AudioFlinger.cpp revision 45faf7e02791993a487d6e038d16ff46395f1975 
1/*
2**
3** Copyright 2007, The Android Open Source Project
4**
5** Licensed under the Apache License, Version 2.0 (the "License");
6** you may not use this file except in compliance with the License.
7** You may obtain a copy of the License at
8**
9**     http://www.apache.org/licenses/LICENSE-2.0
10**
11** Unless required by applicable law or agreed to in writing, software
12** distributed under the License is distributed on an "AS IS" BASIS,
13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
14** See the License for the specific language governing permissions and
15** limitations under the License.
16*/
17
18
19#define LOG_TAG "AudioFlinger"
20//#define LOG_NDEBUG 0
21
22#include "Configuration.h"
23#include <dirent.h>
24#include <math.h>
25#include <signal.h>
26#include <sys/time.h>
27#include <sys/resource.h>
28
29#include <binder/IPCThreadState.h>
30#include <binder/IServiceManager.h>
31#include <utils/Log.h>
32#include <utils/Trace.h>
33#include <binder/Parcel.h>
34#include <utils/String16.h>
35#include <utils/threads.h>
36#include <utils/Atomic.h>
37
38#include <cutils/bitops.h>
39#include <cutils/properties.h>
40
41#include <system/audio.h>
42#include <hardware/audio.h>
43
44#include "AudioMixer.h"
45#include "AudioFlinger.h"
46#include "ServiceUtilities.h"
47
48#include <media/EffectsFactoryApi.h>
49#include <audio_effects/effect_visualizer.h>
50#include <audio_effects/effect_ns.h>
51#include <audio_effects/effect_aec.h>
52
53#include <audio_utils/primitives.h>
54
55#include <powermanager/PowerManager.h>
56
57#include <common_time/cc_helper.h>
58
59#include <media/IMediaLogService.h>
60
61#include <media/nbaio/Pipe.h>
62#include <media/nbaio/PipeReader.h>
63#include <media/AudioParameter.h>
64#include <private/android_filesystem_config.h>
65
66// ----------------------------------------------------------------------------
67
68// Note: the following macro is used for extremely verbose logging message.  In
69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
70// 0; but one side effect of this is to turn all LOGV's as well.  Some messages
71// are so verbose that we want to suppress them even when we have ALOG_ASSERT
72// turned on.  Do not uncomment the #def below unless you really know what you
73// are doing and want to see all of the extremely verbose messages.
74//#define VERY_VERY_VERBOSE_LOGGING
75#ifdef VERY_VERY_VERBOSE_LOGGING
76#define ALOGVV ALOGV
77#else
78#define ALOGVV(a...) do { } while(0)
79#endif
80
81namespace android {
82
83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
84static const char kHardwareLockedString[] = "Hardware lock is taken\n";
85
86
87nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
88
89uint32_t AudioFlinger::mScreenState;
90
91#ifdef TEE_SINK
92bool AudioFlinger::mTeeSinkInputEnabled = false;
93bool AudioFlinger::mTeeSinkOutputEnabled = false;
94bool AudioFlinger::mTeeSinkTrackEnabled = false;
95
96size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
97size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
98size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
99#endif
100
101// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
102// we define a minimum time during which a global effect is considered enabled.
103static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
104
105// ----------------------------------------------------------------------------
106
107const char *formatToString(audio_format_t format) {
108    switch(format) {
109    case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8";
110    case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16";
111    case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32";
112    case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24";
113    case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24";
114    case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat";
115    case AUDIO_FORMAT_MP3: return "mp3";
116    case AUDIO_FORMAT_AMR_NB: return "amr-nb";
117    case AUDIO_FORMAT_AMR_WB: return "amr-wb";
118    case AUDIO_FORMAT_AAC: return "aac";
119    case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
120    case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
121    case AUDIO_FORMAT_VORBIS: return "vorbis";
122    default:
123        break;
124    }
125    return "unknown";
126}
127
128static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
129{
130    const hw_module_t *mod;
131    int rc;
132
133    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
134    ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
135                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
136    if (rc) {
137        goto out;
138    }
139    rc = audio_hw_device_open(mod, dev);
140    ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
141                 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
142    if (rc) {
143        goto out;
144    }
145    if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
146        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
147        rc = BAD_VALUE;
148        goto out;
149    }
150    return 0;
151
152out:
153    *dev = NULL;
154    return rc;
155}
156
157// ----------------------------------------------------------------------------
158
159AudioFlinger::AudioFlinger()
160    : BnAudioFlinger(),
161      mPrimaryHardwareDev(NULL),
162      mHardwareStatus(AUDIO_HW_IDLE),
163      mMasterVolume(1.0f),
164      mMasterMute(false),
165      mNextUniqueId(1),
166      mMode(AUDIO_MODE_INVALID),
167      mBtNrecIsOff(false),
168      mIsLowRamDevice(true),
169      mIsDeviceTypeKnown(false),
170      mGlobalEffectEnableTime(0)
171{
172    getpid_cached = getpid();
173    char value[PROPERTY_VALUE_MAX];
174    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
175    if (doLog) {
176        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
177    }
178#ifdef TEE_SINK
179    (void) property_get("ro.debuggable", value, "0");
180    int debuggable = atoi(value);
181    int teeEnabled = 0;
182    if (debuggable) {
183        (void) property_get("af.tee", value, "0");
184        teeEnabled = atoi(value);
185    }
186    // FIXME symbolic constants here
187    if (teeEnabled & 1) {
188        mTeeSinkInputEnabled = true;
189    }
190    if (teeEnabled & 2) {
191        mTeeSinkOutputEnabled = true;
192    }
193    if (teeEnabled & 4) {
194        mTeeSinkTrackEnabled = true;
195    }
196#endif
197}
198
199void AudioFlinger::onFirstRef()
200{
201    int rc = 0;
202
203    Mutex::Autolock _l(mLock);
204
205    /* TODO: move all this work into an Init() function */
206    char val_str[PROPERTY_VALUE_MAX] = { 0 };
207    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
208        uint32_t int_val;
209        if (1 == sscanf(val_str, "%u", &int_val)) {
210            mStandbyTimeInNsecs = milliseconds(int_val);
211            ALOGI("Using %u mSec as standby time.", int_val);
212        } else {
213            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
214            ALOGI("Using default %u mSec as standby time.",
215                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
216        }
217    }
218
219    mMode = AUDIO_MODE_NORMAL;
220}
221
222AudioFlinger::~AudioFlinger()
223{
224    while (!mRecordThreads.isEmpty()) {
225        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
226        closeInput_nonvirtual(mRecordThreads.keyAt(0));
227    }
228    while (!mPlaybackThreads.isEmpty()) {
229        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
230        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
231    }
232
233    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
234        // no mHardwareLock needed, as there are no other references to this
235        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
236        delete mAudioHwDevs.valueAt(i);
237    }
238
239    // Tell media.log service about any old writers that still need to be unregistered
240    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
241    if (binder != 0) {
242        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
243        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
244            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
245            mUnregisteredWriters.pop();
246            mediaLogService->unregisterWriter(iMemory);
247        }
248    }
249
250}
251
252static const char * const audio_interfaces[] = {
253    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
254    AUDIO_HARDWARE_MODULE_ID_A2DP,
255    AUDIO_HARDWARE_MODULE_ID_USB,
256};
257#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
258
259AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
260        audio_module_handle_t module,
261        audio_devices_t devices)
262{
263    // if module is 0, the request comes from an old policy manager and we should load
264    // well known modules
265    if (module == 0) {
266        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
267        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
268            loadHwModule_l(audio_interfaces[i]);
269        }
270        // then try to find a module supporting the requested device.
271        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
272            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
273            audio_hw_device_t *dev = audioHwDevice->hwDevice();
274            if ((dev->get_supported_devices != NULL) &&
275                    (dev->get_supported_devices(dev) & devices) == devices)
276                return audioHwDevice;
277        }
278    } else {
279        // check a match for the requested module handle
280        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
281        if (audioHwDevice != NULL) {
282            return audioHwDevice;
283        }
284    }
285
286    return NULL;
287}
288
289void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
290{
291    const size_t SIZE = 256;
292    char buffer[SIZE];
293    String8 result;
294
295    result.append("Clients:\n");
296    for (size_t i = 0; i < mClients.size(); ++i) {
297        sp<Client> client = mClients.valueAt(i).promote();
298        if (client != 0) {
299            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
300            result.append(buffer);
301        }
302    }
303
304    result.append("Notification Clients:\n");
305    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
306        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
307        result.append(buffer);
308    }
309
310    result.append("Global session refs:\n");
311    result.append("  session   pid count\n");
312    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
313        AudioSessionRef *r = mAudioSessionRefs[i];
314        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
315        result.append(buffer);
316    }
317    write(fd, result.string(), result.size());
318}
319
320
321void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
322{
323    const size_t SIZE = 256;
324    char buffer[SIZE];
325    String8 result;
326    hardware_call_state hardwareStatus = mHardwareStatus;
327
328    snprintf(buffer, SIZE, "Hardware status: %d\n"
329                           "Standby Time mSec: %u\n",
330                            hardwareStatus,
331                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
332    result.append(buffer);
333    write(fd, result.string(), result.size());
334}
335
336void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
337{
338    const size_t SIZE = 256;
339    char buffer[SIZE];
340    String8 result;
341    snprintf(buffer, SIZE, "Permission Denial: "
342            "can't dump AudioFlinger from pid=%d, uid=%d\n",
343            IPCThreadState::self()->getCallingPid(),
344            IPCThreadState::self()->getCallingUid());
345    result.append(buffer);
346    write(fd, result.string(), result.size());
347}
348
349bool AudioFlinger::dumpTryLock(Mutex& mutex)
350{
351    bool locked = false;
352    for (int i = 0; i < kDumpLockRetries; ++i) {
353        if (mutex.tryLock() == NO_ERROR) {
354            locked = true;
355            break;
356        }
357        usleep(kDumpLockSleepUs);
358    }
359    return locked;
360}
361
362status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
363{
364    if (!dumpAllowed()) {
365        dumpPermissionDenial(fd, args);
366    } else {
367        // get state of hardware lock
368        bool hardwareLocked = dumpTryLock(mHardwareLock);
369        if (!hardwareLocked) {
370            String8 result(kHardwareLockedString);
371            write(fd, result.string(), result.size());
372        } else {
373            mHardwareLock.unlock();
374        }
375
376        bool locked = dumpTryLock(mLock);
377
378        // failed to lock - AudioFlinger is probably deadlocked
379        if (!locked) {
380            String8 result(kDeadlockedString);
381            write(fd, result.string(), result.size());
382        }
383
384        dumpClients(fd, args);
385        dumpInternals(fd, args);
386
387        // dump playback threads
388        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
389            mPlaybackThreads.valueAt(i)->dump(fd, args);
390        }
391
392        // dump record threads
393        for (size_t i = 0; i < mRecordThreads.size(); i++) {
394            mRecordThreads.valueAt(i)->dump(fd, args);
395        }
396
397        // dump all hardware devs
398        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
399            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
400            dev->dump(dev, fd);
401        }
402
403#ifdef TEE_SINK
404        // dump the serially shared record tee sink
405        if (mRecordTeeSource != 0) {
406            dumpTee(fd, mRecordTeeSource);
407        }
408#endif
409
410        if (locked) {
411            mLock.unlock();
412        }
413
414        // append a copy of media.log here by forwarding fd to it, but don't attempt
415        // to lookup the service if it's not running, as it will block for a second
416        if (mLogMemoryDealer != 0) {
417            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
418            if (binder != 0) {
419                fdprintf(fd, "\nmedia.log:\n");
420                Vector<String16> args;
421                binder->dump(fd, args);
422            }
423        }
424    }
425    return NO_ERROR;
426}
427
428sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
429{
430    // If pid is already in the mClients wp<> map, then use that entry
431    // (for which promote() is always != 0), otherwise create a new entry and Client.
432    sp<Client> client = mClients.valueFor(pid).promote();
433    if (client == 0) {
434        client = new Client(this, pid);
435        mClients.add(pid, client);
436    }
437
438    return client;
439}
440
441sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
442{
443    // If there is no memory allocated for logs, return a dummy writer that does nothing
444    if (mLogMemoryDealer == 0) {
445        return new NBLog::Writer();
446    }
447    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
448    // Similarly if we can't contact the media.log service, also return a dummy writer
449    if (binder == 0) {
450        return new NBLog::Writer();
451    }
452    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
453    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
454    // If allocation fails, consult the vector of previously unregistered writers
455    // and garbage-collect one or more them until an allocation succeeds
456    if (shared == 0) {
457        Mutex::Autolock _l(mUnregisteredWritersLock);
458        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
459            {
460                // Pick the oldest stale writer to garbage-collect
461                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
462                mUnregisteredWriters.removeAt(0);
463                mediaLogService->unregisterWriter(iMemory);
464                // Now the media.log remote reference to IMemory is gone.  When our last local
465                // reference to IMemory also drops to zero at end of this block,
466                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
467            }
468            // Re-attempt the allocation
469            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
470            if (shared != 0) {
471                goto success;
472            }
473        }
474        // Even after garbage-collecting all old writers, there is still not enough memory,
475        // so return a dummy writer
476        return new NBLog::Writer();
477    }
478success:
479    mediaLogService->registerWriter(shared, size, name);
480    return new NBLog::Writer(size, shared);
481}
482
483void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
484{
485    if (writer == 0) {
486        return;
487    }
488    sp<IMemory> iMemory(writer->getIMemory());
489    if (iMemory == 0) {
490        return;
491    }
492    // Rather than removing the writer immediately, append it to a queue of old writers to
493    // be garbage-collected later.  This allows us to continue to view old logs for a while.
494    Mutex::Autolock _l(mUnregisteredWritersLock);
495    mUnregisteredWriters.push(writer);
496}
497
498// IAudioFlinger interface
499
500
501sp<IAudioTrack> AudioFlinger::createTrack(
502        audio_stream_type_t streamType,
503        uint32_t sampleRate,
504        audio_format_t format,
505        audio_channel_mask_t channelMask,
506        size_t *frameCount,
507        IAudioFlinger::track_flags_t *flags,
508        const sp<IMemory>& sharedBuffer,
509        audio_io_handle_t output,
510        pid_t tid,
511        int *sessionId,
512        int clientUid,
513        status_t *status)
514{
515    sp<PlaybackThread::Track> track;
516    sp<TrackHandle> trackHandle;
517    sp<Client> client;
518    status_t lStatus;
519    int lSessionId;
520
521    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
522    // but if someone uses binder directly they could bypass that and cause us to crash
523    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
524        ALOGE("createTrack() invalid stream type %d", streamType);
525        lStatus = BAD_VALUE;
526        goto Exit;
527    }
528
529    // further sample rate checks are performed by createTrack_l() depending on the thread type
530    if (sampleRate == 0) {
531        ALOGE("createTrack() invalid sample rate %u", sampleRate);
532        lStatus = BAD_VALUE;
533        goto Exit;
534    }
535
536    // further channel mask checks are performed by createTrack_l() depending on the thread type
537    if (!audio_is_output_channel(channelMask)) {
538        ALOGE("createTrack() invalid channel mask %#x", channelMask);
539        lStatus = BAD_VALUE;
540        goto Exit;
541    }
542
543    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
544    // and we don't yet support 8.24 or 32-bit PCM
545    if (!audio_is_valid_format(format) ||
546            (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT)) {
547        ALOGE("createTrack() invalid format %#x", format);
548        lStatus = BAD_VALUE;
549        goto Exit;
550    }
551
552    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
553        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
554        lStatus = BAD_VALUE;
555        goto Exit;
556    }
557
558    {
559        Mutex::Autolock _l(mLock);
560        PlaybackThread *thread = checkPlaybackThread_l(output);
561        if (thread == NULL) {
562            ALOGE("no playback thread found for output handle %d", output);
563            lStatus = BAD_VALUE;
564            goto Exit;
565        }
566
567        pid_t pid = IPCThreadState::self()->getCallingPid();
568        client = registerPid_l(pid);
569
570        PlaybackThread *effectThread = NULL;
571        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
572            lSessionId = *sessionId;
573            // check if an effect chain with the same session ID is present on another
574            // output thread and move it here.
575            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
576                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
577                if (mPlaybackThreads.keyAt(i) != output) {
578                    uint32_t sessions = t->hasAudioSession(lSessionId);
579                    if (sessions & PlaybackThread::EFFECT_SESSION) {
580                        effectThread = t.get();
581                        break;
582                    }
583                }
584            }
585        } else {
586            // if no audio session id is provided, create one here
587            lSessionId = nextUniqueId();
588            if (sessionId != NULL) {
589                *sessionId = lSessionId;
590            }
591        }
592        ALOGV("createTrack() lSessionId: %d", lSessionId);
593
594        track = thread->createTrack_l(client, streamType, sampleRate, format,
595                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
596        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
597        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
598
599        // move effect chain to this output thread if an effect on same session was waiting
600        // for a track to be created
601        if (lStatus == NO_ERROR && effectThread != NULL) {
602            // no risk of deadlock because AudioFlinger::mLock is held
603            Mutex::Autolock _dl(thread->mLock);
604            Mutex::Autolock _sl(effectThread->mLock);
605            moveEffectChain_l(lSessionId, effectThread, thread, true);
606        }
607
608        // Look for sync events awaiting for a session to be used.
609        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
610            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
611                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
612                    if (lStatus == NO_ERROR) {
613                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
614                    } else {
615                        mPendingSyncEvents[i]->cancel();
616                    }
617                    mPendingSyncEvents.removeAt(i);
618                    i--;
619                }
620            }
621        }
622
623    }
624
625    if (lStatus != NO_ERROR) {
626        // remove local strong reference to Client before deleting the Track so that the
627        // Client destructor is called by the TrackBase destructor with mLock held
628        client.clear();
629        track.clear();
630        goto Exit;
631    }
632
633    // return handle to client
634    trackHandle = new TrackHandle(track);
635
636Exit:
637    *status = lStatus;
638    return trackHandle;
639}
640
641uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
642{
643    Mutex::Autolock _l(mLock);
644    PlaybackThread *thread = checkPlaybackThread_l(output);
645    if (thread == NULL) {
646        ALOGW("sampleRate() unknown thread %d", output);
647        return 0;
648    }
649    return thread->sampleRate();
650}
651
652int AudioFlinger::channelCount(audio_io_handle_t output) const
653{
654    Mutex::Autolock _l(mLock);
655    PlaybackThread *thread = checkPlaybackThread_l(output);
656    if (thread == NULL) {
657        ALOGW("channelCount() unknown thread %d", output);
658        return 0;
659    }
660    return thread->channelCount();
661}
662
663audio_format_t AudioFlinger::format(audio_io_handle_t output) const
664{
665    Mutex::Autolock _l(mLock);
666    PlaybackThread *thread = checkPlaybackThread_l(output);
667    if (thread == NULL) {
668        ALOGW("format() unknown thread %d", output);
669        return AUDIO_FORMAT_INVALID;
670    }
671    return thread->format();
672}
673
674size_t AudioFlinger::frameCount(audio_io_handle_t output) const
675{
676    Mutex::Autolock _l(mLock);
677    PlaybackThread *thread = checkPlaybackThread_l(output);
678    if (thread == NULL) {
679        ALOGW("frameCount() unknown thread %d", output);
680        return 0;
681    }
682    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
683    //       should examine all callers and fix them to handle smaller counts
684    return thread->frameCount();
685}
686
687uint32_t AudioFlinger::latency(audio_io_handle_t output) const
688{
689    Mutex::Autolock _l(mLock);
690    PlaybackThread *thread = checkPlaybackThread_l(output);
691    if (thread == NULL) {
692        ALOGW("latency(): no playback thread found for output handle %d", output);
693        return 0;
694    }
695    return thread->latency();
696}
697
698status_t AudioFlinger::setMasterVolume(float value)
699{
700    status_t ret = initCheck();
701    if (ret != NO_ERROR) {
702        return ret;
703    }
704
705    // check calling permissions
706    if (!settingsAllowed()) {
707        return PERMISSION_DENIED;
708    }
709
710    Mutex::Autolock _l(mLock);
711    mMasterVolume = value;
712
713    // Set master volume in the HALs which support it.
714    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
715        AutoMutex lock(mHardwareLock);
716        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
717
718        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
719        if (dev->canSetMasterVolume()) {
720            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
721        }
722        mHardwareStatus = AUDIO_HW_IDLE;
723    }
724
725    // Now set the master volume in each playback thread.  Playback threads
726    // assigned to HALs which do not have master volume support will apply
727    // master volume during the mix operation.  Threads with HALs which do
728    // support master volume will simply ignore the setting.
729    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
730        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
731
732    return NO_ERROR;
733}
734
735status_t AudioFlinger::setMode(audio_mode_t mode)
736{
737    status_t ret = initCheck();
738    if (ret != NO_ERROR) {
739        return ret;
740    }
741
742    // check calling permissions
743    if (!settingsAllowed()) {
744        return PERMISSION_DENIED;
745    }
746    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
747        ALOGW("Illegal value: setMode(%d)", mode);
748        return BAD_VALUE;
749    }
750
751    { // scope for the lock
752        AutoMutex lock(mHardwareLock);
753        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
754        mHardwareStatus = AUDIO_HW_SET_MODE;
755        ret = dev->set_mode(dev, mode);
756        mHardwareStatus = AUDIO_HW_IDLE;
757    }
758
759    if (NO_ERROR == ret) {
760        Mutex::Autolock _l(mLock);
761        mMode = mode;
762        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
763            mPlaybackThreads.valueAt(i)->setMode(mode);
764    }
765
766    return ret;
767}
768
769status_t AudioFlinger::setMicMute(bool state)
770{
771    status_t ret = initCheck();
772    if (ret != NO_ERROR) {
773        return ret;
774    }
775
776    // check calling permissions
777    if (!settingsAllowed()) {
778        return PERMISSION_DENIED;
779    }
780
781    AutoMutex lock(mHardwareLock);
782    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
783    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
784    ret = dev->set_mic_mute(dev, state);
785    mHardwareStatus = AUDIO_HW_IDLE;
786    return ret;
787}
788
789bool AudioFlinger::getMicMute() const
790{
791    status_t ret = initCheck();
792    if (ret != NO_ERROR) {
793        return false;
794    }
795
796    bool state = AUDIO_MODE_INVALID;
797    AutoMutex lock(mHardwareLock);
798    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
799    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
800    dev->get_mic_mute(dev, &state);
801    mHardwareStatus = AUDIO_HW_IDLE;
802    return state;
803}
804
805status_t AudioFlinger::setMasterMute(bool muted)
806{
807    status_t ret = initCheck();
808    if (ret != NO_ERROR) {
809        return ret;
810    }
811
812    // check calling permissions
813    if (!settingsAllowed()) {
814        return PERMISSION_DENIED;
815    }
816
817    Mutex::Autolock _l(mLock);
818    mMasterMute = muted;
819
820    // Set master mute in the HALs which support it.
821    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
822        AutoMutex lock(mHardwareLock);
823        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
824
825        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
826        if (dev->canSetMasterMute()) {
827            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
828        }
829        mHardwareStatus = AUDIO_HW_IDLE;
830    }
831
832    // Now set the master mute in each playback thread.  Playback threads
833    // assigned to HALs which do not have master mute support will apply master
834    // mute during the mix operation.  Threads with HALs which do support master
835    // mute will simply ignore the setting.
836    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
837        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
838
839    return NO_ERROR;
840}
841
842float AudioFlinger::masterVolume() const
843{
844    Mutex::Autolock _l(mLock);
845    return masterVolume_l();
846}
847
848bool AudioFlinger::masterMute() const
849{
850    Mutex::Autolock _l(mLock);
851    return masterMute_l();
852}
853
854float AudioFlinger::masterVolume_l() const
855{
856    return mMasterVolume;
857}
858
859bool AudioFlinger::masterMute_l() const
860{
861    return mMasterMute;
862}
863
864status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
865        audio_io_handle_t output)
866{
867    // check calling permissions
868    if (!settingsAllowed()) {
869        return PERMISSION_DENIED;
870    }
871
872    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
873        ALOGE("setStreamVolume() invalid stream %d", stream);
874        return BAD_VALUE;
875    }
876
877    AutoMutex lock(mLock);
878    PlaybackThread *thread = NULL;
879    if (output) {
880        thread = checkPlaybackThread_l(output);
881        if (thread == NULL) {
882            return BAD_VALUE;
883        }
884    }
885
886    mStreamTypes[stream].volume = value;
887
888    if (thread == NULL) {
889        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
890            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
891        }
892    } else {
893        thread->setStreamVolume(stream, value);
894    }
895
896    return NO_ERROR;
897}
898
899status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
900{
901    // check calling permissions
902    if (!settingsAllowed()) {
903        return PERMISSION_DENIED;
904    }
905
906    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
907        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
908        ALOGE("setStreamMute() invalid stream %d", stream);
909        return BAD_VALUE;
910    }
911
912    AutoMutex lock(mLock);
913    mStreamTypes[stream].mute = muted;
914    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
915        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
916
917    return NO_ERROR;
918}
919
920float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
921{
922    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
923        return 0.0f;
924    }
925
926    AutoMutex lock(mLock);
927    float volume;
928    if (output) {
929        PlaybackThread *thread = checkPlaybackThread_l(output);
930        if (thread == NULL) {
931            return 0.0f;
932        }
933        volume = thread->streamVolume(stream);
934    } else {
935        volume = streamVolume_l(stream);
936    }
937
938    return volume;
939}
940
941bool AudioFlinger::streamMute(audio_stream_type_t stream) const
942{
943    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
944        return true;
945    }
946
947    AutoMutex lock(mLock);
948    return streamMute_l(stream);
949}
950
951status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
952{
953    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
954            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
955
956    // check calling permissions
957    if (!settingsAllowed()) {
958        return PERMISSION_DENIED;
959    }
960
961    // ioHandle == 0 means the parameters are global to the audio hardware interface
962    if (ioHandle == 0) {
963        Mutex::Autolock _l(mLock);
964        status_t final_result = NO_ERROR;
965        {
966            AutoMutex lock(mHardwareLock);
967            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
968            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
969                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
970                status_t result = dev->set_parameters(dev, keyValuePairs.string());
971                final_result = result ?: final_result;
972            }
973            mHardwareStatus = AUDIO_HW_IDLE;
974        }
975        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
976        AudioParameter param = AudioParameter(keyValuePairs);
977        String8 value;
978        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
979            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
980            if (mBtNrecIsOff != btNrecIsOff) {
981                for (size_t i = 0; i < mRecordThreads.size(); i++) {
982                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
983                    audio_devices_t device = thread->inDevice();
984                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
985                    // collect all of the thread's session IDs
986                    KeyedVector<int, bool> ids = thread->sessionIds();
987                    // suspend effects associated with those session IDs
988                    for (size_t j = 0; j < ids.size(); ++j) {
989                        int sessionId = ids.keyAt(j);
990                        thread->setEffectSuspended(FX_IID_AEC,
991                                                   suspend,
992                                                   sessionId);
993                        thread->setEffectSuspended(FX_IID_NS,
994                                                   suspend,
995                                                   sessionId);
996                    }
997                }
998                mBtNrecIsOff = btNrecIsOff;
999            }
1000        }
1001        String8 screenState;
1002        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1003            bool isOff = screenState == "off";
1004            if (isOff != (AudioFlinger::mScreenState & 1)) {
1005                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1006            }
1007        }
1008        return final_result;
1009    }
1010
1011    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1012    // and the thread is exited once the lock is released
1013    sp<ThreadBase> thread;
1014    {
1015        Mutex::Autolock _l(mLock);
1016        thread = checkPlaybackThread_l(ioHandle);
1017        if (thread == 0) {
1018            thread = checkRecordThread_l(ioHandle);
1019        } else if (thread == primaryPlaybackThread_l()) {
1020            // indicate output device change to all input threads for pre processing
1021            AudioParameter param = AudioParameter(keyValuePairs);
1022            int value;
1023            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1024                    (value != 0)) {
1025                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1026                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1027                }
1028            }
1029        }
1030    }
1031    if (thread != 0) {
1032        return thread->setParameters(keyValuePairs);
1033    }
1034    return BAD_VALUE;
1035}
1036
1037String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1038{
1039    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1040            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1041
1042    Mutex::Autolock _l(mLock);
1043
1044    if (ioHandle == 0) {
1045        String8 out_s8;
1046
1047        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1048            char *s;
1049            {
1050            AutoMutex lock(mHardwareLock);
1051            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1052            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1053            s = dev->get_parameters(dev, keys.string());
1054            mHardwareStatus = AUDIO_HW_IDLE;
1055            }
1056            out_s8 += String8(s ? s : "");
1057            free(s);
1058        }
1059        return out_s8;
1060    }
1061
1062    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1063    if (playbackThread != NULL) {
1064        return playbackThread->getParameters(keys);
1065    }
1066    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1067    if (recordThread != NULL) {
1068        return recordThread->getParameters(keys);
1069    }
1070    return String8("");
1071}
1072
1073size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1074        audio_channel_mask_t channelMask) const
1075{
1076    status_t ret = initCheck();
1077    if (ret != NO_ERROR) {
1078        return 0;
1079    }
1080
1081    AutoMutex lock(mHardwareLock);
1082    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1083    struct audio_config config;
1084    memset(&config, 0, sizeof(config));
1085    config.sample_rate = sampleRate;
1086    config.channel_mask = channelMask;
1087    config.format = format;
1088
1089    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1090    size_t size = dev->get_input_buffer_size(dev, &config);
1091    mHardwareStatus = AUDIO_HW_IDLE;
1092    return size;
1093}
1094
1095uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1096{
1097    Mutex::Autolock _l(mLock);
1098
1099    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1100    if (recordThread != NULL) {
1101        return recordThread->getInputFramesLost();
1102    }
1103    return 0;
1104}
1105
1106status_t AudioFlinger::setVoiceVolume(float value)
1107{
1108    status_t ret = initCheck();
1109    if (ret != NO_ERROR) {
1110        return ret;
1111    }
1112
1113    // check calling permissions
1114    if (!settingsAllowed()) {
1115        return PERMISSION_DENIED;
1116    }
1117
1118    AutoMutex lock(mHardwareLock);
1119    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1120    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1121    ret = dev->set_voice_volume(dev, value);
1122    mHardwareStatus = AUDIO_HW_IDLE;
1123
1124    return ret;
1125}
1126
1127status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1128        audio_io_handle_t output) const
1129{
1130    status_t status;
1131
1132    Mutex::Autolock _l(mLock);
1133
1134    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1135    if (playbackThread != NULL) {
1136        return playbackThread->getRenderPosition(halFrames, dspFrames);
1137    }
1138
1139    return BAD_VALUE;
1140}
1141
1142void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1143{
1144
1145    Mutex::Autolock _l(mLock);
1146
1147    pid_t pid = IPCThreadState::self()->getCallingPid();
1148    if (mNotificationClients.indexOfKey(pid) < 0) {
1149        sp<NotificationClient> notificationClient = new NotificationClient(this,
1150                                                                            client,
1151                                                                            pid);
1152        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1153
1154        mNotificationClients.add(pid, notificationClient);
1155
1156        sp<IBinder> binder = client->asBinder();
1157        binder->linkToDeath(notificationClient);
1158
1159        // the config change is always sent from playback or record threads to avoid deadlock
1160        // with AudioSystem::gLock
1161        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1162            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1163        }
1164
1165        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1166            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1167        }
1168    }
1169}
1170
1171void AudioFlinger::removeNotificationClient(pid_t pid)
1172{
1173    Mutex::Autolock _l(mLock);
1174
1175    mNotificationClients.removeItem(pid);
1176
1177    ALOGV("%d died, releasing its sessions", pid);
1178    size_t num = mAudioSessionRefs.size();
1179    bool removed = false;
1180    for (size_t i = 0; i< num; ) {
1181        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1182        ALOGV(" pid %d @ %d", ref->mPid, i);
1183        if (ref->mPid == pid) {
1184            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1185            mAudioSessionRefs.removeAt(i);
1186            delete ref;
1187            removed = true;
1188            num--;
1189        } else {
1190            i++;
1191        }
1192    }
1193    if (removed) {
1194        purgeStaleEffects_l();
1195    }
1196}
1197
1198// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1199void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1200{
1201    size_t size = mNotificationClients.size();
1202    for (size_t i = 0; i < size; i++) {
1203        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1204                                                                               param2);
1205    }
1206}
1207
1208// removeClient_l() must be called with AudioFlinger::mLock held
1209void AudioFlinger::removeClient_l(pid_t pid)
1210{
1211    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1212            IPCThreadState::self()->getCallingPid());
1213    mClients.removeItem(pid);
1214}
1215
1216// getEffectThread_l() must be called with AudioFlinger::mLock held
1217sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1218{
1219    sp<PlaybackThread> thread;
1220
1221    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1222        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1223            ALOG_ASSERT(thread == 0);
1224            thread = mPlaybackThreads.valueAt(i);
1225        }
1226    }
1227
1228    return thread;
1229}
1230
1231
1232
1233// ----------------------------------------------------------------------------
1234
1235AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1236    :   RefBase(),
1237        mAudioFlinger(audioFlinger),
1238        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1239        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1240        mPid(pid),
1241        mTimedTrackCount(0)
1242{
1243    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1244}
1245
1246// Client destructor must be called with AudioFlinger::mLock held
1247AudioFlinger::Client::~Client()
1248{
1249    mAudioFlinger->removeClient_l(mPid);
1250}
1251
1252sp<MemoryDealer> AudioFlinger::Client::heap() const
1253{
1254    return mMemoryDealer;
1255}
1256
1257// Reserve one of the limited slots for a timed audio track associated
1258// with this client
1259bool AudioFlinger::Client::reserveTimedTrack()
1260{
1261    const int kMaxTimedTracksPerClient = 4;
1262
1263    Mutex::Autolock _l(mTimedTrackLock);
1264
1265    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1266        ALOGW("can not create timed track - pid %d has exceeded the limit",
1267             mPid);
1268        return false;
1269    }
1270
1271    mTimedTrackCount++;
1272    return true;
1273}
1274
1275// Release a slot for a timed audio track
1276void AudioFlinger::Client::releaseTimedTrack()
1277{
1278    Mutex::Autolock _l(mTimedTrackLock);
1279    mTimedTrackCount--;
1280}
1281
1282// ----------------------------------------------------------------------------
1283
1284AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1285                                                     const sp<IAudioFlingerClient>& client,
1286                                                     pid_t pid)
1287    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1288{
1289}
1290
1291AudioFlinger::NotificationClient::~NotificationClient()
1292{
1293}
1294
1295void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1296{
1297    sp<NotificationClient> keep(this);
1298    mAudioFlinger->removeNotificationClient(mPid);
1299}
1300
1301
1302// ----------------------------------------------------------------------------
1303
1304static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1305    return audio_is_remote_submix_device(inDevice);
1306}
1307
1308sp<IAudioRecord> AudioFlinger::openRecord(
1309        audio_io_handle_t input,
1310        uint32_t sampleRate,
1311        audio_format_t format,
1312        audio_channel_mask_t channelMask,
1313        size_t *frameCount,
1314        IAudioFlinger::track_flags_t *flags,
1315        pid_t tid,
1316        int *sessionId,
1317        status_t *status)
1318{
1319    sp<RecordThread::RecordTrack> recordTrack;
1320    sp<RecordHandle> recordHandle;
1321    sp<Client> client;
1322    status_t lStatus;
1323    int lSessionId;
1324
1325    // check calling permissions
1326    if (!recordingAllowed()) {
1327        ALOGE("openRecord() permission denied: recording not allowed");
1328        lStatus = PERMISSION_DENIED;
1329        goto Exit;
1330    }
1331
1332    // further sample rate checks are performed by createRecordTrack_l()
1333    if (sampleRate == 0) {
1334        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1335        lStatus = BAD_VALUE;
1336        goto Exit;
1337    }
1338
1339    // we don't yet support anything other than 16-bit PCM
1340    if (!(audio_is_valid_format(format) &&
1341            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
1342        ALOGE("openRecord() invalid format %#x", format);
1343        lStatus = BAD_VALUE;
1344        goto Exit;
1345    }
1346
1347    // further channel mask checks are performed by createRecordTrack_l()
1348    if (!audio_is_input_channel(channelMask)) {
1349        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1350        lStatus = BAD_VALUE;
1351        goto Exit;
1352    }
1353
1354    {
1355        Mutex::Autolock _l(mLock);
1356        RecordThread *thread = checkRecordThread_l(input);
1357        if (thread == NULL) {
1358            ALOGE("openRecord() checkRecordThread_l failed");
1359            lStatus = BAD_VALUE;
1360            goto Exit;
1361        }
1362
1363        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1364                && !captureAudioOutputAllowed()) {
1365            ALOGE("openRecord() permission denied: capture not allowed");
1366            lStatus = PERMISSION_DENIED;
1367            goto Exit;
1368        }
1369
1370        pid_t pid = IPCThreadState::self()->getCallingPid();
1371        client = registerPid_l(pid);
1372
1373        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1374            lSessionId = *sessionId;
1375        } else {
1376            // if no audio session id is provided, create one here
1377            lSessionId = nextUniqueId();
1378            if (sessionId != NULL) {
1379                *sessionId = lSessionId;
1380            }
1381        }
1382        ALOGV("openRecord() lSessionId: %d", lSessionId);
1383
1384        // TODO: the uid should be passed in as a parameter to openRecord
1385        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1386                                                  frameCount, lSessionId,
1387                                                  IPCThreadState::self()->getCallingUid(),
1388                                                  flags, tid, &lStatus);
1389        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1390    }
1391
1392    if (lStatus != NO_ERROR) {
1393        // remove local strong reference to Client before deleting the RecordTrack so that the
1394        // Client destructor is called by the TrackBase destructor with mLock held
1395        client.clear();
1396        recordTrack.clear();
1397        goto Exit;
1398    }
1399
1400    // return handle to client
1401    recordHandle = new RecordHandle(recordTrack);
1402
1403Exit:
1404    *status = lStatus;
1405    return recordHandle;
1406}
1407
1408
1409
1410// ----------------------------------------------------------------------------
1411
1412audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1413{
1414    if (!settingsAllowed()) {
1415        return 0;
1416    }
1417    Mutex::Autolock _l(mLock);
1418    return loadHwModule_l(name);
1419}
1420
1421// loadHwModule_l() must be called with AudioFlinger::mLock held
1422audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1423{
1424    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1425        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1426            ALOGW("loadHwModule() module %s already loaded", name);
1427            return mAudioHwDevs.keyAt(i);
1428        }
1429    }
1430
1431    audio_hw_device_t *dev;
1432
1433    int rc = load_audio_interface(name, &dev);
1434    if (rc) {
1435        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1436        return 0;
1437    }
1438
1439    mHardwareStatus = AUDIO_HW_INIT;
1440    rc = dev->init_check(dev);
1441    mHardwareStatus = AUDIO_HW_IDLE;
1442    if (rc) {
1443        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1444        return 0;
1445    }
1446
1447    // Check and cache this HAL's level of support for master mute and master
1448    // volume.  If this is the first HAL opened, and it supports the get
1449    // methods, use the initial values provided by the HAL as the current
1450    // master mute and volume settings.
1451
1452    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1453    {  // scope for auto-lock pattern
1454        AutoMutex lock(mHardwareLock);
1455
1456        if (0 == mAudioHwDevs.size()) {
1457            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1458            if (NULL != dev->get_master_volume) {
1459                float mv;
1460                if (OK == dev->get_master_volume(dev, &mv)) {
1461                    mMasterVolume = mv;
1462                }
1463            }
1464
1465            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1466            if (NULL != dev->get_master_mute) {
1467                bool mm;
1468                if (OK == dev->get_master_mute(dev, &mm)) {
1469                    mMasterMute = mm;
1470                }
1471            }
1472        }
1473
1474        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1475        if ((NULL != dev->set_master_volume) &&
1476            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1477            flags = static_cast<AudioHwDevice::Flags>(flags |
1478                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1479        }
1480
1481        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1482        if ((NULL != dev->set_master_mute) &&
1483            (OK == dev->set_master_mute(dev, mMasterMute))) {
1484            flags = static_cast<AudioHwDevice::Flags>(flags |
1485                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1486        }
1487
1488        mHardwareStatus = AUDIO_HW_IDLE;
1489    }
1490
1491    audio_module_handle_t handle = nextUniqueId();
1492    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1493
1494    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1495          name, dev->common.module->name, dev->common.module->id, handle);
1496
1497    return handle;
1498
1499}
1500
1501// ----------------------------------------------------------------------------
1502
1503uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1504{
1505    Mutex::Autolock _l(mLock);
1506    PlaybackThread *thread = primaryPlaybackThread_l();
1507    return thread != NULL ? thread->sampleRate() : 0;
1508}
1509
1510size_t AudioFlinger::getPrimaryOutputFrameCount()
1511{
1512    Mutex::Autolock _l(mLock);
1513    PlaybackThread *thread = primaryPlaybackThread_l();
1514    return thread != NULL ? thread->frameCountHAL() : 0;
1515}
1516
1517// ----------------------------------------------------------------------------
1518
1519status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1520{
1521    uid_t uid = IPCThreadState::self()->getCallingUid();
1522    if (uid != AID_SYSTEM) {
1523        return PERMISSION_DENIED;
1524    }
1525    Mutex::Autolock _l(mLock);
1526    if (mIsDeviceTypeKnown) {
1527        return INVALID_OPERATION;
1528    }
1529    mIsLowRamDevice = isLowRamDevice;
1530    mIsDeviceTypeKnown = true;
1531    return NO_ERROR;
1532}
1533
1534// ----------------------------------------------------------------------------
1535
1536audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1537                                           audio_devices_t *pDevices,
1538                                           uint32_t *pSamplingRate,
1539                                           audio_format_t *pFormat,
1540                                           audio_channel_mask_t *pChannelMask,
1541                                           uint32_t *pLatencyMs,
1542                                           audio_output_flags_t flags,
1543                                           const audio_offload_info_t *offloadInfo)
1544{
1545    struct audio_config config;
1546    memset(&config, 0, sizeof(config));
1547    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1548    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1549    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1550    if (offloadInfo != NULL) {
1551        config.offload_info = *offloadInfo;
1552    }
1553
1554    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1555              module,
1556              (pDevices != NULL) ? *pDevices : 0,
1557              config.sample_rate,
1558              config.format,
1559              config.channel_mask,
1560              flags);
1561    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1562          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1563
1564    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1565        return 0;
1566    }
1567
1568    Mutex::Autolock _l(mLock);
1569
1570    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1571    if (outHwDev == NULL) {
1572        return 0;
1573    }
1574
1575    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1576    audio_io_handle_t id = nextUniqueId();
1577
1578    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1579
1580    audio_stream_out_t *outStream = NULL;
1581    status_t status = hwDevHal->open_output_stream(hwDevHal,
1582                                          id,
1583                                          *pDevices,
1584                                          (audio_output_flags_t)flags,
1585                                          &config,
1586                                          &outStream);
1587
1588    mHardwareStatus = AUDIO_HW_IDLE;
1589    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1590            "Channels %x, status %d",
1591            outStream,
1592            config.sample_rate,
1593            config.format,
1594            config.channel_mask,
1595            status);
1596
1597    if (status == NO_ERROR && outStream != NULL) {
1598        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1599
1600        PlaybackThread *thread;
1601        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1602            thread = new OffloadThread(this, output, id, *pDevices);
1603            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1604        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1605            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1606            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1607            thread = new DirectOutputThread(this, output, id, *pDevices);
1608            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1609        } else {
1610            thread = new MixerThread(this, output, id, *pDevices);
1611            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1612        }
1613        mPlaybackThreads.add(id, thread);
1614
1615        if (pSamplingRate != NULL) {
1616            *pSamplingRate = config.sample_rate;
1617        }
1618        if (pFormat != NULL) {
1619            *pFormat = config.format;
1620        }
1621        if (pChannelMask != NULL) {
1622            *pChannelMask = config.channel_mask;
1623        }
1624        if (pLatencyMs != NULL) {
1625            *pLatencyMs = thread->latency();
1626        }
1627
1628        // notify client processes of the new output creation
1629        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1630
1631        // the first primary output opened designates the primary hw device
1632        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1633            ALOGI("Using module %d has the primary audio interface", module);
1634            mPrimaryHardwareDev = outHwDev;
1635
1636            AutoMutex lock(mHardwareLock);
1637            mHardwareStatus = AUDIO_HW_SET_MODE;
1638            hwDevHal->set_mode(hwDevHal, mMode);
1639            mHardwareStatus = AUDIO_HW_IDLE;
1640        }
1641        return id;
1642    }
1643
1644    return 0;
1645}
1646
1647audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1648        audio_io_handle_t output2)
1649{
1650    Mutex::Autolock _l(mLock);
1651    MixerThread *thread1 = checkMixerThread_l(output1);
1652    MixerThread *thread2 = checkMixerThread_l(output2);
1653
1654    if (thread1 == NULL || thread2 == NULL) {
1655        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1656                output2);
1657        return 0;
1658    }
1659
1660    audio_io_handle_t id = nextUniqueId();
1661    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1662    thread->addOutputTrack(thread2);
1663    mPlaybackThreads.add(id, thread);
1664    // notify client processes of the new output creation
1665    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1666    return id;
1667}
1668
1669status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1670{
1671    return closeOutput_nonvirtual(output);
1672}
1673
1674status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1675{
1676    // keep strong reference on the playback thread so that
1677    // it is not destroyed while exit() is executed
1678    sp<PlaybackThread> thread;
1679    {
1680        Mutex::Autolock _l(mLock);
1681        thread = checkPlaybackThread_l(output);
1682        if (thread == NULL) {
1683            return BAD_VALUE;
1684        }
1685
1686        ALOGV("closeOutput() %d", output);
1687
1688        if (thread->type() == ThreadBase::MIXER) {
1689            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1690                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1691                    DuplicatingThread *dupThread =
1692                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1693                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1694
1695                }
1696            }
1697        }
1698
1699
1700        mPlaybackThreads.removeItem(output);
1701        // save all effects to the default thread
1702        if (mPlaybackThreads.size()) {
1703            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1704            if (dstThread != NULL) {
1705                // audioflinger lock is held here so the acquisition order of thread locks does not
1706                // matter
1707                Mutex::Autolock _dl(dstThread->mLock);
1708                Mutex::Autolock _sl(thread->mLock);
1709                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1710                for (size_t i = 0; i < effectChains.size(); i ++) {
1711                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1712                }
1713            }
1714        }
1715        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1716    }
1717    thread->exit();
1718    // The thread entity (active unit of execution) is no longer running here,
1719    // but the ThreadBase container still exists.
1720
1721    if (thread->type() != ThreadBase::DUPLICATING) {
1722        AudioStreamOut *out = thread->clearOutput();
1723        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1724        // from now on thread->mOutput is NULL
1725        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1726        delete out;
1727    }
1728    return NO_ERROR;
1729}
1730
1731status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1732{
1733    Mutex::Autolock _l(mLock);
1734    PlaybackThread *thread = checkPlaybackThread_l(output);
1735
1736    if (thread == NULL) {
1737        return BAD_VALUE;
1738    }
1739
1740    ALOGV("suspendOutput() %d", output);
1741    thread->suspend();
1742
1743    return NO_ERROR;
1744}
1745
1746status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1747{
1748    Mutex::Autolock _l(mLock);
1749    PlaybackThread *thread = checkPlaybackThread_l(output);
1750
1751    if (thread == NULL) {
1752        return BAD_VALUE;
1753    }
1754
1755    ALOGV("restoreOutput() %d", output);
1756
1757    thread->restore();
1758
1759    return NO_ERROR;
1760}
1761
1762audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1763                                          audio_devices_t *pDevices,
1764                                          uint32_t *pSamplingRate,
1765                                          audio_format_t *pFormat,
1766                                          audio_channel_mask_t *pChannelMask)
1767{
1768    struct audio_config config;
1769    memset(&config, 0, sizeof(config));
1770    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1771    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1772    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1773
1774    uint32_t reqSamplingRate = config.sample_rate;
1775    audio_format_t reqFormat = config.format;
1776    audio_channel_mask_t reqChannelMask = config.channel_mask;
1777
1778    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1779        return 0;
1780    }
1781
1782    Mutex::Autolock _l(mLock);
1783
1784    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1785    if (inHwDev == NULL) {
1786        return 0;
1787    }
1788
1789    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1790    audio_io_handle_t id = nextUniqueId();
1791
1792    audio_stream_in_t *inStream = NULL;
1793    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1794                                        &inStream);
1795    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
1796            "status %d",
1797            inStream,
1798            config.sample_rate,
1799            config.format,
1800            config.channel_mask,
1801            status);
1802
1803    // If the input could not be opened with the requested parameters and we can handle the
1804    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1805    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1806    if (status == BAD_VALUE &&
1807        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1808        (config.sample_rate <= 2 * reqSamplingRate) &&
1809        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1810        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1811        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1812        inStream = NULL;
1813        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1814        // FIXME log this new status; HAL should not propose any further changes
1815    }
1816
1817    if (status == NO_ERROR && inStream != NULL) {
1818
1819#ifdef TEE_SINK
1820        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1821        // or (re-)create if current Pipe is idle and does not match the new format
1822        sp<NBAIO_Sink> teeSink;
1823        enum {
1824            TEE_SINK_NO,    // don't copy input
1825            TEE_SINK_NEW,   // copy input using a new pipe
1826            TEE_SINK_OLD,   // copy input using an existing pipe
1827        } kind;
1828        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1829                                        popcount(inStream->common.get_channels(&inStream->common)));
1830        if (!mTeeSinkInputEnabled) {
1831            kind = TEE_SINK_NO;
1832        } else if (!Format_isValid(format)) {
1833            kind = TEE_SINK_NO;
1834        } else if (mRecordTeeSink == 0) {
1835            kind = TEE_SINK_NEW;
1836        } else if (mRecordTeeSink->getStrongCount() != 1) {
1837            kind = TEE_SINK_NO;
1838        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
1839            kind = TEE_SINK_OLD;
1840        } else {
1841            kind = TEE_SINK_NEW;
1842        }
1843        switch (kind) {
1844        case TEE_SINK_NEW: {
1845            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1846            size_t numCounterOffers = 0;
1847            const NBAIO_Format offers[1] = {format};
1848            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1849            ALOG_ASSERT(index == 0);
1850            PipeReader *pipeReader = new PipeReader(*pipe);
1851            numCounterOffers = 0;
1852            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1853            ALOG_ASSERT(index == 0);
1854            mRecordTeeSink = pipe;
1855            mRecordTeeSource = pipeReader;
1856            teeSink = pipe;
1857            }
1858            break;
1859        case TEE_SINK_OLD:
1860            teeSink = mRecordTeeSink;
1861            break;
1862        case TEE_SINK_NO:
1863        default:
1864            break;
1865        }
1866#endif
1867
1868        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1869
1870        // Start record thread
1871        // RecordThread requires both input and output device indication to forward to audio
1872        // pre processing modules
1873        RecordThread *thread = new RecordThread(this,
1874                                  input,
1875                                  id,
1876                                  primaryOutputDevice_l(),
1877                                  *pDevices
1878#ifdef TEE_SINK
1879                                  , teeSink
1880#endif
1881                                  );
1882        mRecordThreads.add(id, thread);
1883        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1884        if (pSamplingRate != NULL) {
1885            *pSamplingRate = reqSamplingRate;
1886        }
1887        if (pFormat != NULL) {
1888            *pFormat = config.format;
1889        }
1890        if (pChannelMask != NULL) {
1891            *pChannelMask = reqChannelMask;
1892        }
1893
1894        // notify client processes of the new input creation
1895        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1896        return id;
1897    }
1898
1899    return 0;
1900}
1901
1902status_t AudioFlinger::closeInput(audio_io_handle_t input)
1903{
1904    return closeInput_nonvirtual(input);
1905}
1906
1907status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1908{
1909    // keep strong reference on the record thread so that
1910    // it is not destroyed while exit() is executed
1911    sp<RecordThread> thread;
1912    {
1913        Mutex::Autolock _l(mLock);
1914        thread = checkRecordThread_l(input);
1915        if (thread == 0) {
1916            return BAD_VALUE;
1917        }
1918
1919        ALOGV("closeInput() %d", input);
1920        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1921        mRecordThreads.removeItem(input);
1922    }
1923    thread->exit();
1924    // The thread entity (active unit of execution) is no longer running here,
1925    // but the ThreadBase container still exists.
1926
1927    AudioStreamIn *in = thread->clearInput();
1928    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1929    // from now on thread->mInput is NULL
1930    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1931    delete in;
1932
1933    return NO_ERROR;
1934}
1935
1936status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
1937{
1938    Mutex::Autolock _l(mLock);
1939    ALOGV("invalidateStream() stream %d", stream);
1940
1941    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1942        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1943        thread->invalidateTracks(stream);
1944    }
1945
1946    return NO_ERROR;
1947}
1948
1949
1950int AudioFlinger::newAudioSessionId()
1951{
1952    return nextUniqueId();
1953}
1954
1955void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
1956{
1957    Mutex::Autolock _l(mLock);
1958    pid_t caller = IPCThreadState::self()->getCallingPid();
1959    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
1960    if (pid != -1 && (caller == getpid_cached)) {
1961        caller = pid;
1962    }
1963
1964    // Ignore requests received from processes not known as notification client. The request
1965    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1966    // called from a different pid leaving a stale session reference.  Also we don't know how
1967    // to clear this reference if the client process dies.
1968    if (mNotificationClients.indexOfKey(caller) < 0) {
1969        ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1970        return;
1971    }
1972
1973    size_t num = mAudioSessionRefs.size();
1974    for (size_t i = 0; i< num; i++) {
1975        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1976        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1977            ref->mCnt++;
1978            ALOGV(" incremented refcount to %d", ref->mCnt);
1979            return;
1980        }
1981    }
1982    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1983    ALOGV(" added new entry for %d", audioSession);
1984}
1985
1986void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
1987{
1988    Mutex::Autolock _l(mLock);
1989    pid_t caller = IPCThreadState::self()->getCallingPid();
1990    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
1991    if (pid != -1 && (caller == getpid_cached)) {
1992        caller = pid;
1993    }
1994    size_t num = mAudioSessionRefs.size();
1995    for (size_t i = 0; i< num; i++) {
1996        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1997        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1998            ref->mCnt--;
1999            ALOGV(" decremented refcount to %d", ref->mCnt);
2000            if (ref->mCnt == 0) {
2001                mAudioSessionRefs.removeAt(i);
2002                delete ref;
2003                purgeStaleEffects_l();
2004            }
2005            return;
2006        }
2007    }
2008    // If the caller is mediaserver it is likely that the session being released was acquired
2009    // on behalf of a process not in notification clients and we ignore the warning.
2010    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2011}
2012
2013void AudioFlinger::purgeStaleEffects_l() {
2014
2015    ALOGV("purging stale effects");
2016
2017    Vector< sp<EffectChain> > chains;
2018
2019    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2020        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2021        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2022            sp<EffectChain> ec = t->mEffectChains[j];
2023            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2024                chains.push(ec);
2025            }
2026        }
2027    }
2028    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2029        sp<RecordThread> t = mRecordThreads.valueAt(i);
2030        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2031            sp<EffectChain> ec = t->mEffectChains[j];
2032            chains.push(ec);
2033        }
2034    }
2035
2036    for (size_t i = 0; i < chains.size(); i++) {
2037        sp<EffectChain> ec = chains[i];
2038        int sessionid = ec->sessionId();
2039        sp<ThreadBase> t = ec->mThread.promote();
2040        if (t == 0) {
2041            continue;
2042        }
2043        size_t numsessionrefs = mAudioSessionRefs.size();
2044        bool found = false;
2045        for (size_t k = 0; k < numsessionrefs; k++) {
2046            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2047            if (ref->mSessionid == sessionid) {
2048                ALOGV(" session %d still exists for %d with %d refs",
2049                    sessionid, ref->mPid, ref->mCnt);
2050                found = true;
2051                break;
2052            }
2053        }
2054        if (!found) {
2055            Mutex::Autolock _l(t->mLock);
2056            // remove all effects from the chain
2057            while (ec->mEffects.size()) {
2058                sp<EffectModule> effect = ec->mEffects[0];
2059                effect->unPin();
2060                t->removeEffect_l(effect);
2061                if (effect->purgeHandles()) {
2062                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2063                }
2064                AudioSystem::unregisterEffect(effect->id());
2065            }
2066        }
2067    }
2068    return;
2069}
2070
2071// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2072AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2073{
2074    return mPlaybackThreads.valueFor(output).get();
2075}
2076
2077// checkMixerThread_l() must be called with AudioFlinger::mLock held
2078AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2079{
2080    PlaybackThread *thread = checkPlaybackThread_l(output);
2081    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2082}
2083
2084// checkRecordThread_l() must be called with AudioFlinger::mLock held
2085AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2086{
2087    return mRecordThreads.valueFor(input).get();
2088}
2089
2090uint32_t AudioFlinger::nextUniqueId()
2091{
2092    return android_atomic_inc(&mNextUniqueId);
2093}
2094
2095AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2096{
2097    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2098        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2099        AudioStreamOut *output = thread->getOutput();
2100        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2101            return thread;
2102        }
2103    }
2104    return NULL;
2105}
2106
2107audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2108{
2109    PlaybackThread *thread = primaryPlaybackThread_l();
2110
2111    if (thread == NULL) {
2112        return 0;
2113    }
2114
2115    return thread->outDevice();
2116}
2117
2118sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2119                                    int triggerSession,
2120                                    int listenerSession,
2121                                    sync_event_callback_t callBack,
2122                                    wp<RefBase> cookie)
2123{
2124    Mutex::Autolock _l(mLock);
2125
2126    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2127    status_t playStatus = NAME_NOT_FOUND;
2128    status_t recStatus = NAME_NOT_FOUND;
2129    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2130        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2131        if (playStatus == NO_ERROR) {
2132            return event;
2133        }
2134    }
2135    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2136        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2137        if (recStatus == NO_ERROR) {
2138            return event;
2139        }
2140    }
2141    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2142        mPendingSyncEvents.add(event);
2143    } else {
2144        ALOGV("createSyncEvent() invalid event %d", event->type());
2145        event.clear();
2146    }
2147    return event;
2148}
2149
2150// ----------------------------------------------------------------------------
2151//  Effect management
2152// ----------------------------------------------------------------------------
2153
2154
2155status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2156{
2157    Mutex::Autolock _l(mLock);
2158    return EffectQueryNumberEffects(numEffects);
2159}
2160
2161status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2162{
2163    Mutex::Autolock _l(mLock);
2164    return EffectQueryEffect(index, descriptor);
2165}
2166
2167status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2168        effect_descriptor_t *descriptor) const
2169{
2170    Mutex::Autolock _l(mLock);
2171    return EffectGetDescriptor(pUuid, descriptor);
2172}
2173
2174
2175sp<IEffect> AudioFlinger::createEffect(
2176        effect_descriptor_t *pDesc,
2177        const sp<IEffectClient>& effectClient,
2178        int32_t priority,
2179        audio_io_handle_t io,
2180        int sessionId,
2181        status_t *status,
2182        int *id,
2183        int *enabled)
2184{
2185    status_t lStatus = NO_ERROR;
2186    sp<EffectHandle> handle;
2187    effect_descriptor_t desc;
2188
2189    pid_t pid = IPCThreadState::self()->getCallingPid();
2190    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2191            pid, effectClient.get(), priority, sessionId, io);
2192
2193    if (pDesc == NULL) {
2194        lStatus = BAD_VALUE;
2195        goto Exit;
2196    }
2197
2198    // check audio settings permission for global effects
2199    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2200        lStatus = PERMISSION_DENIED;
2201        goto Exit;
2202    }
2203
2204    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2205    // that can only be created by audio policy manager (running in same process)
2206    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2207        lStatus = PERMISSION_DENIED;
2208        goto Exit;
2209    }
2210
2211    {
2212        if (!EffectIsNullUuid(&pDesc->uuid)) {
2213            // if uuid is specified, request effect descriptor
2214            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2215            if (lStatus < 0) {
2216                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2217                goto Exit;
2218            }
2219        } else {
2220            // if uuid is not specified, look for an available implementation
2221            // of the required type in effect factory
2222            if (EffectIsNullUuid(&pDesc->type)) {
2223                ALOGW("createEffect() no effect type");
2224                lStatus = BAD_VALUE;
2225                goto Exit;
2226            }
2227            uint32_t numEffects = 0;
2228            effect_descriptor_t d;
2229            d.flags = 0; // prevent compiler warning
2230            bool found = false;
2231
2232            lStatus = EffectQueryNumberEffects(&numEffects);
2233            if (lStatus < 0) {
2234                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2235                goto Exit;
2236            }
2237            for (uint32_t i = 0; i < numEffects; i++) {
2238                lStatus = EffectQueryEffect(i, &desc);
2239                if (lStatus < 0) {
2240                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2241                    continue;
2242                }
2243                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2244                    // If matching type found save effect descriptor. If the session is
2245                    // 0 and the effect is not auxiliary, continue enumeration in case
2246                    // an auxiliary version of this effect type is available
2247                    found = true;
2248                    d = desc;
2249                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2250                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2251                        break;
2252                    }
2253                }
2254            }
2255            if (!found) {
2256                lStatus = BAD_VALUE;
2257                ALOGW("createEffect() effect not found");
2258                goto Exit;
2259            }
2260            // For same effect type, chose auxiliary version over insert version if
2261            // connect to output mix (Compliance to OpenSL ES)
2262            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2263                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2264                desc = d;
2265            }
2266        }
2267
2268        // Do not allow auxiliary effects on a session different from 0 (output mix)
2269        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2270             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2271            lStatus = INVALID_OPERATION;
2272            goto Exit;
2273        }
2274
2275        // check recording permission for visualizer
2276        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2277            !recordingAllowed()) {
2278            lStatus = PERMISSION_DENIED;
2279            goto Exit;
2280        }
2281
2282        // return effect descriptor
2283        *pDesc = desc;
2284        if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2285            // if the output returned by getOutputForEffect() is removed before we lock the
2286            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2287            // and we will exit safely
2288            io = AudioSystem::getOutputForEffect(&desc);
2289            ALOGV("createEffect got output %d", io);
2290        }
2291
2292        Mutex::Autolock _l(mLock);
2293
2294        // If output is not specified try to find a matching audio session ID in one of the
2295        // output threads.
2296        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2297        // because of code checking output when entering the function.
2298        // Note: io is never 0 when creating an effect on an input
2299        if (io == 0) {
2300            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2301                // output must be specified by AudioPolicyManager when using session
2302                // AUDIO_SESSION_OUTPUT_STAGE
2303                lStatus = BAD_VALUE;
2304                goto Exit;
2305            }
2306            // look for the thread where the specified audio session is present
2307            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2308                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2309                    io = mPlaybackThreads.keyAt(i);
2310                    break;
2311                }
2312            }
2313            if (io == 0) {
2314                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2315                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2316                        io = mRecordThreads.keyAt(i);
2317                        break;
2318                    }
2319                }
2320            }
2321            // If no output thread contains the requested session ID, default to
2322            // first output. The effect chain will be moved to the correct output
2323            // thread when a track with the same session ID is created
2324            if (io == 0 && mPlaybackThreads.size()) {
2325                io = mPlaybackThreads.keyAt(0);
2326            }
2327            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2328        }
2329        ThreadBase *thread = checkRecordThread_l(io);
2330        if (thread == NULL) {
2331            thread = checkPlaybackThread_l(io);
2332            if (thread == NULL) {
2333                ALOGE("createEffect() unknown output thread");
2334                lStatus = BAD_VALUE;
2335                goto Exit;
2336            }
2337        }
2338
2339        sp<Client> client = registerPid_l(pid);
2340
2341        // create effect on selected output thread
2342        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2343                &desc, enabled, &lStatus);
2344        if (handle != 0 && id != NULL) {
2345            *id = handle->id();
2346        }
2347    }
2348
2349Exit:
2350    *status = lStatus;
2351    return handle;
2352}
2353
2354status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2355        audio_io_handle_t dstOutput)
2356{
2357    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2358            sessionId, srcOutput, dstOutput);
2359    Mutex::Autolock _l(mLock);
2360    if (srcOutput == dstOutput) {
2361        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2362        return NO_ERROR;
2363    }
2364    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2365    if (srcThread == NULL) {
2366        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2367        return BAD_VALUE;
2368    }
2369    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2370    if (dstThread == NULL) {
2371        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2372        return BAD_VALUE;
2373    }
2374
2375    Mutex::Autolock _dl(dstThread->mLock);
2376    Mutex::Autolock _sl(srcThread->mLock);
2377    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2378}
2379
2380// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2381status_t AudioFlinger::moveEffectChain_l(int sessionId,
2382                                   AudioFlinger::PlaybackThread *srcThread,
2383                                   AudioFlinger::PlaybackThread *dstThread,
2384                                   bool reRegister)
2385{
2386    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2387            sessionId, srcThread, dstThread);
2388
2389    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2390    if (chain == 0) {
2391        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2392                sessionId, srcThread);
2393        return INVALID_OPERATION;
2394    }
2395
2396    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2397    // so that a new chain is created with correct parameters when first effect is added. This is
2398    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2399    // removed.
2400    srcThread->removeEffectChain_l(chain);
2401
2402    // transfer all effects one by one so that new effect chain is created on new thread with
2403    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2404    sp<EffectChain> dstChain;
2405    uint32_t strategy = 0; // prevent compiler warning
2406    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2407    Vector< sp<EffectModule> > removed;
2408    status_t status = NO_ERROR;
2409    while (effect != 0) {
2410        srcThread->removeEffect_l(effect);
2411        removed.add(effect);
2412        status = dstThread->addEffect_l(effect);
2413        if (status != NO_ERROR) {
2414            break;
2415        }
2416        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2417        if (effect->state() == EffectModule::ACTIVE ||
2418                effect->state() == EffectModule::STOPPING) {
2419            effect->start();
2420        }
2421        // if the move request is not received from audio policy manager, the effect must be
2422        // re-registered with the new strategy and output
2423        if (dstChain == 0) {
2424            dstChain = effect->chain().promote();
2425            if (dstChain == 0) {
2426                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2427                status = NO_INIT;
2428                break;
2429            }
2430            strategy = dstChain->strategy();
2431        }
2432        if (reRegister) {
2433            AudioSystem::unregisterEffect(effect->id());
2434            AudioSystem::registerEffect(&effect->desc(),
2435                                        dstThread->id(),
2436                                        strategy,
2437                                        sessionId,
2438                                        effect->id());
2439            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2440        }
2441        effect = chain->getEffectFromId_l(0);
2442    }
2443
2444    if (status != NO_ERROR) {
2445        for (size_t i = 0; i < removed.size(); i++) {
2446            srcThread->addEffect_l(removed[i]);
2447            if (dstChain != 0 && reRegister) {
2448                AudioSystem::unregisterEffect(removed[i]->id());
2449                AudioSystem::registerEffect(&removed[i]->desc(),
2450                                            srcThread->id(),
2451                                            strategy,
2452                                            sessionId,
2453                                            removed[i]->id());
2454                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2455            }
2456        }
2457    }
2458
2459    return status;
2460}
2461
2462bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2463{
2464    if (mGlobalEffectEnableTime != 0 &&
2465            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2466        return true;
2467    }
2468
2469    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2470        sp<EffectChain> ec =
2471                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2472        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2473            return true;
2474        }
2475    }
2476    return false;
2477}
2478
2479void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2480{
2481    Mutex::Autolock _l(mLock);
2482
2483    mGlobalEffectEnableTime = systemTime();
2484
2485    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2486        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2487        if (t->mType == ThreadBase::OFFLOAD) {
2488            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2489        }
2490    }
2491
2492}
2493
2494struct Entry {
2495#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2496    char mName[MAX_NAME];
2497};
2498
2499int comparEntry(const void *p1, const void *p2)
2500{
2501    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2502}
2503
2504#ifdef TEE_SINK
2505void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2506{
2507    NBAIO_Source *teeSource = source.get();
2508    if (teeSource != NULL) {
2509        // .wav rotation
2510        // There is a benign race condition if 2 threads call this simultaneously.
2511        // They would both traverse the directory, but the result would simply be
2512        // failures at unlink() which are ignored.  It's also unlikely since
2513        // normally dumpsys is only done by bugreport or from the command line.
2514        char teePath[32+256];
2515        strcpy(teePath, "/data/misc/media");
2516        size_t teePathLen = strlen(teePath);
2517        DIR *dir = opendir(teePath);
2518        teePath[teePathLen++] = '/';
2519        if (dir != NULL) {
2520#define MAX_SORT 20 // number of entries to sort
2521#define MAX_KEEP 10 // number of entries to keep
2522            struct Entry entries[MAX_SORT];
2523            size_t entryCount = 0;
2524            while (entryCount < MAX_SORT) {
2525                struct dirent de;
2526                struct dirent *result = NULL;
2527                int rc = readdir_r(dir, &de, &result);
2528                if (rc != 0) {
2529                    ALOGW("readdir_r failed %d", rc);
2530                    break;
2531                }
2532                if (result == NULL) {
2533                    break;
2534                }
2535                if (result != &de) {
2536                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2537                    break;
2538                }
2539                // ignore non .wav file entries
2540                size_t nameLen = strlen(de.d_name);
2541                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2542                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2543                    continue;
2544                }
2545                strcpy(entries[entryCount++].mName, de.d_name);
2546            }
2547            (void) closedir(dir);
2548            if (entryCount > MAX_KEEP) {
2549                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2550                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2551                    strcpy(&teePath[teePathLen], entries[i].mName);
2552                    (void) unlink(teePath);
2553                }
2554            }
2555        } else {
2556            if (fd >= 0) {
2557                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2558            }
2559        }
2560        char teeTime[16];
2561        struct timeval tv;
2562        gettimeofday(&tv, NULL);
2563        struct tm tm;
2564        localtime_r(&tv.tv_sec, &tm);
2565        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2566        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2567        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2568        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2569        if (teeFd >= 0) {
2570            char wavHeader[44];
2571            memcpy(wavHeader,
2572                "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",
2573                sizeof(wavHeader));
2574            NBAIO_Format format = teeSource->format();
2575            unsigned channelCount = Format_channelCount(format);
2576            ALOG_ASSERT(channelCount <= FCC_2);
2577            uint32_t sampleRate = Format_sampleRate(format);
2578            wavHeader[22] = channelCount;       // number of channels
2579            wavHeader[24] = sampleRate;         // sample rate
2580            wavHeader[25] = sampleRate >> 8;
2581            wavHeader[32] = channelCount * 2;   // block alignment
2582            write(teeFd, wavHeader, sizeof(wavHeader));
2583            size_t total = 0;
2584            bool firstRead = true;
2585            for (;;) {
2586#define TEE_SINK_READ 1024
2587                short buffer[TEE_SINK_READ * FCC_2];
2588                size_t count = TEE_SINK_READ;
2589                ssize_t actual = teeSource->read(buffer, count,
2590                        AudioBufferProvider::kInvalidPTS);
2591                bool wasFirstRead = firstRead;
2592                firstRead = false;
2593                if (actual <= 0) {
2594                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2595                        continue;
2596                    }
2597                    break;
2598                }
2599                ALOG_ASSERT(actual <= (ssize_t)count);
2600                write(teeFd, buffer, actual * channelCount * sizeof(short));
2601                total += actual;
2602            }
2603            lseek(teeFd, (off_t) 4, SEEK_SET);
2604            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2605            write(teeFd, &temp, sizeof(temp));
2606            lseek(teeFd, (off_t) 40, SEEK_SET);
2607            temp =  total * channelCount * sizeof(short);
2608            write(teeFd, &temp, sizeof(temp));
2609            close(teeFd);
2610            if (fd >= 0) {
2611                fdprintf(fd, "tee copied to %s\n", teePath);
2612            }
2613        } else {
2614            if (fd >= 0) {
2615                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2616            }
2617        }
2618    }
2619}
2620#endif
2621
2622// ----------------------------------------------------------------------------
2623
2624status_t AudioFlinger::onTransact(
2625        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2626{
2627    return BnAudioFlinger::onTransact(code, data, reply, flags);
2628}
2629
2630}; // namespace android
2631