005                                                   suspend,
1006                                                   sessionId);
1007                    }
1008                }
1009                mBtNrecIsOff = btNrecIsOff;
1010            }
1011        }
1012        String8 screenState;
1013        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1014            bool isOff = screenState == "off";
1015            if (isOff != (AudioFlinger::mScreenState & 1)) {
1016                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1017            }
1018        }
1019        return final_result;
1020    }
1021
1022    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1023    // and the thread is exited once the lock is released
1024    sp<ThreadBase> thread;
1025    {
1026        Mutex::Autolock _l(mLock);
1027        thread = checkPlaybackThread_l(ioHandle);
1028        if (thread == 0) {
1029            thread = checkRecordThread_l(ioHandle);
1030        } else if (thread == primaryPlaybackThread_l()) {
1031            // indicate output device change to all input threads for pre processing
1032            AudioParameter param = AudioParameter(keyValuePairs);
1033            int value;
1034            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1035                    (value != 0)) {
1036                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1037                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1038                }
1039            }
1040        }
1041    }
1042    if (thread != 0) {
1043        return thread->setParameters(keyValuePairs);
1044    }
1045    return BAD_VALUE;
1046}
1047
1048String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1049{
1050    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1051            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1052
1053    Mutex::Autolock _l(mLock);
1054
1055    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1056        String8 out_s8;
1057
1058        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1059            char *s;
1060            {
1061            AutoMutex lock(mHardwareLock);
1062            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1063            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1064            s = dev->get_parameters(dev, keys.string());
1065            mHardwareStatus = AUDIO_HW_IDLE;
1066            }
1067            out_s8 += String8(s ? s : "");
1068            free(s);
1069        }
1070        return out_s8;
1071    }
1072
1073    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1074    if (playbackThread != NULL) {
1075        return playbackThread->getParameters(keys);
1076    }
1077    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1078    if (recordThread != NULL) {
1079        return recordThread->getParameters(keys);
1080    }
1081    return String8("");
1082}
1083
1084size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1085        audio_channel_mask_t channelMask) const
1086{
1087    status_t ret = initCheck();
1088    if (ret != NO_ERROR) {
1089        return 0;
1090    }
1091
1092    AutoMutex lock(mHardwareLock);
1093    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1094    struct audio_config config;
1095    memset(&config, 0, sizeof(config));
1096    config.sample_rate = sampleRate;
1097    config.channel_mask = channelMask;
1098    config.format = format;
1099
1100    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1101    size_t size = dev->get_input_buffer_size(dev, &config);
1102    mHardwareStatus = AUDIO_HW_IDLE;
1103    return size;
1104}
1105
1106uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1107{
1108    Mutex::Autolock _l(mLock);
1109
1110    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1111    if (recordThread != NULL) {
1112        return recordThread->getInputFramesLost();
1113    }
1114    return 0;
1115}
1116
1117status_t AudioFlinger::setVoiceVolume(float value)
1118{
1119    status_t ret = initCheck();
1120    if (ret != NO_ERROR) {
1121        return ret;
1122    }
1123
1124    // check calling permissions
1125    if (!settingsAllowed()) {
1126        return PERMISSION_DENIED;
1127    }
1128
1129    AutoMutex lock(mHardwareLock);
1130    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1131    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1132    ret = dev->set_voice_volume(dev, value);
1133    mHardwareStatus = AUDIO_HW_IDLE;
1134
1135    return ret;
1136}
1137
1138status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1139        audio_io_handle_t output) const
1140{
1141    status_t status;
1142
1143    Mutex::Autolock _l(mLock);
1144
1145    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1146    if (playbackThread != NULL) {
1147        return playbackThread->getRenderPosition(halFrames, dspFrames);
1148    }
1149
1150    return BAD_VALUE;
1151}
1152
1153void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1154{
1155    Mutex::Autolock _l(mLock);
1156    bool clientAdded = false;
1157    {
1158        Mutex::Autolock _cl(mClientLock);
1159
1160        pid_t pid = IPCThreadState::self()->getCallingPid();
1161        if (mNotificationClients.indexOfKey(pid) < 0) {
1162            sp<NotificationClient> notificationClient = new NotificationClient(this,
1163                                                                                client,
1164                                                                                pid);
1165            ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1166
1167            mNotificationClients.add(pid, notificationClient);
1168
1169            sp<IBinder> binder = client->asBinder();
1170            binder->linkToDeath(notificationClient);
1171            clientAdded = true;
1172        }
1173    }
1174
1175    // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
1176    // ThreadBase mutex and teh locknig order is ThreadBase::mLock then AudioFlinger::mClientLock.
1177    if (clientAdded) {
1178        // the config change is always sent from playback or record threads to avoid deadlock
1179        // with AudioSystem::gLock
1180        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1181            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1182        }
1183
1184        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1185            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1186        }
1187    }
1188}
1189
1190void AudioFlinger::removeNotificationClient(pid_t pid)
1191{
1192    Mutex::Autolock _l(mLock);
1193    {
1194        Mutex::Autolock _cl(mClientLock);
1195        mNotificationClients.removeItem(pid);
1196    }
1197
1198    ALOGV("%d died, releasing its sessions", pid);
1199    size_t num = mAudioSessionRefs.size();
1200    bool removed = false;
1201    for (size_t i = 0; i< num; ) {
1202        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1203        ALOGV(" pid %d @ %d", ref->mPid, i);
1204        if (ref->mPid == pid) {
1205            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1206            mAudioSessionRefs.removeAt(i);
1207            delete ref;
1208            removed = true;
1209            num--;
1210        } else {
1211            i++;
1212        }
1213    }
1214    if (removed) {
1215        purgeStaleEffects_l();
1216    }
1217}
1218
1219void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2)
1220{
1221    Mutex::Autolock _l(mClientLock);
1222    size_t size = mNotificationClients.size();
1223    for (size_t i = 0; i < size; i++) {
1224        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event,
1225                                                                              ioHandle,
1226                                                                              param2);
1227    }
1228}
1229
1230// removeClient_l() must be called with AudioFlinger::mClientLock held
1231void AudioFlinger::removeClient_l(pid_t pid)
1232{
1233    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1234            IPCThreadState::self()->getCallingPid());
1235    mClients.removeItem(pid);
1236}
1237
1238// getEffectThread_l() must be called with AudioFlinger::mLock held
1239sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1240{
1241    sp<PlaybackThread> thread;
1242
1243    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1244        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1245            ALOG_ASSERT(thread == 0);
1246            thread = mPlaybackThreads.valueAt(i);
1247        }
1248    }
1249
1250    return thread;
1251}
1252
1253
1254
1255// ----------------------------------------------------------------------------
1256
1257AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1258    :   RefBase(),
1259        mAudioFlinger(audioFlinger),
1260        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1261        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1262        mPid(pid),
1263        mTimedTrackCount(0)
1264{
1265    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1266}
1267
1268// Client destructor must be called with AudioFlinger::mClientLock held
1269AudioFlinger::Client::~Client()
1270{
1271    mAudioFlinger->removeClient_l(mPid);
1272}
1273
1274sp<MemoryDealer> AudioFlinger::Client::heap() const
1275{
1276    return mMemoryDealer;
1277}
1278
1279// Reserve one of the limited slots for a timed audio track associated
1280// with this client
1281bool AudioFlinger::Client::reserveTimedTrack()
1282{
1283    const int kMaxTimedTracksPerClient = 4;
1284
1285    Mutex::Autolock _l(mTimedTrackLock);
1286
1287    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1288        ALOGW("can not create timed track - pid %d has exceeded the limit",
1289             mPid);
1290        return false;
1291    }
1292
1293    mTimedTrackCount++;
1294    return true;
1295}
1296
1297// Release a slot for a timed audio track
1298void AudioFlinger::Client::releaseTimedTrack()
1299{
1300    Mutex::Autolock _l(mTimedTrackLock);
1301    mTimedTrackCount--;
1302}
1303
1304// ----------------------------------------------------------------------------
1305
1306AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1307                                                     const sp<IAudioFlingerClient>& client,
1308                                                     pid_t pid)
1309    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1310{
1311}
1312
1313AudioFlinger::NotificationClient::~NotificationClient()
1314{
1315}
1316
1317void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1318{
1319    sp<NotificationClient> keep(this);
1320    mAudioFlinger->removeNotificationClient(mPid);
1321}
1322
1323
1324// ----------------------------------------------------------------------------
1325
1326static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1327    return audio_is_remote_submix_device(inDevice);
1328}
1329
1330sp<IAudioRecord> AudioFlinger::openRecord(
1331        audio_io_handle_t input,
1332        uint32_t sampleRate,
1333        audio_format_t format,
1334        audio_channel_mask_t channelMask,
1335        size_t *frameCount,
1336        IAudioFlinger::track_flags_t *flags,
1337        pid_t tid,
1338        int *sessionId,
1339        sp<IMemory>& cblk,
1340        sp<IMemory>& buffers,
1341        status_t *status)
1342{
1343    sp<RecordThread::RecordTrack> recordTrack;
1344    sp<RecordHandle> recordHandle;
1345    sp<Client> client;
1346    status_t lStatus;
1347    int lSessionId;
1348
1349    cblk.clear();
1350    buffers.clear();
1351
1352    // check calling permissions
1353    if (!recordingAllowed()) {
1354        ALOGE("openRecord() permission denied: recording not allowed");
1355        lStatus = PERMISSION_DENIED;
1356        goto Exit;
1357    }
1358
1359    // further sample rate checks are performed by createRecordTrack_l()
1360    if (sampleRate == 0) {
1361        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1362        lStatus = BAD_VALUE;
1363        goto Exit;
1364    }
1365
1366    // we don't yet support anything other than 16-bit PCM
1367    if (!(audio_is_valid_format(format) &&
1368            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
1369        ALOGE("openRecord() invalid format %#x", format);
1370        lStatus = BAD_VALUE;
1371        goto Exit;
1372    }
1373
1374    // further channel mask checks are performed by createRecordTrack_l()
1375    if (!audio_is_input_channel(channelMask)) {
1376        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1377        lStatus = BAD_VALUE;
1378        goto Exit;
1379    }
1380
1381    {
1382        Mutex::Autolock _l(mLock);
1383        RecordThread *thread = checkRecordThread_l(input);
1384        if (thread == NULL) {
1385            ALOGE("openRecord() checkRecordThread_l failed");
1386            lStatus = BAD_VALUE;
1387            goto Exit;
1388        }
1389
1390        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1391                && !captureAudioOutputAllowed()) {
1392            ALOGE("openRecord() permission denied: capture not allowed");
1393            lStatus = PERMISSION_DENIED;
1394            goto Exit;
1395        }
1396
1397        pid_t pid = IPCThreadState::self()->getCallingPid();
1398        client = registerPid(pid);
1399
1400        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1401            lSessionId = *sessionId;
1402        } else {
1403            // if no audio session id is provided, create one here
1404            lSessionId = nextUniqueId();
1405            if (sessionId != NULL) {
1406                *sessionId = lSessionId;
1407            }
1408        }
1409        ALOGV("openRecord() lSessionId: %d", lSessionId);
1410
1411        // TODO: the uid should be passed in as a parameter to openRecord
1412        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1413                                                  frameCount, lSessionId,
1414                                                  IPCThreadState::self()->getCallingUid(),
1415                                                  flags, tid, &lStatus);
1416        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1417    }
1418
1419    if (lStatus != NO_ERROR) {
1420        // remove local strong reference to Client before deleting the RecordTrack so that the
1421        // Client destructor is called by the TrackBase destructor with mClientLock held
1422        Mutex::Autolock _cl(mClientLock);
1423        client.clear();
1424        recordTrack.clear();
1425        goto Exit;
1426    }
1427
1428    cblk = recordTrack->getCblk();
1429    buffers = recordTrack->getBuffers();
1430
1431    // return handle to client
1432    recordHandle = new RecordHandle(recordTrack);
1433
1434Exit:
1435    *status = lStatus;
1436    return recordHandle;
1437}
1438
1439
1440
1441// ----------------------------------------------------------------------------
1442
1443audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1444{
1445    if (!settingsAllowed()) {
1446        return 0;
1447    }
1448    Mutex::Autolock _l(mLock);
1449    return loadHwModule_l(name);
1450}
1451
1452// loadHwModule_l() must be called with AudioFlinger::mLock held
1453audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1454{
1455    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1456        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1457            ALOGW("loadHwModule() module %s already loaded", name);
1458            return mAudioHwDevs.keyAt(i);
1459        }
1460    }
1461
1462    audio_hw_device_t *dev;
1463
1464    int rc = load_audio_interface(name, &dev);
1465    if (rc) {
1466        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1467        return 0;
1468    }
1469
1470    mHardwareStatus = AUDIO_HW_INIT;
1471    rc = dev->init_check(dev);
1472    mHardwareStatus = AUDIO_HW_IDLE;
1473    if (rc) {
1474        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1475        return 0;
1476    }
1477
1478    // Check and cache this HAL's level of support for master mute and master
1479    // volume.  If this is the first HAL opened, and it supports the get
1480    // methods, use the initial values provided by the HAL as the current
1481    // master mute and volume settings.
1482
1483    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1484    {  // scope for auto-lock pattern
1485        AutoMutex lock(mHardwareLock);
1486
1487        if (0 == mAudioHwDevs.size()) {
1488            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1489            if (NULL != dev->get_master_volume) {
1490                float mv;
1491                if (OK == dev->get_master_volume(dev, &mv)) {
1492                    mMasterVolume = mv;
1493                }
1494            }
1495
1496            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1497            if (NULL != dev->get_master_mute) {
1498                bool mm;
1499                if (OK == dev->get_master_mute(dev, &mm)) {
1500                    mMasterMute = mm;
1501                }
1502            }
1503        }
1504
1505        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1506        if ((NULL != dev->set_master_volume) &&
1507            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1508            flags = static_cast<AudioHwDevice::Flags>(flags |
1509                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1510        }
1511
1512        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1513        if ((NULL != dev->set_master_mute) &&
1514            (OK == dev->set_master_mute(dev, mMasterMute))) {
1515            flags = static_cast<AudioHwDevice::Flags>(flags |
1516                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1517        }
1518
1519        mHardwareStatus = AUDIO_HW_IDLE;
1520    }
1521
1522    audio_module_handle_t handle = nextUniqueId();
1523    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1524
1525    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1526          name, dev->common.module->name, dev->common.module->id, handle);
1527
1528    return handle;
1529
1530}
1531
1532// ----------------------------------------------------------------------------
1533
1534uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1535{
1536    Mutex::Autolock _l(mLock);
1537    PlaybackThread *thread = primaryPlaybackThread_l();
1538    return thread != NULL ? thread->sampleRate() : 0;
1539}
1540
1541size_t AudioFlinger::getPrimaryOutputFrameCount()
1542{
1543    Mutex::Autolock _l(mLock);
1544    PlaybackThread *thread = primaryPlaybackThread_l();
1545    return thread != NULL ? thread->frameCountHAL() : 0;
1546}
1547
1548// ----------------------------------------------------------------------------
1549
1550status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1551{
1552    uid_t uid = IPCThreadState::self()->getCallingUid();
1553    if (uid != AID_SYSTEM) {
1554        return PERMISSION_DENIED;
1555    }
1556    Mutex::Autolock _l(mLock);
1557    if (mIsDeviceTypeKnown) {
1558        return INVALID_OPERATION;
1559    }
1560    mIsLowRamDevice = isLowRamDevice;
1561    mIsDeviceTypeKnown = true;
1562    return NO_ERROR;
1563}
1564
1565// ----------------------------------------------------------------------------
1566
1567audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1568                                           audio_devices_t *pDevices,
1569                                           uint32_t *pSamplingRate,
1570                                           audio_format_t *pFormat,
1571                                           audio_channel_mask_t *pChannelMask,
1572                                           uint32_t *pLatencyMs,
1573                                           audio_output_flags_t flags,
1574                                           const audio_offload_info_t *offloadInfo)
1575{
1576    struct audio_config config;
1577    memset(&config, 0, sizeof(config));
1578    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1579    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1580    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1581    if (offloadInfo != NULL) {
1582        config.offload_info = *offloadInfo;
1583    }
1584
1585    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1586              module,
1587              (pDevices != NULL) ? *pDevices : 0,
1588              config.sample_rate,
1589              config.format,
1590              config.channel_mask,
1591              flags);
1592    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1593          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1594
1595    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1596        return AUDIO_IO_HANDLE_NONE;
1597    }
1598
1599    Mutex::Autolock _l(mLock);
1600
1601    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1602    if (outHwDev == NULL) {
1603        return AUDIO_IO_HANDLE_NONE;
1604    }
1605
1606    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1607    audio_io_handle_t id = nextUniqueId();
1608
1609    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1610
1611    audio_stream_out_t *outStream = NULL;
1612    status_t status = hwDevHal->open_output_stream(hwDevHal,
1613                                          id,
1614                                          *pDevices,
1615                                          (audio_output_flags_t)flags,
1616                                          &config,
1617                                          &outStream);
1618
1619    mHardwareStatus = AUDIO_HW_IDLE;
1620    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1621            "Channels %x, status %d",
1622            outStream,
1623            config.sample_rate,
1624            config.format,
1625            config.channel_mask,
1626            status);
1627
1628    if (status == NO_ERROR && outStream != NULL) {
1629        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1630
1631        PlaybackThread *thread;
1632        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1633            thread = new OffloadThread(this, output, id, *pDevices);
1634            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1635        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1636            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1637            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1638            thread = new DirectOutputThread(this, output, id, *pDevices);
1639            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1640        } else {
1641            thread = new MixerThread(this, output, id, *pDevices);
1642            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1643        }
1644        mPlaybackThreads.add(id, thread);
1645
1646        if (pSamplingRate != NULL) {
1647            *pSamplingRate = config.sample_rate;
1648        }
1649        if (pFormat != NULL) {
1650            *pFormat = config.format;
1651        }
1652        if (pChannelMask != NULL) {
1653            *pChannelMask = config.channel_mask;
1654        }
1655        if (pLatencyMs != NULL) {
1656            *pLatencyMs = thread->latency();
1657        }
1658
1659        // notify client processes of the new output creation
1660        thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED);
1661
1662        // the first primary output opened designates the primary hw device
1663        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1664            ALOGI("Using module %d has the primary audio interface", module);
1665            mPrimaryHardwareDev = outHwDev;
1666
1667            AutoMutex lock(mHardwareLock);
1668            mHardwareStatus = AUDIO_HW_SET_MODE;
1669            hwDevHal->set_mode(hwDevHal, mMode);
1670            mHardwareStatus = AUDIO_HW_IDLE;
1671        }
1672        return id;
1673    }
1674
1675    return AUDIO_IO_HANDLE_NONE;
1676}
1677
1678audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1679        audio_io_handle_t output2)
1680{
1681    Mutex::Autolock _l(mLock);
1682    MixerThread *thread1 = checkMixerThread_l(output1);
1683    MixerThread *thread2 = checkMixerThread_l(output2);
1684
1685    if (thread1 == NULL || thread2 == NULL) {
1686        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1687                output2);
1688        return AUDIO_IO_HANDLE_NONE;
1689    }
1690
1691    audio_io_handle_t id = nextUniqueId();
1692    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1693    thread->addOutputTrack(thread2);
1694    mPlaybackThreads.add(id, thread);
1695    // notify client processes of the new output creation
1696    thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED);
1697    return id;
1698}
1699
1700status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1701{
1702    return closeOutput_nonvirtual(output);
1703}
1704
1705status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1706{
1707    // keep strong reference on the playback thread so that
1708    // it is not destroyed while exit() is executed
1709    sp<PlaybackThread> thread;
1710    {
1711        Mutex::Autolock _l(mLock);
1712        thread = checkPlaybackThread_l(output);
1713        if (thread == NULL) {
1714            return BAD_VALUE;
1715        }
1716
1717        ALOGV("closeOutput() %d", output);
1718
1719        if (thread->type() == ThreadBase::MIXER) {
1720            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1721                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1722                    DuplicatingThread *dupThread =
1723                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1724                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1725
1726                }
1727            }
1728        }
1729
1730
1731        mPlaybackThreads.removeItem(output);
1732        // save all effects to the default thread
1733        if (mPlaybackThreads.size()) {
1734            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1735            if (dstThread != NULL) {
1736                // audioflinger lock is held here so the acquisition order of thread locks does not
1737                // matter
1738                Mutex::Autolock _dl(dstThread->mLock);
1739                Mutex::Autolock _sl(thread->mLock);
1740                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1741                for (size_t i = 0; i < effectChains.size(); i ++) {
1742                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1743                }
1744            }
1745        }
1746        audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL);
1747    }
1748    thread->exit();
1749    // The thread entity (active unit of execution) is no longer running here,
1750    // but the ThreadBase container still exists.
1751
1752    if (thread->type() != ThreadBase::DUPLICATING) {
1753        AudioStreamOut *out = thread->clearOutput();
1754        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1755        // from now on thread->mOutput is NULL
1756        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1757        delete out;
1758    }
1759    return NO_ERROR;
1760}
1761
1762status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1763{
1764    Mutex::Autolock _l(mLock);
1765    PlaybackThread *thread = checkPlaybackThread_l(output);
1766
1767    if (thread == NULL) {
1768        return BAD_VALUE;
1769    }
1770
1771    ALOGV("suspendOutput() %d", output);
1772    thread->suspend();
1773
1774    return NO_ERROR;
1775}
1776
1777status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1778{
1779    Mutex::Autolock _l(mLock);
1780    PlaybackThread *thread = checkPlaybackThread_l(output);
1781
1782    if (thread == NULL) {
1783        return BAD_VALUE;
1784    }
1785
1786    ALOGV("restoreOutput() %d", output);
1787
1788    thread->restore();
1789
1790    return NO_ERROR;
1791}
1792
1793audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1794                                          audio_devices_t *pDevices,
1795                                          uint32_t *pSamplingRate,
1796                                          audio_format_t *pFormat,
1797                                          audio_channel_mask_t *pChannelMask)
1798{
1799    struct audio_config config;
1800    memset(&config, 0, sizeof(config));
1801    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1802    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1803    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1804
1805    uint32_t reqSamplingRate = config.sample_rate;
1806    audio_format_t reqFormat = config.format;
1807    audio_channel_mask_t reqChannelMask = config.channel_mask;
1808
1809    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1810        return 0;
1811    }
1812
1813    Mutex::Autolock _l(mLock);
1814
1815    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1816    if (inHwDev == NULL) {
1817        return 0;
1818    }
1819
1820    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1821    audio_io_handle_t id = nextUniqueId();
1822
1823    audio_stream_in_t *inStream = NULL;
1824    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1825                                        &inStream);
1826    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
1827            "status %d",
1828            inStream,
1829            config.sample_rate,
1830            config.format,
1831            config.channel_mask,
1832            status);
1833
1834    // If the input could not be opened with the requested parameters and we can handle the
1835    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1836    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1837    if (status == BAD_VALUE &&
1838        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1839        (config.sample_rate <= 2 * reqSamplingRate) &&
1840        (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) &&
1841        (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) {
1842        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1843        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1844        inStream = NULL;
1845        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1846        // FIXME log this new status; HAL should not propose any further changes
1847    }
1848
1849    if (status == NO_ERROR && inStream != NULL) {
1850
1851#ifdef TEE_SINK
1852        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1853        // or (re-)create if current Pipe is idle and does not match the new format
1854        sp<NBAIO_Sink> teeSink;
1855        enum {
1856            TEE_SINK_NO,    // don't copy input
1857            TEE_SINK_NEW,   // copy input using a new pipe
1858            TEE_SINK_OLD,   // copy input using an existing pipe
1859        } kind;
1860        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1861                audio_channel_count_from_in_mask(
1862                        inStream->common.get_channels(&inStream->common)));
1863        if (!mTeeSinkInputEnabled) {
1864            kind = TEE_SINK_NO;
1865        } else if (!Format_isValid(format)) {
1866            kind = TEE_SINK_NO;
1867        } else if (mRecordTeeSink == 0) {
1868            kind = TEE_SINK_NEW;
1869        } else if (mRecordTeeSink->getStrongCount() != 1) {
1870            kind = TEE_SINK_NO;
1871        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
1872            kind = TEE_SINK_OLD;
1873        } else {
1874            kind = TEE_SINK_NEW;
1875        }
1876        switch (kind) {
1877        case TEE_SINK_NEW: {
1878            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1879            size_t numCounterOffers = 0;
1880            const NBAIO_Format offers[1] = {format};
1881            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1882            ALOG_ASSERT(index == 0);
1883            PipeReader *pipeReader = new PipeReader(*pipe);
1884            numCounterOffers = 0;
1885            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1886            ALOG_ASSERT(index == 0);
1887            mRecordTeeSink = pipe;
1888            mRecordTeeSource = pipeReader;
1889            teeSink = pipe;
1890            }
1891            break;
1892        case TEE_SINK_OLD:
1893            teeSink = mRecordTeeSink;
1894            break;
1895        case TEE_SINK_NO:
1896        default:
1897            break;
1898        }
1899#endif
1900
1901        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1902
1903        // Start record thread
1904        // RecordThread requires both input and output device indication to forward to audio
1905        // pre processing modules
1906        RecordThread *thread = new RecordThread(this,
1907                                  input,
1908                                  id,
1909                                  primaryOutputDevice_l(),
1910                                  *pDevices
1911#ifdef TEE_SINK
1912                                  , teeSink
1913#endif
1914                                  );
1915        mRecordThreads.add(id, thread);
1916        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1917        if (pSamplingRate != NULL) {
1918            *pSamplingRate = reqSamplingRate;
1919        }
1920        if (pFormat != NULL) {
1921            *pFormat = config.format;
1922        }
1923        if (pChannelMask != NULL) {
1924            *pChannelMask = reqChannelMask;
1925        }
1926
1927        // notify client processes of the new input creation
1928        thread->audioConfigChanged(AudioSystem::INPUT_OPENED);
1929        return id;
1930    }
1931
1932    return 0;
1933}
1934
1935status_t AudioFlinger::closeInput(audio_io_handle_t input)
1936{
1937    return closeInput_nonvirtual(input);
1938}
1939
1940status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1941{
1942    // keep strong reference on the record thread so that
1943    // it is not destroyed while exit() is executed
1944    sp<RecordThread> thread;
1945    {
1946        Mutex::Autolock _l(mLock);
1947        thread = checkRecordThread_l(input);
1948        if (thread == 0) {
1949            return BAD_VALUE;
1950        }
1951
1952        ALOGV("closeInput() %d", input);
1953        audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL);
1954        mRecordThreads.removeItem(input);
1955    }
1956    thread->exit();
1957    // The thread entity (active unit of execution) is no longer running here,
1958    // but the ThreadBase container still exists.
1959
1960    AudioStreamIn *in = thread->clearInput();
1961    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1962    // from now on thread->mInput is NULL
1963    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1964    delete in;
1965
1966    return NO_ERROR;
1967}
1968
1969status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
1970{
1971    Mutex::Autolock _l(mLock);
1972    ALOGV("invalidateStream() stream %d", stream);
1973
1974    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1975        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1976        thread->invalidateTracks(stream);
1977    }
1978
1979    return NO_ERROR;
1980}
1981
1982
1983int AudioFlinger::newAudioSessionId()
1984{
1985    return nextUniqueId();
1986}
1987
1988void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
1989{
1990    Mutex::Autolock _l(mLock);
1991    pid_t caller = IPCThreadState::self()->getCallingPid();
1992    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
1993    if (pid != -1 && (caller == getpid_cached)) {
1994        caller = pid;
1995    }
1996
1997    {
1998        Mutex::Autolock _cl(mClientLock);
1999        // Ignore requests received from processes not known as notification client. The request
2000        // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
2001        // called from a different pid leaving a stale session reference.  Also we don't know how
2002        // to clear this reference if the client process dies.
2003        if (mNotificationClients.indexOfKey(caller) < 0) {
2004            ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
2005            return;
2006        }
2007    }
2008
2009    size_t num = mAudioSessionRefs.size();
2010    for (size_t i = 0; i< num; i++) {
2011        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
2012        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2013            ref->mCnt++;
2014            ALOGV(" incremented refcount to %d", ref->mCnt);
2015            return;
2016        }
2017    }
2018    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
2019    ALOGV(" added new entry for %d", audioSession);
2020}
2021
2022void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
2023{
2024    Mutex::Autolock _l(mLock);
2025    pid_t caller = IPCThreadState::self()->getCallingPid();
2026    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
2027    if (pid != -1 && (caller == getpid_cached)) {
2028        caller = pid;
2029    }
2030    size_t num = mAudioSessionRefs.size();
2031    for (size_t i = 0; i< num; i++) {
2032        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2033        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2034            ref->mCnt--;
2035            ALOGV(" decremented refcount to %d", ref->mCnt);
2036            if (ref->mCnt == 0) {
2037                mAudioSessionRefs.removeAt(i);
2038                delete ref;
2039                purgeStaleEffects_l();
2040            }
2041            return;
2042        }
2043    }
2044    // If the caller is mediaserver it is likely that the session being released was acquired
2045    // on behalf of a process not in notification clients and we ignore the warning.
2046    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2047}
2048
2049void AudioFlinger::purgeStaleEffects_l() {
2050
2051    ALOGV("purging stale effects");
2052
2053    Vector< sp<EffectChain> > chains;
2054
2055    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2056        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2057        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2058            sp<EffectChain> ec = t->mEffectChains[j];
2059            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2060                chains.push(ec);
2061            }
2062        }
2063    }
2064    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2065        sp<RecordThread> t = mRecordThreads.valueAt(i);
2066        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2067            sp<EffectChain> ec = t->mEffectChains[j];
2068            chains.push(ec);
2069        }
2070    }
2071
2072    for (size_t i = 0; i < chains.size(); i++) {
2073        sp<EffectChain> ec = chains[i];
2074        int sessionid = ec->sessionId();
2075        sp<ThreadBase> t = ec->mThread.promote();
2076        if (t == 0) {
2077            continue;
2078        }
2079        size_t numsessionrefs = mAudioSessionRefs.size();
2080        bool found = false;
2081        for (size_t k = 0; k < numsessionrefs; k++) {
2082            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2083            if (ref->mSessionid == sessionid) {
2084                ALOGV(" session %d still exists for %d with %d refs",
2085                    sessionid, ref->mPid, ref->mCnt);
2086                found = true;
2087                break;
2088            }
2089        }
2090        if (!found) {
2091            Mutex::Autolock _l(t->mLock);
2092            // remove all effects from the chain
2093            while (ec->mEffects.size()) {
2094                sp<EffectModule> effect = ec->mEffects[0];
2095                effect->unPin();
2096                t->removeEffect_l(effect);
2097                if (effect->purgeHandles()) {
2098                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2099                }
2100                AudioSystem::unregisterEffect(effect->id());
2101            }
2102        }
2103    }
2104    return;
2105}
2106
2107// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2108AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2109{
2110    return mPlaybackThreads.valueFor(output).get();
2111}
2112
2113// checkMixerThread_l() must be called with AudioFlinger::mLock held
2114AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2115{
2116    PlaybackThread *thread = checkPlaybackThread_l(output);
2117    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2118}
2119
2120// checkRecordThread_l() must be called with AudioFlinger::mLock held
2121AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2122{
2123    return mRecordThreads.valueFor(input).get();
2124}
2125
2126uint32_t AudioFlinger::nextUniqueId()
2127{
2128    return (uint32_t) android_atomic_inc(&mNextUniqueId);
2129}
2130
2131AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2132{
2133    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2134        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2135        AudioStreamOut *output = thread->getOutput();
2136        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2137            return thread;
2138        }
2139    }
2140    return NULL;
2141}
2142
2143audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2144{
2145    PlaybackThread *thread = primaryPlaybackThread_l();
2146
2147    if (thread == NULL) {
2148        return 0;
2149    }
2150
2151    return thread->outDevice();
2152}
2153
2154sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2155                                    int triggerSession,
2156                                    int listenerSession,
2157                                    sync_event_callback_t callBack,
2158                                    wp<RefBase> cookie)
2159{
2160    Mutex::Autolock _l(mLock);
2161
2162    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2163    status_t playStatus = NAME_NOT_FOUND;
2164    status_t recStatus = NAME_NOT_FOUND;
2165    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2166        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2167        if (playStatus == NO_ERROR) {
2168            return event;
2169        }
2170    }
2171    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2172        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2173        if (recStatus == NO_ERROR) {
2174            return event;
2175        }
2176    }
2177    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2178        mPendingSyncEvents.add(event);
2179    } else {
2180        ALOGV("createSyncEvent() invalid event %d", event->type());
2181        event.clear();
2182    }
2183    return event;
2184}
2185
2186// ----------------------------------------------------------------------------
2187//  Effect management
2188// ----------------------------------------------------------------------------
2189
2190
2191status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2192{
2193    Mutex::Autolock _l(mLock);
2194    return EffectQueryNumberEffects(numEffects);
2195}
2196
2197status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2198{
2199    Mutex::Autolock _l(mLock);
2200    return EffectQueryEffect(index, descriptor);
2201}
2202
2203status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2204        effect_descriptor_t *descriptor) const
2205{
2206    Mutex::Autolock _l(mLock);
2207    return EffectGetDescriptor(pUuid, descriptor);
2208}
2209
2210
2211sp<IEffect> AudioFlinger::createEffect(
2212        effect_descriptor_t *pDesc,
2213        const sp<IEffectClient>& effectClient,
2214        int32_t priority,
2215        audio_io_handle_t io,
2216        int sessionId,
2217        status_t *status,
2218        int *id,
2219        int *enabled)
2220{
2221    status_t lStatus = NO_ERROR;
2222    sp<EffectHandle> handle;
2223    effect_descriptor_t desc;
2224
2225    pid_t pid = IPCThreadState::self()->getCallingPid();
2226    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2227            pid, effectClient.get(), priority, sessionId, io);
2228
2229    if (pDesc == NULL) {
2230        lStatus = BAD_VALUE;
2231        goto Exit;
2232    }
2233
2234    // check audio settings permission for global effects
2235    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2236        lStatus = PERMISSION_DENIED;
2237        goto Exit;
2238    }
2239
2240    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2241    // that can only be created by audio policy manager (running in same process)
2242    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2243        lStatus = PERMISSION_DENIED;
2244        goto Exit;
2245    }
2246
2247    {
2248        if (!EffectIsNullUuid(&pDesc->uuid)) {
2249            // if uuid is specified, request effect descriptor
2250            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2251            if (lStatus < 0) {
2252                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2253                goto Exit;
2254            }
2255        } else {
2256            // if uuid is not specified, look for an available implementation
2257            // of the required type in effect factory
2258            if (EffectIsNullUuid(&pDesc->type)) {
2259                ALOGW("createEffect() no effect type");
2260                lStatus = BAD_VALUE;
2261                goto Exit;
2262            }
2263            uint32_t numEffects = 0;
2264            effect_descriptor_t d;
2265            d.flags = 0; // prevent compiler warning
2266            bool found = false;
2267
2268            lStatus = EffectQueryNumberEffects(&numEffects);
2269            if (lStatus < 0) {
2270                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2271                goto Exit;
2272            }
2273            for (uint32_t i = 0; i < numEffects; i++) {
2274                lStatus = EffectQueryEffect(i, &desc);
2275                if (lStatus < 0) {
2276                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2277                    continue;
2278                }
2279                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2280                    // If matching type found save effect descriptor. If the session is
2281                    // 0 and the effect is not auxiliary, continue enumeration in case
2282                    // an auxiliary version of this effect type is available
2283                    found = true;
2284                    d = desc;
2285                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2286                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2287                        break;
2288                    }
2289                }
2290            }
2291            if (!found) {
2292                lStatus = BAD_VALUE;
2293                ALOGW("createEffect() effect not found");
2294                goto Exit;
2295            }
2296            // For same effect type, chose auxiliary version over insert version if
2297            // connect to output mix (Compliance to OpenSL ES)
2298            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2299                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2300                desc = d;
2301            }
2302        }
2303
2304        // Do not allow auxiliary effects on a session different from 0 (output mix)
2305        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2306             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2307            lStatus = INVALID_OPERATION;
2308            goto Exit;
2309        }
2310
2311        // check recording permission for visualizer
2312        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2313            !recordingAllowed()) {
2314            lStatus = PERMISSION_DENIED;
2315            goto Exit;
2316        }
2317
2318        // return effect descriptor
2319        *pDesc = desc;
2320        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2321            // if the output returned by getOutputForEffect() is removed before we lock the
2322            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2323            // and we will exit safely
2324            io = AudioSystem::getOutputForEffect(&desc);
2325            ALOGV("createEffect got output %d", io);
2326        }
2327
2328        Mutex::Autolock _l(mLock);
2329
2330        // If output is not specified try to find a matching audio session ID in one of the
2331        // output threads.
2332        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2333        // because of code checking output when entering the function.
2334        // Note: io is never 0 when creating an effect on an input
2335        if (io == AUDIO_IO_HANDLE_NONE) {
2336            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2337                // output must be specified by AudioPolicyManager when using session
2338                // AUDIO_SESSION_OUTPUT_STAGE
2339                lStatus = BAD_VALUE;
2340                goto Exit;
2341            }
2342            // look for the thread where the specified audio session is present
2343            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2344                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2345                    io = mPlaybackThreads.keyAt(i);
2346                    break;
2347                }
2348            }
2349            if (io == 0) {
2350                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2351                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2352                        io = mRecordThreads.keyAt(i);
2353                        break;
2354                    }
2355                }
2356            }
2357            // If no output thread contains the requested session ID, default to
2358            // first output. The effect chain will be moved to the correct output
2359            // thread when a track with the same session ID is created
2360            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2361                io = mPlaybackThreads.keyAt(0);
2362            }
2363            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2364        }
2365        ThreadBase *thread = checkRecordThread_l(io);
2366        if (thread == NULL) {
2367            thread = checkPlaybackThread_l(io);
2368            if (thread == NULL) {
2369                ALOGE("createEffect() unknown output thread");
2370                lStatus = BAD_VALUE;
2371                goto Exit;
2372            }
2373        }
2374
2375        sp<Client> client = registerPid(pid);
2376
2377        // create effect on selected output thread
2378        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2379                &desc, enabled, &lStatus);
2380        if (handle != 0 && id != NULL) {
2381            *id = handle->id();
2382        }
2383    }
2384
2385Exit:
2386    *status = lStatus;
2387    return handle;
2388}
2389
2390status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2391        audio_io_handle_t dstOutput)
2392{
2393    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2394            sessionId, srcOutput, dstOutput);
2395    Mutex::Autolock _l(mLock);
2396    if (srcOutput == dstOutput) {
2397        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2398        return NO_ERROR;
2399    }
2400    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2401    if (srcThread == NULL) {
2402        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2403        return BAD_VALUE;
2404    }
2405    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2406    if (dstThread == NULL) {
2407        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2408        return BAD_VALUE;
2409    }
2410
2411    Mutex::Autolock _dl(dstThread->mLock);
2412    Mutex::Autolock _sl(srcThread->mLock);
2413    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2414}
2415
2416// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2417status_t AudioFlinger::moveEffectChain_l(int sessionId,
2418                                   AudioFlinger::PlaybackThread *srcThread,
2419                                   AudioFlinger::PlaybackThread *dstThread,
2420                                   bool reRegister)
2421{
2422    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2423            sessionId, srcThread, dstThread);
2424
2425    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2426    if (chain == 0) {
2427        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2428                sessionId, srcThread);
2429        return INVALID_OPERATION;
2430    }
2431
2432    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2433    // so that a new chain is created with correct parameters when first effect is added. This is
2434    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2435    // removed.
2436    srcThread->removeEffectChain_l(chain);
2437
2438    // transfer all effects one by one so that new effect chain is created on new thread with
2439    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2440    sp<EffectChain> dstChain;
2441    uint32_t strategy = 0; // prevent compiler warning
2442    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2443    Vector< sp<EffectModule> > removed;
2444    status_t status = NO_ERROR;
2445    while (effect != 0) {
2446        srcThread->removeEffect_l(effect);
2447        removed.add(effect);
2448        status = dstThread->addEffect_l(effect);
2449        if (status != NO_ERROR) {
2450            break;
2451        }
2452        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2453        if (effect->state() == EffectModule::ACTIVE ||
2454                effect->state() == EffectModule::STOPPING) {
2455            effect->start();
2456        }
2457        // if the move request is not received from audio policy manager, the effect must be
2458        // re-registered with the new strategy and output
2459        if (dstChain == 0) {
2460            dstChain = effect->chain().promote();
2461            if (dstChain == 0) {
2462                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2463                status = NO_INIT;
2464                break;
2465            }
2466            strategy = dstChain->strategy();
2467        }
2468        if (reRegister) {
2469            AudioSystem::unregisterEffect(effect->id());
2470            AudioSystem::registerEffect(&effect->desc(),
2471                                        dstThread->id(),
2472                                        strategy,
2473                                        sessionId,
2474                                        effect->id());
2475            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2476        }
2477        effect = chain->getEffectFromId_l(0);
2478    }
2479
2480    if (status != NO_ERROR) {
2481        for (size_t i = 0; i < removed.size(); i++) {
2482            srcThread->addEffect_l(removed[i]);
2483            if (dstChain != 0 && reRegister) {
2484                AudioSystem::unregisterEffect(removed[i]->id());
2485                AudioSystem::registerEffect(&removed[i]->desc(),
2486                                            srcThread->id(),
2487                                            strategy,
2488                                            sessionId,
2489                                            removed[i]->id());
2490                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2491            }
2492        }
2493    }
2494
2495    return status;
2496}
2497
2498bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2499{
2500    if (mGlobalEffectEnableTime != 0 &&
2501            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2502        return true;
2503    }
2504
2505    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2506        sp<EffectChain> ec =
2507                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2508        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2509            return true;
2510        }
2511    }
2512    return false;
2513}
2514
2515void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2516{
2517    Mutex::Autolock _l(mLock);
2518
2519    mGlobalEffectEnableTime = systemTime();
2520
2521    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2522        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2523        if (t->mType == ThreadBase::OFFLOAD) {
2524            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2525        }
2526    }
2527
2528}
2529
2530struct Entry {
2531#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2532    char mName[MAX_NAME];
2533};
2534
2535int comparEntry(const void *p1, const void *p2)
2536{
2537    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2538}
2539
2540#ifdef TEE_SINK
2541void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2542{
2543    NBAIO_Source *teeSource = source.get();
2544    if (teeSource != NULL) {
2545        // .wav rotation
2546        // There is a benign race condition if 2 threads call this simultaneously.
2547        // They would both traverse the directory, but the result would simply be
2548        // failures at unlink() which are ignored.  It's also unlikely since
2549        // normally dumpsys is only done by bugreport or from the command line.
2550        char teePath[32+256];
2551        strcpy(teePath, "/data/misc/media");
2552        size_t teePathLen = strlen(teePath);
2553        DIR *dir = opendir(teePath);
2554        teePath[teePathLen++] = '/';
2555        if (dir != NULL) {
2556#define MAX_SORT 20 // number of entries to sort
2557#define MAX_KEEP 10 // number of entries to keep
2558            struct Entry entries[MAX_SORT];
2559            size_t entryCount = 0;
2560            while (entryCount < MAX_SORT) {
2561                struct dirent de;
2562                struct dirent *result = NULL;
2563                int rc = readdir_r(dir, &de, &result);
2564                if (rc != 0) {
2565                    ALOGW("readdir_r failed %d", rc);
2566                    break;
2567                }
2568                if (result == NULL) {
2569                    break;
2570                }
2571                if (result != &de) {
2572                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2573                    break;
2574                }
2575                // ignore non .wav file entries
2576                size_t nameLen = strlen(de.d_name);
2577                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2578                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2579                    continue;
2580                }
2581                strcpy(entries[entryCount++].mName, de.d_name);
2582            }
2583            (void) closedir(dir);
2584            if (entryCount > MAX_KEEP) {
2585                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2586                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2587                    strcpy(&teePath[teePathLen], entries[i].mName);
2588                    (void) unlink(teePath);
2589                }
2590            }
2591        } else {
2592            if (fd >= 0) {
2593                dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2594            }
2595        }
2596        char teeTime[16];
2597        struct timeval tv;
2598        gettimeofday(&tv, NULL);
2599        struct tm tm;
2600        localtime_r(&tv.tv_sec, &tm);
2601        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2602        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2603        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2604        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2605        if (teeFd >= 0) {
2606            char wavHeader[44];
2607            memcpy(wavHeader,
2608                "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",
2609                sizeof(wavHeader));
2610            NBAIO_Format format = teeSource->format();
2611            unsigned channelCount = Format_channelCount(format);
2612            ALOG_ASSERT(channelCount <= FCC_2);
2613            uint32_t sampleRate = Format_sampleRate(format);
2614            wavHeader[22] = channelCount;       // number of channels
2615            wavHeader[24] = sampleRate;         // sample rate
2616            wavHeader[25] = sampleRate >> 8;
2617            wavHeader[32] = channelCount * 2;   // block alignment
2618            write(teeFd, wavHeader, sizeof(wavHeader));
2619            size_t total = 0;
2620            bool firstRead = true;
2621            for (;;) {
2622#define TEE_SINK_READ 1024
2623                short buffer[TEE_SINK_READ * FCC_2];
2624                size_t count = TEE_SINK_READ;
2625                ssize_t actual = teeSource->read(buffer, count,
2626                        AudioBufferProvider::kInvalidPTS);
2627                bool wasFirstRead = firstRead;
2628                firstRead = false;
2629                if (actual <= 0) {
2630                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2631                        continue;
2632                    }
2633                    break;
2634                }
2635                ALOG_ASSERT(actual <= (ssize_t)count);
2636                write(teeFd, buffer, actual * channelCount * sizeof(short));
2637                total += actual;
2638            }
2639            lseek(teeFd, (off_t) 4, SEEK_SET);
2640            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2641            write(teeFd, &temp, sizeof(temp));
2642            lseek(teeFd, (off_t) 40, SEEK_SET);
2643            temp =  total * channelCount * sizeof(short);
2644            write(teeFd, &temp, sizeof(temp));
2645            close(teeFd);
2646            if (fd >= 0) {
2647                dprintf(fd, "tee copied to %s\n", teePath);
2648            }
2649        } else {
2650            if (fd >= 0) {
2651                dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2652            }
2653        }
2654    }
2655}
2656#endif
2657
2658// ----------------------------------------------------------------------------
2659
2660status_t AudioFlinger::onTransact(
2661        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2662{
2663    return BnAudioFlinger::onTransact(code, data, reply, flags);
2664}
2665
2666}; // namespace android
2667