AudioFlinger.cpp revision d2304db2fcb5112292105a0949a55986a4c9875f 
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        String8& name,
513        int clientUid,
514        status_t *status)
515{
516    sp<PlaybackThread::Track> track;
517    sp<TrackHandle> trackHandle;
518    sp<Client> client;
519    status_t lStatus;
520    int lSessionId;
521
522    // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
523    // but if someone uses binder directly they could bypass that and cause us to crash
524    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
525        ALOGE("createTrack() invalid stream type %d", streamType);
526        lStatus = BAD_VALUE;
527        goto Exit;
528    }
529
530    // further sample rate checks are performed by createTrack_l() depending on the thread type
531    if (sampleRate == 0) {
532        ALOGE("createTrack() invalid sample rate %u", sampleRate);
533        lStatus = BAD_VALUE;
534        goto Exit;
535    }
536
537    // further channel mask checks are performed by createTrack_l() depending on the thread type
538    if (!audio_is_output_channel(channelMask)) {
539        ALOGE("createTrack() invalid channel mask %#x", channelMask);
540        lStatus = BAD_VALUE;
541        goto Exit;
542    }
543
544    // client is responsible for conversion of 8-bit PCM to 16-bit PCM,
545    // and we don't yet support 8.24 or 32-bit PCM
546    if (!audio_is_valid_format(format) ||
547            (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT)) {
548        ALOGE("createTrack() invalid format %#x", format);
549        lStatus = BAD_VALUE;
550        goto Exit;
551    }
552
553    if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
554        ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
555        lStatus = BAD_VALUE;
556        goto Exit;
557    }
558
559    {
560        Mutex::Autolock _l(mLock);
561        PlaybackThread *thread = checkPlaybackThread_l(output);
562        PlaybackThread *effectThread = NULL;
563        if (thread == NULL) {
564            ALOGE("no playback thread found for output handle %d", output);
565            lStatus = BAD_VALUE;
566            goto Exit;
567        }
568
569        pid_t pid = IPCThreadState::self()->getCallingPid();
570
571        client = registerPid_l(pid);
572
573        ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
574        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
575            // check if an effect chain with the same session ID is present on another
576            // output thread and move it here.
577            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
578                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
579                if (mPlaybackThreads.keyAt(i) != output) {
580                    uint32_t sessions = t->hasAudioSession(*sessionId);
581                    if (sessions & PlaybackThread::EFFECT_SESSION) {
582                        effectThread = t.get();
583                        break;
584                    }
585                }
586            }
587            lSessionId = *sessionId;
588        } else {
589            // if no audio session id is provided, create one here
590            lSessionId = nextUniqueId();
591            if (sessionId != NULL) {
592                *sessionId = lSessionId;
593            }
594        }
595        ALOGV("createTrack() lSessionId: %d", lSessionId);
596
597        track = thread->createTrack_l(client, streamType, sampleRate, format,
598                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
599        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
600        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
601
602        // move effect chain to this output thread if an effect on same session was waiting
603        // for a track to be created
604        if (lStatus == NO_ERROR && effectThread != NULL) {
605            // no risk of deadlock because AudioFlinger::mLock is held
606            Mutex::Autolock _dl(thread->mLock);
607            Mutex::Autolock _sl(effectThread->mLock);
608            moveEffectChain_l(lSessionId, effectThread, thread, true);
609        }
610
611        // Look for sync events awaiting for a session to be used.
612        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
613            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
614                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
615                    if (lStatus == NO_ERROR) {
616                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
617                    } else {
618                        mPendingSyncEvents[i]->cancel();
619                    }
620                    mPendingSyncEvents.removeAt(i);
621                    i--;
622                }
623            }
624        }
625
626    }
627
628    if (lStatus == NO_ERROR) {
629        // s for server's pid, n for normal mixer name, f for fast index
630        name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0,
631                track->fastIndex());
632        trackHandle = new TrackHandle(track);
633    } else {
634        // remove local strong reference to Client before deleting the Track so that the Client
635        // destructor is called by the TrackBase destructor with mLock held
636        client.clear();
637        track.clear();
638    }
639
640Exit:
641    *status = lStatus;
642    return trackHandle;
643}
644
645uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
646{
647    Mutex::Autolock _l(mLock);
648    PlaybackThread *thread = checkPlaybackThread_l(output);
649    if (thread == NULL) {
650        ALOGW("sampleRate() unknown thread %d", output);
651        return 0;
652    }
653    return thread->sampleRate();
654}
655
656int AudioFlinger::channelCount(audio_io_handle_t output) const
657{
658    Mutex::Autolock _l(mLock);
659    PlaybackThread *thread = checkPlaybackThread_l(output);
660    if (thread == NULL) {
661        ALOGW("channelCount() unknown thread %d", output);
662        return 0;
663    }
664    return thread->channelCount();
665}
666
667audio_format_t AudioFlinger::format(audio_io_handle_t output) const
668{
669    Mutex::Autolock _l(mLock);
670    PlaybackThread *thread = checkPlaybackThread_l(output);
671    if (thread == NULL) {
672        ALOGW("format() unknown thread %d", output);
673        return AUDIO_FORMAT_INVALID;
674    }
675    return thread->format();
676}
677
678size_t AudioFlinger::frameCount(audio_io_handle_t output) const
679{
680    Mutex::Autolock _l(mLock);
681    PlaybackThread *thread = checkPlaybackThread_l(output);
682    if (thread == NULL) {
683        ALOGW("frameCount() unknown thread %d", output);
684        return 0;
685    }
686    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
687    //       should examine all callers and fix them to handle smaller counts
688    return thread->frameCount();
689}
690
691uint32_t AudioFlinger::latency(audio_io_handle_t output) const
692{
693    Mutex::Autolock _l(mLock);
694    PlaybackThread *thread = checkPlaybackThread_l(output);
695    if (thread == NULL) {
696        ALOGW("latency(): no playback thread found for output handle %d", output);
697        return 0;
698    }
699    return thread->latency();
700}
701
702status_t AudioFlinger::setMasterVolume(float value)
703{
704    status_t ret = initCheck();
705    if (ret != NO_ERROR) {
706        return ret;
707    }
708
709    // check calling permissions
710    if (!settingsAllowed()) {
711        return PERMISSION_DENIED;
712    }
713
714    Mutex::Autolock _l(mLock);
715    mMasterVolume = value;
716
717    // Set master volume in the HALs which support it.
718    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
719        AutoMutex lock(mHardwareLock);
720        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
721
722        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
723        if (dev->canSetMasterVolume()) {
724            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
725        }
726        mHardwareStatus = AUDIO_HW_IDLE;
727    }
728
729    // Now set the master volume in each playback thread.  Playback threads
730    // assigned to HALs which do not have master volume support will apply
731    // master volume during the mix operation.  Threads with HALs which do
732    // support master volume will simply ignore the setting.
733    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
734        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
735
736    return NO_ERROR;
737}
738
739status_t AudioFlinger::setMode(audio_mode_t mode)
740{
741    status_t ret = initCheck();
742    if (ret != NO_ERROR) {
743        return ret;
744    }
745
746    // check calling permissions
747    if (!settingsAllowed()) {
748        return PERMISSION_DENIED;
749    }
750    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
751        ALOGW("Illegal value: setMode(%d)", mode);
752        return BAD_VALUE;
753    }
754
755    { // scope for the lock
756        AutoMutex lock(mHardwareLock);
757        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
758        mHardwareStatus = AUDIO_HW_SET_MODE;
759        ret = dev->set_mode(dev, mode);
760        mHardwareStatus = AUDIO_HW_IDLE;
761    }
762
763    if (NO_ERROR == ret) {
764        Mutex::Autolock _l(mLock);
765        mMode = mode;
766        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
767            mPlaybackThreads.valueAt(i)->setMode(mode);
768    }
769
770    return ret;
771}
772
773status_t AudioFlinger::setMicMute(bool state)
774{
775    status_t ret = initCheck();
776    if (ret != NO_ERROR) {
777        return ret;
778    }
779
780    // check calling permissions
781    if (!settingsAllowed()) {
782        return PERMISSION_DENIED;
783    }
784
785    AutoMutex lock(mHardwareLock);
786    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
787    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
788    ret = dev->set_mic_mute(dev, state);
789    mHardwareStatus = AUDIO_HW_IDLE;
790    return ret;
791}
792
793bool AudioFlinger::getMicMute() const
794{
795    status_t ret = initCheck();
796    if (ret != NO_ERROR) {
797        return false;
798    }
799
800    bool state = AUDIO_MODE_INVALID;
801    AutoMutex lock(mHardwareLock);
802    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
803    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
804    dev->get_mic_mute(dev, &state);
805    mHardwareStatus = AUDIO_HW_IDLE;
806    return state;
807}
808
809status_t AudioFlinger::setMasterMute(bool muted)
810{
811    status_t ret = initCheck();
812    if (ret != NO_ERROR) {
813        return ret;
814    }
815
816    // check calling permissions
817    if (!settingsAllowed()) {
818        return PERMISSION_DENIED;
819    }
820
821    Mutex::Autolock _l(mLock);
822    mMasterMute = muted;
823
824    // Set master mute in the HALs which support it.
825    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
826        AutoMutex lock(mHardwareLock);
827        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
828
829        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
830        if (dev->canSetMasterMute()) {
831            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
832        }
833        mHardwareStatus = AUDIO_HW_IDLE;
834    }
835
836    // Now set the master mute in each playback thread.  Playback threads
837    // assigned to HALs which do not have master mute support will apply master
838    // mute during the mix operation.  Threads with HALs which do support master
839    // mute will simply ignore the setting.
840    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
841        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
842
843    return NO_ERROR;
844}
845
846float AudioFlinger::masterVolume() const
847{
848    Mutex::Autolock _l(mLock);
849    return masterVolume_l();
850}
851
852bool AudioFlinger::masterMute() const
853{
854    Mutex::Autolock _l(mLock);
855    return masterMute_l();
856}
857
858float AudioFlinger::masterVolume_l() const
859{
860    return mMasterVolume;
861}
862
863bool AudioFlinger::masterMute_l() const
864{
865    return mMasterMute;
866}
867
868status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
869        audio_io_handle_t output)
870{
871    // check calling permissions
872    if (!settingsAllowed()) {
873        return PERMISSION_DENIED;
874    }
875
876    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
877        ALOGE("setStreamVolume() invalid stream %d", stream);
878        return BAD_VALUE;
879    }
880
881    AutoMutex lock(mLock);
882    PlaybackThread *thread = NULL;
883    if (output) {
884        thread = checkPlaybackThread_l(output);
885        if (thread == NULL) {
886            return BAD_VALUE;
887        }
888    }
889
890    mStreamTypes[stream].volume = value;
891
892    if (thread == NULL) {
893        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
894            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
895        }
896    } else {
897        thread->setStreamVolume(stream, value);
898    }
899
900    return NO_ERROR;
901}
902
903status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
904{
905    // check calling permissions
906    if (!settingsAllowed()) {
907        return PERMISSION_DENIED;
908    }
909
910    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
911        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
912        ALOGE("setStreamMute() invalid stream %d", stream);
913        return BAD_VALUE;
914    }
915
916    AutoMutex lock(mLock);
917    mStreamTypes[stream].mute = muted;
918    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
919        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
920
921    return NO_ERROR;
922}
923
924float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
925{
926    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
927        return 0.0f;
928    }
929
930    AutoMutex lock(mLock);
931    float volume;
932    if (output) {
933        PlaybackThread *thread = checkPlaybackThread_l(output);
934        if (thread == NULL) {
935            return 0.0f;
936        }
937        volume = thread->streamVolume(stream);
938    } else {
939        volume = streamVolume_l(stream);
940    }
941
942    return volume;
943}
944
945bool AudioFlinger::streamMute(audio_stream_type_t stream) const
946{
947    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
948        return true;
949    }
950
951    AutoMutex lock(mLock);
952    return streamMute_l(stream);
953}
954
955status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
956{
957    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
958            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
959
960    // check calling permissions
961    if (!settingsAllowed()) {
962        return PERMISSION_DENIED;
963    }
964
965    // ioHandle == 0 means the parameters are global to the audio hardware interface
966    if (ioHandle == 0) {
967        Mutex::Autolock _l(mLock);
968        status_t final_result = NO_ERROR;
969        {
970            AutoMutex lock(mHardwareLock);
971            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
972            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
973                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
974                status_t result = dev->set_parameters(dev, keyValuePairs.string());
975                final_result = result ?: final_result;
976            }
977            mHardwareStatus = AUDIO_HW_IDLE;
978        }
979        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
980        AudioParameter param = AudioParameter(keyValuePairs);
981        String8 value;
982        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
983            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
984            if (mBtNrecIsOff != btNrecIsOff) {
985                for (size_t i = 0; i < mRecordThreads.size(); i++) {
986                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
987                    audio_devices_t device = thread->inDevice();
988                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
989                    // collect all of the thread's session IDs
990                    KeyedVector<int, bool> ids = thread->sessionIds();
991                    // suspend effects associated with those session IDs
992                    for (size_t j = 0; j < ids.size(); ++j) {
993                        int sessionId = ids.keyAt(j);
994                        thread->setEffectSuspended(FX_IID_AEC,
995                                                   suspend,
996                                                   sessionId);
997                        thread->setEffectSuspended(FX_IID_NS,
998                                                   suspend,
999                                                   sessionId);
1000                    }
1001                }
1002                mBtNrecIsOff = btNrecIsOff;
1003            }
1004        }
1005        String8 screenState;
1006        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1007            bool isOff = screenState == "off";
1008            if (isOff != (AudioFlinger::mScreenState & 1)) {
1009                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1010            }
1011        }
1012        return final_result;
1013    }
1014
1015    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1016    // and the thread is exited once the lock is released
1017    sp<ThreadBase> thread;
1018    {
1019        Mutex::Autolock _l(mLock);
1020        thread = checkPlaybackThread_l(ioHandle);
1021        if (thread == 0) {
1022            thread = checkRecordThread_l(ioHandle);
1023        } else if (thread == primaryPlaybackThread_l()) {
1024            // indicate output device change to all input threads for pre processing
1025            AudioParameter param = AudioParameter(keyValuePairs);
1026            int value;
1027            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1028                    (value != 0)) {
1029                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1030                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1031                }
1032            }
1033        }
1034    }
1035    if (thread != 0) {
1036        return thread->setParameters(keyValuePairs);
1037    }
1038    return BAD_VALUE;
1039}
1040
1041String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1042{
1043    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1044            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1045
1046    Mutex::Autolock _l(mLock);
1047
1048    if (ioHandle == 0) {
1049        String8 out_s8;
1050
1051        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1052            char *s;
1053            {
1054            AutoMutex lock(mHardwareLock);
1055            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1056            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1057            s = dev->get_parameters(dev, keys.string());
1058            mHardwareStatus = AUDIO_HW_IDLE;
1059            }
1060            out_s8 += String8(s ? s : "");
1061            free(s);
1062        }
1063        return out_s8;
1064    }
1065
1066    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1067    if (playbackThread != NULL) {
1068        return playbackThread->getParameters(keys);
1069    }
1070    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1071    if (recordThread != NULL) {
1072        return recordThread->getParameters(keys);
1073    }
1074    return String8("");
1075}
1076
1077size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1078        audio_channel_mask_t channelMask) const
1079{
1080    status_t ret = initCheck();
1081    if (ret != NO_ERROR) {
1082        return 0;
1083    }
1084
1085    AutoMutex lock(mHardwareLock);
1086    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1087    struct audio_config config;
1088    memset(&config, 0, sizeof(config));
1089    config.sample_rate = sampleRate;
1090    config.channel_mask = channelMask;
1091    config.format = format;
1092
1093    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1094    size_t size = dev->get_input_buffer_size(dev, &config);
1095    mHardwareStatus = AUDIO_HW_IDLE;
1096    return size;
1097}
1098
1099uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1100{
1101    Mutex::Autolock _l(mLock);
1102
1103    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1104    if (recordThread != NULL) {
1105        return recordThread->getInputFramesLost();
1106    }
1107    return 0;
1108}
1109
1110status_t AudioFlinger::setVoiceVolume(float value)
1111{
1112    status_t ret = initCheck();
1113    if (ret != NO_ERROR) {
1114        return ret;
1115    }
1116
1117    // check calling permissions
1118    if (!settingsAllowed()) {
1119        return PERMISSION_DENIED;
1120    }
1121
1122    AutoMutex lock(mHardwareLock);
1123    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1124    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1125    ret = dev->set_voice_volume(dev, value);
1126    mHardwareStatus = AUDIO_HW_IDLE;
1127
1128    return ret;
1129}
1130
1131status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1132        audio_io_handle_t output) const
1133{
1134    status_t status;
1135
1136    Mutex::Autolock _l(mLock);
1137
1138    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1139    if (playbackThread != NULL) {
1140        return playbackThread->getRenderPosition(halFrames, dspFrames);
1141    }
1142
1143    return BAD_VALUE;
1144}
1145
1146void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1147{
1148
1149    Mutex::Autolock _l(mLock);
1150
1151    pid_t pid = IPCThreadState::self()->getCallingPid();
1152    if (mNotificationClients.indexOfKey(pid) < 0) {
1153        sp<NotificationClient> notificationClient = new NotificationClient(this,
1154                                                                            client,
1155                                                                            pid);
1156        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1157
1158        mNotificationClients.add(pid, notificationClient);
1159
1160        sp<IBinder> binder = client->asBinder();
1161        binder->linkToDeath(notificationClient);
1162
1163        // the config change is always sent from playback or record threads to avoid deadlock
1164        // with AudioSystem::gLock
1165        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1166            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1167        }
1168
1169        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1170            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1171        }
1172    }
1173}
1174
1175void AudioFlinger::removeNotificationClient(pid_t pid)
1176{
1177    Mutex::Autolock _l(mLock);
1178
1179    mNotificationClients.removeItem(pid);
1180
1181    ALOGV("%d died, releasing its sessions", pid);
1182    size_t num = mAudioSessionRefs.size();
1183    bool removed = false;
1184    for (size_t i = 0; i< num; ) {
1185        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1186        ALOGV(" pid %d @ %d", ref->mPid, i);
1187        if (ref->mPid == pid) {
1188            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1189            mAudioSessionRefs.removeAt(i);
1190            delete ref;
1191            removed = true;
1192            num--;
1193        } else {
1194            i++;
1195        }
1196    }
1197    if (removed) {
1198        purgeStaleEffects_l();
1199    }
1200}
1201
1202// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1203void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1204{
1205    size_t size = mNotificationClients.size();
1206    for (size_t i = 0; i < size; i++) {
1207        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1208                                                                               param2);
1209    }
1210}
1211
1212// removeClient_l() must be called with AudioFlinger::mLock held
1213void AudioFlinger::removeClient_l(pid_t pid)
1214{
1215    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1216            IPCThreadState::self()->getCallingPid());
1217    mClients.removeItem(pid);
1218}
1219
1220// getEffectThread_l() must be called with AudioFlinger::mLock held
1221sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1222{
1223    sp<PlaybackThread> thread;
1224
1225    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1226        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1227            ALOG_ASSERT(thread == 0);
1228            thread = mPlaybackThreads.valueAt(i);
1229        }
1230    }
1231
1232    return thread;
1233}
1234
1235
1236
1237// ----------------------------------------------------------------------------
1238
1239AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1240    :   RefBase(),
1241        mAudioFlinger(audioFlinger),
1242        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1243        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1244        mPid(pid),
1245        mTimedTrackCount(0)
1246{
1247    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1248}
1249
1250// Client destructor must be called with AudioFlinger::mLock held
1251AudioFlinger::Client::~Client()
1252{
1253    mAudioFlinger->removeClient_l(mPid);
1254}
1255
1256sp<MemoryDealer> AudioFlinger::Client::heap() const
1257{
1258    return mMemoryDealer;
1259}
1260
1261// Reserve one of the limited slots for a timed audio track associated
1262// with this client
1263bool AudioFlinger::Client::reserveTimedTrack()
1264{
1265    const int kMaxTimedTracksPerClient = 4;
1266
1267    Mutex::Autolock _l(mTimedTrackLock);
1268
1269    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1270        ALOGW("can not create timed track - pid %d has exceeded the limit",
1271             mPid);
1272        return false;
1273    }
1274
1275    mTimedTrackCount++;
1276    return true;
1277}
1278
1279// Release a slot for a timed audio track
1280void AudioFlinger::Client::releaseTimedTrack()
1281{
1282    Mutex::Autolock _l(mTimedTrackLock);
1283    mTimedTrackCount--;
1284}
1285
1286// ----------------------------------------------------------------------------
1287
1288AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1289                                                     const sp<IAudioFlingerClient>& client,
1290                                                     pid_t pid)
1291    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1292{
1293}
1294
1295AudioFlinger::NotificationClient::~NotificationClient()
1296{
1297}
1298
1299void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1300{
1301    sp<NotificationClient> keep(this);
1302    mAudioFlinger->removeNotificationClient(mPid);
1303}
1304
1305
1306// ----------------------------------------------------------------------------
1307
1308static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1309    return audio_is_remote_submix_device(inDevice);
1310}
1311
1312sp<IAudioRecord> AudioFlinger::openRecord(
1313        audio_io_handle_t input,
1314        uint32_t sampleRate,
1315        audio_format_t format,
1316        audio_channel_mask_t channelMask,
1317        size_t *frameCount,
1318        IAudioFlinger::track_flags_t *flags,
1319        pid_t tid,
1320        int *sessionId,
1321        status_t *status)
1322{
1323    sp<RecordThread::RecordTrack> recordTrack;
1324    sp<RecordHandle> recordHandle;
1325    sp<Client> client;
1326    status_t lStatus;
1327    RecordThread *thread;
1328    size_t inFrameCount;
1329    int lSessionId;
1330
1331    // check calling permissions
1332    if (!recordingAllowed()) {
1333        ALOGE("openRecord() permission denied: recording not allowed");
1334        lStatus = PERMISSION_DENIED;
1335        goto Exit;
1336    }
1337
1338    // further sample rate checks are performed by createRecordTrack_l()
1339    if (sampleRate == 0) {
1340        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1341        lStatus = BAD_VALUE;
1342        goto Exit;
1343    }
1344
1345    // FIXME when we support more formats, add audio_is_valid_format(format)
1346    //       and any explicit restrictions if audio_is_linear_pcm(format)
1347    if (format != AUDIO_FORMAT_PCM_16_BIT) {
1348        ALOGE("openRecord() invalid format %#x", format);
1349        lStatus = BAD_VALUE;
1350        goto Exit;
1351    }
1352
1353    // further channel mask checks are performed by createRecordTrack_l()
1354    if (!audio_is_input_channel(channelMask)) {
1355        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1356        lStatus = BAD_VALUE;
1357        goto Exit;
1358    }
1359
1360    // add client to list
1361    { // scope for mLock
1362        Mutex::Autolock _l(mLock);
1363        thread = checkRecordThread_l(input);
1364        if (thread == NULL) {
1365            ALOGE("openRecord() checkRecordThread_l failed");
1366            lStatus = BAD_VALUE;
1367            goto Exit;
1368        }
1369
1370        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1371                && !captureAudioOutputAllowed()) {
1372            ALOGE("openRecord() permission denied: capture not allowed");
1373            lStatus = PERMISSION_DENIED;
1374            goto Exit;
1375        }
1376
1377        pid_t pid = IPCThreadState::self()->getCallingPid();
1378        client = registerPid_l(pid);
1379
1380        // If no audio session id is provided, create one here
1381        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1382            lSessionId = *sessionId;
1383        } else {
1384            lSessionId = nextUniqueId();
1385            if (sessionId != NULL) {
1386                *sessionId = lSessionId;
1387            }
1388        }
1389        // create new record track.
1390        // The record track uses one track in mHardwareMixerThread by convention.
1391        // TODO: the uid should be passed in as a parameter to openRecord
1392        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1393                                                  frameCount, lSessionId,
1394                                                  IPCThreadState::self()->getCallingUid(),
1395                                                  flags, tid, &lStatus);
1396        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1397    }
1398
1399    if (lStatus != NO_ERROR) {
1400        // remove local strong reference to Client before deleting the RecordTrack so that the
1401        // Client destructor is called by the TrackBase destructor with mLock held
1402        client.clear();
1403        recordTrack.clear();
1404        goto Exit;
1405    }
1406
1407    // return handle to client
1408    recordHandle = new RecordHandle(recordTrack);
1409
1410Exit:
1411    *status = lStatus;
1412    return recordHandle;
1413}
1414
1415
1416
1417// ----------------------------------------------------------------------------
1418
1419audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1420{
1421    if (!settingsAllowed()) {
1422        return 0;
1423    }
1424    Mutex::Autolock _l(mLock);
1425    return loadHwModule_l(name);
1426}
1427
1428// loadHwModule_l() must be called with AudioFlinger::mLock held
1429audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1430{
1431    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1432        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1433            ALOGW("loadHwModule() module %s already loaded", name);
1434            return mAudioHwDevs.keyAt(i);
1435        }
1436    }
1437
1438    audio_hw_device_t *dev;
1439
1440    int rc = load_audio_interface(name, &dev);
1441    if (rc) {
1442        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1443        return 0;
1444    }
1445
1446    mHardwareStatus = AUDIO_HW_INIT;
1447    rc = dev->init_check(dev);
1448    mHardwareStatus = AUDIO_HW_IDLE;
1449    if (rc) {
1450        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1451        return 0;
1452    }
1453
1454    // Check and cache this HAL's level of support for master mute and master
1455    // volume.  If this is the first HAL opened, and it supports the get
1456    // methods, use the initial values provided by the HAL as the current
1457    // master mute and volume settings.
1458
1459    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1460    {  // scope for auto-lock pattern
1461        AutoMutex lock(mHardwareLock);
1462
1463        if (0 == mAudioHwDevs.size()) {
1464            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1465            if (NULL != dev->get_master_volume) {
1466                float mv;
1467                if (OK == dev->get_master_volume(dev, &mv)) {
1468                    mMasterVolume = mv;
1469                }
1470            }
1471
1472            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1473            if (NULL != dev->get_master_mute) {
1474                bool mm;
1475                if (OK == dev->get_master_mute(dev, &mm)) {
1476                    mMasterMute = mm;
1477                }
1478            }
1479        }
1480
1481        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1482        if ((NULL != dev->set_master_volume) &&
1483            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1484            flags = static_cast<AudioHwDevice::Flags>(flags |
1485                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1486        }
1487
1488        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1489        if ((NULL != dev->set_master_mute) &&
1490            (OK == dev->set_master_mute(dev, mMasterMute))) {
1491            flags = static_cast<AudioHwDevice::Flags>(flags |
1492                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1493        }
1494
1495        mHardwareStatus = AUDIO_HW_IDLE;
1496    }
1497
1498    audio_module_handle_t handle = nextUniqueId();
1499    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1500
1501    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1502          name, dev->common.module->name, dev->common.module->id, handle);
1503
1504    return handle;
1505
1506}
1507
1508// ----------------------------------------------------------------------------
1509
1510uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1511{
1512    Mutex::Autolock _l(mLock);
1513    PlaybackThread *thread = primaryPlaybackThread_l();
1514    return thread != NULL ? thread->sampleRate() : 0;
1515}
1516
1517size_t AudioFlinger::getPrimaryOutputFrameCount()
1518{
1519    Mutex::Autolock _l(mLock);
1520    PlaybackThread *thread = primaryPlaybackThread_l();
1521    return thread != NULL ? thread->frameCountHAL() : 0;
1522}
1523
1524// ----------------------------------------------------------------------------
1525
1526status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1527{
1528    uid_t uid = IPCThreadState::self()->getCallingUid();
1529    if (uid != AID_SYSTEM) {
1530        return PERMISSION_DENIED;
1531    }
1532    Mutex::Autolock _l(mLock);
1533    if (mIsDeviceTypeKnown) {
1534        return INVALID_OPERATION;
1535    }
1536    mIsLowRamDevice = isLowRamDevice;
1537    mIsDeviceTypeKnown = true;
1538    return NO_ERROR;
1539}
1540
1541// ----------------------------------------------------------------------------
1542
1543audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1544                                           audio_devices_t *pDevices,
1545                                           uint32_t *pSamplingRate,
1546                                           audio_format_t *pFormat,
1547                                           audio_channel_mask_t *pChannelMask,
1548                                           uint32_t *pLatencyMs,
1549                                           audio_output_flags_t flags,
1550                                           const audio_offload_info_t *offloadInfo)
1551{
1552    struct audio_config config;
1553    memset(&config, 0, sizeof(config));
1554    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1555    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1556    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1557    if (offloadInfo != NULL) {
1558        config.offload_info = *offloadInfo;
1559    }
1560
1561    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1562              module,
1563              (pDevices != NULL) ? *pDevices : 0,
1564              config.sample_rate,
1565              config.format,
1566              config.channel_mask,
1567              flags);
1568    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1569          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1570
1571    if (pDevices == NULL || *pDevices == 0) {
1572        return 0;
1573    }
1574
1575    Mutex::Autolock _l(mLock);
1576
1577    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1578    if (outHwDev == NULL) {
1579        return 0;
1580    }
1581
1582    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1583    audio_io_handle_t id = nextUniqueId();
1584
1585    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1586
1587    audio_stream_out_t *outStream = NULL;
1588    status_t status = hwDevHal->open_output_stream(hwDevHal,
1589                                          id,
1590                                          *pDevices,
1591                                          (audio_output_flags_t)flags,
1592                                          &config,
1593                                          &outStream);
1594
1595    mHardwareStatus = AUDIO_HW_IDLE;
1596    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1597            "Channels %x, status %d",
1598            outStream,
1599            config.sample_rate,
1600            config.format,
1601            config.channel_mask,
1602            status);
1603
1604    if (status == NO_ERROR && outStream != NULL) {
1605        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1606
1607        PlaybackThread *thread;
1608        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1609            thread = new OffloadThread(this, output, id, *pDevices);
1610            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1611        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1612            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1613            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1614            thread = new DirectOutputThread(this, output, id, *pDevices);
1615            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1616        } else {
1617            thread = new MixerThread(this, output, id, *pDevices);
1618            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1619        }
1620        mPlaybackThreads.add(id, thread);
1621
1622        if (pSamplingRate != NULL) {
1623            *pSamplingRate = config.sample_rate;
1624        }
1625        if (pFormat != NULL) {
1626            *pFormat = config.format;
1627        }
1628        if (pChannelMask != NULL) {
1629            *pChannelMask = config.channel_mask;
1630        }
1631        if (pLatencyMs != NULL) {
1632            *pLatencyMs = thread->latency();
1633        }
1634
1635        // notify client processes of the new output creation
1636        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1637
1638        // the first primary output opened designates the primary hw device
1639        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1640            ALOGI("Using module %d has the primary audio interface", module);
1641            mPrimaryHardwareDev = outHwDev;
1642
1643            AutoMutex lock(mHardwareLock);
1644            mHardwareStatus = AUDIO_HW_SET_MODE;
1645            hwDevHal->set_mode(hwDevHal, mMode);
1646            mHardwareStatus = AUDIO_HW_IDLE;
1647        }
1648        return id;
1649    }
1650
1651    return 0;
1652}
1653
1654audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1655        audio_io_handle_t output2)
1656{
1657    Mutex::Autolock _l(mLock);
1658    MixerThread *thread1 = checkMixerThread_l(output1);
1659    MixerThread *thread2 = checkMixerThread_l(output2);
1660
1661    if (thread1 == NULL || thread2 == NULL) {
1662        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1663                output2);
1664        return 0;
1665    }
1666
1667    audio_io_handle_t id = nextUniqueId();
1668    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1669    thread->addOutputTrack(thread2);
1670    mPlaybackThreads.add(id, thread);
1671    // notify client processes of the new output creation
1672    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1673    return id;
1674}
1675
1676status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1677{
1678    return closeOutput_nonvirtual(output);
1679}
1680
1681status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1682{
1683    // keep strong reference on the playback thread so that
1684    // it is not destroyed while exit() is executed
1685    sp<PlaybackThread> thread;
1686    {
1687        Mutex::Autolock _l(mLock);
1688        thread = checkPlaybackThread_l(output);
1689        if (thread == NULL) {
1690            return BAD_VALUE;
1691        }
1692
1693        ALOGV("closeOutput() %d", output);
1694
1695        if (thread->type() == ThreadBase::MIXER) {
1696            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1697                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1698                    DuplicatingThread *dupThread =
1699                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1700                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1701
1702                }
1703            }
1704        }
1705
1706
1707        mPlaybackThreads.removeItem(output);
1708        // save all effects to the default thread
1709        if (mPlaybackThreads.size()) {
1710            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1711            if (dstThread != NULL) {
1712                // audioflinger lock is held here so the acquisition order of thread locks does not
1713                // matter
1714                Mutex::Autolock _dl(dstThread->mLock);
1715                Mutex::Autolock _sl(thread->mLock);
1716                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1717                for (size_t i = 0; i < effectChains.size(); i ++) {
1718                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1719                }
1720            }
1721        }
1722        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1723    }
1724    thread->exit();
1725    // The thread entity (active unit of execution) is no longer running here,
1726    // but the ThreadBase container still exists.
1727
1728    if (thread->type() != ThreadBase::DUPLICATING) {
1729        AudioStreamOut *out = thread->clearOutput();
1730        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1731        // from now on thread->mOutput is NULL
1732        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1733        delete out;
1734    }
1735    return NO_ERROR;
1736}
1737
1738status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1739{
1740    Mutex::Autolock _l(mLock);
1741    PlaybackThread *thread = checkPlaybackThread_l(output);
1742
1743    if (thread == NULL) {
1744        return BAD_VALUE;
1745    }
1746
1747    ALOGV("suspendOutput() %d", output);
1748    thread->suspend();
1749
1750    return NO_ERROR;
1751}
1752
1753status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1754{
1755    Mutex::Autolock _l(mLock);
1756    PlaybackThread *thread = checkPlaybackThread_l(output);
1757
1758    if (thread == NULL) {
1759        return BAD_VALUE;
1760    }
1761
1762    ALOGV("restoreOutput() %d", output);
1763
1764    thread->restore();
1765
1766    return NO_ERROR;
1767}
1768
1769audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1770                                          audio_devices_t *pDevices,
1771                                          uint32_t *pSamplingRate,
1772                                          audio_format_t *pFormat,
1773                                          audio_channel_mask_t *pChannelMask)
1774{
1775    struct audio_config config;
1776    memset(&config, 0, sizeof(config));
1777    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1778    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1779    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1780
1781    uint32_t reqSamplingRate = config.sample_rate;
1782    audio_format_t reqFormat = config.format;
1783    audio_channel_mask_t reqChannelMask = config.channel_mask;
1784
1785    if (pDevices == NULL || *pDevices == 0) {
1786        return 0;
1787    }
1788
1789    Mutex::Autolock _l(mLock);
1790
1791    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1792    if (inHwDev == NULL) {
1793        return 0;
1794    }
1795
1796    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1797    audio_io_handle_t id = nextUniqueId();
1798
1799    audio_stream_in_t *inStream = NULL;
1800    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1801                                        &inStream);
1802    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
1803            "status %d",
1804            inStream,
1805            config.sample_rate,
1806            config.format,
1807            config.channel_mask,
1808            status);
1809
1810    // If the input could not be opened with the requested parameters and we can handle the
1811    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1812    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1813    if (status == BAD_VALUE &&
1814        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1815        (config.sample_rate <= 2 * reqSamplingRate) &&
1816        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1817        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1818        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1819        inStream = NULL;
1820        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1821        // FIXME log this new status; HAL should not propose any further changes
1822    }
1823
1824    if (status == NO_ERROR && inStream != NULL) {
1825
1826#ifdef TEE_SINK
1827        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1828        // or (re-)create if current Pipe is idle and does not match the new format
1829        sp<NBAIO_Sink> teeSink;
1830        enum {
1831            TEE_SINK_NO,    // don't copy input
1832            TEE_SINK_NEW,   // copy input using a new pipe
1833            TEE_SINK_OLD,   // copy input using an existing pipe
1834        } kind;
1835        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1836                                        popcount(inStream->common.get_channels(&inStream->common)));
1837        if (!mTeeSinkInputEnabled) {
1838            kind = TEE_SINK_NO;
1839        } else if (!Format_isValid(format)) {
1840            kind = TEE_SINK_NO;
1841        } else if (mRecordTeeSink == 0) {
1842            kind = TEE_SINK_NEW;
1843        } else if (mRecordTeeSink->getStrongCount() != 1) {
1844            kind = TEE_SINK_NO;
1845        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
1846            kind = TEE_SINK_OLD;
1847        } else {
1848            kind = TEE_SINK_NEW;
1849        }
1850        switch (kind) {
1851        case TEE_SINK_NEW: {
1852            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1853            size_t numCounterOffers = 0;
1854            const NBAIO_Format offers[1] = {format};
1855            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1856            ALOG_ASSERT(index == 0);
1857            PipeReader *pipeReader = new PipeReader(*pipe);
1858            numCounterOffers = 0;
1859            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1860            ALOG_ASSERT(index == 0);
1861            mRecordTeeSink = pipe;
1862            mRecordTeeSource = pipeReader;
1863            teeSink = pipe;
1864            }
1865            break;
1866        case TEE_SINK_OLD:
1867            teeSink = mRecordTeeSink;
1868            break;
1869        case TEE_SINK_NO:
1870        default:
1871            break;
1872        }
1873#endif
1874
1875        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1876
1877        // Start record thread
1878        // RecordThread requires both input and output device indication to forward to audio
1879        // pre processing modules
1880        RecordThread *thread = new RecordThread(this,
1881                                  input,
1882                                  id,
1883                                  primaryOutputDevice_l(),
1884                                  *pDevices
1885#ifdef TEE_SINK
1886                                  , teeSink
1887#endif
1888                                  );
1889        mRecordThreads.add(id, thread);
1890        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1891        if (pSamplingRate != NULL) {
1892            *pSamplingRate = reqSamplingRate;
1893        }
1894        if (pFormat != NULL) {
1895            *pFormat = config.format;
1896        }
1897        if (pChannelMask != NULL) {
1898            *pChannelMask = reqChannelMask;
1899        }
1900
1901        // notify client processes of the new input creation
1902        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1903        return id;
1904    }
1905
1906    return 0;
1907}
1908
1909status_t AudioFlinger::closeInput(audio_io_handle_t input)
1910{
1911    return closeInput_nonvirtual(input);
1912}
1913
1914status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1915{
1916    // keep strong reference on the record thread so that
1917    // it is not destroyed while exit() is executed
1918    sp<RecordThread> thread;
1919    {
1920        Mutex::Autolock _l(mLock);
1921        thread = checkRecordThread_l(input);
1922        if (thread == 0) {
1923            return BAD_VALUE;
1924        }
1925
1926        ALOGV("closeInput() %d", input);
1927        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1928        mRecordThreads.removeItem(input);
1929    }
1930    thread->exit();
1931    // The thread entity (active unit of execution) is no longer running here,
1932    // but the ThreadBase container still exists.
1933
1934    AudioStreamIn *in = thread->clearInput();
1935    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1936    // from now on thread->mInput is NULL
1937    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1938    delete in;
1939
1940    return NO_ERROR;
1941}
1942
1943status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
1944{
1945    Mutex::Autolock _l(mLock);
1946    ALOGV("invalidateStream() stream %d", stream);
1947
1948    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1949        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1950        thread->invalidateTracks(stream);
1951    }
1952
1953    return NO_ERROR;
1954}
1955
1956
1957int AudioFlinger::newAudioSessionId()
1958{
1959    return nextUniqueId();
1960}
1961
1962void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
1963{
1964    Mutex::Autolock _l(mLock);
1965    pid_t caller = IPCThreadState::self()->getCallingPid();
1966    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
1967    if (pid != -1 && (caller == getpid_cached)) {
1968        caller = pid;
1969    }
1970
1971    // Ignore requests received from processes not known as notification client. The request
1972    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1973    // called from a different pid leaving a stale session reference.  Also we don't know how
1974    // to clear this reference if the client process dies.
1975    if (mNotificationClients.indexOfKey(caller) < 0) {
1976        ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1977        return;
1978    }
1979
1980    size_t num = mAudioSessionRefs.size();
1981    for (size_t i = 0; i< num; i++) {
1982        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1983        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1984            ref->mCnt++;
1985            ALOGV(" incremented refcount to %d", ref->mCnt);
1986            return;
1987        }
1988    }
1989    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1990    ALOGV(" added new entry for %d", audioSession);
1991}
1992
1993void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
1994{
1995    Mutex::Autolock _l(mLock);
1996    pid_t caller = IPCThreadState::self()->getCallingPid();
1997    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
1998    if (pid != -1 && (caller == getpid_cached)) {
1999        caller = pid;
2000    }
2001    size_t num = mAudioSessionRefs.size();
2002    for (size_t i = 0; i< num; i++) {
2003        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
2004        if (ref->mSessionid == audioSession && ref->mPid == caller) {
2005            ref->mCnt--;
2006            ALOGV(" decremented refcount to %d", ref->mCnt);
2007            if (ref->mCnt == 0) {
2008                mAudioSessionRefs.removeAt(i);
2009                delete ref;
2010                purgeStaleEffects_l();
2011            }
2012            return;
2013        }
2014    }
2015    // If the caller is mediaserver it is likely that the session being released was acquired
2016    // on behalf of a process not in notification clients and we ignore the warning.
2017    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2018}
2019
2020void AudioFlinger::purgeStaleEffects_l() {
2021
2022    ALOGV("purging stale effects");
2023
2024    Vector< sp<EffectChain> > chains;
2025
2026    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2027        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2028        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2029            sp<EffectChain> ec = t->mEffectChains[j];
2030            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2031                chains.push(ec);
2032            }
2033        }
2034    }
2035    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2036        sp<RecordThread> t = mRecordThreads.valueAt(i);
2037        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2038            sp<EffectChain> ec = t->mEffectChains[j];
2039            chains.push(ec);
2040        }
2041    }
2042
2043    for (size_t i = 0; i < chains.size(); i++) {
2044        sp<EffectChain> ec = chains[i];
2045        int sessionid = ec->sessionId();
2046        sp<ThreadBase> t = ec->mThread.promote();
2047        if (t == 0) {
2048            continue;
2049        }
2050        size_t numsessionrefs = mAudioSessionRefs.size();
2051        bool found = false;
2052        for (size_t k = 0; k < numsessionrefs; k++) {
2053            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2054            if (ref->mSessionid == sessionid) {
2055                ALOGV(" session %d still exists for %d with %d refs",
2056                    sessionid, ref->mPid, ref->mCnt);
2057                found = true;
2058                break;
2059            }
2060        }
2061        if (!found) {
2062            Mutex::Autolock _l(t->mLock);
2063            // remove all effects from the chain
2064            while (ec->mEffects.size()) {
2065                sp<EffectModule> effect = ec->mEffects[0];
2066                effect->unPin();
2067                t->removeEffect_l(effect);
2068                if (effect->purgeHandles()) {
2069                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2070                }
2071                AudioSystem::unregisterEffect(effect->id());
2072            }
2073        }
2074    }
2075    return;
2076}
2077
2078// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2079AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2080{
2081    return mPlaybackThreads.valueFor(output).get();
2082}
2083
2084// checkMixerThread_l() must be called with AudioFlinger::mLock held
2085AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2086{
2087    PlaybackThread *thread = checkPlaybackThread_l(output);
2088    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2089}
2090
2091// checkRecordThread_l() must be called with AudioFlinger::mLock held
2092AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2093{
2094    return mRecordThreads.valueFor(input).get();
2095}
2096
2097uint32_t AudioFlinger::nextUniqueId()
2098{
2099    return android_atomic_inc(&mNextUniqueId);
2100}
2101
2102AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2103{
2104    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2105        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2106        AudioStreamOut *output = thread->getOutput();
2107        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2108            return thread;
2109        }
2110    }
2111    return NULL;
2112}
2113
2114audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2115{
2116    PlaybackThread *thread = primaryPlaybackThread_l();
2117
2118    if (thread == NULL) {
2119        return 0;
2120    }
2121
2122    return thread->outDevice();
2123}
2124
2125sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2126                                    int triggerSession,
2127                                    int listenerSession,
2128                                    sync_event_callback_t callBack,
2129                                    wp<RefBase> cookie)
2130{
2131    Mutex::Autolock _l(mLock);
2132
2133    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2134    status_t playStatus = NAME_NOT_FOUND;
2135    status_t recStatus = NAME_NOT_FOUND;
2136    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2137        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2138        if (playStatus == NO_ERROR) {
2139            return event;
2140        }
2141    }
2142    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2143        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2144        if (recStatus == NO_ERROR) {
2145            return event;
2146        }
2147    }
2148    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2149        mPendingSyncEvents.add(event);
2150    } else {
2151        ALOGV("createSyncEvent() invalid event %d", event->type());
2152        event.clear();
2153    }
2154    return event;
2155}
2156
2157// ----------------------------------------------------------------------------
2158//  Effect management
2159// ----------------------------------------------------------------------------
2160
2161
2162status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2163{
2164    Mutex::Autolock _l(mLock);
2165    return EffectQueryNumberEffects(numEffects);
2166}
2167
2168status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2169{
2170    Mutex::Autolock _l(mLock);
2171    return EffectQueryEffect(index, descriptor);
2172}
2173
2174status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2175        effect_descriptor_t *descriptor) const
2176{
2177    Mutex::Autolock _l(mLock);
2178    return EffectGetDescriptor(pUuid, descriptor);
2179}
2180
2181
2182sp<IEffect> AudioFlinger::createEffect(
2183        effect_descriptor_t *pDesc,
2184        const sp<IEffectClient>& effectClient,
2185        int32_t priority,
2186        audio_io_handle_t io,
2187        int sessionId,
2188        status_t *status,
2189        int *id,
2190        int *enabled)
2191{
2192    status_t lStatus = NO_ERROR;
2193    sp<EffectHandle> handle;
2194    effect_descriptor_t desc;
2195
2196    pid_t pid = IPCThreadState::self()->getCallingPid();
2197    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2198            pid, effectClient.get(), priority, sessionId, io);
2199
2200    if (pDesc == NULL) {
2201        lStatus = BAD_VALUE;
2202        goto Exit;
2203    }
2204
2205    // check audio settings permission for global effects
2206    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2207        lStatus = PERMISSION_DENIED;
2208        goto Exit;
2209    }
2210
2211    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2212    // that can only be created by audio policy manager (running in same process)
2213    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2214        lStatus = PERMISSION_DENIED;
2215        goto Exit;
2216    }
2217
2218    {
2219        if (!EffectIsNullUuid(&pDesc->uuid)) {
2220            // if uuid is specified, request effect descriptor
2221            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2222            if (lStatus < 0) {
2223                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2224                goto Exit;
2225            }
2226        } else {
2227            // if uuid is not specified, look for an available implementation
2228            // of the required type in effect factory
2229            if (EffectIsNullUuid(&pDesc->type)) {
2230                ALOGW("createEffect() no effect type");
2231                lStatus = BAD_VALUE;
2232                goto Exit;
2233            }
2234            uint32_t numEffects = 0;
2235            effect_descriptor_t d;
2236            d.flags = 0; // prevent compiler warning
2237            bool found = false;
2238
2239            lStatus = EffectQueryNumberEffects(&numEffects);
2240            if (lStatus < 0) {
2241                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2242                goto Exit;
2243            }
2244            for (uint32_t i = 0; i < numEffects; i++) {
2245                lStatus = EffectQueryEffect(i, &desc);
2246                if (lStatus < 0) {
2247                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2248                    continue;
2249                }
2250                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2251                    // If matching type found save effect descriptor. If the session is
2252                    // 0 and the effect is not auxiliary, continue enumeration in case
2253                    // an auxiliary version of this effect type is available
2254                    found = true;
2255                    d = desc;
2256                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2257                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2258                        break;
2259                    }
2260                }
2261            }
2262            if (!found) {
2263                lStatus = BAD_VALUE;
2264                ALOGW("createEffect() effect not found");
2265                goto Exit;
2266            }
2267            // For same effect type, chose auxiliary version over insert version if
2268            // connect to output mix (Compliance to OpenSL ES)
2269            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2270                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2271                desc = d;
2272            }
2273        }
2274
2275        // Do not allow auxiliary effects on a session different from 0 (output mix)
2276        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2277             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2278            lStatus = INVALID_OPERATION;
2279            goto Exit;
2280        }
2281
2282        // check recording permission for visualizer
2283        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2284            !recordingAllowed()) {
2285            lStatus = PERMISSION_DENIED;
2286            goto Exit;
2287        }
2288
2289        // return effect descriptor
2290        *pDesc = desc;
2291        if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2292            // if the output returned by getOutputForEffect() is removed before we lock the
2293            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2294            // and we will exit safely
2295            io = AudioSystem::getOutputForEffect(&desc);
2296            ALOGV("createEffect got output %d", io);
2297        }
2298
2299        Mutex::Autolock _l(mLock);
2300
2301        // If output is not specified try to find a matching audio session ID in one of the
2302        // output threads.
2303        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2304        // because of code checking output when entering the function.
2305        // Note: io is never 0 when creating an effect on an input
2306        if (io == 0) {
2307            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2308                // output must be specified by AudioPolicyManager when using session
2309                // AUDIO_SESSION_OUTPUT_STAGE
2310                lStatus = BAD_VALUE;
2311                goto Exit;
2312            }
2313            // look for the thread where the specified audio session is present
2314            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2315                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2316                    io = mPlaybackThreads.keyAt(i);
2317                    break;
2318                }
2319            }
2320            if (io == 0) {
2321                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2322                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2323                        io = mRecordThreads.keyAt(i);
2324                        break;
2325                    }
2326                }
2327            }
2328            // If no output thread contains the requested session ID, default to
2329            // first output. The effect chain will be moved to the correct output
2330            // thread when a track with the same session ID is created
2331            if (io == 0 && mPlaybackThreads.size()) {
2332                io = mPlaybackThreads.keyAt(0);
2333            }
2334            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2335        }
2336        ThreadBase *thread = checkRecordThread_l(io);
2337        if (thread == NULL) {
2338            thread = checkPlaybackThread_l(io);
2339            if (thread == NULL) {
2340                ALOGE("createEffect() unknown output thread");
2341                lStatus = BAD_VALUE;
2342                goto Exit;
2343            }
2344        }
2345
2346        sp<Client> client = registerPid_l(pid);
2347
2348        // create effect on selected output thread
2349        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2350                &desc, enabled, &lStatus);
2351        if (handle != 0 && id != NULL) {
2352            *id = handle->id();
2353        }
2354    }
2355
2356Exit:
2357    *status = lStatus;
2358    return handle;
2359}
2360
2361status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2362        audio_io_handle_t dstOutput)
2363{
2364    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2365            sessionId, srcOutput, dstOutput);
2366    Mutex::Autolock _l(mLock);
2367    if (srcOutput == dstOutput) {
2368        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2369        return NO_ERROR;
2370    }
2371    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2372    if (srcThread == NULL) {
2373        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2374        return BAD_VALUE;
2375    }
2376    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2377    if (dstThread == NULL) {
2378        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2379        return BAD_VALUE;
2380    }
2381
2382    Mutex::Autolock _dl(dstThread->mLock);
2383    Mutex::Autolock _sl(srcThread->mLock);
2384    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2385}
2386
2387// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2388status_t AudioFlinger::moveEffectChain_l(int sessionId,
2389                                   AudioFlinger::PlaybackThread *srcThread,
2390                                   AudioFlinger::PlaybackThread *dstThread,
2391                                   bool reRegister)
2392{
2393    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2394            sessionId, srcThread, dstThread);
2395
2396    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2397    if (chain == 0) {
2398        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2399                sessionId, srcThread);
2400        return INVALID_OPERATION;
2401    }
2402
2403    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2404    // so that a new chain is created with correct parameters when first effect is added. This is
2405    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2406    // removed.
2407    srcThread->removeEffectChain_l(chain);
2408
2409    // transfer all effects one by one so that new effect chain is created on new thread with
2410    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2411    sp<EffectChain> dstChain;
2412    uint32_t strategy = 0; // prevent compiler warning
2413    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2414    Vector< sp<EffectModule> > removed;
2415    status_t status = NO_ERROR;
2416    while (effect != 0) {
2417        srcThread->removeEffect_l(effect);
2418        removed.add(effect);
2419        status = dstThread->addEffect_l(effect);
2420        if (status != NO_ERROR) {
2421            break;
2422        }
2423        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2424        if (effect->state() == EffectModule::ACTIVE ||
2425                effect->state() == EffectModule::STOPPING) {
2426            effect->start();
2427        }
2428        // if the move request is not received from audio policy manager, the effect must be
2429        // re-registered with the new strategy and output
2430        if (dstChain == 0) {
2431            dstChain = effect->chain().promote();
2432            if (dstChain == 0) {
2433                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2434                status = NO_INIT;
2435                break;
2436            }
2437            strategy = dstChain->strategy();
2438        }
2439        if (reRegister) {
2440            AudioSystem::unregisterEffect(effect->id());
2441            AudioSystem::registerEffect(&effect->desc(),
2442                                        dstThread->id(),
2443                                        strategy,
2444                                        sessionId,
2445                                        effect->id());
2446            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2447        }
2448        effect = chain->getEffectFromId_l(0);
2449    }
2450
2451    if (status != NO_ERROR) {
2452        for (size_t i = 0; i < removed.size(); i++) {
2453            srcThread->addEffect_l(removed[i]);
2454            if (dstChain != 0 && reRegister) {
2455                AudioSystem::unregisterEffect(removed[i]->id());
2456                AudioSystem::registerEffect(&removed[i]->desc(),
2457                                            srcThread->id(),
2458                                            strategy,
2459                                            sessionId,
2460                                            removed[i]->id());
2461                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2462            }
2463        }
2464    }
2465
2466    return status;
2467}
2468
2469bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2470{
2471    if (mGlobalEffectEnableTime != 0 &&
2472            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2473        return true;
2474    }
2475
2476    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2477        sp<EffectChain> ec =
2478                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2479        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2480            return true;
2481        }
2482    }
2483    return false;
2484}
2485
2486void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2487{
2488    Mutex::Autolock _l(mLock);
2489
2490    mGlobalEffectEnableTime = systemTime();
2491
2492    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2493        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2494        if (t->mType == ThreadBase::OFFLOAD) {
2495            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2496        }
2497    }
2498
2499}
2500
2501struct Entry {
2502#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2503    char mName[MAX_NAME];
2504};
2505
2506int comparEntry(const void *p1, const void *p2)
2507{
2508    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2509}
2510
2511#ifdef TEE_SINK
2512void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2513{
2514    NBAIO_Source *teeSource = source.get();
2515    if (teeSource != NULL) {
2516        // .wav rotation
2517        // There is a benign race condition if 2 threads call this simultaneously.
2518        // They would both traverse the directory, but the result would simply be
2519        // failures at unlink() which are ignored.  It's also unlikely since
2520        // normally dumpsys is only done by bugreport or from the command line.
2521        char teePath[32+256];
2522        strcpy(teePath, "/data/misc/media");
2523        size_t teePathLen = strlen(teePath);
2524        DIR *dir = opendir(teePath);
2525        teePath[teePathLen++] = '/';
2526        if (dir != NULL) {
2527#define MAX_SORT 20 // number of entries to sort
2528#define MAX_KEEP 10 // number of entries to keep
2529            struct Entry entries[MAX_SORT];
2530            size_t entryCount = 0;
2531            while (entryCount < MAX_SORT) {
2532                struct dirent de;
2533                struct dirent *result = NULL;
2534                int rc = readdir_r(dir, &de, &result);
2535                if (rc != 0) {
2536                    ALOGW("readdir_r failed %d", rc);
2537                    break;
2538                }
2539                if (result == NULL) {
2540                    break;
2541                }
2542                if (result != &de) {
2543                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2544                    break;
2545                }
2546                // ignore non .wav file entries
2547                size_t nameLen = strlen(de.d_name);
2548                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2549                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2550                    continue;
2551                }
2552                strcpy(entries[entryCount++].mName, de.d_name);
2553            }
2554            (void) closedir(dir);
2555            if (entryCount > MAX_KEEP) {
2556                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2557                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2558                    strcpy(&teePath[teePathLen], entries[i].mName);
2559                    (void) unlink(teePath);
2560                }
2561            }
2562        } else {
2563            if (fd >= 0) {
2564                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2565            }
2566        }
2567        char teeTime[16];
2568        struct timeval tv;
2569        gettimeofday(&tv, NULL);
2570        struct tm tm;
2571        localtime_r(&tv.tv_sec, &tm);
2572        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2573        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2574        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2575        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2576        if (teeFd >= 0) {
2577            char wavHeader[44];
2578            memcpy(wavHeader,
2579                "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",
2580                sizeof(wavHeader));
2581            NBAIO_Format format = teeSource->format();
2582            unsigned channelCount = Format_channelCount(format);
2583            ALOG_ASSERT(channelCount <= FCC_2);
2584            uint32_t sampleRate = Format_sampleRate(format);
2585            wavHeader[22] = channelCount;       // number of channels
2586            wavHeader[24] = sampleRate;         // sample rate
2587            wavHeader[25] = sampleRate >> 8;
2588            wavHeader[32] = channelCount * 2;   // block alignment
2589            write(teeFd, wavHeader, sizeof(wavHeader));
2590            size_t total = 0;
2591            bool firstRead = true;
2592            for (;;) {
2593#define TEE_SINK_READ 1024
2594                short buffer[TEE_SINK_READ * FCC_2];
2595                size_t count = TEE_SINK_READ;
2596                ssize_t actual = teeSource->read(buffer, count,
2597                        AudioBufferProvider::kInvalidPTS);
2598                bool wasFirstRead = firstRead;
2599                firstRead = false;
2600                if (actual <= 0) {
2601                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2602                        continue;
2603                    }
2604                    break;
2605                }
2606                ALOG_ASSERT(actual <= (ssize_t)count);
2607                write(teeFd, buffer, actual * channelCount * sizeof(short));
2608                total += actual;
2609            }
2610            lseek(teeFd, (off_t) 4, SEEK_SET);
2611            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2612            write(teeFd, &temp, sizeof(temp));
2613            lseek(teeFd, (off_t) 40, SEEK_SET);
2614            temp =  total * channelCount * sizeof(short);
2615            write(teeFd, &temp, sizeof(temp));
2616            close(teeFd);
2617            if (fd >= 0) {
2618                fdprintf(fd, "tee copied to %s\n", teePath);
2619            }
2620        } else {
2621            if (fd >= 0) {
2622                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2623            }
2624        }
2625    }
2626}
2627#endif
2628
2629// ----------------------------------------------------------------------------
2630
2631status_t AudioFlinger::onTransact(
2632        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2633{
2634    return BnAudioFlinger::onTransact(code, data, reply, flags);
2635}
2636
2637}; // namespace android
2638