AudioFlinger.cpp revision 142f519aa1acd5804d111e60d100f170fed28405 
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      mAudioHwDevs(NULL),
163      mHardwareStatus(AUDIO_HW_IDLE),
164      mMasterVolume(1.0f),
165      mMasterMute(false),
166      mNextUniqueId(1),
167      mMode(AUDIO_MODE_INVALID),
168      mBtNrecIsOff(false),
169      mIsLowRamDevice(true),
170      mIsDeviceTypeKnown(false),
171      mGlobalEffectEnableTime(0)
172{
173    getpid_cached = getpid();
174    char value[PROPERTY_VALUE_MAX];
175    bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
176    if (doLog) {
177        mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
178    }
179#ifdef TEE_SINK
180    (void) property_get("ro.debuggable", value, "0");
181    int debuggable = atoi(value);
182    int teeEnabled = 0;
183    if (debuggable) {
184        (void) property_get("af.tee", value, "0");
185        teeEnabled = atoi(value);
186    }
187    // FIXME symbolic constants here
188    if (teeEnabled & 1) {
189        mTeeSinkInputEnabled = true;
190    }
191    if (teeEnabled & 2) {
192        mTeeSinkOutputEnabled = true;
193    }
194    if (teeEnabled & 4) {
195        mTeeSinkTrackEnabled = true;
196    }
197#endif
198}
199
200void AudioFlinger::onFirstRef()
201{
202    int rc = 0;
203
204    Mutex::Autolock _l(mLock);
205
206    /* TODO: move all this work into an Init() function */
207    char val_str[PROPERTY_VALUE_MAX] = { 0 };
208    if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
209        uint32_t int_val;
210        if (1 == sscanf(val_str, "%u", &int_val)) {
211            mStandbyTimeInNsecs = milliseconds(int_val);
212            ALOGI("Using %u mSec as standby time.", int_val);
213        } else {
214            mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
215            ALOGI("Using default %u mSec as standby time.",
216                    (uint32_t)(mStandbyTimeInNsecs / 1000000));
217        }
218    }
219
220    mMode = AUDIO_MODE_NORMAL;
221}
222
223AudioFlinger::~AudioFlinger()
224{
225    while (!mRecordThreads.isEmpty()) {
226        // closeInput_nonvirtual() will remove specified entry from mRecordThreads
227        closeInput_nonvirtual(mRecordThreads.keyAt(0));
228    }
229    while (!mPlaybackThreads.isEmpty()) {
230        // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
231        closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
232    }
233
234    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
235        // no mHardwareLock needed, as there are no other references to this
236        audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
237        delete mAudioHwDevs.valueAt(i);
238    }
239
240    // Tell media.log service about any old writers that still need to be unregistered
241    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
242    if (binder != 0) {
243        sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
244        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
245            sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
246            mUnregisteredWriters.pop();
247            mediaLogService->unregisterWriter(iMemory);
248        }
249    }
250
251}
252
253static const char * const audio_interfaces[] = {
254    AUDIO_HARDWARE_MODULE_ID_PRIMARY,
255    AUDIO_HARDWARE_MODULE_ID_A2DP,
256    AUDIO_HARDWARE_MODULE_ID_USB,
257};
258#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
259
260AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
261        audio_module_handle_t module,
262        audio_devices_t devices)
263{
264    // if module is 0, the request comes from an old policy manager and we should load
265    // well known modules
266    if (module == 0) {
267        ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
268        for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
269            loadHwModule_l(audio_interfaces[i]);
270        }
271        // then try to find a module supporting the requested device.
272        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
273            AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
274            audio_hw_device_t *dev = audioHwDevice->hwDevice();
275            if ((dev->get_supported_devices != NULL) &&
276                    (dev->get_supported_devices(dev) & devices) == devices)
277                return audioHwDevice;
278        }
279    } else {
280        // check a match for the requested module handle
281        AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
282        if (audioHwDevice != NULL) {
283            return audioHwDevice;
284        }
285    }
286
287    return NULL;
288}
289
290void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
291{
292    const size_t SIZE = 256;
293    char buffer[SIZE];
294    String8 result;
295
296    result.append("Clients:\n");
297    for (size_t i = 0; i < mClients.size(); ++i) {
298        sp<Client> client = mClients.valueAt(i).promote();
299        if (client != 0) {
300            snprintf(buffer, SIZE, "  pid: %d\n", client->pid());
301            result.append(buffer);
302        }
303    }
304
305    result.append("Notification Clients:\n");
306    for (size_t i = 0; i < mNotificationClients.size(); ++i) {
307        snprintf(buffer, SIZE, "  pid: %d\n", mNotificationClients.keyAt(i));
308        result.append(buffer);
309    }
310
311    result.append("Global session refs:\n");
312    result.append("  session   pid count\n");
313    for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
314        AudioSessionRef *r = mAudioSessionRefs[i];
315        snprintf(buffer, SIZE, "  %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
316        result.append(buffer);
317    }
318    write(fd, result.string(), result.size());
319}
320
321
322void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
323{
324    const size_t SIZE = 256;
325    char buffer[SIZE];
326    String8 result;
327    hardware_call_state hardwareStatus = mHardwareStatus;
328
329    snprintf(buffer, SIZE, "Hardware status: %d\n"
330                           "Standby Time mSec: %u\n",
331                            hardwareStatus,
332                            (uint32_t)(mStandbyTimeInNsecs / 1000000));
333    result.append(buffer);
334    write(fd, result.string(), result.size());
335}
336
337void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
338{
339    const size_t SIZE = 256;
340    char buffer[SIZE];
341    String8 result;
342    snprintf(buffer, SIZE, "Permission Denial: "
343            "can't dump AudioFlinger from pid=%d, uid=%d\n",
344            IPCThreadState::self()->getCallingPid(),
345            IPCThreadState::self()->getCallingUid());
346    result.append(buffer);
347    write(fd, result.string(), result.size());
348}
349
350bool AudioFlinger::dumpTryLock(Mutex& mutex)
351{
352    bool locked = false;
353    for (int i = 0; i < kDumpLockRetries; ++i) {
354        if (mutex.tryLock() == NO_ERROR) {
355            locked = true;
356            break;
357        }
358        usleep(kDumpLockSleepUs);
359    }
360    return locked;
361}
362
363status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
364{
365    if (!dumpAllowed()) {
366        dumpPermissionDenial(fd, args);
367    } else {
368        // get state of hardware lock
369        bool hardwareLocked = dumpTryLock(mHardwareLock);
370        if (!hardwareLocked) {
371            String8 result(kHardwareLockedString);
372            write(fd, result.string(), result.size());
373        } else {
374            mHardwareLock.unlock();
375        }
376
377        bool locked = dumpTryLock(mLock);
378
379        // failed to lock - AudioFlinger is probably deadlocked
380        if (!locked) {
381            String8 result(kDeadlockedString);
382            write(fd, result.string(), result.size());
383        }
384
385        dumpClients(fd, args);
386        dumpInternals(fd, args);
387
388        // dump playback threads
389        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
390            mPlaybackThreads.valueAt(i)->dump(fd, args);
391        }
392
393        // dump record threads
394        for (size_t i = 0; i < mRecordThreads.size(); i++) {
395            mRecordThreads.valueAt(i)->dump(fd, args);
396        }
397
398        // dump all hardware devs
399        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
400            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
401            dev->dump(dev, fd);
402        }
403
404#ifdef TEE_SINK
405        // dump the serially shared record tee sink
406        if (mRecordTeeSource != 0) {
407            dumpTee(fd, mRecordTeeSource);
408        }
409#endif
410
411        if (locked) {
412            mLock.unlock();
413        }
414
415        // append a copy of media.log here by forwarding fd to it, but don't attempt
416        // to lookup the service if it's not running, as it will block for a second
417        if (mLogMemoryDealer != 0) {
418            sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
419            if (binder != 0) {
420                fdprintf(fd, "\nmedia.log:\n");
421                Vector<String16> args;
422                binder->dump(fd, args);
423            }
424        }
425    }
426    return NO_ERROR;
427}
428
429sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
430{
431    // If pid is already in the mClients wp<> map, then use that entry
432    // (for which promote() is always != 0), otherwise create a new entry and Client.
433    sp<Client> client = mClients.valueFor(pid).promote();
434    if (client == 0) {
435        client = new Client(this, pid);
436        mClients.add(pid, client);
437    }
438
439    return client;
440}
441
442sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
443{
444    // If there is no memory allocated for logs, return a dummy writer that does nothing
445    if (mLogMemoryDealer == 0) {
446        return new NBLog::Writer();
447    }
448    sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
449    // Similarly if we can't contact the media.log service, also return a dummy writer
450    if (binder == 0) {
451        return new NBLog::Writer();
452    }
453    sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
454    sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
455    // If allocation fails, consult the vector of previously unregistered writers
456    // and garbage-collect one or more them until an allocation succeeds
457    if (shared == 0) {
458        Mutex::Autolock _l(mUnregisteredWritersLock);
459        for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
460            {
461                // Pick the oldest stale writer to garbage-collect
462                sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
463                mUnregisteredWriters.removeAt(0);
464                mediaLogService->unregisterWriter(iMemory);
465                // Now the media.log remote reference to IMemory is gone.  When our last local
466                // reference to IMemory also drops to zero at end of this block,
467                // the IMemory destructor will deallocate the region from mLogMemoryDealer.
468            }
469            // Re-attempt the allocation
470            shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
471            if (shared != 0) {
472                goto success;
473            }
474        }
475        // Even after garbage-collecting all old writers, there is still not enough memory,
476        // so return a dummy writer
477        return new NBLog::Writer();
478    }
479success:
480    mediaLogService->registerWriter(shared, size, name);
481    return new NBLog::Writer(size, shared);
482}
483
484void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
485{
486    if (writer == 0) {
487        return;
488    }
489    sp<IMemory> iMemory(writer->getIMemory());
490    if (iMemory == 0) {
491        return;
492    }
493    // Rather than removing the writer immediately, append it to a queue of old writers to
494    // be garbage-collected later.  This allows us to continue to view old logs for a while.
495    Mutex::Autolock _l(mUnregisteredWritersLock);
496    mUnregisteredWriters.push(writer);
497}
498
499// IAudioFlinger interface
500
501
502sp<IAudioTrack> AudioFlinger::createTrack(
503        audio_stream_type_t streamType,
504        uint32_t sampleRate,
505        audio_format_t format,
506        audio_channel_mask_t channelMask,
507        size_t *frameCount,
508        IAudioFlinger::track_flags_t *flags,
509        const sp<IMemory>& sharedBuffer,
510        audio_io_handle_t output,
511        pid_t tid,
512        int *sessionId,
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        if (thread == NULL) {
563            ALOGE("no playback thread found for output handle %d", output);
564            lStatus = BAD_VALUE;
565            goto Exit;
566        }
567
568        pid_t pid = IPCThreadState::self()->getCallingPid();
569        client = registerPid_l(pid);
570
571        PlaybackThread *effectThread = NULL;
572        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
573            lSessionId = *sessionId;
574            // check if an effect chain with the same session ID is present on another
575            // output thread and move it here.
576            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
577                sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
578                if (mPlaybackThreads.keyAt(i) != output) {
579                    uint32_t sessions = t->hasAudioSession(lSessionId);
580                    if (sessions & PlaybackThread::EFFECT_SESSION) {
581                        effectThread = t.get();
582                        break;
583                    }
584                }
585            }
586        } else {
587            // if no audio session id is provided, create one here
588            lSessionId = nextUniqueId();
589            if (sessionId != NULL) {
590                *sessionId = lSessionId;
591            }
592        }
593        ALOGV("createTrack() lSessionId: %d", lSessionId);
594
595        track = thread->createTrack_l(client, streamType, sampleRate, format,
596                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
597        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
598        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
599
600        // move effect chain to this output thread if an effect on same session was waiting
601        // for a track to be created
602        if (lStatus == NO_ERROR && effectThread != NULL) {
603            // no risk of deadlock because AudioFlinger::mLock is held
604            Mutex::Autolock _dl(thread->mLock);
605            Mutex::Autolock _sl(effectThread->mLock);
606            moveEffectChain_l(lSessionId, effectThread, thread, true);
607        }
608
609        // Look for sync events awaiting for a session to be used.
610        for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
611            if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
612                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
613                    if (lStatus == NO_ERROR) {
614                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
615                    } else {
616                        mPendingSyncEvents[i]->cancel();
617                    }
618                    mPendingSyncEvents.removeAt(i);
619                    i--;
620                }
621            }
622        }
623
624    }
625
626    if (lStatus != NO_ERROR) {
627        // remove local strong reference to Client before deleting the Track so that the
628        // Client destructor is called by the TrackBase destructor with mLock held
629        client.clear();
630        track.clear();
631        goto Exit;
632    }
633
634    // return handle to client
635    trackHandle = new TrackHandle(track);
636
637Exit:
638    *status = lStatus;
639    return trackHandle;
640}
641
642uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
643{
644    Mutex::Autolock _l(mLock);
645    PlaybackThread *thread = checkPlaybackThread_l(output);
646    if (thread == NULL) {
647        ALOGW("sampleRate() unknown thread %d", output);
648        return 0;
649    }
650    return thread->sampleRate();
651}
652
653int AudioFlinger::channelCount(audio_io_handle_t output) const
654{
655    Mutex::Autolock _l(mLock);
656    PlaybackThread *thread = checkPlaybackThread_l(output);
657    if (thread == NULL) {
658        ALOGW("channelCount() unknown thread %d", output);
659        return 0;
660    }
661    return thread->channelCount();
662}
663
664audio_format_t AudioFlinger::format(audio_io_handle_t output) const
665{
666    Mutex::Autolock _l(mLock);
667    PlaybackThread *thread = checkPlaybackThread_l(output);
668    if (thread == NULL) {
669        ALOGW("format() unknown thread %d", output);
670        return AUDIO_FORMAT_INVALID;
671    }
672    return thread->format();
673}
674
675size_t AudioFlinger::frameCount(audio_io_handle_t output) const
676{
677    Mutex::Autolock _l(mLock);
678    PlaybackThread *thread = checkPlaybackThread_l(output);
679    if (thread == NULL) {
680        ALOGW("frameCount() unknown thread %d", output);
681        return 0;
682    }
683    // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
684    //       should examine all callers and fix them to handle smaller counts
685    return thread->frameCount();
686}
687
688uint32_t AudioFlinger::latency(audio_io_handle_t output) const
689{
690    Mutex::Autolock _l(mLock);
691    PlaybackThread *thread = checkPlaybackThread_l(output);
692    if (thread == NULL) {
693        ALOGW("latency(): no playback thread found for output handle %d", output);
694        return 0;
695    }
696    return thread->latency();
697}
698
699status_t AudioFlinger::setMasterVolume(float value)
700{
701    status_t ret = initCheck();
702    if (ret != NO_ERROR) {
703        return ret;
704    }
705
706    // check calling permissions
707    if (!settingsAllowed()) {
708        return PERMISSION_DENIED;
709    }
710
711    Mutex::Autolock _l(mLock);
712    mMasterVolume = value;
713
714    // Set master volume in the HALs which support it.
715    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
716        AutoMutex lock(mHardwareLock);
717        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
718
719        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
720        if (dev->canSetMasterVolume()) {
721            dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
722        }
723        mHardwareStatus = AUDIO_HW_IDLE;
724    }
725
726    // Now set the master volume in each playback thread.  Playback threads
727    // assigned to HALs which do not have master volume support will apply
728    // master volume during the mix operation.  Threads with HALs which do
729    // support master volume will simply ignore the setting.
730    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
731        mPlaybackThreads.valueAt(i)->setMasterVolume(value);
732
733    return NO_ERROR;
734}
735
736status_t AudioFlinger::setMode(audio_mode_t mode)
737{
738    status_t ret = initCheck();
739    if (ret != NO_ERROR) {
740        return ret;
741    }
742
743    // check calling permissions
744    if (!settingsAllowed()) {
745        return PERMISSION_DENIED;
746    }
747    if (uint32_t(mode) >= AUDIO_MODE_CNT) {
748        ALOGW("Illegal value: setMode(%d)", mode);
749        return BAD_VALUE;
750    }
751
752    { // scope for the lock
753        AutoMutex lock(mHardwareLock);
754        audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
755        mHardwareStatus = AUDIO_HW_SET_MODE;
756        ret = dev->set_mode(dev, mode);
757        mHardwareStatus = AUDIO_HW_IDLE;
758    }
759
760    if (NO_ERROR == ret) {
761        Mutex::Autolock _l(mLock);
762        mMode = mode;
763        for (size_t i = 0; i < mPlaybackThreads.size(); i++)
764            mPlaybackThreads.valueAt(i)->setMode(mode);
765    }
766
767    return ret;
768}
769
770status_t AudioFlinger::setMicMute(bool state)
771{
772    status_t ret = initCheck();
773    if (ret != NO_ERROR) {
774        return ret;
775    }
776
777    // check calling permissions
778    if (!settingsAllowed()) {
779        return PERMISSION_DENIED;
780    }
781
782    AutoMutex lock(mHardwareLock);
783    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
784    mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
785    ret = dev->set_mic_mute(dev, state);
786    mHardwareStatus = AUDIO_HW_IDLE;
787    return ret;
788}
789
790bool AudioFlinger::getMicMute() const
791{
792    status_t ret = initCheck();
793    if (ret != NO_ERROR) {
794        return false;
795    }
796
797    bool state = AUDIO_MODE_INVALID;
798    AutoMutex lock(mHardwareLock);
799    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
800    mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
801    dev->get_mic_mute(dev, &state);
802    mHardwareStatus = AUDIO_HW_IDLE;
803    return state;
804}
805
806status_t AudioFlinger::setMasterMute(bool muted)
807{
808    status_t ret = initCheck();
809    if (ret != NO_ERROR) {
810        return ret;
811    }
812
813    // check calling permissions
814    if (!settingsAllowed()) {
815        return PERMISSION_DENIED;
816    }
817
818    Mutex::Autolock _l(mLock);
819    mMasterMute = muted;
820
821    // Set master mute in the HALs which support it.
822    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
823        AutoMutex lock(mHardwareLock);
824        AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
825
826        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
827        if (dev->canSetMasterMute()) {
828            dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
829        }
830        mHardwareStatus = AUDIO_HW_IDLE;
831    }
832
833    // Now set the master mute in each playback thread.  Playback threads
834    // assigned to HALs which do not have master mute support will apply master
835    // mute during the mix operation.  Threads with HALs which do support master
836    // mute will simply ignore the setting.
837    for (size_t i = 0; i < mPlaybackThreads.size(); i++)
838        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
839
840    return NO_ERROR;
841}
842
843float AudioFlinger::masterVolume() const
844{
845    Mutex::Autolock _l(mLock);
846    return masterVolume_l();
847}
848
849bool AudioFlinger::masterMute() const
850{
851    Mutex::Autolock _l(mLock);
852    return masterMute_l();
853}
854
855float AudioFlinger::masterVolume_l() const
856{
857    return mMasterVolume;
858}
859
860bool AudioFlinger::masterMute_l() const
861{
862    return mMasterMute;
863}
864
865status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
866        audio_io_handle_t output)
867{
868    // check calling permissions
869    if (!settingsAllowed()) {
870        return PERMISSION_DENIED;
871    }
872
873    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
874        ALOGE("setStreamVolume() invalid stream %d", stream);
875        return BAD_VALUE;
876    }
877
878    AutoMutex lock(mLock);
879    PlaybackThread *thread = NULL;
880    if (output != AUDIO_IO_HANDLE_NONE) {
881        thread = checkPlaybackThread_l(output);
882        if (thread == NULL) {
883            return BAD_VALUE;
884        }
885    }
886
887    mStreamTypes[stream].volume = value;
888
889    if (thread == NULL) {
890        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
891            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
892        }
893    } else {
894        thread->setStreamVolume(stream, value);
895    }
896
897    return NO_ERROR;
898}
899
900status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
901{
902    // check calling permissions
903    if (!settingsAllowed()) {
904        return PERMISSION_DENIED;
905    }
906
907    if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
908        uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
909        ALOGE("setStreamMute() invalid stream %d", stream);
910        return BAD_VALUE;
911    }
912
913    AutoMutex lock(mLock);
914    mStreamTypes[stream].mute = muted;
915    for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
916        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
917
918    return NO_ERROR;
919}
920
921float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
922{
923    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
924        return 0.0f;
925    }
926
927    AutoMutex lock(mLock);
928    float volume;
929    if (output != AUDIO_IO_HANDLE_NONE) {
930        PlaybackThread *thread = checkPlaybackThread_l(output);
931        if (thread == NULL) {
932            return 0.0f;
933        }
934        volume = thread->streamVolume(stream);
935    } else {
936        volume = streamVolume_l(stream);
937    }
938
939    return volume;
940}
941
942bool AudioFlinger::streamMute(audio_stream_type_t stream) const
943{
944    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
945        return true;
946    }
947
948    AutoMutex lock(mLock);
949    return streamMute_l(stream);
950}
951
952status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
953{
954    ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
955            ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
956
957    // check calling permissions
958    if (!settingsAllowed()) {
959        return PERMISSION_DENIED;
960    }
961
962    // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
963    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
964        Mutex::Autolock _l(mLock);
965        status_t final_result = NO_ERROR;
966        {
967            AutoMutex lock(mHardwareLock);
968            mHardwareStatus = AUDIO_HW_SET_PARAMETER;
969            for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
970                audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
971                status_t result = dev->set_parameters(dev, keyValuePairs.string());
972                final_result = result ?: final_result;
973            }
974            mHardwareStatus = AUDIO_HW_IDLE;
975        }
976        // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
977        AudioParameter param = AudioParameter(keyValuePairs);
978        String8 value;
979        if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
980            bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
981            if (mBtNrecIsOff != btNrecIsOff) {
982                for (size_t i = 0; i < mRecordThreads.size(); i++) {
983                    sp<RecordThread> thread = mRecordThreads.valueAt(i);
984                    audio_devices_t device = thread->inDevice();
985                    bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
986                    // collect all of the thread's session IDs
987                    KeyedVector<int, bool> ids = thread->sessionIds();
988                    // suspend effects associated with those session IDs
989                    for (size_t j = 0; j < ids.size(); ++j) {
990                        int sessionId = ids.keyAt(j);
991                        thread->setEffectSuspended(FX_IID_AEC,
992                                                   suspend,
993                                                   sessionId);
994                        thread->setEffectSuspended(FX_IID_NS,
995                                                   suspend,
996                                                   sessionId);
997                    }
998                }
999                mBtNrecIsOff = btNrecIsOff;
1000            }
1001        }
1002        String8 screenState;
1003        if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
1004            bool isOff = screenState == "off";
1005            if (isOff != (AudioFlinger::mScreenState & 1)) {
1006                AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
1007            }
1008        }
1009        return final_result;
1010    }
1011
1012    // hold a strong ref on thread in case closeOutput() or closeInput() is called
1013    // and the thread is exited once the lock is released
1014    sp<ThreadBase> thread;
1015    {
1016        Mutex::Autolock _l(mLock);
1017        thread = checkPlaybackThread_l(ioHandle);
1018        if (thread == 0) {
1019            thread = checkRecordThread_l(ioHandle);
1020        } else if (thread == primaryPlaybackThread_l()) {
1021            // indicate output device change to all input threads for pre processing
1022            AudioParameter param = AudioParameter(keyValuePairs);
1023            int value;
1024            if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
1025                    (value != 0)) {
1026                for (size_t i = 0; i < mRecordThreads.size(); i++) {
1027                    mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
1028                }
1029            }
1030        }
1031    }
1032    if (thread != 0) {
1033        return thread->setParameters(keyValuePairs);
1034    }
1035    return BAD_VALUE;
1036}
1037
1038String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
1039{
1040    ALOGVV("getParameters() io %d, keys %s, calling pid %d",
1041            ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
1042
1043    Mutex::Autolock _l(mLock);
1044
1045    if (ioHandle == AUDIO_IO_HANDLE_NONE) {
1046        String8 out_s8;
1047
1048        for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1049            char *s;
1050            {
1051            AutoMutex lock(mHardwareLock);
1052            mHardwareStatus = AUDIO_HW_GET_PARAMETER;
1053            audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
1054            s = dev->get_parameters(dev, keys.string());
1055            mHardwareStatus = AUDIO_HW_IDLE;
1056            }
1057            out_s8 += String8(s ? s : "");
1058            free(s);
1059        }
1060        return out_s8;
1061    }
1062
1063    PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
1064    if (playbackThread != NULL) {
1065        return playbackThread->getParameters(keys);
1066    }
1067    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1068    if (recordThread != NULL) {
1069        return recordThread->getParameters(keys);
1070    }
1071    return String8("");
1072}
1073
1074size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
1075        audio_channel_mask_t channelMask) const
1076{
1077    status_t ret = initCheck();
1078    if (ret != NO_ERROR) {
1079        return 0;
1080    }
1081
1082    AutoMutex lock(mHardwareLock);
1083    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
1084    struct audio_config config;
1085    memset(&config, 0, sizeof(config));
1086    config.sample_rate = sampleRate;
1087    config.channel_mask = channelMask;
1088    config.format = format;
1089
1090    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1091    size_t size = dev->get_input_buffer_size(dev, &config);
1092    mHardwareStatus = AUDIO_HW_IDLE;
1093    return size;
1094}
1095
1096uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
1097{
1098    Mutex::Autolock _l(mLock);
1099
1100    RecordThread *recordThread = checkRecordThread_l(ioHandle);
1101    if (recordThread != NULL) {
1102        return recordThread->getInputFramesLost();
1103    }
1104    return 0;
1105}
1106
1107status_t AudioFlinger::setVoiceVolume(float value)
1108{
1109    status_t ret = initCheck();
1110    if (ret != NO_ERROR) {
1111        return ret;
1112    }
1113
1114    // check calling permissions
1115    if (!settingsAllowed()) {
1116        return PERMISSION_DENIED;
1117    }
1118
1119    AutoMutex lock(mHardwareLock);
1120    audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
1121    mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
1122    ret = dev->set_voice_volume(dev, value);
1123    mHardwareStatus = AUDIO_HW_IDLE;
1124
1125    return ret;
1126}
1127
1128status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
1129        audio_io_handle_t output) const
1130{
1131    status_t status;
1132
1133    Mutex::Autolock _l(mLock);
1134
1135    PlaybackThread *playbackThread = checkPlaybackThread_l(output);
1136    if (playbackThread != NULL) {
1137        return playbackThread->getRenderPosition(halFrames, dspFrames);
1138    }
1139
1140    return BAD_VALUE;
1141}
1142
1143void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
1144{
1145
1146    Mutex::Autolock _l(mLock);
1147
1148    pid_t pid = IPCThreadState::self()->getCallingPid();
1149    if (mNotificationClients.indexOfKey(pid) < 0) {
1150        sp<NotificationClient> notificationClient = new NotificationClient(this,
1151                                                                            client,
1152                                                                            pid);
1153        ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
1154
1155        mNotificationClients.add(pid, notificationClient);
1156
1157        sp<IBinder> binder = client->asBinder();
1158        binder->linkToDeath(notificationClient);
1159
1160        // the config change is always sent from playback or record threads to avoid deadlock
1161        // with AudioSystem::gLock
1162        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1163            mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
1164        }
1165
1166        for (size_t i = 0; i < mRecordThreads.size(); i++) {
1167            mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
1168        }
1169    }
1170}
1171
1172void AudioFlinger::removeNotificationClient(pid_t pid)
1173{
1174    Mutex::Autolock _l(mLock);
1175
1176    mNotificationClients.removeItem(pid);
1177
1178    ALOGV("%d died, releasing its sessions", pid);
1179    size_t num = mAudioSessionRefs.size();
1180    bool removed = false;
1181    for (size_t i = 0; i< num; ) {
1182        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1183        ALOGV(" pid %d @ %d", ref->mPid, i);
1184        if (ref->mPid == pid) {
1185            ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
1186            mAudioSessionRefs.removeAt(i);
1187            delete ref;
1188            removed = true;
1189            num--;
1190        } else {
1191            i++;
1192        }
1193    }
1194    if (removed) {
1195        purgeStaleEffects_l();
1196    }
1197}
1198
1199// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1200void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
1201{
1202    size_t size = mNotificationClients.size();
1203    for (size_t i = 0; i < size; i++) {
1204        mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
1205                                                                               param2);
1206    }
1207}
1208
1209// removeClient_l() must be called with AudioFlinger::mLock held
1210void AudioFlinger::removeClient_l(pid_t pid)
1211{
1212    ALOGV("removeClient_l() pid %d, calling pid %d", pid,
1213            IPCThreadState::self()->getCallingPid());
1214    mClients.removeItem(pid);
1215}
1216
1217// getEffectThread_l() must be called with AudioFlinger::mLock held
1218sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
1219{
1220    sp<PlaybackThread> thread;
1221
1222    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1223        if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
1224            ALOG_ASSERT(thread == 0);
1225            thread = mPlaybackThreads.valueAt(i);
1226        }
1227    }
1228
1229    return thread;
1230}
1231
1232
1233
1234// ----------------------------------------------------------------------------
1235
1236AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
1237    :   RefBase(),
1238        mAudioFlinger(audioFlinger),
1239        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
1240        mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
1241        mPid(pid),
1242        mTimedTrackCount(0)
1243{
1244    // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
1245}
1246
1247// Client destructor must be called with AudioFlinger::mLock held
1248AudioFlinger::Client::~Client()
1249{
1250    mAudioFlinger->removeClient_l(mPid);
1251}
1252
1253sp<MemoryDealer> AudioFlinger::Client::heap() const
1254{
1255    return mMemoryDealer;
1256}
1257
1258// Reserve one of the limited slots for a timed audio track associated
1259// with this client
1260bool AudioFlinger::Client::reserveTimedTrack()
1261{
1262    const int kMaxTimedTracksPerClient = 4;
1263
1264    Mutex::Autolock _l(mTimedTrackLock);
1265
1266    if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
1267        ALOGW("can not create timed track - pid %d has exceeded the limit",
1268             mPid);
1269        return false;
1270    }
1271
1272    mTimedTrackCount++;
1273    return true;
1274}
1275
1276// Release a slot for a timed audio track
1277void AudioFlinger::Client::releaseTimedTrack()
1278{
1279    Mutex::Autolock _l(mTimedTrackLock);
1280    mTimedTrackCount--;
1281}
1282
1283// ----------------------------------------------------------------------------
1284
1285AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
1286                                                     const sp<IAudioFlingerClient>& client,
1287                                                     pid_t pid)
1288    : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
1289{
1290}
1291
1292AudioFlinger::NotificationClient::~NotificationClient()
1293{
1294}
1295
1296void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
1297{
1298    sp<NotificationClient> keep(this);
1299    mAudioFlinger->removeNotificationClient(mPid);
1300}
1301
1302
1303// ----------------------------------------------------------------------------
1304
1305static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) {
1306    return audio_is_remote_submix_device(inDevice);
1307}
1308
1309sp<IAudioRecord> AudioFlinger::openRecord(
1310        audio_io_handle_t input,
1311        uint32_t sampleRate,
1312        audio_format_t format,
1313        audio_channel_mask_t channelMask,
1314        size_t *frameCount,
1315        IAudioFlinger::track_flags_t *flags,
1316        pid_t tid,
1317        int *sessionId,
1318        status_t *status)
1319{
1320    sp<RecordThread::RecordTrack> recordTrack;
1321    sp<RecordHandle> recordHandle;
1322    sp<Client> client;
1323    status_t lStatus;
1324    int lSessionId;
1325
1326    // check calling permissions
1327    if (!recordingAllowed()) {
1328        ALOGE("openRecord() permission denied: recording not allowed");
1329        lStatus = PERMISSION_DENIED;
1330        goto Exit;
1331    }
1332
1333    // further sample rate checks are performed by createRecordTrack_l()
1334    if (sampleRate == 0) {
1335        ALOGE("openRecord() invalid sample rate %u", sampleRate);
1336        lStatus = BAD_VALUE;
1337        goto Exit;
1338    }
1339
1340    // we don't yet support anything other than 16-bit PCM
1341    if (!(audio_is_valid_format(format) &&
1342            audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) {
1343        ALOGE("openRecord() invalid format %#x", format);
1344        lStatus = BAD_VALUE;
1345        goto Exit;
1346    }
1347
1348    // further channel mask checks are performed by createRecordTrack_l()
1349    if (!audio_is_input_channel(channelMask)) {
1350        ALOGE("openRecord() invalid channel mask %#x", channelMask);
1351        lStatus = BAD_VALUE;
1352        goto Exit;
1353    }
1354
1355    {
1356        Mutex::Autolock _l(mLock);
1357        RecordThread *thread = checkRecordThread_l(input);
1358        if (thread == NULL) {
1359            ALOGE("openRecord() checkRecordThread_l failed");
1360            lStatus = BAD_VALUE;
1361            goto Exit;
1362        }
1363
1364        if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice())
1365                && !captureAudioOutputAllowed()) {
1366            ALOGE("openRecord() permission denied: capture not allowed");
1367            lStatus = PERMISSION_DENIED;
1368            goto Exit;
1369        }
1370
1371        pid_t pid = IPCThreadState::self()->getCallingPid();
1372        client = registerPid_l(pid);
1373
1374        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
1375            lSessionId = *sessionId;
1376        } else {
1377            // if no audio session id is provided, create one here
1378            lSessionId = nextUniqueId();
1379            if (sessionId != NULL) {
1380                *sessionId = lSessionId;
1381            }
1382        }
1383        ALOGV("openRecord() lSessionId: %d", lSessionId);
1384
1385        // TODO: the uid should be passed in as a parameter to openRecord
1386        recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
1387                                                  frameCount, lSessionId,
1388                                                  IPCThreadState::self()->getCallingUid(),
1389                                                  flags, tid, &lStatus);
1390        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
1391    }
1392
1393    if (lStatus != NO_ERROR) {
1394        // remove local strong reference to Client before deleting the RecordTrack so that the
1395        // Client destructor is called by the TrackBase destructor with mLock held
1396        client.clear();
1397        recordTrack.clear();
1398        goto Exit;
1399    }
1400
1401    // return handle to client
1402    recordHandle = new RecordHandle(recordTrack);
1403
1404Exit:
1405    *status = lStatus;
1406    return recordHandle;
1407}
1408
1409
1410
1411// ----------------------------------------------------------------------------
1412
1413audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
1414{
1415    if (!settingsAllowed()) {
1416        return 0;
1417    }
1418    Mutex::Autolock _l(mLock);
1419    return loadHwModule_l(name);
1420}
1421
1422// loadHwModule_l() must be called with AudioFlinger::mLock held
1423audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
1424{
1425    for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
1426        if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
1427            ALOGW("loadHwModule() module %s already loaded", name);
1428            return mAudioHwDevs.keyAt(i);
1429        }
1430    }
1431
1432    audio_hw_device_t *dev;
1433
1434    int rc = load_audio_interface(name, &dev);
1435    if (rc) {
1436        ALOGI("loadHwModule() error %d loading module %s ", rc, name);
1437        return 0;
1438    }
1439
1440    mHardwareStatus = AUDIO_HW_INIT;
1441    rc = dev->init_check(dev);
1442    mHardwareStatus = AUDIO_HW_IDLE;
1443    if (rc) {
1444        ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
1445        return 0;
1446    }
1447
1448    // Check and cache this HAL's level of support for master mute and master
1449    // volume.  If this is the first HAL opened, and it supports the get
1450    // methods, use the initial values provided by the HAL as the current
1451    // master mute and volume settings.
1452
1453    AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
1454    {  // scope for auto-lock pattern
1455        AutoMutex lock(mHardwareLock);
1456
1457        if (0 == mAudioHwDevs.size()) {
1458            mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
1459            if (NULL != dev->get_master_volume) {
1460                float mv;
1461                if (OK == dev->get_master_volume(dev, &mv)) {
1462                    mMasterVolume = mv;
1463                }
1464            }
1465
1466            mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
1467            if (NULL != dev->get_master_mute) {
1468                bool mm;
1469                if (OK == dev->get_master_mute(dev, &mm)) {
1470                    mMasterMute = mm;
1471                }
1472            }
1473        }
1474
1475        mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
1476        if ((NULL != dev->set_master_volume) &&
1477            (OK == dev->set_master_volume(dev, mMasterVolume))) {
1478            flags = static_cast<AudioHwDevice::Flags>(flags |
1479                    AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
1480        }
1481
1482        mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
1483        if ((NULL != dev->set_master_mute) &&
1484            (OK == dev->set_master_mute(dev, mMasterMute))) {
1485            flags = static_cast<AudioHwDevice::Flags>(flags |
1486                    AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
1487        }
1488
1489        mHardwareStatus = AUDIO_HW_IDLE;
1490    }
1491
1492    audio_module_handle_t handle = nextUniqueId();
1493    mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
1494
1495    ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
1496          name, dev->common.module->name, dev->common.module->id, handle);
1497
1498    return handle;
1499
1500}
1501
1502// ----------------------------------------------------------------------------
1503
1504uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
1505{
1506    Mutex::Autolock _l(mLock);
1507    PlaybackThread *thread = primaryPlaybackThread_l();
1508    return thread != NULL ? thread->sampleRate() : 0;
1509}
1510
1511size_t AudioFlinger::getPrimaryOutputFrameCount()
1512{
1513    Mutex::Autolock _l(mLock);
1514    PlaybackThread *thread = primaryPlaybackThread_l();
1515    return thread != NULL ? thread->frameCountHAL() : 0;
1516}
1517
1518// ----------------------------------------------------------------------------
1519
1520status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
1521{
1522    uid_t uid = IPCThreadState::self()->getCallingUid();
1523    if (uid != AID_SYSTEM) {
1524        return PERMISSION_DENIED;
1525    }
1526    Mutex::Autolock _l(mLock);
1527    if (mIsDeviceTypeKnown) {
1528        return INVALID_OPERATION;
1529    }
1530    mIsLowRamDevice = isLowRamDevice;
1531    mIsDeviceTypeKnown = true;
1532    return NO_ERROR;
1533}
1534
1535// ----------------------------------------------------------------------------
1536
1537audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
1538                                           audio_devices_t *pDevices,
1539                                           uint32_t *pSamplingRate,
1540                                           audio_format_t *pFormat,
1541                                           audio_channel_mask_t *pChannelMask,
1542                                           uint32_t *pLatencyMs,
1543                                           audio_output_flags_t flags,
1544                                           const audio_offload_info_t *offloadInfo)
1545{
1546    struct audio_config config;
1547    memset(&config, 0, sizeof(config));
1548    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1549    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1550    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1551    if (offloadInfo != NULL) {
1552        config.offload_info = *offloadInfo;
1553    }
1554
1555    ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
1556              module,
1557              (pDevices != NULL) ? *pDevices : 0,
1558              config.sample_rate,
1559              config.format,
1560              config.channel_mask,
1561              flags);
1562    ALOGV("openOutput(), offloadInfo %p version 0x%04x",
1563          offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version);
1564
1565    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1566        return AUDIO_IO_HANDLE_NONE;
1567    }
1568
1569    Mutex::Autolock _l(mLock);
1570
1571    AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices);
1572    if (outHwDev == NULL) {
1573        return AUDIO_IO_HANDLE_NONE;
1574    }
1575
1576    audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
1577    audio_io_handle_t id = nextUniqueId();
1578
1579    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
1580
1581    audio_stream_out_t *outStream = NULL;
1582    status_t status = hwDevHal->open_output_stream(hwDevHal,
1583                                          id,
1584                                          *pDevices,
1585                                          (audio_output_flags_t)flags,
1586                                          &config,
1587                                          &outStream);
1588
1589    mHardwareStatus = AUDIO_HW_IDLE;
1590    ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, "
1591            "Channels %x, status %d",
1592            outStream,
1593            config.sample_rate,
1594            config.format,
1595            config.channel_mask,
1596            status);
1597
1598    if (status == NO_ERROR && outStream != NULL) {
1599        AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags);
1600
1601        PlaybackThread *thread;
1602        if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
1603            thread = new OffloadThread(this, output, id, *pDevices);
1604            ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
1605        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
1606            (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
1607            (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
1608            thread = new DirectOutputThread(this, output, id, *pDevices);
1609            ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
1610        } else {
1611            thread = new MixerThread(this, output, id, *pDevices);
1612            ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
1613        }
1614        mPlaybackThreads.add(id, thread);
1615
1616        if (pSamplingRate != NULL) {
1617            *pSamplingRate = config.sample_rate;
1618        }
1619        if (pFormat != NULL) {
1620            *pFormat = config.format;
1621        }
1622        if (pChannelMask != NULL) {
1623            *pChannelMask = config.channel_mask;
1624        }
1625        if (pLatencyMs != NULL) {
1626            *pLatencyMs = thread->latency();
1627        }
1628
1629        // notify client processes of the new output creation
1630        thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1631
1632        // the first primary output opened designates the primary hw device
1633        if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
1634            ALOGI("Using module %d has the primary audio interface", module);
1635            mPrimaryHardwareDev = outHwDev;
1636
1637            AutoMutex lock(mHardwareLock);
1638            mHardwareStatus = AUDIO_HW_SET_MODE;
1639            hwDevHal->set_mode(hwDevHal, mMode);
1640            mHardwareStatus = AUDIO_HW_IDLE;
1641        }
1642        return id;
1643    }
1644
1645    return AUDIO_IO_HANDLE_NONE;
1646}
1647
1648audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
1649        audio_io_handle_t output2)
1650{
1651    Mutex::Autolock _l(mLock);
1652    MixerThread *thread1 = checkMixerThread_l(output1);
1653    MixerThread *thread2 = checkMixerThread_l(output2);
1654
1655    if (thread1 == NULL || thread2 == NULL) {
1656        ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
1657                output2);
1658        return AUDIO_IO_HANDLE_NONE;
1659    }
1660
1661    audio_io_handle_t id = nextUniqueId();
1662    DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
1663    thread->addOutputTrack(thread2);
1664    mPlaybackThreads.add(id, thread);
1665    // notify client processes of the new output creation
1666    thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
1667    return id;
1668}
1669
1670status_t AudioFlinger::closeOutput(audio_io_handle_t output)
1671{
1672    return closeOutput_nonvirtual(output);
1673}
1674
1675status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
1676{
1677    // keep strong reference on the playback thread so that
1678    // it is not destroyed while exit() is executed
1679    sp<PlaybackThread> thread;
1680    {
1681        Mutex::Autolock _l(mLock);
1682        thread = checkPlaybackThread_l(output);
1683        if (thread == NULL) {
1684            return BAD_VALUE;
1685        }
1686
1687        ALOGV("closeOutput() %d", output);
1688
1689        if (thread->type() == ThreadBase::MIXER) {
1690            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1691                if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
1692                    DuplicatingThread *dupThread =
1693                            (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
1694                    dupThread->removeOutputTrack((MixerThread *)thread.get());
1695
1696                }
1697            }
1698        }
1699
1700
1701        mPlaybackThreads.removeItem(output);
1702        // save all effects to the default thread
1703        if (mPlaybackThreads.size()) {
1704            PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
1705            if (dstThread != NULL) {
1706                // audioflinger lock is held here so the acquisition order of thread locks does not
1707                // matter
1708                Mutex::Autolock _dl(dstThread->mLock);
1709                Mutex::Autolock _sl(thread->mLock);
1710                Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
1711                for (size_t i = 0; i < effectChains.size(); i ++) {
1712                    moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
1713                }
1714            }
1715        }
1716        audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
1717    }
1718    thread->exit();
1719    // The thread entity (active unit of execution) is no longer running here,
1720    // but the ThreadBase container still exists.
1721
1722    if (thread->type() != ThreadBase::DUPLICATING) {
1723        AudioStreamOut *out = thread->clearOutput();
1724        ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
1725        // from now on thread->mOutput is NULL
1726        out->hwDev()->close_output_stream(out->hwDev(), out->stream);
1727        delete out;
1728    }
1729    return NO_ERROR;
1730}
1731
1732status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
1733{
1734    Mutex::Autolock _l(mLock);
1735    PlaybackThread *thread = checkPlaybackThread_l(output);
1736
1737    if (thread == NULL) {
1738        return BAD_VALUE;
1739    }
1740
1741    ALOGV("suspendOutput() %d", output);
1742    thread->suspend();
1743
1744    return NO_ERROR;
1745}
1746
1747status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
1748{
1749    Mutex::Autolock _l(mLock);
1750    PlaybackThread *thread = checkPlaybackThread_l(output);
1751
1752    if (thread == NULL) {
1753        return BAD_VALUE;
1754    }
1755
1756    ALOGV("restoreOutput() %d", output);
1757
1758    thread->restore();
1759
1760    return NO_ERROR;
1761}
1762
1763audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
1764                                          audio_devices_t *pDevices,
1765                                          uint32_t *pSamplingRate,
1766                                          audio_format_t *pFormat,
1767                                          audio_channel_mask_t *pChannelMask)
1768{
1769    struct audio_config config;
1770    memset(&config, 0, sizeof(config));
1771    config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
1772    config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
1773    config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
1774
1775    uint32_t reqSamplingRate = config.sample_rate;
1776    audio_format_t reqFormat = config.format;
1777    audio_channel_mask_t reqChannelMask = config.channel_mask;
1778
1779    if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) {
1780        return 0;
1781    }
1782
1783    Mutex::Autolock _l(mLock);
1784
1785    AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices);
1786    if (inHwDev == NULL) {
1787        return 0;
1788    }
1789
1790    audio_hw_device_t *inHwHal = inHwDev->hwDevice();
1791    audio_io_handle_t id = nextUniqueId();
1792
1793    audio_stream_in_t *inStream = NULL;
1794    status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
1795                                        &inStream);
1796    ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, "
1797            "status %d",
1798            inStream,
1799            config.sample_rate,
1800            config.format,
1801            config.channel_mask,
1802            status);
1803
1804    // If the input could not be opened with the requested parameters and we can handle the
1805    // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
1806    // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
1807    if (status == BAD_VALUE &&
1808        reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
1809        (config.sample_rate <= 2 * reqSamplingRate) &&
1810        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
1811        // FIXME describe the change proposed by HAL (save old values so we can log them here)
1812        ALOGV("openInput() reopening with proposed sampling rate and channel mask");
1813        inStream = NULL;
1814        status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
1815        // FIXME log this new status; HAL should not propose any further changes
1816    }
1817
1818    if (status == NO_ERROR && inStream != NULL) {
1819
1820#ifdef TEE_SINK
1821        // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
1822        // or (re-)create if current Pipe is idle and does not match the new format
1823        sp<NBAIO_Sink> teeSink;
1824        enum {
1825            TEE_SINK_NO,    // don't copy input
1826            TEE_SINK_NEW,   // copy input using a new pipe
1827            TEE_SINK_OLD,   // copy input using an existing pipe
1828        } kind;
1829        NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
1830                                        popcount(inStream->common.get_channels(&inStream->common)));
1831        if (!mTeeSinkInputEnabled) {
1832            kind = TEE_SINK_NO;
1833        } else if (!Format_isValid(format)) {
1834            kind = TEE_SINK_NO;
1835        } else if (mRecordTeeSink == 0) {
1836            kind = TEE_SINK_NEW;
1837        } else if (mRecordTeeSink->getStrongCount() != 1) {
1838            kind = TEE_SINK_NO;
1839        } else if (Format_isEqual(format, mRecordTeeSink->format())) {
1840            kind = TEE_SINK_OLD;
1841        } else {
1842            kind = TEE_SINK_NEW;
1843        }
1844        switch (kind) {
1845        case TEE_SINK_NEW: {
1846            Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
1847            size_t numCounterOffers = 0;
1848            const NBAIO_Format offers[1] = {format};
1849            ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
1850            ALOG_ASSERT(index == 0);
1851            PipeReader *pipeReader = new PipeReader(*pipe);
1852            numCounterOffers = 0;
1853            index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
1854            ALOG_ASSERT(index == 0);
1855            mRecordTeeSink = pipe;
1856            mRecordTeeSource = pipeReader;
1857            teeSink = pipe;
1858            }
1859            break;
1860        case TEE_SINK_OLD:
1861            teeSink = mRecordTeeSink;
1862            break;
1863        case TEE_SINK_NO:
1864        default:
1865            break;
1866        }
1867#endif
1868
1869        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
1870
1871        // Start record thread
1872        // RecordThread requires both input and output device indication to forward to audio
1873        // pre processing modules
1874        RecordThread *thread = new RecordThread(this,
1875                                  input,
1876                                  id,
1877                                  primaryOutputDevice_l(),
1878                                  *pDevices
1879#ifdef TEE_SINK
1880                                  , teeSink
1881#endif
1882                                  );
1883        mRecordThreads.add(id, thread);
1884        ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
1885        if (pSamplingRate != NULL) {
1886            *pSamplingRate = reqSamplingRate;
1887        }
1888        if (pFormat != NULL) {
1889            *pFormat = config.format;
1890        }
1891        if (pChannelMask != NULL) {
1892            *pChannelMask = reqChannelMask;
1893        }
1894
1895        // notify client processes of the new input creation
1896        thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
1897        return id;
1898    }
1899
1900    return 0;
1901}
1902
1903status_t AudioFlinger::closeInput(audio_io_handle_t input)
1904{
1905    return closeInput_nonvirtual(input);
1906}
1907
1908status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
1909{
1910    // keep strong reference on the record thread so that
1911    // it is not destroyed while exit() is executed
1912    sp<RecordThread> thread;
1913    {
1914        Mutex::Autolock _l(mLock);
1915        thread = checkRecordThread_l(input);
1916        if (thread == 0) {
1917            return BAD_VALUE;
1918        }
1919
1920        ALOGV("closeInput() %d", input);
1921        audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
1922        mRecordThreads.removeItem(input);
1923    }
1924    thread->exit();
1925    // The thread entity (active unit of execution) is no longer running here,
1926    // but the ThreadBase container still exists.
1927
1928    AudioStreamIn *in = thread->clearInput();
1929    ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
1930    // from now on thread->mInput is NULL
1931    in->hwDev()->close_input_stream(in->hwDev(), in->stream);
1932    delete in;
1933
1934    return NO_ERROR;
1935}
1936
1937status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
1938{
1939    Mutex::Autolock _l(mLock);
1940    ALOGV("invalidateStream() stream %d", stream);
1941
1942    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
1943        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
1944        thread->invalidateTracks(stream);
1945    }
1946
1947    return NO_ERROR;
1948}
1949
1950
1951int AudioFlinger::newAudioSessionId()
1952{
1953    return nextUniqueId();
1954}
1955
1956void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid)
1957{
1958    Mutex::Autolock _l(mLock);
1959    pid_t caller = IPCThreadState::self()->getCallingPid();
1960    ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
1961    if (pid != -1 && (caller == getpid_cached)) {
1962        caller = pid;
1963    }
1964
1965    // Ignore requests received from processes not known as notification client. The request
1966    // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
1967    // called from a different pid leaving a stale session reference.  Also we don't know how
1968    // to clear this reference if the client process dies.
1969    if (mNotificationClients.indexOfKey(caller) < 0) {
1970        ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
1971        return;
1972    }
1973
1974    size_t num = mAudioSessionRefs.size();
1975    for (size_t i = 0; i< num; i++) {
1976        AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
1977        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1978            ref->mCnt++;
1979            ALOGV(" incremented refcount to %d", ref->mCnt);
1980            return;
1981        }
1982    }
1983    mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
1984    ALOGV(" added new entry for %d", audioSession);
1985}
1986
1987void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid)
1988{
1989    Mutex::Autolock _l(mLock);
1990    pid_t caller = IPCThreadState::self()->getCallingPid();
1991    ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
1992    if (pid != -1 && (caller == getpid_cached)) {
1993        caller = pid;
1994    }
1995    size_t num = mAudioSessionRefs.size();
1996    for (size_t i = 0; i< num; i++) {
1997        AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
1998        if (ref->mSessionid == audioSession && ref->mPid == caller) {
1999            ref->mCnt--;
2000            ALOGV(" decremented refcount to %d", ref->mCnt);
2001            if (ref->mCnt == 0) {
2002                mAudioSessionRefs.removeAt(i);
2003                delete ref;
2004                purgeStaleEffects_l();
2005            }
2006            return;
2007        }
2008    }
2009    // If the caller is mediaserver it is likely that the session being released was acquired
2010    // on behalf of a process not in notification clients and we ignore the warning.
2011    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
2012}
2013
2014void AudioFlinger::purgeStaleEffects_l() {
2015
2016    ALOGV("purging stale effects");
2017
2018    Vector< sp<EffectChain> > chains;
2019
2020    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2021        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2022        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2023            sp<EffectChain> ec = t->mEffectChains[j];
2024            if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
2025                chains.push(ec);
2026            }
2027        }
2028    }
2029    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2030        sp<RecordThread> t = mRecordThreads.valueAt(i);
2031        for (size_t j = 0; j < t->mEffectChains.size(); j++) {
2032            sp<EffectChain> ec = t->mEffectChains[j];
2033            chains.push(ec);
2034        }
2035    }
2036
2037    for (size_t i = 0; i < chains.size(); i++) {
2038        sp<EffectChain> ec = chains[i];
2039        int sessionid = ec->sessionId();
2040        sp<ThreadBase> t = ec->mThread.promote();
2041        if (t == 0) {
2042            continue;
2043        }
2044        size_t numsessionrefs = mAudioSessionRefs.size();
2045        bool found = false;
2046        for (size_t k = 0; k < numsessionrefs; k++) {
2047            AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
2048            if (ref->mSessionid == sessionid) {
2049                ALOGV(" session %d still exists for %d with %d refs",
2050                    sessionid, ref->mPid, ref->mCnt);
2051                found = true;
2052                break;
2053            }
2054        }
2055        if (!found) {
2056            Mutex::Autolock _l(t->mLock);
2057            // remove all effects from the chain
2058            while (ec->mEffects.size()) {
2059                sp<EffectModule> effect = ec->mEffects[0];
2060                effect->unPin();
2061                t->removeEffect_l(effect);
2062                if (effect->purgeHandles()) {
2063                    t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
2064                }
2065                AudioSystem::unregisterEffect(effect->id());
2066            }
2067        }
2068    }
2069    return;
2070}
2071
2072// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
2073AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
2074{
2075    return mPlaybackThreads.valueFor(output).get();
2076}
2077
2078// checkMixerThread_l() must be called with AudioFlinger::mLock held
2079AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
2080{
2081    PlaybackThread *thread = checkPlaybackThread_l(output);
2082    return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
2083}
2084
2085// checkRecordThread_l() must be called with AudioFlinger::mLock held
2086AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
2087{
2088    return mRecordThreads.valueFor(input).get();
2089}
2090
2091uint32_t AudioFlinger::nextUniqueId()
2092{
2093    return (uint32_t) android_atomic_inc(&mNextUniqueId);
2094}
2095
2096AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
2097{
2098    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2099        PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
2100        AudioStreamOut *output = thread->getOutput();
2101        if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
2102            return thread;
2103        }
2104    }
2105    return NULL;
2106}
2107
2108audio_devices_t AudioFlinger::primaryOutputDevice_l() const
2109{
2110    PlaybackThread *thread = primaryPlaybackThread_l();
2111
2112    if (thread == NULL) {
2113        return 0;
2114    }
2115
2116    return thread->outDevice();
2117}
2118
2119sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
2120                                    int triggerSession,
2121                                    int listenerSession,
2122                                    sync_event_callback_t callBack,
2123                                    wp<RefBase> cookie)
2124{
2125    Mutex::Autolock _l(mLock);
2126
2127    sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
2128    status_t playStatus = NAME_NOT_FOUND;
2129    status_t recStatus = NAME_NOT_FOUND;
2130    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2131        playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
2132        if (playStatus == NO_ERROR) {
2133            return event;
2134        }
2135    }
2136    for (size_t i = 0; i < mRecordThreads.size(); i++) {
2137        recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
2138        if (recStatus == NO_ERROR) {
2139            return event;
2140        }
2141    }
2142    if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
2143        mPendingSyncEvents.add(event);
2144    } else {
2145        ALOGV("createSyncEvent() invalid event %d", event->type());
2146        event.clear();
2147    }
2148    return event;
2149}
2150
2151// ----------------------------------------------------------------------------
2152//  Effect management
2153// ----------------------------------------------------------------------------
2154
2155
2156status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
2157{
2158    Mutex::Autolock _l(mLock);
2159    return EffectQueryNumberEffects(numEffects);
2160}
2161
2162status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
2163{
2164    Mutex::Autolock _l(mLock);
2165    return EffectQueryEffect(index, descriptor);
2166}
2167
2168status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
2169        effect_descriptor_t *descriptor) const
2170{
2171    Mutex::Autolock _l(mLock);
2172    return EffectGetDescriptor(pUuid, descriptor);
2173}
2174
2175
2176sp<IEffect> AudioFlinger::createEffect(
2177        effect_descriptor_t *pDesc,
2178        const sp<IEffectClient>& effectClient,
2179        int32_t priority,
2180        audio_io_handle_t io,
2181        int sessionId,
2182        status_t *status,
2183        int *id,
2184        int *enabled)
2185{
2186    status_t lStatus = NO_ERROR;
2187    sp<EffectHandle> handle;
2188    effect_descriptor_t desc;
2189
2190    pid_t pid = IPCThreadState::self()->getCallingPid();
2191    ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
2192            pid, effectClient.get(), priority, sessionId, io);
2193
2194    if (pDesc == NULL) {
2195        lStatus = BAD_VALUE;
2196        goto Exit;
2197    }
2198
2199    // check audio settings permission for global effects
2200    if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
2201        lStatus = PERMISSION_DENIED;
2202        goto Exit;
2203    }
2204
2205    // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
2206    // that can only be created by audio policy manager (running in same process)
2207    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
2208        lStatus = PERMISSION_DENIED;
2209        goto Exit;
2210    }
2211
2212    {
2213        if (!EffectIsNullUuid(&pDesc->uuid)) {
2214            // if uuid is specified, request effect descriptor
2215            lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
2216            if (lStatus < 0) {
2217                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
2218                goto Exit;
2219            }
2220        } else {
2221            // if uuid is not specified, look for an available implementation
2222            // of the required type in effect factory
2223            if (EffectIsNullUuid(&pDesc->type)) {
2224                ALOGW("createEffect() no effect type");
2225                lStatus = BAD_VALUE;
2226                goto Exit;
2227            }
2228            uint32_t numEffects = 0;
2229            effect_descriptor_t d;
2230            d.flags = 0; // prevent compiler warning
2231            bool found = false;
2232
2233            lStatus = EffectQueryNumberEffects(&numEffects);
2234            if (lStatus < 0) {
2235                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
2236                goto Exit;
2237            }
2238            for (uint32_t i = 0; i < numEffects; i++) {
2239                lStatus = EffectQueryEffect(i, &desc);
2240                if (lStatus < 0) {
2241                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
2242                    continue;
2243                }
2244                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
2245                    // If matching type found save effect descriptor. If the session is
2246                    // 0 and the effect is not auxiliary, continue enumeration in case
2247                    // an auxiliary version of this effect type is available
2248                    found = true;
2249                    d = desc;
2250                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
2251                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2252                        break;
2253                    }
2254                }
2255            }
2256            if (!found) {
2257                lStatus = BAD_VALUE;
2258                ALOGW("createEffect() effect not found");
2259                goto Exit;
2260            }
2261            // For same effect type, chose auxiliary version over insert version if
2262            // connect to output mix (Compliance to OpenSL ES)
2263            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
2264                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
2265                desc = d;
2266            }
2267        }
2268
2269        // Do not allow auxiliary effects on a session different from 0 (output mix)
2270        if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
2271             (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2272            lStatus = INVALID_OPERATION;
2273            goto Exit;
2274        }
2275
2276        // check recording permission for visualizer
2277        if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
2278            !recordingAllowed()) {
2279            lStatus = PERMISSION_DENIED;
2280            goto Exit;
2281        }
2282
2283        // return effect descriptor
2284        *pDesc = desc;
2285        if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2286            // if the output returned by getOutputForEffect() is removed before we lock the
2287            // mutex below, the call to checkPlaybackThread_l(io) below will detect it
2288            // and we will exit safely
2289            io = AudioSystem::getOutputForEffect(&desc);
2290            ALOGV("createEffect got output %d", io);
2291        }
2292
2293        Mutex::Autolock _l(mLock);
2294
2295        // If output is not specified try to find a matching audio session ID in one of the
2296        // output threads.
2297        // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
2298        // because of code checking output when entering the function.
2299        // Note: io is never 0 when creating an effect on an input
2300        if (io == AUDIO_IO_HANDLE_NONE) {
2301            if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
2302                // output must be specified by AudioPolicyManager when using session
2303                // AUDIO_SESSION_OUTPUT_STAGE
2304                lStatus = BAD_VALUE;
2305                goto Exit;
2306            }
2307            // look for the thread where the specified audio session is present
2308            for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2309                if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2310                    io = mPlaybackThreads.keyAt(i);
2311                    break;
2312                }
2313            }
2314            if (io == 0) {
2315                for (size_t i = 0; i < mRecordThreads.size(); i++) {
2316                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
2317                        io = mRecordThreads.keyAt(i);
2318                        break;
2319                    }
2320                }
2321            }
2322            // If no output thread contains the requested session ID, default to
2323            // first output. The effect chain will be moved to the correct output
2324            // thread when a track with the same session ID is created
2325            if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
2326                io = mPlaybackThreads.keyAt(0);
2327            }
2328            ALOGV("createEffect() got io %d for effect %s", io, desc.name);
2329        }
2330        ThreadBase *thread = checkRecordThread_l(io);
2331        if (thread == NULL) {
2332            thread = checkPlaybackThread_l(io);
2333            if (thread == NULL) {
2334                ALOGE("createEffect() unknown output thread");
2335                lStatus = BAD_VALUE;
2336                goto Exit;
2337            }
2338        }
2339
2340        sp<Client> client = registerPid_l(pid);
2341
2342        // create effect on selected output thread
2343        handle = thread->createEffect_l(client, effectClient, priority, sessionId,
2344                &desc, enabled, &lStatus);
2345        if (handle != 0 && id != NULL) {
2346            *id = handle->id();
2347        }
2348    }
2349
2350Exit:
2351    *status = lStatus;
2352    return handle;
2353}
2354
2355status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
2356        audio_io_handle_t dstOutput)
2357{
2358    ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
2359            sessionId, srcOutput, dstOutput);
2360    Mutex::Autolock _l(mLock);
2361    if (srcOutput == dstOutput) {
2362        ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
2363        return NO_ERROR;
2364    }
2365    PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
2366    if (srcThread == NULL) {
2367        ALOGW("moveEffects() bad srcOutput %d", srcOutput);
2368        return BAD_VALUE;
2369    }
2370    PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
2371    if (dstThread == NULL) {
2372        ALOGW("moveEffects() bad dstOutput %d", dstOutput);
2373        return BAD_VALUE;
2374    }
2375
2376    Mutex::Autolock _dl(dstThread->mLock);
2377    Mutex::Autolock _sl(srcThread->mLock);
2378    return moveEffectChain_l(sessionId, srcThread, dstThread, false);
2379}
2380
2381// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
2382status_t AudioFlinger::moveEffectChain_l(int sessionId,
2383                                   AudioFlinger::PlaybackThread *srcThread,
2384                                   AudioFlinger::PlaybackThread *dstThread,
2385                                   bool reRegister)
2386{
2387    ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
2388            sessionId, srcThread, dstThread);
2389
2390    sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
2391    if (chain == 0) {
2392        ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
2393                sessionId, srcThread);
2394        return INVALID_OPERATION;
2395    }
2396
2397    // remove chain first. This is useful only if reconfiguring effect chain on same output thread,
2398    // so that a new chain is created with correct parameters when first effect is added. This is
2399    // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
2400    // removed.
2401    srcThread->removeEffectChain_l(chain);
2402
2403    // transfer all effects one by one so that new effect chain is created on new thread with
2404    // correct buffer sizes and audio parameters and effect engines reconfigured accordingly
2405    sp<EffectChain> dstChain;
2406    uint32_t strategy = 0; // prevent compiler warning
2407    sp<EffectModule> effect = chain->getEffectFromId_l(0);
2408    Vector< sp<EffectModule> > removed;
2409    status_t status = NO_ERROR;
2410    while (effect != 0) {
2411        srcThread->removeEffect_l(effect);
2412        removed.add(effect);
2413        status = dstThread->addEffect_l(effect);
2414        if (status != NO_ERROR) {
2415            break;
2416        }
2417        // removeEffect_l() has stopped the effect if it was active so it must be restarted
2418        if (effect->state() == EffectModule::ACTIVE ||
2419                effect->state() == EffectModule::STOPPING) {
2420            effect->start();
2421        }
2422        // if the move request is not received from audio policy manager, the effect must be
2423        // re-registered with the new strategy and output
2424        if (dstChain == 0) {
2425            dstChain = effect->chain().promote();
2426            if (dstChain == 0) {
2427                ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
2428                status = NO_INIT;
2429                break;
2430            }
2431            strategy = dstChain->strategy();
2432        }
2433        if (reRegister) {
2434            AudioSystem::unregisterEffect(effect->id());
2435            AudioSystem::registerEffect(&effect->desc(),
2436                                        dstThread->id(),
2437                                        strategy,
2438                                        sessionId,
2439                                        effect->id());
2440            AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2441        }
2442        effect = chain->getEffectFromId_l(0);
2443    }
2444
2445    if (status != NO_ERROR) {
2446        for (size_t i = 0; i < removed.size(); i++) {
2447            srcThread->addEffect_l(removed[i]);
2448            if (dstChain != 0 && reRegister) {
2449                AudioSystem::unregisterEffect(removed[i]->id());
2450                AudioSystem::registerEffect(&removed[i]->desc(),
2451                                            srcThread->id(),
2452                                            strategy,
2453                                            sessionId,
2454                                            removed[i]->id());
2455                AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
2456            }
2457        }
2458    }
2459
2460    return status;
2461}
2462
2463bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
2464{
2465    if (mGlobalEffectEnableTime != 0 &&
2466            ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
2467        return true;
2468    }
2469
2470    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2471        sp<EffectChain> ec =
2472                mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2473        if (ec != 0 && ec->isNonOffloadableEnabled()) {
2474            return true;
2475        }
2476    }
2477    return false;
2478}
2479
2480void AudioFlinger::onNonOffloadableGlobalEffectEnable()
2481{
2482    Mutex::Autolock _l(mLock);
2483
2484    mGlobalEffectEnableTime = systemTime();
2485
2486    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
2487        sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
2488        if (t->mType == ThreadBase::OFFLOAD) {
2489            t->invalidateTracks(AUDIO_STREAM_MUSIC);
2490        }
2491    }
2492
2493}
2494
2495struct Entry {
2496#define MAX_NAME 32     // %Y%m%d%H%M%S_%d.wav
2497    char mName[MAX_NAME];
2498};
2499
2500int comparEntry(const void *p1, const void *p2)
2501{
2502    return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
2503}
2504
2505#ifdef TEE_SINK
2506void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
2507{
2508    NBAIO_Source *teeSource = source.get();
2509    if (teeSource != NULL) {
2510        // .wav rotation
2511        // There is a benign race condition if 2 threads call this simultaneously.
2512        // They would both traverse the directory, but the result would simply be
2513        // failures at unlink() which are ignored.  It's also unlikely since
2514        // normally dumpsys is only done by bugreport or from the command line.
2515        char teePath[32+256];
2516        strcpy(teePath, "/data/misc/media");
2517        size_t teePathLen = strlen(teePath);
2518        DIR *dir = opendir(teePath);
2519        teePath[teePathLen++] = '/';
2520        if (dir != NULL) {
2521#define MAX_SORT 20 // number of entries to sort
2522#define MAX_KEEP 10 // number of entries to keep
2523            struct Entry entries[MAX_SORT];
2524            size_t entryCount = 0;
2525            while (entryCount < MAX_SORT) {
2526                struct dirent de;
2527                struct dirent *result = NULL;
2528                int rc = readdir_r(dir, &de, &result);
2529                if (rc != 0) {
2530                    ALOGW("readdir_r failed %d", rc);
2531                    break;
2532                }
2533                if (result == NULL) {
2534                    break;
2535                }
2536                if (result != &de) {
2537                    ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
2538                    break;
2539                }
2540                // ignore non .wav file entries
2541                size_t nameLen = strlen(de.d_name);
2542                if (nameLen <= 4 || nameLen >= MAX_NAME ||
2543                        strcmp(&de.d_name[nameLen - 4], ".wav")) {
2544                    continue;
2545                }
2546                strcpy(entries[entryCount++].mName, de.d_name);
2547            }
2548            (void) closedir(dir);
2549            if (entryCount > MAX_KEEP) {
2550                qsort(entries, entryCount, sizeof(Entry), comparEntry);
2551                for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
2552                    strcpy(&teePath[teePathLen], entries[i].mName);
2553                    (void) unlink(teePath);
2554                }
2555            }
2556        } else {
2557            if (fd >= 0) {
2558                fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
2559            }
2560        }
2561        char teeTime[16];
2562        struct timeval tv;
2563        gettimeofday(&tv, NULL);
2564        struct tm tm;
2565        localtime_r(&tv.tv_sec, &tm);
2566        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
2567        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
2568        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
2569        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
2570        if (teeFd >= 0) {
2571            char wavHeader[44];
2572            memcpy(wavHeader,
2573                "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",
2574                sizeof(wavHeader));
2575            NBAIO_Format format = teeSource->format();
2576            unsigned channelCount = Format_channelCount(format);
2577            ALOG_ASSERT(channelCount <= FCC_2);
2578            uint32_t sampleRate = Format_sampleRate(format);
2579            wavHeader[22] = channelCount;       // number of channels
2580            wavHeader[24] = sampleRate;         // sample rate
2581            wavHeader[25] = sampleRate >> 8;
2582            wavHeader[32] = channelCount * 2;   // block alignment
2583            write(teeFd, wavHeader, sizeof(wavHeader));
2584            size_t total = 0;
2585            bool firstRead = true;
2586            for (;;) {
2587#define TEE_SINK_READ 1024
2588                short buffer[TEE_SINK_READ * FCC_2];
2589                size_t count = TEE_SINK_READ;
2590                ssize_t actual = teeSource->read(buffer, count,
2591                        AudioBufferProvider::kInvalidPTS);
2592                bool wasFirstRead = firstRead;
2593                firstRead = false;
2594                if (actual <= 0) {
2595                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
2596                        continue;
2597                    }
2598                    break;
2599                }
2600                ALOG_ASSERT(actual <= (ssize_t)count);
2601                write(teeFd, buffer, actual * channelCount * sizeof(short));
2602                total += actual;
2603            }
2604            lseek(teeFd, (off_t) 4, SEEK_SET);
2605            uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
2606            write(teeFd, &temp, sizeof(temp));
2607            lseek(teeFd, (off_t) 40, SEEK_SET);
2608            temp =  total * channelCount * sizeof(short);
2609            write(teeFd, &temp, sizeof(temp));
2610            close(teeFd);
2611            if (fd >= 0) {
2612                fdprintf(fd, "tee copied to %s\n", teePath);
2613            }
2614        } else {
2615            if (fd >= 0) {
2616                fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
2617            }
2618        }
2619    }
2620}
2621#endif
2622
2623// ----------------------------------------------------------------------------
2624
2625status_t AudioFlinger::onTransact(
2626        uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
2627{
2628    return BnAudioFlinger::onTransact(code, data, reply, flags);
2629}
2630
2631}; // namespace android
2632