blob: 8cc5f6c4d03c376d07600ffb37b696f16019bf9c [file] [log] [blame]
Eric Laurent81784c32012-11-19 14:55:58 -08001/*
2**
3** Copyright 2012, 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
Alex Ray371eb972012-11-30 11:11:54 -080021#define ATRACE_TAG ATRACE_TAG_AUDIO
Eric Laurent81784c32012-11-19 14:55:58 -080022
23#include <math.h>
24#include <fcntl.h>
25#include <sys/stat.h>
26#include <cutils/properties.h>
27#include <cutils/compiler.h>
28#include <utils/Log.h>
Alex Ray371eb972012-11-30 11:11:54 -080029#include <utils/Trace.h>
Eric Laurent81784c32012-11-19 14:55:58 -080030
31#include <private/media/AudioTrackShared.h>
32#include <hardware/audio.h>
33#include <audio_effects/effect_ns.h>
34#include <audio_effects/effect_aec.h>
35#include <audio_utils/primitives.h>
36
37// NBAIO implementations
38#include <media/nbaio/AudioStreamOutSink.h>
39#include <media/nbaio/MonoPipe.h>
40#include <media/nbaio/MonoPipeReader.h>
41#include <media/nbaio/Pipe.h>
42#include <media/nbaio/PipeReader.h>
43#include <media/nbaio/SourceAudioBufferProvider.h>
44
45#include <powermanager/PowerManager.h>
46
47#include <common_time/cc_helper.h>
48#include <common_time/local_clock.h>
49
50#include "AudioFlinger.h"
51#include "AudioMixer.h"
52#include "FastMixer.h"
53#include "ServiceUtilities.h"
54#include "SchedulingPolicyService.h"
55
56#undef ADD_BATTERY_DATA
57
58#ifdef ADD_BATTERY_DATA
59#include <media/IMediaPlayerService.h>
60#include <media/IMediaDeathNotifier.h>
61#endif
62
63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds
64#ifdef DEBUG_CPU_USAGE
65#include <cpustats/CentralTendencyStatistics.h>
66#include <cpustats/ThreadCpuUsage.h>
67#endif
68
69// ----------------------------------------------------------------------------
70
71// Note: the following macro is used for extremely verbose logging message. In
72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
73// 0; but one side effect of this is to turn all LOGV's as well. Some messages
74// are so verbose that we want to suppress them even when we have ALOG_ASSERT
75// turned on. Do not uncomment the #def below unless you really know what you
76// are doing and want to see all of the extremely verbose messages.
77//#define VERY_VERY_VERBOSE_LOGGING
78#ifdef VERY_VERY_VERBOSE_LOGGING
79#define ALOGVV ALOGV
80#else
81#define ALOGVV(a...) do { } while(0)
82#endif
83
84namespace android {
85
86// retry counts for buffer fill timeout
87// 50 * ~20msecs = 1 second
88static const int8_t kMaxTrackRetries = 50;
89static const int8_t kMaxTrackStartupRetries = 50;
90// allow less retry attempts on direct output thread.
91// direct outputs can be a scarce resource in audio hardware and should
92// be released as quickly as possible.
93static const int8_t kMaxTrackRetriesDirect = 2;
94
95// don't warn about blocked writes or record buffer overflows more often than this
96static const nsecs_t kWarningThrottleNs = seconds(5);
97
98// RecordThread loop sleep time upon application overrun or audio HAL read error
99static const int kRecordThreadSleepUs = 5000;
100
101// maximum time to wait for setParameters to complete
102static const nsecs_t kSetParametersTimeoutNs = seconds(2);
103
104// minimum sleep time for the mixer thread loop when tracks are active but in underrun
105static const uint32_t kMinThreadSleepTimeUs = 5000;
106// maximum divider applied to the active sleep time in the mixer thread loop
107static const uint32_t kMaxThreadSleepTimeShift = 2;
108
109// minimum normal mix buffer size, expressed in milliseconds rather than frames
110static const uint32_t kMinNormalMixBufferSizeMs = 20;
111// maximum normal mix buffer size
112static const uint32_t kMaxNormalMixBufferSizeMs = 24;
113
114// Whether to use fast mixer
115static const enum {
116 FastMixer_Never, // never initialize or use: for debugging only
117 FastMixer_Always, // always initialize and use, even if not needed: for debugging only
118 // normal mixer multiplier is 1
119 FastMixer_Static, // initialize if needed, then use all the time if initialized,
120 // multiplier is calculated based on min & max normal mixer buffer size
121 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load,
122 // multiplier is calculated based on min & max normal mixer buffer size
123 // FIXME for FastMixer_Dynamic:
124 // Supporting this option will require fixing HALs that can't handle large writes.
125 // For example, one HAL implementation returns an error from a large write,
126 // and another HAL implementation corrupts memory, possibly in the sample rate converter.
127 // We could either fix the HAL implementations, or provide a wrapper that breaks
128 // up large writes into smaller ones, and the wrapper would need to deal with scheduler.
129} kUseFastMixer = FastMixer_Static;
130
131// Priorities for requestPriority
132static const int kPriorityAudioApp = 2;
133static const int kPriorityFastMixer = 3;
134
135// IAudioFlinger::createTrack() reports back to client the total size of shared memory area
136// for the track. The client then sub-divides this into smaller buffers for its use.
137// Currently the client uses double-buffering by default, but doesn't tell us about that.
138// So for now we just assume that client is double-buffered.
139// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or
140// N-buffering, so AudioFlinger could allocate the right amount of memory.
141// See the client's minBufCount and mNotificationFramesAct calculations for details.
142static const int kFastTrackMultiplier = 2;
143
144// ----------------------------------------------------------------------------
145
146#ifdef ADD_BATTERY_DATA
147// To collect the amplifier usage
148static void addBatteryData(uint32_t params) {
149 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
150 if (service == NULL) {
151 // it already logged
152 return;
153 }
154
155 service->addBatteryData(params);
156}
157#endif
158
159
160// ----------------------------------------------------------------------------
161// CPU Stats
162// ----------------------------------------------------------------------------
163
164class CpuStats {
165public:
166 CpuStats();
167 void sample(const String8 &title);
168#ifdef DEBUG_CPU_USAGE
169private:
170 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns
171 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
172
173 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
174
175 int mCpuNum; // thread's current CPU number
176 int mCpukHz; // frequency of thread's current CPU in kHz
177#endif
178};
179
180CpuStats::CpuStats()
181#ifdef DEBUG_CPU_USAGE
182 : mCpuNum(-1), mCpukHz(-1)
183#endif
184{
185}
186
187void CpuStats::sample(const String8 &title) {
188#ifdef DEBUG_CPU_USAGE
189 // get current thread's delta CPU time in wall clock ns
190 double wcNs;
191 bool valid = mCpuUsage.sampleAndEnable(wcNs);
192
193 // record sample for wall clock statistics
194 if (valid) {
195 mWcStats.sample(wcNs);
196 }
197
198 // get the current CPU number
199 int cpuNum = sched_getcpu();
200
201 // get the current CPU frequency in kHz
202 int cpukHz = mCpuUsage.getCpukHz(cpuNum);
203
204 // check if either CPU number or frequency changed
205 if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
206 mCpuNum = cpuNum;
207 mCpukHz = cpukHz;
208 // ignore sample for purposes of cycles
209 valid = false;
210 }
211
212 // if no change in CPU number or frequency, then record sample for cycle statistics
213 if (valid && mCpukHz > 0) {
214 double cycles = wcNs * cpukHz * 0.000001;
215 mHzStats.sample(cycles);
216 }
217
218 unsigned n = mWcStats.n();
219 // mCpuUsage.elapsed() is expensive, so don't call it every loop
220 if ((n & 127) == 1) {
221 long long elapsed = mCpuUsage.elapsed();
222 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
223 double perLoop = elapsed / (double) n;
224 double perLoop100 = perLoop * 0.01;
225 double perLoop1k = perLoop * 0.001;
226 double mean = mWcStats.mean();
227 double stddev = mWcStats.stddev();
228 double minimum = mWcStats.minimum();
229 double maximum = mWcStats.maximum();
230 double meanCycles = mHzStats.mean();
231 double stddevCycles = mHzStats.stddev();
232 double minCycles = mHzStats.minimum();
233 double maxCycles = mHzStats.maximum();
234 mCpuUsage.resetElapsed();
235 mWcStats.reset();
236 mHzStats.reset();
237 ALOGD("CPU usage for %s over past %.1f secs\n"
238 " (%u mixer loops at %.1f mean ms per loop):\n"
239 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
240 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
241 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
242 title.string(),
243 elapsed * .000000001, n, perLoop * .000001,
244 mean * .001,
245 stddev * .001,
246 minimum * .001,
247 maximum * .001,
248 mean / perLoop100,
249 stddev / perLoop100,
250 minimum / perLoop100,
251 maximum / perLoop100,
252 meanCycles / perLoop1k,
253 stddevCycles / perLoop1k,
254 minCycles / perLoop1k,
255 maxCycles / perLoop1k);
256
257 }
258 }
259#endif
260};
261
262// ----------------------------------------------------------------------------
263// ThreadBase
264// ----------------------------------------------------------------------------
265
266AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
267 audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
268 : Thread(false /*canCallJava*/),
269 mType(type),
270 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
271 // mChannelMask
272 mChannelCount(0),
273 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
274 mParamStatus(NO_ERROR),
275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
277 // mName will be set by concrete (non-virtual) subclass
278 mDeathRecipient(new PMDeathRecipient(this))
279{
280}
281
282AudioFlinger::ThreadBase::~ThreadBase()
283{
284 mParamCond.broadcast();
285 // do not lock the mutex in destructor
286 releaseWakeLock_l();
287 if (mPowerManager != 0) {
288 sp<IBinder> binder = mPowerManager->asBinder();
289 binder->unlinkToDeath(mDeathRecipient);
290 }
291}
292
293void AudioFlinger::ThreadBase::exit()
294{
295 ALOGV("ThreadBase::exit");
296 // do any cleanup required for exit to succeed
297 preExit();
298 {
299 // This lock prevents the following race in thread (uniprocessor for illustration):
300 // if (!exitPending()) {
301 // // context switch from here to exit()
302 // // exit() calls requestExit(), what exitPending() observes
303 // // exit() calls signal(), which is dropped since no waiters
304 // // context switch back from exit() to here
305 // mWaitWorkCV.wait(...);
306 // // now thread is hung
307 // }
308 AutoMutex lock(mLock);
309 requestExit();
310 mWaitWorkCV.broadcast();
311 }
312 // When Thread::requestExitAndWait is made virtual and this method is renamed to
313 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
314 requestExitAndWait();
315}
316
317status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
318{
319 status_t status;
320
321 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
322 Mutex::Autolock _l(mLock);
323
324 mNewParameters.add(keyValuePairs);
325 mWaitWorkCV.signal();
326 // wait condition with timeout in case the thread loop has exited
327 // before the request could be processed
328 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
329 status = mParamStatus;
330 mWaitWorkCV.signal();
331 } else {
332 status = TIMED_OUT;
333 }
334 return status;
335}
336
337void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param)
338{
339 Mutex::Autolock _l(mLock);
340 sendIoConfigEvent_l(event, param);
341}
342
343// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
344void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param)
345{
346 IoConfigEvent *ioEvent = new IoConfigEvent(event, param);
347 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent));
348 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event,
349 param);
350 mWaitWorkCV.signal();
351}
352
353// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
354void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
355{
356 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio);
357 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent));
358 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d",
359 mConfigEvents.size(), pid, tid, prio);
360 mWaitWorkCV.signal();
361}
362
363void AudioFlinger::ThreadBase::processConfigEvents()
364{
365 mLock.lock();
366 while (!mConfigEvents.isEmpty()) {
367 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
368 ConfigEvent *event = mConfigEvents[0];
369 mConfigEvents.removeAt(0);
370 // release mLock before locking AudioFlinger mLock: lock order is always
371 // AudioFlinger then ThreadBase to avoid cross deadlock
372 mLock.unlock();
373 switch(event->type()) {
374 case CFG_EVENT_PRIO: {
375 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event);
376 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio());
377 if (err != 0) {
378 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; "
379 "error %d",
380 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err);
381 }
382 } break;
383 case CFG_EVENT_IO: {
384 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event);
385 mAudioFlinger->mLock.lock();
386 audioConfigChanged_l(ioEvent->event(), ioEvent->param());
387 mAudioFlinger->mLock.unlock();
388 } break;
389 default:
390 ALOGE("processConfigEvents() unknown event type %d", event->type());
391 break;
392 }
393 delete event;
394 mLock.lock();
395 }
396 mLock.unlock();
397}
398
399void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
400{
401 const size_t SIZE = 256;
402 char buffer[SIZE];
403 String8 result;
404
405 bool locked = AudioFlinger::dumpTryLock(mLock);
406 if (!locked) {
407 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
408 write(fd, buffer, strlen(buffer));
409 }
410
411 snprintf(buffer, SIZE, "io handle: %d\n", mId);
412 result.append(buffer);
413 snprintf(buffer, SIZE, "TID: %d\n", getTid());
414 result.append(buffer);
415 snprintf(buffer, SIZE, "standby: %d\n", mStandby);
416 result.append(buffer);
417 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
418 result.append(buffer);
419 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
420 result.append(buffer);
421 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
422 result.append(buffer);
423 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
424 result.append(buffer);
425 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
426 result.append(buffer);
427 snprintf(buffer, SIZE, "Format: %d\n", mFormat);
428 result.append(buffer);
429 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
430 result.append(buffer);
431
432 snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
433 result.append(buffer);
434 result.append(" Index Command");
435 for (size_t i = 0; i < mNewParameters.size(); ++i) {
436 snprintf(buffer, SIZE, "\n %02d ", i);
437 result.append(buffer);
438 result.append(mNewParameters[i]);
439 }
440
441 snprintf(buffer, SIZE, "\n\nPending config events: \n");
442 result.append(buffer);
443 for (size_t i = 0; i < mConfigEvents.size(); i++) {
444 mConfigEvents[i]->dump(buffer, SIZE);
445 result.append(buffer);
446 }
447 result.append("\n");
448
449 write(fd, result.string(), result.size());
450
451 if (locked) {
452 mLock.unlock();
453 }
454}
455
456void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
457{
458 const size_t SIZE = 256;
459 char buffer[SIZE];
460 String8 result;
461
462 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
463 write(fd, buffer, strlen(buffer));
464
465 for (size_t i = 0; i < mEffectChains.size(); ++i) {
466 sp<EffectChain> chain = mEffectChains[i];
467 if (chain != 0) {
468 chain->dump(fd, args);
469 }
470 }
471}
472
473void AudioFlinger::ThreadBase::acquireWakeLock()
474{
475 Mutex::Autolock _l(mLock);
476 acquireWakeLock_l();
477}
478
479void AudioFlinger::ThreadBase::acquireWakeLock_l()
480{
481 if (mPowerManager == 0) {
482 // use checkService() to avoid blocking if power service is not up yet
483 sp<IBinder> binder =
484 defaultServiceManager()->checkService(String16("power"));
485 if (binder == 0) {
486 ALOGW("Thread %s cannot connect to the power manager service", mName);
487 } else {
488 mPowerManager = interface_cast<IPowerManager>(binder);
489 binder->linkToDeath(mDeathRecipient);
490 }
491 }
492 if (mPowerManager != 0) {
493 sp<IBinder> binder = new BBinder();
494 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
495 binder,
496 String16(mName));
497 if (status == NO_ERROR) {
498 mWakeLockToken = binder;
499 }
500 ALOGV("acquireWakeLock_l() %s status %d", mName, status);
501 }
502}
503
504void AudioFlinger::ThreadBase::releaseWakeLock()
505{
506 Mutex::Autolock _l(mLock);
507 releaseWakeLock_l();
508}
509
510void AudioFlinger::ThreadBase::releaseWakeLock_l()
511{
512 if (mWakeLockToken != 0) {
513 ALOGV("releaseWakeLock_l() %s", mName);
514 if (mPowerManager != 0) {
515 mPowerManager->releaseWakeLock(mWakeLockToken, 0);
516 }
517 mWakeLockToken.clear();
518 }
519}
520
521void AudioFlinger::ThreadBase::clearPowerManager()
522{
523 Mutex::Autolock _l(mLock);
524 releaseWakeLock_l();
525 mPowerManager.clear();
526}
527
528void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
529{
530 sp<ThreadBase> thread = mThread.promote();
531 if (thread != 0) {
532 thread->clearPowerManager();
533 }
534 ALOGW("power manager service died !!!");
535}
536
537void AudioFlinger::ThreadBase::setEffectSuspended(
538 const effect_uuid_t *type, bool suspend, int sessionId)
539{
540 Mutex::Autolock _l(mLock);
541 setEffectSuspended_l(type, suspend, sessionId);
542}
543
544void AudioFlinger::ThreadBase::setEffectSuspended_l(
545 const effect_uuid_t *type, bool suspend, int sessionId)
546{
547 sp<EffectChain> chain = getEffectChain_l(sessionId);
548 if (chain != 0) {
549 if (type != NULL) {
550 chain->setEffectSuspended_l(type, suspend);
551 } else {
552 chain->setEffectSuspendedAll_l(suspend);
553 }
554 }
555
556 updateSuspendedSessions_l(type, suspend, sessionId);
557}
558
559void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
560{
561 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
562 if (index < 0) {
563 return;
564 }
565
566 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects =
567 mSuspendedSessions.valueAt(index);
568
569 for (size_t i = 0; i < sessionEffects.size(); i++) {
570 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
571 for (int j = 0; j < desc->mRefCount; j++) {
572 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
573 chain->setEffectSuspendedAll_l(true);
574 } else {
575 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
576 desc->mType.timeLow);
577 chain->setEffectSuspended_l(&desc->mType, true);
578 }
579 }
580 }
581}
582
583void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
584 bool suspend,
585 int sessionId)
586{
587 ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
588
589 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
590
591 if (suspend) {
592 if (index >= 0) {
593 sessionEffects = mSuspendedSessions.valueAt(index);
594 } else {
595 mSuspendedSessions.add(sessionId, sessionEffects);
596 }
597 } else {
598 if (index < 0) {
599 return;
600 }
601 sessionEffects = mSuspendedSessions.valueAt(index);
602 }
603
604
605 int key = EffectChain::kKeyForSuspendAll;
606 if (type != NULL) {
607 key = type->timeLow;
608 }
609 index = sessionEffects.indexOfKey(key);
610
611 sp<SuspendedSessionDesc> desc;
612 if (suspend) {
613 if (index >= 0) {
614 desc = sessionEffects.valueAt(index);
615 } else {
616 desc = new SuspendedSessionDesc();
617 if (type != NULL) {
618 desc->mType = *type;
619 }
620 sessionEffects.add(key, desc);
621 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
622 }
623 desc->mRefCount++;
624 } else {
625 if (index < 0) {
626 return;
627 }
628 desc = sessionEffects.valueAt(index);
629 if (--desc->mRefCount == 0) {
630 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
631 sessionEffects.removeItemsAt(index);
632 if (sessionEffects.isEmpty()) {
633 ALOGV("updateSuspendedSessions_l() restore removing session %d",
634 sessionId);
635 mSuspendedSessions.removeItem(sessionId);
636 }
637 }
638 }
639 if (!sessionEffects.isEmpty()) {
640 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
641 }
642}
643
644void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
645 bool enabled,
646 int sessionId)
647{
648 Mutex::Autolock _l(mLock);
649 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
650}
651
652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
653 bool enabled,
654 int sessionId)
655{
656 if (mType != RECORD) {
657 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
658 // another session. This gives the priority to well behaved effect control panels
659 // and applications not using global effects.
660 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
661 // global effects
662 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
663 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
664 }
665 }
666
667 sp<EffectChain> chain = getEffectChain_l(sessionId);
668 if (chain != 0) {
669 chain->checkSuspendOnEffectEnabled(effect, enabled);
670 }
671}
672
673// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held
674sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
675 const sp<AudioFlinger::Client>& client,
676 const sp<IEffectClient>& effectClient,
677 int32_t priority,
678 int sessionId,
679 effect_descriptor_t *desc,
680 int *enabled,
681 status_t *status
682 )
683{
684 sp<EffectModule> effect;
685 sp<EffectHandle> handle;
686 status_t lStatus;
687 sp<EffectChain> chain;
688 bool chainCreated = false;
689 bool effectCreated = false;
690 bool effectRegistered = false;
691
692 lStatus = initCheck();
693 if (lStatus != NO_ERROR) {
694 ALOGW("createEffect_l() Audio driver not initialized.");
695 goto Exit;
696 }
697
698 // Do not allow effects with session ID 0 on direct output or duplicating threads
699 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format
700 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
701 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
702 desc->name, sessionId);
703 lStatus = BAD_VALUE;
704 goto Exit;
705 }
706 // Only Pre processor effects are allowed on input threads and only on input threads
707 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
708 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
709 desc->name, desc->flags, mType);
710 lStatus = BAD_VALUE;
711 goto Exit;
712 }
713
714 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
715
716 { // scope for mLock
717 Mutex::Autolock _l(mLock);
718
719 // check for existing effect chain with the requested audio session
720 chain = getEffectChain_l(sessionId);
721 if (chain == 0) {
722 // create a new chain for this session
723 ALOGV("createEffect_l() new effect chain for session %d", sessionId);
724 chain = new EffectChain(this, sessionId);
725 addEffectChain_l(chain);
726 chain->setStrategy(getStrategyForSession_l(sessionId));
727 chainCreated = true;
728 } else {
729 effect = chain->getEffectFromDesc_l(desc);
730 }
731
732 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
733
734 if (effect == 0) {
735 int id = mAudioFlinger->nextUniqueId();
736 // Check CPU and memory usage
737 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
738 if (lStatus != NO_ERROR) {
739 goto Exit;
740 }
741 effectRegistered = true;
742 // create a new effect module if none present in the chain
743 effect = new EffectModule(this, chain, desc, id, sessionId);
744 lStatus = effect->status();
745 if (lStatus != NO_ERROR) {
746 goto Exit;
747 }
748 lStatus = chain->addEffect_l(effect);
749 if (lStatus != NO_ERROR) {
750 goto Exit;
751 }
752 effectCreated = true;
753
754 effect->setDevice(mOutDevice);
755 effect->setDevice(mInDevice);
756 effect->setMode(mAudioFlinger->getMode());
757 effect->setAudioSource(mAudioSource);
758 }
759 // create effect handle and connect it to effect module
760 handle = new EffectHandle(effect, client, effectClient, priority);
761 lStatus = effect->addHandle(handle.get());
762 if (enabled != NULL) {
763 *enabled = (int)effect->isEnabled();
764 }
765 }
766
767Exit:
768 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
769 Mutex::Autolock _l(mLock);
770 if (effectCreated) {
771 chain->removeEffect_l(effect);
772 }
773 if (effectRegistered) {
774 AudioSystem::unregisterEffect(effect->id());
775 }
776 if (chainCreated) {
777 removeEffectChain_l(chain);
778 }
779 handle.clear();
780 }
781
782 if (status != NULL) {
783 *status = lStatus;
784 }
785 return handle;
786}
787
788sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId)
789{
790 Mutex::Autolock _l(mLock);
791 return getEffect_l(sessionId, effectId);
792}
793
794sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
795{
796 sp<EffectChain> chain = getEffectChain_l(sessionId);
797 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
798}
799
800// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
801// PlaybackThread::mLock held
802status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
803{
804 // check for existing effect chain with the requested audio session
805 int sessionId = effect->sessionId();
806 sp<EffectChain> chain = getEffectChain_l(sessionId);
807 bool chainCreated = false;
808
809 if (chain == 0) {
810 // create a new chain for this session
811 ALOGV("addEffect_l() new effect chain for session %d", sessionId);
812 chain = new EffectChain(this, sessionId);
813 addEffectChain_l(chain);
814 chain->setStrategy(getStrategyForSession_l(sessionId));
815 chainCreated = true;
816 }
817 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
818
819 if (chain->getEffectFromId_l(effect->id()) != 0) {
820 ALOGW("addEffect_l() %p effect %s already present in chain %p",
821 this, effect->desc().name, chain.get());
822 return BAD_VALUE;
823 }
824
825 status_t status = chain->addEffect_l(effect);
826 if (status != NO_ERROR) {
827 if (chainCreated) {
828 removeEffectChain_l(chain);
829 }
830 return status;
831 }
832
833 effect->setDevice(mOutDevice);
834 effect->setDevice(mInDevice);
835 effect->setMode(mAudioFlinger->getMode());
836 effect->setAudioSource(mAudioSource);
837 return NO_ERROR;
838}
839
840void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
841
842 ALOGV("removeEffect_l() %p effect %p", this, effect.get());
843 effect_descriptor_t desc = effect->desc();
844 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
845 detachAuxEffect_l(effect->id());
846 }
847
848 sp<EffectChain> chain = effect->chain().promote();
849 if (chain != 0) {
850 // remove effect chain if removing last effect
851 if (chain->removeEffect_l(effect) == 0) {
852 removeEffectChain_l(chain);
853 }
854 } else {
855 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
856 }
857}
858
859void AudioFlinger::ThreadBase::lockEffectChains_l(
860 Vector< sp<AudioFlinger::EffectChain> >& effectChains)
861{
862 effectChains = mEffectChains;
863 for (size_t i = 0; i < mEffectChains.size(); i++) {
864 mEffectChains[i]->lock();
865 }
866}
867
868void AudioFlinger::ThreadBase::unlockEffectChains(
869 const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
870{
871 for (size_t i = 0; i < effectChains.size(); i++) {
872 effectChains[i]->unlock();
873 }
874}
875
876sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
877{
878 Mutex::Autolock _l(mLock);
879 return getEffectChain_l(sessionId);
880}
881
882sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const
883{
884 size_t size = mEffectChains.size();
885 for (size_t i = 0; i < size; i++) {
886 if (mEffectChains[i]->sessionId() == sessionId) {
887 return mEffectChains[i];
888 }
889 }
890 return 0;
891}
892
893void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
894{
895 Mutex::Autolock _l(mLock);
896 size_t size = mEffectChains.size();
897 for (size_t i = 0; i < size; i++) {
898 mEffectChains[i]->setMode_l(mode);
899 }
900}
901
902void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
903 EffectHandle *handle,
904 bool unpinIfLast) {
905
906 Mutex::Autolock _l(mLock);
907 ALOGV("disconnectEffect() %p effect %p", this, effect.get());
908 // delete the effect module if removing last handle on it
909 if (effect->removeHandle(handle) == 0) {
910 if (!effect->isPinned() || unpinIfLast) {
911 removeEffect_l(effect);
912 AudioSystem::unregisterEffect(effect->id());
913 }
914 }
915}
916
917// ----------------------------------------------------------------------------
918// Playback
919// ----------------------------------------------------------------------------
920
921AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
922 AudioStreamOut* output,
923 audio_io_handle_t id,
924 audio_devices_t device,
925 type_t type)
926 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
927 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
928 // mStreamTypes[] initialized in constructor body
929 mOutput(output),
930 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
931 mMixerStatus(MIXER_IDLE),
932 mMixerStatusIgnoringFastTracks(MIXER_IDLE),
933 standbyDelay(AudioFlinger::mStandbyTimeInNsecs),
934 mScreenState(AudioFlinger::mScreenState),
935 // index 0 is reserved for normal mixer's submix
936 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
937{
938 snprintf(mName, kNameLength, "AudioOut_%X", id);
Glenn Kasten9e58b552013-01-18 15:09:48 -0800939 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
Eric Laurent81784c32012-11-19 14:55:58 -0800940
941 // Assumes constructor is called by AudioFlinger with it's mLock held, but
942 // it would be safer to explicitly pass initial masterVolume/masterMute as
943 // parameter.
944 //
945 // If the HAL we are using has support for master volume or master mute,
946 // then do not attenuate or mute during mixing (just leave the volume at 1.0
947 // and the mute set to false).
948 mMasterVolume = audioFlinger->masterVolume_l();
949 mMasterMute = audioFlinger->masterMute_l();
950 if (mOutput && mOutput->audioHwDev) {
951 if (mOutput->audioHwDev->canSetMasterVolume()) {
952 mMasterVolume = 1.0;
953 }
954
955 if (mOutput->audioHwDev->canSetMasterMute()) {
956 mMasterMute = false;
957 }
958 }
959
960 readOutputParameters();
961
962 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
963 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
964 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
965 stream = (audio_stream_type_t) (stream + 1)) {
966 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
967 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
968 }
969 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
970 // because mAudioFlinger doesn't have one to copy from
971}
972
973AudioFlinger::PlaybackThread::~PlaybackThread()
974{
Glenn Kasten9e58b552013-01-18 15:09:48 -0800975 mAudioFlinger->unregisterWriter(mNBLogWriter);
Eric Laurent81784c32012-11-19 14:55:58 -0800976 delete [] mMixBuffer;
977}
978
979void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
980{
981 dumpInternals(fd, args);
982 dumpTracks(fd, args);
983 dumpEffectChains(fd, args);
984}
985
986void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
987{
988 const size_t SIZE = 256;
989 char buffer[SIZE];
990 String8 result;
991
992 result.appendFormat("Output thread %p stream volumes in dB:\n ", this);
993 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
994 const stream_type_t *st = &mStreamTypes[i];
995 if (i > 0) {
996 result.appendFormat(", ");
997 }
998 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
999 if (st->mute) {
1000 result.append("M");
1001 }
1002 }
1003 result.append("\n");
1004 write(fd, result.string(), result.length());
1005 result.clear();
1006
1007 snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
1008 result.append(buffer);
1009 Track::appendDumpHeader(result);
1010 for (size_t i = 0; i < mTracks.size(); ++i) {
1011 sp<Track> track = mTracks[i];
1012 if (track != 0) {
1013 track->dump(buffer, SIZE);
1014 result.append(buffer);
1015 }
1016 }
1017
1018 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
1019 result.append(buffer);
1020 Track::appendDumpHeader(result);
1021 for (size_t i = 0; i < mActiveTracks.size(); ++i) {
1022 sp<Track> track = mActiveTracks[i].promote();
1023 if (track != 0) {
1024 track->dump(buffer, SIZE);
1025 result.append(buffer);
1026 }
1027 }
1028 write(fd, result.string(), result.size());
1029
1030 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way.
1031 FastTrackUnderruns underruns = getFastTrackUnderruns(0);
1032 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n",
1033 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
1034}
1035
1036void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1037{
1038 const size_t SIZE = 256;
1039 char buffer[SIZE];
1040 String8 result;
1041
1042 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
1043 result.append(buffer);
1044 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",
1045 ns2ms(systemTime() - mLastWriteTime));
1046 result.append(buffer);
1047 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
1048 result.append(buffer);
1049 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
1050 result.append(buffer);
1051 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
1052 result.append(buffer);
1053 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
1054 result.append(buffer);
1055 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
1056 result.append(buffer);
1057 write(fd, result.string(), result.size());
1058 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask);
1059
1060 dumpBase(fd, args);
1061}
1062
1063// Thread virtuals
1064status_t AudioFlinger::PlaybackThread::readyToRun()
1065{
1066 status_t status = initCheck();
1067 if (status == NO_ERROR) {
1068 ALOGI("AudioFlinger's thread %p ready to run", this);
1069 } else {
1070 ALOGE("No working audio driver found.");
1071 }
1072 return status;
1073}
1074
1075void AudioFlinger::PlaybackThread::onFirstRef()
1076{
1077 run(mName, ANDROID_PRIORITY_URGENT_AUDIO);
1078}
1079
1080// ThreadBase virtuals
1081void AudioFlinger::PlaybackThread::preExit()
1082{
1083 ALOGV(" preExit()");
1084 // FIXME this is using hard-coded strings but in the future, this functionality will be
1085 // converted to use audio HAL extensions required to support tunneling
1086 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1");
1087}
1088
1089// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1090sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
1091 const sp<AudioFlinger::Client>& client,
1092 audio_stream_type_t streamType,
1093 uint32_t sampleRate,
1094 audio_format_t format,
1095 audio_channel_mask_t channelMask,
1096 size_t frameCount,
1097 const sp<IMemory>& sharedBuffer,
1098 int sessionId,
1099 IAudioFlinger::track_flags_t *flags,
1100 pid_t tid,
1101 status_t *status)
1102{
1103 sp<Track> track;
1104 status_t lStatus;
1105
1106 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0;
1107
1108 // client expresses a preference for FAST, but we get the final say
1109 if (*flags & IAudioFlinger::TRACK_FAST) {
1110 if (
1111 // not timed
1112 (!isTimed) &&
1113 // either of these use cases:
1114 (
1115 // use case 1: shared buffer with any frame count
1116 (
1117 (sharedBuffer != 0)
1118 ) ||
1119 // use case 2: callback handler and frame count is default or at least as large as HAL
1120 (
1121 (tid != -1) &&
1122 ((frameCount == 0) ||
1123 (frameCount >= (mFrameCount * kFastTrackMultiplier)))
1124 )
1125 ) &&
1126 // PCM data
1127 audio_is_linear_pcm(format) &&
1128 // mono or stereo
1129 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
1130 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
1131#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
1132 // hardware sample rate
1133 (sampleRate == mSampleRate) &&
1134#endif
1135 // normal mixer has an associated fast mixer
1136 hasFastMixer() &&
1137 // there are sufficient fast track slots available
1138 (mFastTrackAvailMask != 0)
1139 // FIXME test that MixerThread for this fast track has a capable output HAL
1140 // FIXME add a permission test also?
1141 ) {
1142 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count
1143 if (frameCount == 0) {
1144 frameCount = mFrameCount * kFastTrackMultiplier;
1145 }
1146 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
1147 frameCount, mFrameCount);
1148 } else {
1149 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
1150 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
1151 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
1152 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
1153 audio_is_linear_pcm(format),
1154 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
1155 *flags &= ~IAudioFlinger::TRACK_FAST;
1156 // For compatibility with AudioTrack calculation, buffer depth is forced
1157 // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
1158 // This is probably too conservative, but legacy application code may depend on it.
1159 // If you change this calculation, also review the start threshold which is related.
1160 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
1161 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
1162 if (minBufCount < 2) {
1163 minBufCount = 2;
1164 }
1165 size_t minFrameCount = mNormalFrameCount * minBufCount;
1166 if (frameCount < minFrameCount) {
1167 frameCount = minFrameCount;
1168 }
1169 }
1170 }
1171
1172 if (mType == DIRECT) {
1173 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
1174 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
1175 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
1176 "for output %p with format %d",
1177 sampleRate, format, channelMask, mOutput, mFormat);
1178 lStatus = BAD_VALUE;
1179 goto Exit;
1180 }
1181 }
1182 } else {
1183 // Resampler implementation limits input sampling rate to 2 x output sampling rate.
1184 if (sampleRate > mSampleRate*2) {
1185 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
1186 lStatus = BAD_VALUE;
1187 goto Exit;
1188 }
1189 }
1190
1191 lStatus = initCheck();
1192 if (lStatus != NO_ERROR) {
1193 ALOGE("Audio driver not initialized.");
1194 goto Exit;
1195 }
1196
1197 { // scope for mLock
1198 Mutex::Autolock _l(mLock);
Glenn Kasten3051df22013-02-12 12:12:42 -08001199 mNBLogWriter->logf("createTrack_l isFast=%d caller=%d",
1200 (*flags & IAudioFlinger::TRACK_FAST) != 0, IPCThreadState::self()->getCallingPid());
Eric Laurent81784c32012-11-19 14:55:58 -08001201
1202 // all tracks in same audio session must share the same routing strategy otherwise
1203 // conflicts will happen when tracks are moved from one output to another by audio policy
1204 // manager
1205 uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
1206 for (size_t i = 0; i < mTracks.size(); ++i) {
1207 sp<Track> t = mTracks[i];
1208 if (t != 0 && !t->isOutputTrack()) {
1209 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
1210 if (sessionId == t->sessionId() && strategy != actual) {
1211 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1212 strategy, actual);
1213 lStatus = BAD_VALUE;
1214 goto Exit;
1215 }
1216 }
1217 }
1218
1219 if (!isTimed) {
1220 track = new Track(this, client, streamType, sampleRate, format,
1221 channelMask, frameCount, sharedBuffer, sessionId, *flags);
1222 } else {
1223 track = TimedTrack::create(this, client, streamType, sampleRate, format,
1224 channelMask, frameCount, sharedBuffer, sessionId);
1225 }
1226 if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
1227 lStatus = NO_MEMORY;
1228 goto Exit;
1229 }
1230 mTracks.add(track);
1231
1232 sp<EffectChain> chain = getEffectChain_l(sessionId);
1233 if (chain != 0) {
1234 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1235 track->setMainBuffer(chain->inBuffer());
1236 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
1237 chain->incTrackCnt();
1238 }
1239
1240 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
1241 pid_t callingPid = IPCThreadState::self()->getCallingPid();
1242 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
1243 // so ask activity manager to do this on our behalf
1244 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
1245 }
1246 }
1247
1248 lStatus = NO_ERROR;
1249
1250Exit:
1251 if (status) {
1252 *status = lStatus;
1253 }
1254 return track;
1255}
1256
1257uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
1258{
1259 return latency;
1260}
1261
1262uint32_t AudioFlinger::PlaybackThread::latency() const
1263{
1264 Mutex::Autolock _l(mLock);
1265 return latency_l();
1266}
1267uint32_t AudioFlinger::PlaybackThread::latency_l() const
1268{
1269 if (initCheck() == NO_ERROR) {
1270 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream));
1271 } else {
1272 return 0;
1273 }
1274}
1275
1276void AudioFlinger::PlaybackThread::setMasterVolume(float value)
1277{
1278 Mutex::Autolock _l(mLock);
1279 // Don't apply master volume in SW if our HAL can do it for us.
1280 if (mOutput && mOutput->audioHwDev &&
1281 mOutput->audioHwDev->canSetMasterVolume()) {
1282 mMasterVolume = 1.0;
1283 } else {
1284 mMasterVolume = value;
1285 }
1286}
1287
1288void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1289{
1290 Mutex::Autolock _l(mLock);
1291 // Don't apply master mute in SW if our HAL can do it for us.
1292 if (mOutput && mOutput->audioHwDev &&
1293 mOutput->audioHwDev->canSetMasterMute()) {
1294 mMasterMute = false;
1295 } else {
1296 mMasterMute = muted;
1297 }
1298}
1299
1300void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
1301{
1302 Mutex::Autolock _l(mLock);
1303 mStreamTypes[stream].volume = value;
1304}
1305
1306void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
1307{
1308 Mutex::Autolock _l(mLock);
1309 mStreamTypes[stream].mute = muted;
1310}
1311
1312float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
1313{
1314 Mutex::Autolock _l(mLock);
1315 return mStreamTypes[stream].volume;
1316}
1317
1318// addTrack_l() must be called with ThreadBase::mLock held
1319status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
1320{
Glenn Kasten3051df22013-02-12 12:12:42 -08001321 mNBLogWriter->logf("addTrack_l mName=%d mFastIndex=%d caller=%d", track->mName,
1322 track->mFastIndex, IPCThreadState::self()->getCallingPid());
Eric Laurent81784c32012-11-19 14:55:58 -08001323 status_t status = ALREADY_EXISTS;
1324
1325 // set retry count for buffer fill
1326 track->mRetryCount = kMaxTrackStartupRetries;
1327 if (mActiveTracks.indexOf(track) < 0) {
1328 // the track is newly added, make sure it fills up all its
1329 // buffers before playing. This is to ensure the client will
1330 // effectively get the latency it requested.
1331 track->mFillingUpStatus = Track::FS_FILLING;
1332 track->mResetDone = false;
1333 track->mPresentationCompleteFrames = 0;
1334 mActiveTracks.add(track);
1335 if (track->mainBuffer() != mMixBuffer) {
1336 sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1337 if (chain != 0) {
1338 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
1339 track->sessionId());
1340 chain->incActiveTrackCnt();
1341 }
1342 }
1343
1344 status = NO_ERROR;
1345 }
1346
1347 ALOGV("mWaitWorkCV.broadcast");
1348 mWaitWorkCV.broadcast();
1349
1350 return status;
1351}
1352
1353// destroyTrack_l() must be called with ThreadBase::mLock held
1354void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
1355{
Glenn Kasten3051df22013-02-12 12:12:42 -08001356 mNBLogWriter->logTimestamp();
1357 mNBLogWriter->logf("destroyTrack_l mName=%d mFastIndex=%d mClientPid=%d", track->mName,
1358 track->mFastIndex, track->mClient != 0 ? track->mClient->pid() : -1);
Eric Laurent81784c32012-11-19 14:55:58 -08001359 track->mState = TrackBase::TERMINATED;
1360 // active tracks are removed by threadLoop()
1361 if (mActiveTracks.indexOf(track) < 0) {
1362 removeTrack_l(track);
1363 }
1364}
1365
1366void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
1367{
Glenn Kasten3051df22013-02-12 12:12:42 -08001368 mNBLogWriter->logTimestamp();
1369 mNBLogWriter->logf("removeTrack_l mName=%d mFastIndex=%d clientPid=%d", track->mName,
1370 track->mFastIndex, track->mClient != 0 ? track->mClient->pid() : -1);
Eric Laurent81784c32012-11-19 14:55:58 -08001371 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
1372 mTracks.remove(track);
1373 deleteTrackName_l(track->name());
1374 // redundant as track is about to be destroyed, for dumpsys only
1375 track->mName = -1;
1376 if (track->isFastTrack()) {
1377 int index = track->mFastIndex;
1378 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks);
1379 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
1380 mFastTrackAvailMask |= 1 << index;
1381 // redundant as track is about to be destroyed, for dumpsys only
1382 track->mFastIndex = -1;
1383 }
1384 sp<EffectChain> chain = getEffectChain_l(track->sessionId());
1385 if (chain != 0) {
1386 chain->decTrackCnt();
1387 }
1388}
1389
1390String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
1391{
1392 String8 out_s8 = String8("");
1393 char *s;
1394
1395 Mutex::Autolock _l(mLock);
1396 if (initCheck() != NO_ERROR) {
1397 return out_s8;
1398 }
1399
1400 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
1401 out_s8 = String8(s);
1402 free(s);
1403 return out_s8;
1404}
1405
1406// audioConfigChanged_l() must be called with AudioFlinger::mLock held
1407void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
1408 AudioSystem::OutputDescriptor desc;
1409 void *param2 = NULL;
1410
1411 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event,
1412 param);
1413
1414 switch (event) {
1415 case AudioSystem::OUTPUT_OPENED:
1416 case AudioSystem::OUTPUT_CONFIG_CHANGED:
1417 desc.channels = mChannelMask;
1418 desc.samplingRate = mSampleRate;
1419 desc.format = mFormat;
1420 desc.frameCount = mNormalFrameCount; // FIXME see
1421 // AudioFlinger::frameCount(audio_io_handle_t)
1422 desc.latency = latency();
1423 param2 = &desc;
1424 break;
1425
1426 case AudioSystem::STREAM_CONFIG_CHANGED:
1427 param2 = &param;
1428 case AudioSystem::OUTPUT_CLOSED:
1429 default:
1430 break;
1431 }
1432 mAudioFlinger->audioConfigChanged_l(event, mId, param2);
1433}
1434
1435void AudioFlinger::PlaybackThread::readOutputParameters()
1436{
1437 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
1438 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
1439 mChannelCount = (uint16_t)popcount(mChannelMask);
1440 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
1441 mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
1442 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
1443 if (mFrameCount & 15) {
1444 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
1445 mFrameCount);
1446 }
1447
1448 // Calculate size of normal mix buffer relative to the HAL output buffer size
1449 double multiplier = 1.0;
1450 if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
1451 kUseFastMixer == FastMixer_Dynamic)) {
1452 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000;
1453 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000;
1454 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
1455 minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
1456 maxNormalFrameCount = maxNormalFrameCount & ~15;
1457 if (maxNormalFrameCount < minNormalFrameCount) {
1458 maxNormalFrameCount = minNormalFrameCount;
1459 }
1460 multiplier = (double) minNormalFrameCount / (double) mFrameCount;
1461 if (multiplier <= 1.0) {
1462 multiplier = 1.0;
1463 } else if (multiplier <= 2.0) {
1464 if (2 * mFrameCount <= maxNormalFrameCount) {
1465 multiplier = 2.0;
1466 } else {
1467 multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
1468 }
1469 } else {
1470 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL
1471 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast
1472 // track, but we sometimes have to do this to satisfy the maximum frame count
1473 // constraint)
1474 // FIXME this rounding up should not be done if no HAL SRC
1475 uint32_t truncMult = (uint32_t) multiplier;
1476 if ((truncMult & 1)) {
1477 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) {
1478 ++truncMult;
1479 }
1480 }
1481 multiplier = (double) truncMult;
1482 }
1483 }
1484 mNormalFrameCount = multiplier * mFrameCount;
1485 // round up to nearest 16 frames to satisfy AudioMixer
1486 mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
1487 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
1488 mNormalFrameCount);
1489
1490 delete[] mMixBuffer;
1491 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount];
1492 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
1493
1494 // force reconfiguration of effect chains and engines to take new buffer size and audio
1495 // parameters into account
1496 // Note that mLock is not held when readOutputParameters() is called from the constructor
1497 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
1498 // matter.
1499 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
1500 Vector< sp<EffectChain> > effectChains = mEffectChains;
1501 for (size_t i = 0; i < effectChains.size(); i ++) {
1502 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
1503 }
1504}
1505
1506
1507status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames)
1508{
1509 if (halFrames == NULL || dspFrames == NULL) {
1510 return BAD_VALUE;
1511 }
1512 Mutex::Autolock _l(mLock);
1513 if (initCheck() != NO_ERROR) {
1514 return INVALID_OPERATION;
1515 }
1516 size_t framesWritten = mBytesWritten / mFrameSize;
1517 *halFrames = framesWritten;
1518
1519 if (isSuspended()) {
1520 // return an estimation of rendered frames when the output is suspended
1521 size_t latencyFrames = (latency_l() * mSampleRate) / 1000;
1522 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0;
1523 return NO_ERROR;
1524 } else {
1525 return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
1526 }
1527}
1528
1529uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const
1530{
1531 Mutex::Autolock _l(mLock);
1532 uint32_t result = 0;
1533 if (getEffectChain_l(sessionId) != 0) {
1534 result = EFFECT_SESSION;
1535 }
1536
1537 for (size_t i = 0; i < mTracks.size(); ++i) {
1538 sp<Track> track = mTracks[i];
Glenn Kasten5736c352012-12-04 12:12:34 -08001539 if (sessionId == track->sessionId() && !track->isInvalid()) {
Eric Laurent81784c32012-11-19 14:55:58 -08001540 result |= TRACK_SESSION;
1541 break;
1542 }
1543 }
1544
1545 return result;
1546}
1547
1548uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
1549{
1550 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
1551 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
1552 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1553 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1554 }
1555 for (size_t i = 0; i < mTracks.size(); i++) {
1556 sp<Track> track = mTracks[i];
Glenn Kasten5736c352012-12-04 12:12:34 -08001557 if (sessionId == track->sessionId() && !track->isInvalid()) {
Eric Laurent81784c32012-11-19 14:55:58 -08001558 return AudioSystem::getStrategyForStream(track->streamType());
1559 }
1560 }
1561 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
1562}
1563
1564
1565AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
1566{
1567 Mutex::Autolock _l(mLock);
1568 return mOutput;
1569}
1570
1571AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
1572{
1573 Mutex::Autolock _l(mLock);
1574 AudioStreamOut *output = mOutput;
1575 mOutput = NULL;
1576 // FIXME FastMixer might also have a raw ptr to mOutputSink;
1577 // must push a NULL and wait for ack
1578 mOutputSink.clear();
1579 mPipeSink.clear();
1580 mNormalSink.clear();
1581 return output;
1582}
1583
1584// this method must always be called either with ThreadBase mLock held or inside the thread loop
1585audio_stream_t* AudioFlinger::PlaybackThread::stream() const
1586{
1587 if (mOutput == NULL) {
1588 return NULL;
1589 }
1590 return &mOutput->stream->common;
1591}
1592
1593uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
1594{
1595 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
1596}
1597
1598status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
1599{
1600 if (!isValidSyncEvent(event)) {
1601 return BAD_VALUE;
1602 }
1603
1604 Mutex::Autolock _l(mLock);
1605
1606 for (size_t i = 0; i < mTracks.size(); ++i) {
1607 sp<Track> track = mTracks[i];
1608 if (event->triggerSession() == track->sessionId()) {
1609 (void) track->setSyncEvent(event);
1610 return NO_ERROR;
1611 }
1612 }
1613
1614 return NAME_NOT_FOUND;
1615}
1616
1617bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
1618{
1619 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
1620}
1621
1622void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
1623 const Vector< sp<Track> >& tracksToRemove)
1624{
1625 size_t count = tracksToRemove.size();
1626 if (CC_UNLIKELY(count)) {
1627 for (size_t i = 0 ; i < count ; i++) {
1628 const sp<Track>& track = tracksToRemove.itemAt(i);
1629 if ((track->sharedBuffer() != 0) &&
1630 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) {
1631 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId());
1632 }
1633 }
1634 }
1635
1636}
1637
1638void AudioFlinger::PlaybackThread::checkSilentMode_l()
1639{
1640 if (!mMasterMute) {
1641 char value[PROPERTY_VALUE_MAX];
1642 if (property_get("ro.audio.silent", value, "0") > 0) {
1643 char *endptr;
1644 unsigned long ul = strtoul(value, &endptr, 0);
1645 if (*endptr == '\0' && ul != 0) {
1646 ALOGD("Silence is golden");
1647 // The setprop command will not allow a property to be changed after
1648 // the first time it is set, so we don't have to worry about un-muting.
1649 setMasterMute_l(true);
1650 }
1651 }
1652 }
1653}
1654
1655// shared by MIXER and DIRECT, overridden by DUPLICATING
1656void AudioFlinger::PlaybackThread::threadLoop_write()
1657{
1658 // FIXME rewrite to reduce number of system calls
1659 mLastWriteTime = systemTime();
1660 mInWrite = true;
1661 int bytesWritten;
1662
1663 // If an NBAIO sink is present, use it to write the normal mixer's submix
1664 if (mNormalSink != 0) {
1665#define mBitShift 2 // FIXME
1666 size_t count = mixBufferSize >> mBitShift;
Simon Wilson2d590962012-11-29 15:18:50 -08001667 ATRACE_BEGIN("write");
Eric Laurent81784c32012-11-19 14:55:58 -08001668 // update the setpoint when AudioFlinger::mScreenState changes
1669 uint32_t screenState = AudioFlinger::mScreenState;
1670 if (screenState != mScreenState) {
1671 mScreenState = screenState;
1672 MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
1673 if (pipe != NULL) {
1674 pipe->setAvgFrames((mScreenState & 1) ?
1675 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
1676 }
1677 }
1678 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count);
Simon Wilson2d590962012-11-29 15:18:50 -08001679 ATRACE_END();
Eric Laurent81784c32012-11-19 14:55:58 -08001680 if (framesWritten > 0) {
1681 bytesWritten = framesWritten << mBitShift;
1682 } else {
1683 bytesWritten = framesWritten;
1684 }
1685 // otherwise use the HAL / AudioStreamOut directly
1686 } else {
1687 // Direct output thread.
1688 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
1689 }
1690
1691 if (bytesWritten > 0) {
1692 mBytesWritten += mixBufferSize;
1693 }
1694 mNumWrites++;
1695 mInWrite = false;
1696}
1697
1698/*
1699The derived values that are cached:
1700 - mixBufferSize from frame count * frame size
1701 - activeSleepTime from activeSleepTimeUs()
1702 - idleSleepTime from idleSleepTimeUs()
1703 - standbyDelay from mActiveSleepTimeUs (DIRECT only)
1704 - maxPeriod from frame count and sample rate (MIXER only)
1705
1706The parameters that affect these derived values are:
1707 - frame count
1708 - frame size
1709 - sample rate
1710 - device type: A2DP or not
1711 - device latency
1712 - format: PCM or not
1713 - active sleep time
1714 - idle sleep time
1715*/
1716
1717void AudioFlinger::PlaybackThread::cacheParameters_l()
1718{
1719 mixBufferSize = mNormalFrameCount * mFrameSize;
1720 activeSleepTime = activeSleepTimeUs();
1721 idleSleepTime = idleSleepTimeUs();
1722}
1723
1724void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
1725{
1726 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
1727 this, streamType, mTracks.size());
1728 Mutex::Autolock _l(mLock);
1729
1730 size_t size = mTracks.size();
1731 for (size_t i = 0; i < size; i++) {
1732 sp<Track> t = mTracks[i];
1733 if (t->streamType() == streamType) {
Glenn Kasten5736c352012-12-04 12:12:34 -08001734 t->invalidate();
Eric Laurent81784c32012-11-19 14:55:58 -08001735 }
1736 }
1737}
1738
1739status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
1740{
1741 int session = chain->sessionId();
1742 int16_t *buffer = mMixBuffer;
1743 bool ownsBuffer = false;
1744
1745 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
1746 if (session > 0) {
1747 // Only one effect chain can be present in direct output thread and it uses
1748 // the mix buffer as input
1749 if (mType != DIRECT) {
1750 size_t numSamples = mNormalFrameCount * mChannelCount;
1751 buffer = new int16_t[numSamples];
1752 memset(buffer, 0, numSamples * sizeof(int16_t));
1753 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
1754 ownsBuffer = true;
1755 }
1756
1757 // Attach all tracks with same session ID to this chain.
1758 for (size_t i = 0; i < mTracks.size(); ++i) {
1759 sp<Track> track = mTracks[i];
1760 if (session == track->sessionId()) {
1761 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(),
1762 buffer);
1763 track->setMainBuffer(buffer);
1764 chain->incTrackCnt();
1765 }
1766 }
1767
1768 // indicate all active tracks in the chain
1769 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
1770 sp<Track> track = mActiveTracks[i].promote();
1771 if (track == 0) {
1772 continue;
1773 }
1774 if (session == track->sessionId()) {
1775 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
1776 chain->incActiveTrackCnt();
1777 }
1778 }
1779 }
1780
1781 chain->setInBuffer(buffer, ownsBuffer);
1782 chain->setOutBuffer(mMixBuffer);
1783 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
1784 // chains list in order to be processed last as it contains output stage effects
1785 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
1786 // session AUDIO_SESSION_OUTPUT_STAGE to be processed
1787 // after track specific effects and before output stage
1788 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
1789 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX
1790 // Effect chain for other sessions are inserted at beginning of effect
1791 // chains list to be processed before output mix effects. Relative order between other
1792 // sessions is not important
1793 size_t size = mEffectChains.size();
1794 size_t i = 0;
1795 for (i = 0; i < size; i++) {
1796 if (mEffectChains[i]->sessionId() < session) {
1797 break;
1798 }
1799 }
1800 mEffectChains.insertAt(chain, i);
1801 checkSuspendOnAddEffectChain_l(chain);
1802
1803 return NO_ERROR;
1804}
1805
1806size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
1807{
1808 int session = chain->sessionId();
1809
1810 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
1811
1812 for (size_t i = 0; i < mEffectChains.size(); i++) {
1813 if (chain == mEffectChains[i]) {
1814 mEffectChains.removeAt(i);
1815 // detach all active tracks from the chain
1816 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
1817 sp<Track> track = mActiveTracks[i].promote();
1818 if (track == 0) {
1819 continue;
1820 }
1821 if (session == track->sessionId()) {
1822 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
1823 chain.get(), session);
1824 chain->decActiveTrackCnt();
1825 }
1826 }
1827
1828 // detach all tracks with same session ID from this chain
1829 for (size_t i = 0; i < mTracks.size(); ++i) {
1830 sp<Track> track = mTracks[i];
1831 if (session == track->sessionId()) {
1832 track->setMainBuffer(mMixBuffer);
1833 chain->decTrackCnt();
1834 }
1835 }
1836 break;
1837 }
1838 }
1839 return mEffectChains.size();
1840}
1841
1842status_t AudioFlinger::PlaybackThread::attachAuxEffect(
1843 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
1844{
1845 Mutex::Autolock _l(mLock);
1846 return attachAuxEffect_l(track, EffectId);
1847}
1848
1849status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
1850 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
1851{
1852 status_t status = NO_ERROR;
1853
1854 if (EffectId == 0) {
1855 track->setAuxBuffer(0, NULL);
1856 } else {
1857 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
1858 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
1859 if (effect != 0) {
1860 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
1861 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
1862 } else {
1863 status = INVALID_OPERATION;
1864 }
1865 } else {
1866 status = BAD_VALUE;
1867 }
1868 }
1869 return status;
1870}
1871
1872void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
1873{
1874 for (size_t i = 0; i < mTracks.size(); ++i) {
1875 sp<Track> track = mTracks[i];
1876 if (track->auxEffectId() == effectId) {
1877 attachAuxEffect_l(track, 0);
1878 }
1879 }
1880}
1881
1882bool AudioFlinger::PlaybackThread::threadLoop()
1883{
1884 Vector< sp<Track> > tracksToRemove;
1885
1886 standbyTime = systemTime();
1887
1888 // MIXER
1889 nsecs_t lastWarning = 0;
1890
1891 // DUPLICATING
1892 // FIXME could this be made local to while loop?
1893 writeFrames = 0;
1894
1895 cacheParameters_l();
1896 sleepTime = idleSleepTime;
1897
1898 if (mType == MIXER) {
1899 sleepTimeShift = 0;
1900 }
1901
1902 CpuStats cpuStats;
1903 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
1904
1905 acquireWakeLock();
1906
Glenn Kasten9e58b552013-01-18 15:09:48 -08001907 // mNBLogWriter->log can only be called while thread mutex mLock is held.
1908 // So if you need to log when mutex is unlocked, set logString to a non-NULL string,
1909 // and then that string will be logged at the next convenient opportunity.
1910 const char *logString = NULL;
1911
Eric Laurent81784c32012-11-19 14:55:58 -08001912 while (!exitPending())
1913 {
1914 cpuStats.sample(myName);
1915
1916 Vector< sp<EffectChain> > effectChains;
1917
1918 processConfigEvents();
1919
1920 { // scope for mLock
1921
1922 Mutex::Autolock _l(mLock);
1923
Glenn Kasten9e58b552013-01-18 15:09:48 -08001924 if (logString != NULL) {
1925 mNBLogWriter->logTimestamp();
1926 mNBLogWriter->log(logString);
1927 logString = NULL;
1928 }
1929
Eric Laurent81784c32012-11-19 14:55:58 -08001930 if (checkForNewParameters_l()) {
1931 cacheParameters_l();
1932 }
1933
1934 saveOutputTracks();
1935
1936 // put audio hardware into standby after short delay
1937 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
1938 isSuspended())) {
1939 if (!mStandby) {
1940
1941 threadLoop_standby();
1942
Glenn Kasten9e58b552013-01-18 15:09:48 -08001943 mNBLogWriter->log("standby");
Eric Laurent81784c32012-11-19 14:55:58 -08001944 mStandby = true;
1945 }
1946
1947 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
1948 // we're about to wait, flush the binder command buffer
1949 IPCThreadState::self()->flushCommands();
1950
1951 clearOutputTracks();
1952
1953 if (exitPending()) {
1954 break;
1955 }
1956
1957 releaseWakeLock_l();
1958 // wait until we have something to do...
1959 ALOGV("%s going to sleep", myName.string());
1960 mWaitWorkCV.wait(mLock);
1961 ALOGV("%s waking up", myName.string());
1962 acquireWakeLock_l();
1963
1964 mMixerStatus = MIXER_IDLE;
1965 mMixerStatusIgnoringFastTracks = MIXER_IDLE;
1966 mBytesWritten = 0;
1967
1968 checkSilentMode_l();
1969
1970 standbyTime = systemTime() + standbyDelay;
1971 sleepTime = idleSleepTime;
1972 if (mType == MIXER) {
1973 sleepTimeShift = 0;
1974 }
1975
1976 continue;
1977 }
1978 }
1979
1980 // mMixerStatusIgnoringFastTracks is also updated internally
1981 mMixerStatus = prepareTracks_l(&tracksToRemove);
1982
1983 // prevent any changes in effect chain list and in each effect chain
1984 // during mixing and effect process as the audio buffers could be deleted
1985 // or modified if an effect is created or deleted
1986 lockEffectChains_l(effectChains);
1987 }
1988
1989 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) {
1990 threadLoop_mix();
1991 } else {
1992 threadLoop_sleepTime();
1993 }
1994
1995 if (isSuspended()) {
1996 sleepTime = suspendSleepTimeUs();
1997 mBytesWritten += mixBufferSize;
1998 }
1999
2000 // only process effects if we're going to write
2001 if (sleepTime == 0) {
2002 for (size_t i = 0; i < effectChains.size(); i ++) {
2003 effectChains[i]->process_l();
2004 }
2005 }
2006
2007 // enable changes in effect chain
2008 unlockEffectChains(effectChains);
2009
2010 // sleepTime == 0 means we must write to audio hardware
2011 if (sleepTime == 0) {
2012
2013 threadLoop_write();
2014
2015if (mType == MIXER) {
2016 // write blocked detection
2017 nsecs_t now = systemTime();
2018 nsecs_t delta = now - mLastWriteTime;
2019 if (!mStandby && delta > maxPeriod) {
2020 mNumDelayedWrites++;
2021 if ((now - lastWarning) > kWarningThrottleNs) {
Alex Ray371eb972012-11-30 11:11:54 -08002022 ATRACE_NAME("underrun");
Eric Laurent81784c32012-11-19 14:55:58 -08002023 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
2024 ns2ms(delta), mNumDelayedWrites, this);
2025 lastWarning = now;
2026 }
2027 }
2028}
2029
2030 mStandby = false;
2031 } else {
2032 usleep(sleepTime);
2033 }
2034
2035 // Finally let go of removed track(s), without the lock held
2036 // since we can't guarantee the destructors won't acquire that
2037 // same lock. This will also mutate and push a new fast mixer state.
2038 threadLoop_removeTracks(tracksToRemove);
Glenn Kasten9e58b552013-01-18 15:09:48 -08002039 if (tracksToRemove.size() > 0) {
2040 logString = "remove";
2041 }
Eric Laurent81784c32012-11-19 14:55:58 -08002042 tracksToRemove.clear();
2043
2044 // FIXME I don't understand the need for this here;
2045 // it was in the original code but maybe the
2046 // assignment in saveOutputTracks() makes this unnecessary?
2047 clearOutputTracks();
2048
2049 // Effect chains will be actually deleted here if they were removed from
2050 // mEffectChains list during mixing or effects processing
2051 effectChains.clear();
2052
2053 // FIXME Note that the above .clear() is no longer necessary since effectChains
2054 // is now local to this block, but will keep it for now (at least until merge done).
2055 }
2056
2057 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ...
2058 if (mType == MIXER || mType == DIRECT) {
2059 // put output stream into standby mode
2060 if (!mStandby) {
2061 mOutput->stream->common.standby(&mOutput->stream->common);
2062 }
2063 }
2064
2065 releaseWakeLock();
2066
2067 ALOGV("Thread %p type %d exiting", this, mType);
2068 return false;
2069}
2070
2071
2072// ----------------------------------------------------------------------------
2073
2074AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
2075 audio_io_handle_t id, audio_devices_t device, type_t type)
2076 : PlaybackThread(audioFlinger, output, id, device, type),
2077 // mAudioMixer below
2078 // mFastMixer below
2079 mFastMixerFutex(0)
2080 // mOutputSink below
2081 // mPipeSink below
2082 // mNormalSink below
2083{
2084 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
2085 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, "
2086 "mFrameCount=%d, mNormalFrameCount=%d",
2087 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
2088 mNormalFrameCount);
2089 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
2090
2091 // FIXME - Current mixer implementation only supports stereo output
2092 if (mChannelCount != FCC_2) {
2093 ALOGE("Invalid audio hardware channel count %d", mChannelCount);
2094 }
2095
2096 // create an NBAIO sink for the HAL output stream, and negotiate
2097 mOutputSink = new AudioStreamOutSink(output->stream);
2098 size_t numCounterOffers = 0;
2099 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)};
2100 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
2101 ALOG_ASSERT(index == 0);
2102
2103 // initialize fast mixer depending on configuration
2104 bool initFastMixer;
2105 switch (kUseFastMixer) {
2106 case FastMixer_Never:
2107 initFastMixer = false;
2108 break;
2109 case FastMixer_Always:
2110 initFastMixer = true;
2111 break;
2112 case FastMixer_Static:
2113 case FastMixer_Dynamic:
2114 initFastMixer = mFrameCount < mNormalFrameCount;
2115 break;
2116 }
2117 if (initFastMixer) {
2118
2119 // create a MonoPipe to connect our submix to FastMixer
2120 NBAIO_Format format = mOutputSink->format();
2121 // This pipe depth compensates for scheduling latency of the normal mixer thread.
2122 // When it wakes up after a maximum latency, it runs a few cycles quickly before
2123 // finally blocking. Note the pipe implementation rounds up the request to a power of 2.
2124 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
2125 const NBAIO_Format offers[1] = {format};
2126 size_t numCounterOffers = 0;
2127 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
2128 ALOG_ASSERT(index == 0);
2129 monoPipe->setAvgFrames((mScreenState & 1) ?
2130 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
2131 mPipeSink = monoPipe;
2132
2133#ifdef TEE_SINK_FRAMES
2134 // create a Pipe to archive a copy of FastMixer's output for dumpsys
2135 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format);
2136 numCounterOffers = 0;
2137 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers);
2138 ALOG_ASSERT(index == 0);
2139 mTeeSink = teeSink;
2140 PipeReader *teeSource = new PipeReader(*teeSink);
2141 numCounterOffers = 0;
2142 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers);
2143 ALOG_ASSERT(index == 0);
2144 mTeeSource = teeSource;
2145#endif
2146
2147 // create fast mixer and configure it initially with just one fast track for our submix
2148 mFastMixer = new FastMixer();
2149 FastMixerStateQueue *sq = mFastMixer->sq();
2150#ifdef STATE_QUEUE_DUMP
2151 sq->setObserverDump(&mStateQueueObserverDump);
2152 sq->setMutatorDump(&mStateQueueMutatorDump);
2153#endif
2154 FastMixerState *state = sq->begin();
2155 FastTrack *fastTrack = &state->mFastTracks[0];
2156 // wrap the source side of the MonoPipe to make it an AudioBufferProvider
2157 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
2158 fastTrack->mVolumeProvider = NULL;
2159 fastTrack->mGeneration++;
2160 state->mFastTracksGen++;
2161 state->mTrackMask = 1;
2162 // fast mixer will use the HAL output sink
2163 state->mOutputSink = mOutputSink.get();
2164 state->mOutputSinkGen++;
2165 state->mFrameCount = mFrameCount;
2166 state->mCommand = FastMixerState::COLD_IDLE;
2167 // already done in constructor initialization list
2168 //mFastMixerFutex = 0;
2169 state->mColdFutexAddr = &mFastMixerFutex;
2170 state->mColdGen++;
2171 state->mDumpState = &mFastMixerDumpState;
2172 state->mTeeSink = mTeeSink.get();
Glenn Kasten9e58b552013-01-18 15:09:48 -08002173 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
2174 state->mNBLogWriter = mFastMixerNBLogWriter.get();
Eric Laurent81784c32012-11-19 14:55:58 -08002175 sq->end();
2176 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
2177
2178 // start the fast mixer
2179 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
2180 pid_t tid = mFastMixer->getTid();
2181 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
2182 if (err != 0) {
2183 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
2184 kPriorityFastMixer, getpid_cached, tid, err);
2185 }
2186
2187#ifdef AUDIO_WATCHDOG
2188 // create and start the watchdog
2189 mAudioWatchdog = new AudioWatchdog();
2190 mAudioWatchdog->setDump(&mAudioWatchdogDump);
2191 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
2192 tid = mAudioWatchdog->getTid();
2193 err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
2194 if (err != 0) {
2195 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
2196 kPriorityFastMixer, getpid_cached, tid, err);
2197 }
2198#endif
2199
2200 } else {
2201 mFastMixer = NULL;
2202 }
2203
2204 switch (kUseFastMixer) {
2205 case FastMixer_Never:
2206 case FastMixer_Dynamic:
2207 mNormalSink = mOutputSink;
2208 break;
2209 case FastMixer_Always:
2210 mNormalSink = mPipeSink;
2211 break;
2212 case FastMixer_Static:
2213 mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
2214 break;
2215 }
2216}
2217
2218AudioFlinger::MixerThread::~MixerThread()
2219{
2220 if (mFastMixer != NULL) {
2221 FastMixerStateQueue *sq = mFastMixer->sq();
2222 FastMixerState *state = sq->begin();
2223 if (state->mCommand == FastMixerState::COLD_IDLE) {
2224 int32_t old = android_atomic_inc(&mFastMixerFutex);
2225 if (old == -1) {
2226 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
2227 }
2228 }
2229 state->mCommand = FastMixerState::EXIT;
2230 sq->end();
2231 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
2232 mFastMixer->join();
2233 // Though the fast mixer thread has exited, it's state queue is still valid.
2234 // We'll use that extract the final state which contains one remaining fast track
2235 // corresponding to our sub-mix.
2236 state = sq->begin();
2237 ALOG_ASSERT(state->mTrackMask == 1);
2238 FastTrack *fastTrack = &state->mFastTracks[0];
2239 ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
2240 delete fastTrack->mBufferProvider;
2241 sq->end(false /*didModify*/);
2242 delete mFastMixer;
2243#ifdef AUDIO_WATCHDOG
2244 if (mAudioWatchdog != 0) {
2245 mAudioWatchdog->requestExit();
2246 mAudioWatchdog->requestExitAndWait();
2247 mAudioWatchdog.clear();
2248 }
2249#endif
2250 }
Glenn Kasten9e58b552013-01-18 15:09:48 -08002251 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter);
Eric Laurent81784c32012-11-19 14:55:58 -08002252 delete mAudioMixer;
2253}
2254
2255
2256uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
2257{
2258 if (mFastMixer != NULL) {
2259 MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
2260 latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
2261 }
2262 return latency;
2263}
2264
2265
2266void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
2267{
2268 PlaybackThread::threadLoop_removeTracks(tracksToRemove);
2269}
2270
2271void AudioFlinger::MixerThread::threadLoop_write()
2272{
2273 // FIXME we should only do one push per cycle; confirm this is true
2274 // Start the fast mixer if it's not already running
2275 if (mFastMixer != NULL) {
2276 FastMixerStateQueue *sq = mFastMixer->sq();
2277 FastMixerState *state = sq->begin();
2278 if (state->mCommand != FastMixerState::MIX_WRITE &&
2279 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
2280 if (state->mCommand == FastMixerState::COLD_IDLE) {
2281 int32_t old = android_atomic_inc(&mFastMixerFutex);
2282 if (old == -1) {
2283 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
2284 }
2285#ifdef AUDIO_WATCHDOG
2286 if (mAudioWatchdog != 0) {
2287 mAudioWatchdog->resume();
2288 }
2289#endif
2290 }
2291 state->mCommand = FastMixerState::MIX_WRITE;
2292 sq->end();
2293 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
2294 if (kUseFastMixer == FastMixer_Dynamic) {
2295 mNormalSink = mPipeSink;
2296 }
2297 } else {
2298 sq->end(false /*didModify*/);
2299 }
2300 }
2301 PlaybackThread::threadLoop_write();
2302}
2303
2304void AudioFlinger::MixerThread::threadLoop_standby()
2305{
2306 // Idle the fast mixer if it's currently running
2307 if (mFastMixer != NULL) {
2308 FastMixerStateQueue *sq = mFastMixer->sq();
2309 FastMixerState *state = sq->begin();
2310 if (!(state->mCommand & FastMixerState::IDLE)) {
2311 state->mCommand = FastMixerState::COLD_IDLE;
2312 state->mColdFutexAddr = &mFastMixerFutex;
2313 state->mColdGen++;
2314 mFastMixerFutex = 0;
2315 sq->end();
2316 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
2317 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
2318 if (kUseFastMixer == FastMixer_Dynamic) {
2319 mNormalSink = mOutputSink;
2320 }
2321#ifdef AUDIO_WATCHDOG
2322 if (mAudioWatchdog != 0) {
2323 mAudioWatchdog->pause();
2324 }
2325#endif
2326 } else {
2327 sq->end(false /*didModify*/);
2328 }
2329 }
2330 PlaybackThread::threadLoop_standby();
2331}
2332
2333// shared by MIXER and DIRECT, overridden by DUPLICATING
2334void AudioFlinger::PlaybackThread::threadLoop_standby()
2335{
2336 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
2337 mOutput->stream->common.standby(&mOutput->stream->common);
2338}
2339
2340void AudioFlinger::MixerThread::threadLoop_mix()
2341{
2342 // obtain the presentation timestamp of the next output buffer
2343 int64_t pts;
2344 status_t status = INVALID_OPERATION;
2345
2346 if (mNormalSink != 0) {
2347 status = mNormalSink->getNextWriteTimestamp(&pts);
2348 } else {
2349 status = mOutputSink->getNextWriteTimestamp(&pts);
2350 }
2351
2352 if (status != NO_ERROR) {
2353 pts = AudioBufferProvider::kInvalidPTS;
2354 }
2355
2356 // mix buffers...
2357 mAudioMixer->process(pts);
2358 // increase sleep time progressively when application underrun condition clears.
2359 // Only increase sleep time if the mixer is ready for two consecutive times to avoid
2360 // that a steady state of alternating ready/not ready conditions keeps the sleep time
2361 // such that we would underrun the audio HAL.
2362 if ((sleepTime == 0) && (sleepTimeShift > 0)) {
2363 sleepTimeShift--;
2364 }
2365 sleepTime = 0;
2366 standbyTime = systemTime() + standbyDelay;
2367 //TODO: delay standby when effects have a tail
2368}
2369
2370void AudioFlinger::MixerThread::threadLoop_sleepTime()
2371{
2372 // If no tracks are ready, sleep once for the duration of an output
2373 // buffer size, then write 0s to the output
2374 if (sleepTime == 0) {
2375 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
2376 sleepTime = activeSleepTime >> sleepTimeShift;
2377 if (sleepTime < kMinThreadSleepTimeUs) {
2378 sleepTime = kMinThreadSleepTimeUs;
2379 }
2380 // reduce sleep time in case of consecutive application underruns to avoid
2381 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
2382 // duration we would end up writing less data than needed by the audio HAL if
2383 // the condition persists.
2384 if (sleepTimeShift < kMaxThreadSleepTimeShift) {
2385 sleepTimeShift++;
2386 }
2387 } else {
2388 sleepTime = idleSleepTime;
2389 }
2390 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
2391 memset (mMixBuffer, 0, mixBufferSize);
2392 sleepTime = 0;
2393 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
2394 "anticipated start");
2395 }
2396 // TODO add standby time extension fct of effect tail
2397}
2398
2399// prepareTracks_l() must be called with ThreadBase::mLock held
2400AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
2401 Vector< sp<Track> > *tracksToRemove)
2402{
2403
2404 mixer_state mixerStatus = MIXER_IDLE;
2405 // find out which tracks need to be processed
2406 size_t count = mActiveTracks.size();
2407 size_t mixedTracks = 0;
2408 size_t tracksWithEffect = 0;
2409 // counts only _active_ fast tracks
2410 size_t fastTracks = 0;
2411 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
2412
2413 float masterVolume = mMasterVolume;
2414 bool masterMute = mMasterMute;
2415
2416 if (masterMute) {
2417 masterVolume = 0;
2418 }
2419 // Delegate master volume control to effect in output mix effect chain if needed
2420 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
2421 if (chain != 0) {
2422 uint32_t v = (uint32_t)(masterVolume * (1 << 24));
2423 chain->setVolume_l(&v, &v);
2424 masterVolume = (float)((v + (1 << 23)) >> 24);
2425 chain.clear();
2426 }
2427
2428 // prepare a new state to push
2429 FastMixerStateQueue *sq = NULL;
2430 FastMixerState *state = NULL;
2431 bool didModify = false;
2432 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
2433 if (mFastMixer != NULL) {
2434 sq = mFastMixer->sq();
2435 state = sq->begin();
2436 }
2437
2438 for (size_t i=0 ; i<count ; i++) {
2439 sp<Track> t = mActiveTracks[i].promote();
2440 if (t == 0) {
2441 continue;
2442 }
2443
2444 // this const just means the local variable doesn't change
2445 Track* const track = t.get();
2446
2447 // process fast tracks
2448 if (track->isFastTrack()) {
2449
2450 // It's theoretically possible (though unlikely) for a fast track to be created
2451 // and then removed within the same normal mix cycle. This is not a problem, as
2452 // the track never becomes active so it's fast mixer slot is never touched.
2453 // The converse, of removing an (active) track and then creating a new track
2454 // at the identical fast mixer slot within the same normal mix cycle,
2455 // is impossible because the slot isn't marked available until the end of each cycle.
2456 int j = track->mFastIndex;
2457 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks);
2458 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
2459 FastTrack *fastTrack = &state->mFastTracks[j];
2460
2461 // Determine whether the track is currently in underrun condition,
2462 // and whether it had a recent underrun.
2463 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
2464 FastTrackUnderruns underruns = ftDump->mUnderruns;
2465 uint32_t recentFull = (underruns.mBitFields.mFull -
2466 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
2467 uint32_t recentPartial = (underruns.mBitFields.mPartial -
2468 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
2469 uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
2470 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
2471 uint32_t recentUnderruns = recentPartial + recentEmpty;
2472 track->mObservedUnderruns = underruns;
2473 // don't count underruns that occur while stopping or pausing
2474 // or stopped which can occur when flush() is called while active
2475 if (!(track->isStopping() || track->isPausing() || track->isStopped())) {
2476 track->mUnderrunCount += recentUnderruns;
2477 }
2478
2479 // This is similar to the state machine for normal tracks,
2480 // with a few modifications for fast tracks.
2481 bool isActive = true;
2482 switch (track->mState) {
2483 case TrackBase::STOPPING_1:
2484 // track stays active in STOPPING_1 state until first underrun
2485 if (recentUnderruns > 0) {
2486 track->mState = TrackBase::STOPPING_2;
2487 }
2488 break;
2489 case TrackBase::PAUSING:
2490 // ramp down is not yet implemented
2491 track->setPaused();
2492 break;
2493 case TrackBase::RESUMING:
2494 // ramp up is not yet implemented
2495 track->mState = TrackBase::ACTIVE;
2496 break;
2497 case TrackBase::ACTIVE:
2498 if (recentFull > 0 || recentPartial > 0) {
2499 // track has provided at least some frames recently: reset retry count
2500 track->mRetryCount = kMaxTrackRetries;
2501 }
2502 if (recentUnderruns == 0) {
2503 // no recent underruns: stay active
2504 break;
2505 }
2506 // there has recently been an underrun of some kind
2507 if (track->sharedBuffer() == 0) {
2508 // were any of the recent underruns "empty" (no frames available)?
2509 if (recentEmpty == 0) {
2510 // no, then ignore the partial underruns as they are allowed indefinitely
2511 break;
2512 }
2513 // there has recently been an "empty" underrun: decrement the retry counter
2514 if (--(track->mRetryCount) > 0) {
2515 break;
2516 }
2517 // indicate to client process that the track was disabled because of underrun;
2518 // it will then automatically call start() when data is available
2519 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags);
2520 // remove from active list, but state remains ACTIVE [confusing but true]
2521 isActive = false;
2522 break;
2523 }
2524 // fall through
2525 case TrackBase::STOPPING_2:
2526 case TrackBase::PAUSED:
2527 case TrackBase::TERMINATED:
2528 case TrackBase::STOPPED:
2529 case TrackBase::FLUSHED: // flush() while active
2530 // Check for presentation complete if track is inactive
2531 // We have consumed all the buffers of this track.
2532 // This would be incomplete if we auto-paused on underrun
2533 {
2534 size_t audioHALFrames =
2535 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
2536 size_t framesWritten = mBytesWritten / mFrameSize;
2537 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
2538 // track stays in active list until presentation is complete
2539 break;
2540 }
2541 }
2542 if (track->isStopping_2()) {
2543 track->mState = TrackBase::STOPPED;
2544 }
2545 if (track->isStopped()) {
2546 // Can't reset directly, as fast mixer is still polling this track
2547 // track->reset();
2548 // So instead mark this track as needing to be reset after push with ack
2549 resetMask |= 1 << i;
2550 }
2551 isActive = false;
2552 break;
2553 case TrackBase::IDLE:
2554 default:
2555 LOG_FATAL("unexpected track state %d", track->mState);
2556 }
2557
2558 if (isActive) {
2559 // was it previously inactive?
2560 if (!(state->mTrackMask & (1 << j))) {
2561 ExtendedAudioBufferProvider *eabp = track;
2562 VolumeProvider *vp = track;
2563 fastTrack->mBufferProvider = eabp;
2564 fastTrack->mVolumeProvider = vp;
2565 fastTrack->mSampleRate = track->mSampleRate;
2566 fastTrack->mChannelMask = track->mChannelMask;
2567 fastTrack->mGeneration++;
2568 state->mTrackMask |= 1 << j;
2569 didModify = true;
2570 // no acknowledgement required for newly active tracks
2571 }
2572 // cache the combined master volume and stream type volume for fast mixer; this
2573 // lacks any synchronization or barrier so VolumeProvider may read a stale value
Glenn Kastene4756fe2012-11-29 13:38:14 -08002574 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume;
Eric Laurent81784c32012-11-19 14:55:58 -08002575 ++fastTracks;
2576 } else {
2577 // was it previously active?
2578 if (state->mTrackMask & (1 << j)) {
2579 fastTrack->mBufferProvider = NULL;
2580 fastTrack->mGeneration++;
2581 state->mTrackMask &= ~(1 << j);
2582 didModify = true;
2583 // If any fast tracks were removed, we must wait for acknowledgement
2584 // because we're about to decrement the last sp<> on those tracks.
2585 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
2586 } else {
2587 LOG_FATAL("fast track %d should have been active", j);
2588 }
2589 tracksToRemove->add(track);
2590 // Avoids a misleading display in dumpsys
2591 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
2592 }
2593 continue;
2594 }
2595
2596 { // local variable scope to avoid goto warning
2597
2598 audio_track_cblk_t* cblk = track->cblk();
2599
2600 // The first time a track is added we wait
2601 // for all its buffers to be filled before processing it
2602 int name = track->name();
2603 // make sure that we have enough frames to mix one full buffer.
2604 // enforce this condition only once to enable draining the buffer in case the client
2605 // app does not call stop() and relies on underrun to stop:
2606 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
2607 // during last round
2608 uint32_t minFrames = 1;
2609 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
2610 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
2611 if (t->sampleRate() == mSampleRate) {
2612 minFrames = mNormalFrameCount;
2613 } else {
2614 // +1 for rounding and +1 for additional sample needed for interpolation
2615 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
2616 // add frames already consumed but not yet released by the resampler
2617 // because cblk->framesReady() will include these frames
2618 minFrames += mAudioMixer->getUnreleasedFrames(track->name());
2619 // the minimum track buffer size is normally twice the number of frames necessary
2620 // to fill one buffer and the resampler should not leave more than one buffer worth
2621 // of unreleased frames after each pass, but just in case...
Eric Laurent2592f6e2013-01-17 17:36:00 -08002622 ALOG_ASSERT(minFrames <= cblk->frameCount_);
Eric Laurent81784c32012-11-19 14:55:58 -08002623 }
2624 }
2625 if ((track->framesReady() >= minFrames) && track->isReady() &&
2626 !track->isPaused() && !track->isTerminated())
2627 {
2628 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server,
2629 this);
2630
2631 mixedTracks++;
2632
2633 // track->mainBuffer() != mMixBuffer means there is an effect chain
2634 // connected to the track
2635 chain.clear();
2636 if (track->mainBuffer() != mMixBuffer) {
2637 chain = getEffectChain_l(track->sessionId());
2638 // Delegate volume control to effect in track effect chain if needed
2639 if (chain != 0) {
2640 tracksWithEffect++;
2641 } else {
2642 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
2643 "session %d",
2644 name, track->sessionId());
2645 }
2646 }
2647
2648
2649 int param = AudioMixer::VOLUME;
2650 if (track->mFillingUpStatus == Track::FS_FILLED) {
2651 // no ramp for the first volume setting
2652 track->mFillingUpStatus = Track::FS_ACTIVE;
2653 if (track->mState == TrackBase::RESUMING) {
2654 track->mState = TrackBase::ACTIVE;
2655 param = AudioMixer::RAMP_VOLUME;
2656 }
2657 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
2658 } else if (cblk->server != 0) {
2659 // If the track is stopped before the first frame was mixed,
2660 // do not apply ramp
2661 param = AudioMixer::RAMP_VOLUME;
2662 }
2663
2664 // compute volume for this track
2665 uint32_t vl, vr, va;
Glenn Kastene4756fe2012-11-29 13:38:14 -08002666 if (track->isPausing() || mStreamTypes[track->streamType()].mute) {
Eric Laurent81784c32012-11-19 14:55:58 -08002667 vl = vr = va = 0;
2668 if (track->isPausing()) {
2669 track->setPaused();
2670 }
2671 } else {
2672
2673 // read original volumes with volume control
2674 float typeVolume = mStreamTypes[track->streamType()].volume;
2675 float v = masterVolume * typeVolume;
Glenn Kastene3aa6592012-12-04 12:22:46 -08002676 ServerProxy *proxy = track->mServerProxy;
2677 uint32_t vlr = proxy->getVolumeLR();
Eric Laurent81784c32012-11-19 14:55:58 -08002678 vl = vlr & 0xFFFF;
2679 vr = vlr >> 16;
2680 // track volumes come from shared memory, so can't be trusted and must be clamped
2681 if (vl > MAX_GAIN_INT) {
2682 ALOGV("Track left volume out of range: %04X", vl);
2683 vl = MAX_GAIN_INT;
2684 }
2685 if (vr > MAX_GAIN_INT) {
2686 ALOGV("Track right volume out of range: %04X", vr);
2687 vr = MAX_GAIN_INT;
2688 }
2689 // now apply the master volume and stream type volume
2690 vl = (uint32_t)(v * vl) << 12;
2691 vr = (uint32_t)(v * vr) << 12;
2692 // assuming master volume and stream type volume each go up to 1.0,
2693 // vl and vr are now in 8.24 format
2694
Glenn Kastene3aa6592012-12-04 12:22:46 -08002695 uint16_t sendLevel = proxy->getSendLevel_U4_12();
Eric Laurent81784c32012-11-19 14:55:58 -08002696 // send level comes from shared memory and so may be corrupt
2697 if (sendLevel > MAX_GAIN_INT) {
2698 ALOGV("Track send level out of range: %04X", sendLevel);
2699 sendLevel = MAX_GAIN_INT;
2700 }
2701 va = (uint32_t)(v * sendLevel);
2702 }
2703 // Delegate volume control to effect in track effect chain if needed
2704 if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
2705 // Do not ramp volume if volume is controlled by effect
2706 param = AudioMixer::VOLUME;
2707 track->mHasVolumeController = true;
2708 } else {
2709 // force no volume ramp when volume controller was just disabled or removed
2710 // from effect chain to avoid volume spike
2711 if (track->mHasVolumeController) {
2712 param = AudioMixer::VOLUME;
2713 }
2714 track->mHasVolumeController = false;
2715 }
2716
2717 // Convert volumes from 8.24 to 4.12 format
2718 // This additional clamping is needed in case chain->setVolume_l() overshot
2719 vl = (vl + (1 << 11)) >> 12;
2720 if (vl > MAX_GAIN_INT) {
2721 vl = MAX_GAIN_INT;
2722 }
2723 vr = (vr + (1 << 11)) >> 12;
2724 if (vr > MAX_GAIN_INT) {
2725 vr = MAX_GAIN_INT;
2726 }
2727
2728 if (va > MAX_GAIN_INT) {
2729 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for -
2730 }
2731
2732 // XXX: these things DON'T need to be done each time
2733 mAudioMixer->setBufferProvider(name, track);
2734 mAudioMixer->enable(name);
2735
2736 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
2737 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
2738 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
2739 mAudioMixer->setParameter(
2740 name,
2741 AudioMixer::TRACK,
2742 AudioMixer::FORMAT, (void *)track->format());
2743 mAudioMixer->setParameter(
2744 name,
2745 AudioMixer::TRACK,
2746 AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
Glenn Kastene3aa6592012-12-04 12:22:46 -08002747 // limit track sample rate to 2 x output sample rate, which changes at re-configuration
2748 uint32_t maxSampleRate = mSampleRate * 2;
2749 uint32_t reqSampleRate = track->mServerProxy->getSampleRate();
2750 if (reqSampleRate == 0) {
2751 reqSampleRate = mSampleRate;
2752 } else if (reqSampleRate > maxSampleRate) {
2753 reqSampleRate = maxSampleRate;
2754 }
Eric Laurent81784c32012-11-19 14:55:58 -08002755 mAudioMixer->setParameter(
2756 name,
2757 AudioMixer::RESAMPLE,
2758 AudioMixer::SAMPLE_RATE,
Glenn Kastene3aa6592012-12-04 12:22:46 -08002759 (void *)reqSampleRate);
Eric Laurent81784c32012-11-19 14:55:58 -08002760 mAudioMixer->setParameter(
2761 name,
2762 AudioMixer::TRACK,
2763 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
2764 mAudioMixer->setParameter(
2765 name,
2766 AudioMixer::TRACK,
2767 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
2768
2769 // reset retry count
2770 track->mRetryCount = kMaxTrackRetries;
2771
2772 // If one track is ready, set the mixer ready if:
2773 // - the mixer was not ready during previous round OR
2774 // - no other track is not ready
2775 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
2776 mixerStatus != MIXER_TRACKS_ENABLED) {
2777 mixerStatus = MIXER_TRACKS_READY;
2778 }
2779 } else {
2780 // clear effect chain input buffer if an active track underruns to avoid sending
2781 // previous audio buffer again to effects
2782 chain = getEffectChain_l(track->sessionId());
2783 if (chain != 0) {
2784 chain->clearInputBuffer();
2785 }
2786
2787 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user,
2788 cblk->server, this);
2789 if ((track->sharedBuffer() != 0) || track->isTerminated() ||
2790 track->isStopped() || track->isPaused()) {
2791 // We have consumed all the buffers of this track.
2792 // Remove it from the list of active tracks.
2793 // TODO: use actual buffer filling status instead of latency when available from
2794 // audio HAL
2795 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
2796 size_t framesWritten = mBytesWritten / mFrameSize;
2797 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
2798 if (track->isStopped()) {
2799 track->reset();
2800 }
2801 tracksToRemove->add(track);
2802 }
2803 } else {
2804 track->mUnderrunCount++;
2805 // No buffers for this track. Give it a few chances to
2806 // fill a buffer, then remove it from active list.
2807 if (--(track->mRetryCount) <= 0) {
2808 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
2809 tracksToRemove->add(track);
2810 // indicate to client process that the track was disabled because of underrun;
2811 // it will then automatically call start() when data is available
2812 android_atomic_or(CBLK_DISABLED, &cblk->flags);
2813 // If one track is not ready, mark the mixer also not ready if:
2814 // - the mixer was ready during previous round OR
2815 // - no other track is ready
2816 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
2817 mixerStatus != MIXER_TRACKS_READY) {
2818 mixerStatus = MIXER_TRACKS_ENABLED;
2819 }
2820 }
2821 mAudioMixer->disable(name);
2822 }
2823
2824 } // local variable scope to avoid goto warning
2825track_is_ready: ;
2826
2827 }
2828
2829 // Push the new FastMixer state if necessary
2830 bool pauseAudioWatchdog = false;
2831 if (didModify) {
2832 state->mFastTracksGen++;
2833 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
2834 if (kUseFastMixer == FastMixer_Dynamic &&
2835 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
2836 state->mCommand = FastMixerState::COLD_IDLE;
2837 state->mColdFutexAddr = &mFastMixerFutex;
2838 state->mColdGen++;
2839 mFastMixerFutex = 0;
2840 if (kUseFastMixer == FastMixer_Dynamic) {
2841 mNormalSink = mOutputSink;
2842 }
2843 // If we go into cold idle, need to wait for acknowledgement
2844 // so that fast mixer stops doing I/O.
2845 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
2846 pauseAudioWatchdog = true;
2847 }
2848 sq->end();
2849 }
2850 if (sq != NULL) {
2851 sq->end(didModify);
2852 sq->push(block);
2853 }
2854#ifdef AUDIO_WATCHDOG
2855 if (pauseAudioWatchdog && mAudioWatchdog != 0) {
2856 mAudioWatchdog->pause();
2857 }
2858#endif
2859
2860 // Now perform the deferred reset on fast tracks that have stopped
2861 while (resetMask != 0) {
2862 size_t i = __builtin_ctz(resetMask);
2863 ALOG_ASSERT(i < count);
2864 resetMask &= ~(1 << i);
2865 sp<Track> t = mActiveTracks[i].promote();
2866 if (t == 0) {
2867 continue;
2868 }
2869 Track* track = t.get();
2870 ALOG_ASSERT(track->isFastTrack() && track->isStopped());
2871 track->reset();
2872 }
2873
2874 // remove all the tracks that need to be...
2875 count = tracksToRemove->size();
2876 if (CC_UNLIKELY(count)) {
2877 for (size_t i=0 ; i<count ; i++) {
2878 const sp<Track>& track = tracksToRemove->itemAt(i);
Glenn Kasten3051df22013-02-12 12:12:42 -08002879 mNBLogWriter->logTimestamp();
2880 mNBLogWriter->logf("prepareTracks_l remove name=%u mFastIndex=%d", track->name(),
2881 track->mFastIndex);
Eric Laurent81784c32012-11-19 14:55:58 -08002882 mActiveTracks.remove(track);
2883 if (track->mainBuffer() != mMixBuffer) {
2884 chain = getEffectChain_l(track->sessionId());
2885 if (chain != 0) {
2886 ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
2887 track->sessionId());
2888 chain->decActiveTrackCnt();
2889 }
2890 }
2891 if (track->isTerminated()) {
2892 removeTrack_l(track);
2893 }
2894 }
2895 }
2896
2897 // mix buffer must be cleared if all tracks are connected to an
2898 // effect chain as in this case the mixer will not write to
2899 // mix buffer and track effects will accumulate into it
2900 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
2901 (mixedTracks == 0 && fastTracks > 0)) {
2902 // FIXME as a performance optimization, should remember previous zero status
2903 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
2904 }
2905
2906 // if any fast tracks, then status is ready
2907 mMixerStatusIgnoringFastTracks = mixerStatus;
2908 if (fastTracks > 0) {
2909 mixerStatus = MIXER_TRACKS_READY;
2910 }
2911 return mixerStatus;
2912}
2913
2914// getTrackName_l() must be called with ThreadBase::mLock held
2915int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId)
2916{
2917 return mAudioMixer->getTrackName(channelMask, sessionId);
2918}
2919
2920// deleteTrackName_l() must be called with ThreadBase::mLock held
2921void AudioFlinger::MixerThread::deleteTrackName_l(int name)
2922{
2923 ALOGV("remove track (%d) and delete from mixer", name);
2924 mAudioMixer->deleteTrackName(name);
2925}
2926
2927// checkForNewParameters_l() must be called with ThreadBase::mLock held
2928bool AudioFlinger::MixerThread::checkForNewParameters_l()
2929{
2930 // if !&IDLE, holds the FastMixer state to restore after new parameters processed
2931 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE;
2932 bool reconfig = false;
2933
2934 while (!mNewParameters.isEmpty()) {
2935
2936 if (mFastMixer != NULL) {
2937 FastMixerStateQueue *sq = mFastMixer->sq();
2938 FastMixerState *state = sq->begin();
2939 if (!(state->mCommand & FastMixerState::IDLE)) {
2940 previousCommand = state->mCommand;
2941 state->mCommand = FastMixerState::HOT_IDLE;
2942 sq->end();
2943 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
2944 } else {
2945 sq->end(false /*didModify*/);
2946 }
2947 }
2948
2949 status_t status = NO_ERROR;
2950 String8 keyValuePair = mNewParameters[0];
2951 AudioParameter param = AudioParameter(keyValuePair);
2952 int value;
2953
2954 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
2955 reconfig = true;
2956 }
2957 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
2958 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
2959 status = BAD_VALUE;
2960 } else {
2961 reconfig = true;
2962 }
2963 }
2964 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
2965 if (value != AUDIO_CHANNEL_OUT_STEREO) {
2966 status = BAD_VALUE;
2967 } else {
2968 reconfig = true;
2969 }
2970 }
2971 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
2972 // do not accept frame count changes if tracks are open as the track buffer
2973 // size depends on frame count and correct behavior would not be guaranteed
2974 // if frame count is changed after track creation
2975 if (!mTracks.isEmpty()) {
2976 status = INVALID_OPERATION;
2977 } else {
2978 reconfig = true;
2979 }
2980 }
2981 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
2982#ifdef ADD_BATTERY_DATA
2983 // when changing the audio output device, call addBatteryData to notify
2984 // the change
2985 if (mOutDevice != value) {
2986 uint32_t params = 0;
2987 // check whether speaker is on
2988 if (value & AUDIO_DEVICE_OUT_SPEAKER) {
2989 params |= IMediaPlayerService::kBatteryDataSpeakerOn;
2990 }
2991
2992 audio_devices_t deviceWithoutSpeaker
2993 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
2994 // check if any other device (except speaker) is on
2995 if (value & deviceWithoutSpeaker ) {
2996 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
2997 }
2998
2999 if (params != 0) {
3000 addBatteryData(params);
3001 }
3002 }
3003#endif
3004
3005 // forward device change to effects that have requested to be
3006 // aware of attached audio device.
3007 mOutDevice = value;
3008 for (size_t i = 0; i < mEffectChains.size(); i++) {
3009 mEffectChains[i]->setDevice_l(mOutDevice);
3010 }
3011 }
3012
3013 if (status == NO_ERROR) {
3014 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3015 keyValuePair.string());
3016 if (!mStandby && status == INVALID_OPERATION) {
3017 mOutput->stream->common.standby(&mOutput->stream->common);
3018 mStandby = true;
3019 mBytesWritten = 0;
3020 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3021 keyValuePair.string());
3022 }
3023 if (status == NO_ERROR && reconfig) {
3024 delete mAudioMixer;
3025 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
3026 mAudioMixer = NULL;
3027 readOutputParameters();
3028 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
3029 for (size_t i = 0; i < mTracks.size() ; i++) {
3030 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId);
3031 if (name < 0) {
3032 break;
3033 }
3034 mTracks[i]->mName = name;
Eric Laurent81784c32012-11-19 14:55:58 -08003035 }
3036 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
3037 }
3038 }
3039
3040 mNewParameters.removeAt(0);
3041
3042 mParamStatus = status;
3043 mParamCond.signal();
3044 // wait for condition with time out in case the thread calling ThreadBase::setParameters()
3045 // already timed out waiting for the status and will never signal the condition.
3046 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
3047 }
3048
3049 if (!(previousCommand & FastMixerState::IDLE)) {
3050 ALOG_ASSERT(mFastMixer != NULL);
3051 FastMixerStateQueue *sq = mFastMixer->sq();
3052 FastMixerState *state = sq->begin();
3053 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE);
3054 state->mCommand = previousCommand;
3055 sq->end();
3056 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3057 }
3058
3059 return reconfig;
3060}
3061
3062
3063void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
3064{
3065 const size_t SIZE = 256;
3066 char buffer[SIZE];
3067 String8 result;
3068
3069 PlaybackThread::dumpInternals(fd, args);
3070
3071 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
3072 result.append(buffer);
3073 write(fd, result.string(), result.size());
3074
3075 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
3076 FastMixerDumpState copy = mFastMixerDumpState;
3077 copy.dump(fd);
3078
3079#ifdef STATE_QUEUE_DUMP
3080 // Similar for state queue
3081 StateQueueObserverDump observerCopy = mStateQueueObserverDump;
3082 observerCopy.dump(fd);
3083 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
3084 mutatorCopy.dump(fd);
3085#endif
3086
3087 // Write the tee output to a .wav file
3088 dumpTee(fd, mTeeSource, mId);
3089
3090#ifdef AUDIO_WATCHDOG
3091 if (mAudioWatchdog != 0) {
3092 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
3093 AudioWatchdogDump wdCopy = mAudioWatchdogDump;
3094 wdCopy.dump(fd);
3095 }
3096#endif
3097}
3098
3099uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
3100{
3101 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
3102}
3103
3104uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
3105{
3106 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
3107}
3108
3109void AudioFlinger::MixerThread::cacheParameters_l()
3110{
3111 PlaybackThread::cacheParameters_l();
3112
3113 // FIXME: Relaxed timing because of a certain device that can't meet latency
3114 // Should be reduced to 2x after the vendor fixes the driver issue
3115 // increase threshold again due to low power audio mode. The way this warning
3116 // threshold is calculated and its usefulness should be reconsidered anyway.
3117 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
3118}
3119
3120// ----------------------------------------------------------------------------
3121
3122AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
3123 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device)
3124 : PlaybackThread(audioFlinger, output, id, device, DIRECT)
3125 // mLeftVolFloat, mRightVolFloat
3126{
3127}
3128
3129AudioFlinger::DirectOutputThread::~DirectOutputThread()
3130{
3131}
3132
3133AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
3134 Vector< sp<Track> > *tracksToRemove
3135)
3136{
3137 sp<Track> trackToRemove;
3138
3139 mixer_state mixerStatus = MIXER_IDLE;
3140
3141 // find out which tracks need to be processed
3142 if (mActiveTracks.size() != 0) {
3143 sp<Track> t = mActiveTracks[0].promote();
3144 // The track died recently
3145 if (t == 0) {
3146 return MIXER_IDLE;
3147 }
3148
3149 Track* const track = t.get();
3150 audio_track_cblk_t* cblk = track->cblk();
3151
3152 // The first time a track is added we wait
3153 // for all its buffers to be filled before processing it
3154 uint32_t minFrames;
3155 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) {
3156 minFrames = mNormalFrameCount;
3157 } else {
3158 minFrames = 1;
3159 }
3160 if ((track->framesReady() >= minFrames) && track->isReady() &&
3161 !track->isPaused() && !track->isTerminated())
3162 {
3163 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
3164
3165 if (track->mFillingUpStatus == Track::FS_FILLED) {
3166 track->mFillingUpStatus = Track::FS_ACTIVE;
3167 mLeftVolFloat = mRightVolFloat = 0;
3168 if (track->mState == TrackBase::RESUMING) {
3169 track->mState = TrackBase::ACTIVE;
3170 }
3171 }
3172
3173 // compute volume for this track
3174 float left, right;
Glenn Kastene4756fe2012-11-29 13:38:14 -08003175 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) {
Eric Laurent81784c32012-11-19 14:55:58 -08003176 left = right = 0;
3177 if (track->isPausing()) {
3178 track->setPaused();
3179 }
3180 } else {
3181 float typeVolume = mStreamTypes[track->streamType()].volume;
3182 float v = mMasterVolume * typeVolume;
Glenn Kastene3aa6592012-12-04 12:22:46 -08003183 uint32_t vlr = track->mServerProxy->getVolumeLR();
Eric Laurent81784c32012-11-19 14:55:58 -08003184 float v_clamped = v * (vlr & 0xFFFF);
3185 if (v_clamped > MAX_GAIN) {
3186 v_clamped = MAX_GAIN;
3187 }
3188 left = v_clamped/MAX_GAIN;
3189 v_clamped = v * (vlr >> 16);
3190 if (v_clamped > MAX_GAIN) {
3191 v_clamped = MAX_GAIN;
3192 }
3193 right = v_clamped/MAX_GAIN;
3194 }
3195
3196 if (left != mLeftVolFloat || right != mRightVolFloat) {
3197 mLeftVolFloat = left;
3198 mRightVolFloat = right;
3199
3200 // Convert volumes from float to 8.24
3201 uint32_t vl = (uint32_t)(left * (1 << 24));
3202 uint32_t vr = (uint32_t)(right * (1 << 24));
3203
3204 // Delegate volume control to effect in track effect chain if needed
3205 // only one effect chain can be present on DirectOutputThread, so if
3206 // there is one, the track is connected to it
3207 if (!mEffectChains.isEmpty()) {
3208 // Do not ramp volume if volume is controlled by effect
3209 mEffectChains[0]->setVolume_l(&vl, &vr);
3210 left = (float)vl / (1 << 24);
3211 right = (float)vr / (1 << 24);
3212 }
3213 mOutput->stream->set_volume(mOutput->stream, left, right);
3214 }
3215
3216 // reset retry count
3217 track->mRetryCount = kMaxTrackRetriesDirect;
3218 mActiveTrack = t;
3219 mixerStatus = MIXER_TRACKS_READY;
3220 } else {
3221 // clear effect chain input buffer if an active track underruns to avoid sending
3222 // previous audio buffer again to effects
3223 if (!mEffectChains.isEmpty()) {
3224 mEffectChains[0]->clearInputBuffer();
3225 }
3226
3227 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
3228 if ((track->sharedBuffer() != 0) || track->isTerminated() ||
3229 track->isStopped() || track->isPaused()) {
3230 // We have consumed all the buffers of this track.
3231 // Remove it from the list of active tracks.
3232 // TODO: implement behavior for compressed audio
3233 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
3234 size_t framesWritten = mBytesWritten / mFrameSize;
3235 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
3236 if (track->isStopped()) {
3237 track->reset();
3238 }
3239 trackToRemove = track;
3240 }
3241 } else {
3242 // No buffers for this track. Give it a few chances to
3243 // fill a buffer, then remove it from active list.
3244 if (--(track->mRetryCount) <= 0) {
3245 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
3246 trackToRemove = track;
3247 } else {
3248 mixerStatus = MIXER_TRACKS_ENABLED;
3249 }
3250 }
3251 }
3252 }
3253
3254 // FIXME merge this with similar code for removing multiple tracks
3255 // remove all the tracks that need to be...
3256 if (CC_UNLIKELY(trackToRemove != 0)) {
3257 tracksToRemove->add(trackToRemove);
Glenn Kasten9e58b552013-01-18 15:09:48 -08003258#if 0
3259 mNBLogWriter->logf("prepareTracks_l remove name=%u", trackToRemove->name());
3260#endif
Eric Laurent81784c32012-11-19 14:55:58 -08003261 mActiveTracks.remove(trackToRemove);
3262 if (!mEffectChains.isEmpty()) {
3263 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(),
3264 trackToRemove->sessionId());
3265 mEffectChains[0]->decActiveTrackCnt();
3266 }
3267 if (trackToRemove->isTerminated()) {
3268 removeTrack_l(trackToRemove);
3269 }
3270 }
3271
3272 return mixerStatus;
3273}
3274
3275void AudioFlinger::DirectOutputThread::threadLoop_mix()
3276{
3277 AudioBufferProvider::Buffer buffer;
3278 size_t frameCount = mFrameCount;
3279 int8_t *curBuf = (int8_t *)mMixBuffer;
3280 // output audio to hardware
3281 while (frameCount) {
3282 buffer.frameCount = frameCount;
3283 mActiveTrack->getNextBuffer(&buffer);
3284 if (CC_UNLIKELY(buffer.raw == NULL)) {
3285 memset(curBuf, 0, frameCount * mFrameSize);
3286 break;
3287 }
3288 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
3289 frameCount -= buffer.frameCount;
3290 curBuf += buffer.frameCount * mFrameSize;
3291 mActiveTrack->releaseBuffer(&buffer);
3292 }
3293 sleepTime = 0;
3294 standbyTime = systemTime() + standbyDelay;
3295 mActiveTrack.clear();
3296
3297}
3298
3299void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
3300{
3301 if (sleepTime == 0) {
3302 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
3303 sleepTime = activeSleepTime;
3304 } else {
3305 sleepTime = idleSleepTime;
3306 }
3307 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
3308 memset(mMixBuffer, 0, mFrameCount * mFrameSize);
3309 sleepTime = 0;
3310 }
3311}
3312
3313// getTrackName_l() must be called with ThreadBase::mLock held
3314int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask,
3315 int sessionId)
3316{
3317 return 0;
3318}
3319
3320// deleteTrackName_l() must be called with ThreadBase::mLock held
3321void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
3322{
3323}
3324
3325// checkForNewParameters_l() must be called with ThreadBase::mLock held
3326bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
3327{
3328 bool reconfig = false;
3329
3330 while (!mNewParameters.isEmpty()) {
3331 status_t status = NO_ERROR;
3332 String8 keyValuePair = mNewParameters[0];
3333 AudioParameter param = AudioParameter(keyValuePair);
3334 int value;
3335
3336 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
3337 // do not accept frame count changes if tracks are open as the track buffer
3338 // size depends on frame count and correct behavior would not be garantied
3339 // if frame count is changed after track creation
3340 if (!mTracks.isEmpty()) {
3341 status = INVALID_OPERATION;
3342 } else {
3343 reconfig = true;
3344 }
3345 }
3346 if (status == NO_ERROR) {
3347 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3348 keyValuePair.string());
3349 if (!mStandby && status == INVALID_OPERATION) {
3350 mOutput->stream->common.standby(&mOutput->stream->common);
3351 mStandby = true;
3352 mBytesWritten = 0;
3353 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3354 keyValuePair.string());
3355 }
3356 if (status == NO_ERROR && reconfig) {
3357 readOutputParameters();
3358 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
3359 }
3360 }
3361
3362 mNewParameters.removeAt(0);
3363
3364 mParamStatus = status;
3365 mParamCond.signal();
3366 // wait for condition with time out in case the thread calling ThreadBase::setParameters()
3367 // already timed out waiting for the status and will never signal the condition.
3368 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
3369 }
3370 return reconfig;
3371}
3372
3373uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
3374{
3375 uint32_t time;
3376 if (audio_is_linear_pcm(mFormat)) {
3377 time = PlaybackThread::activeSleepTimeUs();
3378 } else {
3379 time = 10000;
3380 }
3381 return time;
3382}
3383
3384uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
3385{
3386 uint32_t time;
3387 if (audio_is_linear_pcm(mFormat)) {
3388 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
3389 } else {
3390 time = 10000;
3391 }
3392 return time;
3393}
3394
3395uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
3396{
3397 uint32_t time;
3398 if (audio_is_linear_pcm(mFormat)) {
3399 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
3400 } else {
3401 time = 10000;
3402 }
3403 return time;
3404}
3405
3406void AudioFlinger::DirectOutputThread::cacheParameters_l()
3407{
3408 PlaybackThread::cacheParameters_l();
3409
3410 // use shorter standby delay as on normal output to release
3411 // hardware resources as soon as possible
3412 standbyDelay = microseconds(activeSleepTime*2);
3413}
3414
3415// ----------------------------------------------------------------------------
3416
3417AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
3418 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
3419 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
3420 DUPLICATING),
3421 mWaitTimeMs(UINT_MAX)
3422{
3423 addOutputTrack(mainThread);
3424}
3425
3426AudioFlinger::DuplicatingThread::~DuplicatingThread()
3427{
3428 for (size_t i = 0; i < mOutputTracks.size(); i++) {
3429 mOutputTracks[i]->destroy();
3430 }
3431}
3432
3433void AudioFlinger::DuplicatingThread::threadLoop_mix()
3434{
3435 // mix buffers...
3436 if (outputsReady(outputTracks)) {
3437 mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
3438 } else {
3439 memset(mMixBuffer, 0, mixBufferSize);
3440 }
3441 sleepTime = 0;
3442 writeFrames = mNormalFrameCount;
3443 standbyTime = systemTime() + standbyDelay;
3444}
3445
3446void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
3447{
3448 if (sleepTime == 0) {
3449 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
3450 sleepTime = activeSleepTime;
3451 } else {
3452 sleepTime = idleSleepTime;
3453 }
3454 } else if (mBytesWritten != 0) {
3455 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
3456 writeFrames = mNormalFrameCount;
3457 memset(mMixBuffer, 0, mixBufferSize);
3458 } else {
3459 // flush remaining overflow buffers in output tracks
3460 writeFrames = 0;
3461 }
3462 sleepTime = 0;
3463 }
3464}
3465
3466void AudioFlinger::DuplicatingThread::threadLoop_write()
3467{
3468 for (size_t i = 0; i < outputTracks.size(); i++) {
3469 outputTracks[i]->write(mMixBuffer, writeFrames);
3470 }
3471 mBytesWritten += mixBufferSize;
3472}
3473
3474void AudioFlinger::DuplicatingThread::threadLoop_standby()
3475{
3476 // DuplicatingThread implements standby by stopping all tracks
3477 for (size_t i = 0; i < outputTracks.size(); i++) {
3478 outputTracks[i]->stop();
3479 }
3480}
3481
3482void AudioFlinger::DuplicatingThread::saveOutputTracks()
3483{
3484 outputTracks = mOutputTracks;
3485}
3486
3487void AudioFlinger::DuplicatingThread::clearOutputTracks()
3488{
3489 outputTracks.clear();
3490}
3491
3492void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
3493{
3494 Mutex::Autolock _l(mLock);
3495 // FIXME explain this formula
3496 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
3497 OutputTrack *outputTrack = new OutputTrack(thread,
3498 this,
3499 mSampleRate,
3500 mFormat,
3501 mChannelMask,
3502 frameCount);
3503 if (outputTrack->cblk() != NULL) {
3504 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
3505 mOutputTracks.add(outputTrack);
3506 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
3507 updateWaitTime_l();
3508 }
3509}
3510
3511void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
3512{
3513 Mutex::Autolock _l(mLock);
3514 for (size_t i = 0; i < mOutputTracks.size(); i++) {
3515 if (mOutputTracks[i]->thread() == thread) {
3516 mOutputTracks[i]->destroy();
3517 mOutputTracks.removeAt(i);
3518 updateWaitTime_l();
3519 return;
3520 }
3521 }
3522 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread);
3523}
3524
3525// caller must hold mLock
3526void AudioFlinger::DuplicatingThread::updateWaitTime_l()
3527{
3528 mWaitTimeMs = UINT_MAX;
3529 for (size_t i = 0; i < mOutputTracks.size(); i++) {
3530 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
3531 if (strong != 0) {
3532 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
3533 if (waitTimeMs < mWaitTimeMs) {
3534 mWaitTimeMs = waitTimeMs;
3535 }
3536 }
3537 }
3538}
3539
3540
3541bool AudioFlinger::DuplicatingThread::outputsReady(
3542 const SortedVector< sp<OutputTrack> > &outputTracks)
3543{
3544 for (size_t i = 0; i < outputTracks.size(); i++) {
3545 sp<ThreadBase> thread = outputTracks[i]->thread().promote();
3546 if (thread == 0) {
3547 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
3548 outputTracks[i].get());
3549 return false;
3550 }
3551 PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
3552 // see note at standby() declaration
3553 if (playbackThread->standby() && !playbackThread->isSuspended()) {
3554 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
3555 thread.get());
3556 return false;
3557 }
3558 }
3559 return true;
3560}
3561
3562uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
3563{
3564 return (mWaitTimeMs * 1000) / 2;
3565}
3566
3567void AudioFlinger::DuplicatingThread::cacheParameters_l()
3568{
3569 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
3570 updateWaitTime_l();
3571
3572 MixerThread::cacheParameters_l();
3573}
3574
3575// ----------------------------------------------------------------------------
3576// Record
3577// ----------------------------------------------------------------------------
3578
3579AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
3580 AudioStreamIn *input,
3581 uint32_t sampleRate,
3582 audio_channel_mask_t channelMask,
3583 audio_io_handle_t id,
Eric Laurentd3922f72013-02-01 17:57:04 -08003584 audio_devices_t outDevice,
3585 audio_devices_t inDevice,
Eric Laurent81784c32012-11-19 14:55:58 -08003586 const sp<NBAIO_Sink>& teeSink) :
Eric Laurentd3922f72013-02-01 17:57:04 -08003587 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
Eric Laurent81784c32012-11-19 14:55:58 -08003588 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
3589 // mRsmpInIndex and mInputBytes set by readInputParameters()
3590 mReqChannelCount(popcount(channelMask)),
3591 mReqSampleRate(sampleRate),
3592 // mBytesRead is only meaningful while active, and so is cleared in start()
3593 // (but might be better to also clear here for dump?)
3594 mTeeSink(teeSink)
3595{
3596 snprintf(mName, kNameLength, "AudioIn_%X", id);
3597
3598 readInputParameters();
3599
3600}
3601
3602
3603AudioFlinger::RecordThread::~RecordThread()
3604{
3605 delete[] mRsmpInBuffer;
3606 delete mResampler;
3607 delete[] mRsmpOutBuffer;
3608}
3609
3610void AudioFlinger::RecordThread::onFirstRef()
3611{
3612 run(mName, PRIORITY_URGENT_AUDIO);
3613}
3614
3615status_t AudioFlinger::RecordThread::readyToRun()
3616{
3617 status_t status = initCheck();
3618 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
3619 return status;
3620}
3621
3622bool AudioFlinger::RecordThread::threadLoop()
3623{
3624 AudioBufferProvider::Buffer buffer;
3625 sp<RecordTrack> activeTrack;
3626 Vector< sp<EffectChain> > effectChains;
3627
3628 nsecs_t lastWarning = 0;
3629
3630 inputStandBy();
3631 acquireWakeLock();
3632
3633 // used to verify we've read at least once before evaluating how many bytes were read
3634 bool readOnce = false;
3635
3636 // start recording
3637 while (!exitPending()) {
3638
3639 processConfigEvents();
3640
3641 { // scope for mLock
3642 Mutex::Autolock _l(mLock);
3643 checkForNewParameters_l();
3644 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
3645 standby();
3646
3647 if (exitPending()) {
3648 break;
3649 }
3650
3651 releaseWakeLock_l();
3652 ALOGV("RecordThread: loop stopping");
3653 // go to sleep
3654 mWaitWorkCV.wait(mLock);
3655 ALOGV("RecordThread: loop starting");
3656 acquireWakeLock_l();
3657 continue;
3658 }
3659 if (mActiveTrack != 0) {
3660 if (mActiveTrack->mState == TrackBase::PAUSING) {
3661 standby();
3662 mActiveTrack.clear();
3663 mStartStopCond.broadcast();
3664 } else if (mActiveTrack->mState == TrackBase::RESUMING) {
3665 if (mReqChannelCount != mActiveTrack->channelCount()) {
3666 mActiveTrack.clear();
3667 mStartStopCond.broadcast();
3668 } else if (readOnce) {
3669 // record start succeeds only if first read from audio input
3670 // succeeds
3671 if (mBytesRead >= 0) {
3672 mActiveTrack->mState = TrackBase::ACTIVE;
3673 } else {
3674 mActiveTrack.clear();
3675 }
3676 mStartStopCond.broadcast();
3677 }
3678 mStandby = false;
3679 } else if (mActiveTrack->mState == TrackBase::TERMINATED) {
3680 removeTrack_l(mActiveTrack);
3681 mActiveTrack.clear();
3682 }
3683 }
3684 lockEffectChains_l(effectChains);
3685 }
3686
3687 if (mActiveTrack != 0) {
3688 if (mActiveTrack->mState != TrackBase::ACTIVE &&
3689 mActiveTrack->mState != TrackBase::RESUMING) {
3690 unlockEffectChains(effectChains);
3691 usleep(kRecordThreadSleepUs);
3692 continue;
3693 }
3694 for (size_t i = 0; i < effectChains.size(); i ++) {
3695 effectChains[i]->process_l();
3696 }
3697
3698 buffer.frameCount = mFrameCount;
3699 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
3700 readOnce = true;
3701 size_t framesOut = buffer.frameCount;
3702 if (mResampler == NULL) {
3703 // no resampling
3704 while (framesOut) {
3705 size_t framesIn = mFrameCount - mRsmpInIndex;
3706 if (framesIn) {
3707 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
3708 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
3709 mActiveTrack->mFrameSize;
3710 if (framesIn > framesOut)
3711 framesIn = framesOut;
3712 mRsmpInIndex += framesIn;
3713 framesOut -= framesIn;
3714 if (mChannelCount == mReqChannelCount ||
3715 mFormat != AUDIO_FORMAT_PCM_16_BIT) {
3716 memcpy(dst, src, framesIn * mFrameSize);
3717 } else {
3718 if (mChannelCount == 1) {
3719 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst,
3720 (int16_t *)src, framesIn);
3721 } else {
3722 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
3723 (int16_t *)src, framesIn);
3724 }
3725 }
3726 }
3727 if (framesOut && mFrameCount == mRsmpInIndex) {
3728 void *readInto;
3729 if (framesOut == mFrameCount &&
3730 (mChannelCount == mReqChannelCount ||
3731 mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
3732 readInto = buffer.raw;
3733 framesOut = 0;
3734 } else {
3735 readInto = mRsmpInBuffer;
3736 mRsmpInIndex = 0;
3737 }
3738 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes);
3739 if (mBytesRead <= 0) {
3740 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE))
3741 {
3742 ALOGE("Error reading audio input");
3743 // Force input into standby so that it tries to
3744 // recover at next read attempt
3745 inputStandBy();
3746 usleep(kRecordThreadSleepUs);
3747 }
3748 mRsmpInIndex = mFrameCount;
3749 framesOut = 0;
3750 buffer.frameCount = 0;
3751 } else if (mTeeSink != 0) {
3752 (void) mTeeSink->write(readInto,
3753 mBytesRead >> Format_frameBitShift(mTeeSink->format()));
3754 }
3755 }
3756 }
3757 } else {
3758 // resampling
3759
3760 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
3761 // alter output frame count as if we were expecting stereo samples
3762 if (mChannelCount == 1 && mReqChannelCount == 1) {
3763 framesOut >>= 1;
3764 }
3765 mResampler->resample(mRsmpOutBuffer, framesOut,
3766 this /* AudioBufferProvider* */);
3767 // ditherAndClamp() works as long as all buffers returned by
3768 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true.
3769 if (mChannelCount == 2 && mReqChannelCount == 1) {
3770 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
3771 // the resampler always outputs stereo samples:
3772 // do post stereo to mono conversion
3773 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer,
3774 framesOut);
3775 } else {
3776 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
3777 }
3778
3779 }
3780 if (mFramestoDrop == 0) {
3781 mActiveTrack->releaseBuffer(&buffer);
3782 } else {
3783 if (mFramestoDrop > 0) {
3784 mFramestoDrop -= buffer.frameCount;
3785 if (mFramestoDrop <= 0) {
3786 clearSyncStartEvent();
3787 }
3788 } else {
3789 mFramestoDrop += buffer.frameCount;
3790 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 ||
3791 mSyncStartEvent->isCancelled()) {
3792 ALOGW("Synced record %s, session %d, trigger session %d",
3793 (mFramestoDrop >= 0) ? "timed out" : "cancelled",
3794 mActiveTrack->sessionId(),
3795 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0);
3796 clearSyncStartEvent();
3797 }
3798 }
3799 }
3800 mActiveTrack->clearOverflow();
3801 }
3802 // client isn't retrieving buffers fast enough
3803 else {
3804 if (!mActiveTrack->setOverflow()) {
3805 nsecs_t now = systemTime();
3806 if ((now - lastWarning) > kWarningThrottleNs) {
3807 ALOGW("RecordThread: buffer overflow");
3808 lastWarning = now;
3809 }
3810 }
3811 // Release the processor for a while before asking for a new buffer.
3812 // This will give the application more chance to read from the buffer and
3813 // clear the overflow.
3814 usleep(kRecordThreadSleepUs);
3815 }
3816 }
3817 // enable changes in effect chain
3818 unlockEffectChains(effectChains);
3819 effectChains.clear();
3820 }
3821
3822 standby();
3823
3824 {
3825 Mutex::Autolock _l(mLock);
3826 mActiveTrack.clear();
3827 mStartStopCond.broadcast();
3828 }
3829
3830 releaseWakeLock();
3831
3832 ALOGV("RecordThread %p exiting", this);
3833 return false;
3834}
3835
3836void AudioFlinger::RecordThread::standby()
3837{
3838 if (!mStandby) {
3839 inputStandBy();
3840 mStandby = true;
3841 }
3842}
3843
3844void AudioFlinger::RecordThread::inputStandBy()
3845{
3846 mInput->stream->common.standby(&mInput->stream->common);
3847}
3848
3849sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
3850 const sp<AudioFlinger::Client>& client,
3851 uint32_t sampleRate,
3852 audio_format_t format,
3853 audio_channel_mask_t channelMask,
3854 size_t frameCount,
3855 int sessionId,
3856 IAudioFlinger::track_flags_t flags,
3857 pid_t tid,
3858 status_t *status)
3859{
3860 sp<RecordTrack> track;
3861 status_t lStatus;
3862
3863 lStatus = initCheck();
3864 if (lStatus != NO_ERROR) {
3865 ALOGE("Audio driver not initialized.");
3866 goto Exit;
3867 }
3868
3869 // FIXME use flags and tid similar to createTrack_l()
3870
3871 { // scope for mLock
3872 Mutex::Autolock _l(mLock);
3873
3874 track = new RecordTrack(this, client, sampleRate,
3875 format, channelMask, frameCount, sessionId);
3876
3877 if (track->getCblk() == 0) {
3878 lStatus = NO_MEMORY;
3879 goto Exit;
3880 }
3881 mTracks.add(track);
3882
3883 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
3884 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
3885 mAudioFlinger->btNrecIsOff();
3886 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
3887 setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
3888 }
3889 lStatus = NO_ERROR;
3890
3891Exit:
3892 if (status) {
3893 *status = lStatus;
3894 }
3895 return track;
3896}
3897
3898status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
3899 AudioSystem::sync_event_t event,
3900 int triggerSession)
3901{
3902 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
3903 sp<ThreadBase> strongMe = this;
3904 status_t status = NO_ERROR;
3905
3906 if (event == AudioSystem::SYNC_EVENT_NONE) {
3907 clearSyncStartEvent();
3908 } else if (event != AudioSystem::SYNC_EVENT_SAME) {
3909 mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
3910 triggerSession,
3911 recordTrack->sessionId(),
3912 syncStartEventCallback,
3913 this);
3914 // Sync event can be cancelled by the trigger session if the track is not in a
3915 // compatible state in which case we start record immediately
3916 if (mSyncStartEvent->isCancelled()) {
3917 clearSyncStartEvent();
3918 } else {
3919 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
3920 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000);
3921 }
3922 }
3923
3924 {
3925 AutoMutex lock(mLock);
3926 if (mActiveTrack != 0) {
3927 if (recordTrack != mActiveTrack.get()) {
3928 status = -EBUSY;
3929 } else if (mActiveTrack->mState == TrackBase::PAUSING) {
3930 mActiveTrack->mState = TrackBase::ACTIVE;
3931 }
3932 return status;
3933 }
3934
3935 recordTrack->mState = TrackBase::IDLE;
3936 mActiveTrack = recordTrack;
3937 mLock.unlock();
3938 status_t status = AudioSystem::startInput(mId);
3939 mLock.lock();
3940 if (status != NO_ERROR) {
3941 mActiveTrack.clear();
3942 clearSyncStartEvent();
3943 return status;
3944 }
3945 mRsmpInIndex = mFrameCount;
3946 mBytesRead = 0;
3947 if (mResampler != NULL) {
3948 mResampler->reset();
3949 }
3950 mActiveTrack->mState = TrackBase::RESUMING;
3951 // signal thread to start
3952 ALOGV("Signal record thread");
3953 mWaitWorkCV.broadcast();
3954 // do not wait for mStartStopCond if exiting
3955 if (exitPending()) {
3956 mActiveTrack.clear();
3957 status = INVALID_OPERATION;
3958 goto startError;
3959 }
3960 mStartStopCond.wait(mLock);
3961 if (mActiveTrack == 0) {
3962 ALOGV("Record failed to start");
3963 status = BAD_VALUE;
3964 goto startError;
3965 }
3966 ALOGV("Record started OK");
3967 return status;
3968 }
3969startError:
3970 AudioSystem::stopInput(mId);
3971 clearSyncStartEvent();
3972 return status;
3973}
3974
3975void AudioFlinger::RecordThread::clearSyncStartEvent()
3976{
3977 if (mSyncStartEvent != 0) {
3978 mSyncStartEvent->cancel();
3979 }
3980 mSyncStartEvent.clear();
3981 mFramestoDrop = 0;
3982}
3983
3984void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
3985{
3986 sp<SyncEvent> strongEvent = event.promote();
3987
3988 if (strongEvent != 0) {
3989 RecordThread *me = (RecordThread *)strongEvent->cookie();
3990 me->handleSyncStartEvent(strongEvent);
3991 }
3992}
3993
3994void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event)
3995{
3996 if (event == mSyncStartEvent) {
3997 // TODO: use actual buffer filling status instead of 2 buffers when info is available
3998 // from audio HAL
3999 mFramestoDrop = mFrameCount * 2;
4000 }
4001}
4002
4003bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) {
4004 ALOGV("RecordThread::stop");
4005 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) {
4006 return false;
4007 }
4008 recordTrack->mState = TrackBase::PAUSING;
4009 // do not wait for mStartStopCond if exiting
4010 if (exitPending()) {
4011 return true;
4012 }
4013 mStartStopCond.wait(mLock);
4014 // if we have been restarted, recordTrack == mActiveTrack.get() here
4015 if (exitPending() || recordTrack != mActiveTrack.get()) {
4016 ALOGV("Record stopped OK");
4017 return true;
4018 }
4019 return false;
4020}
4021
4022bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const
4023{
4024 return false;
4025}
4026
4027status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event)
4028{
4029#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
4030 if (!isValidSyncEvent(event)) {
4031 return BAD_VALUE;
4032 }
4033
4034 int eventSession = event->triggerSession();
4035 status_t ret = NAME_NOT_FOUND;
4036
4037 Mutex::Autolock _l(mLock);
4038
4039 for (size_t i = 0; i < mTracks.size(); i++) {
4040 sp<RecordTrack> track = mTracks[i];
4041 if (eventSession == track->sessionId()) {
4042 (void) track->setSyncEvent(event);
4043 ret = NO_ERROR;
4044 }
4045 }
4046 return ret;
4047#else
4048 return BAD_VALUE;
4049#endif
4050}
4051
4052// destroyTrack_l() must be called with ThreadBase::mLock held
4053void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
4054{
4055 track->mState = TrackBase::TERMINATED;
4056 // active tracks are removed by threadLoop()
4057 if (mActiveTrack != track) {
4058 removeTrack_l(track);
4059 }
4060}
4061
4062void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
4063{
4064 mTracks.remove(track);
4065 // need anything related to effects here?
4066}
4067
4068void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
4069{
4070 dumpInternals(fd, args);
4071 dumpTracks(fd, args);
4072 dumpEffectChains(fd, args);
4073}
4074
4075void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
4076{
4077 const size_t SIZE = 256;
4078 char buffer[SIZE];
4079 String8 result;
4080
4081 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
4082 result.append(buffer);
4083
4084 if (mActiveTrack != 0) {
4085 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
4086 result.append(buffer);
4087 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
4088 result.append(buffer);
4089 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
4090 result.append(buffer);
4091 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount);
4092 result.append(buffer);
4093 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate);
4094 result.append(buffer);
4095 } else {
4096 result.append("No active record client\n");
4097 }
4098
4099 write(fd, result.string(), result.size());
4100
4101 dumpBase(fd, args);
4102}
4103
4104void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args)
4105{
4106 const size_t SIZE = 256;
4107 char buffer[SIZE];
4108 String8 result;
4109
4110 snprintf(buffer, SIZE, "Input thread %p tracks\n", this);
4111 result.append(buffer);
4112 RecordTrack::appendDumpHeader(result);
4113 for (size_t i = 0; i < mTracks.size(); ++i) {
4114 sp<RecordTrack> track = mTracks[i];
4115 if (track != 0) {
4116 track->dump(buffer, SIZE);
4117 result.append(buffer);
4118 }
4119 }
4120
4121 if (mActiveTrack != 0) {
4122 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this);
4123 result.append(buffer);
4124 RecordTrack::appendDumpHeader(result);
4125 mActiveTrack->dump(buffer, SIZE);
4126 result.append(buffer);
4127
4128 }
4129 write(fd, result.string(), result.size());
4130}
4131
4132// AudioBufferProvider interface
4133status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
4134{
4135 size_t framesReq = buffer->frameCount;
4136 size_t framesReady = mFrameCount - mRsmpInIndex;
4137 int channelCount;
4138
4139 if (framesReady == 0) {
4140 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
4141 if (mBytesRead <= 0) {
4142 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) {
4143 ALOGE("RecordThread::getNextBuffer() Error reading audio input");
4144 // Force input into standby so that it tries to
4145 // recover at next read attempt
4146 inputStandBy();
4147 usleep(kRecordThreadSleepUs);
4148 }
4149 buffer->raw = NULL;
4150 buffer->frameCount = 0;
4151 return NOT_ENOUGH_DATA;
4152 }
4153 mRsmpInIndex = 0;
4154 framesReady = mFrameCount;
4155 }
4156
4157 if (framesReq > framesReady) {
4158 framesReq = framesReady;
4159 }
4160
4161 if (mChannelCount == 1 && mReqChannelCount == 2) {
4162 channelCount = 1;
4163 } else {
4164 channelCount = 2;
4165 }
4166 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
4167 buffer->frameCount = framesReq;
4168 return NO_ERROR;
4169}
4170
4171// AudioBufferProvider interface
4172void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
4173{
4174 mRsmpInIndex += buffer->frameCount;
4175 buffer->frameCount = 0;
4176}
4177
4178bool AudioFlinger::RecordThread::checkForNewParameters_l()
4179{
4180 bool reconfig = false;
4181
4182 while (!mNewParameters.isEmpty()) {
4183 status_t status = NO_ERROR;
4184 String8 keyValuePair = mNewParameters[0];
4185 AudioParameter param = AudioParameter(keyValuePair);
4186 int value;
4187 audio_format_t reqFormat = mFormat;
4188 uint32_t reqSamplingRate = mReqSampleRate;
4189 uint32_t reqChannelCount = mReqChannelCount;
4190
4191 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4192 reqSamplingRate = value;
4193 reconfig = true;
4194 }
4195 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
4196 reqFormat = (audio_format_t) value;
4197 reconfig = true;
4198 }
4199 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
4200 reqChannelCount = popcount(value);
4201 reconfig = true;
4202 }
4203 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4204 // do not accept frame count changes if tracks are open as the track buffer
4205 // size depends on frame count and correct behavior would not be guaranteed
4206 // if frame count is changed after track creation
4207 if (mActiveTrack != 0) {
4208 status = INVALID_OPERATION;
4209 } else {
4210 reconfig = true;
4211 }
4212 }
4213 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4214 // forward device change to effects that have requested to be
4215 // aware of attached audio device.
4216 for (size_t i = 0; i < mEffectChains.size(); i++) {
4217 mEffectChains[i]->setDevice_l(value);
4218 }
4219
4220 // store input device and output device but do not forward output device to audio HAL.
4221 // Note that status is ignored by the caller for output device
4222 // (see AudioFlinger::setParameters()
4223 if (audio_is_output_devices(value)) {
4224 mOutDevice = value;
4225 status = BAD_VALUE;
4226 } else {
4227 mInDevice = value;
4228 // disable AEC and NS if the device is a BT SCO headset supporting those
4229 // pre processings
4230 if (mTracks.size() > 0) {
4231 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
4232 mAudioFlinger->btNrecIsOff();
4233 for (size_t i = 0; i < mTracks.size(); i++) {
4234 sp<RecordTrack> track = mTracks[i];
4235 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
4236 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
4237 }
4238 }
4239 }
4240 }
4241 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
4242 mAudioSource != (audio_source_t)value) {
4243 // forward device change to effects that have requested to be
4244 // aware of attached audio device.
4245 for (size_t i = 0; i < mEffectChains.size(); i++) {
4246 mEffectChains[i]->setAudioSource_l((audio_source_t)value);
4247 }
4248 mAudioSource = (audio_source_t)value;
4249 }
4250 if (status == NO_ERROR) {
4251 status = mInput->stream->common.set_parameters(&mInput->stream->common,
4252 keyValuePair.string());
4253 if (status == INVALID_OPERATION) {
4254 inputStandBy();
4255 status = mInput->stream->common.set_parameters(&mInput->stream->common,
4256 keyValuePair.string());
4257 }
4258 if (reconfig) {
4259 if (status == BAD_VALUE &&
4260 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
4261 reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
Glenn Kastenc4974312012-12-14 07:13:28 -08004262 (mInput->stream->common.get_sample_rate(&mInput->stream->common)
Eric Laurent81784c32012-11-19 14:55:58 -08004263 <= (2 * reqSamplingRate)) &&
4264 popcount(mInput->stream->common.get_channels(&mInput->stream->common))
4265 <= FCC_2 &&
4266 (reqChannelCount <= FCC_2)) {
4267 status = NO_ERROR;
4268 }
4269 if (status == NO_ERROR) {
4270 readInputParameters();
4271 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
4272 }
4273 }
4274 }
4275
4276 mNewParameters.removeAt(0);
4277
4278 mParamStatus = status;
4279 mParamCond.signal();
4280 // wait for condition with time out in case the thread calling ThreadBase::setParameters()
4281 // already timed out waiting for the status and will never signal the condition.
4282 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
4283 }
4284 return reconfig;
4285}
4286
4287String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
4288{
4289 char *s;
4290 String8 out_s8 = String8();
4291
4292 Mutex::Autolock _l(mLock);
4293 if (initCheck() != NO_ERROR) {
4294 return out_s8;
4295 }
4296
4297 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
4298 out_s8 = String8(s);
4299 free(s);
4300 return out_s8;
4301}
4302
4303void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
4304 AudioSystem::OutputDescriptor desc;
4305 void *param2 = NULL;
4306
4307 switch (event) {
4308 case AudioSystem::INPUT_OPENED:
4309 case AudioSystem::INPUT_CONFIG_CHANGED:
4310 desc.channels = mChannelMask;
4311 desc.samplingRate = mSampleRate;
4312 desc.format = mFormat;
4313 desc.frameCount = mFrameCount;
4314 desc.latency = 0;
4315 param2 = &desc;
4316 break;
4317
4318 case AudioSystem::INPUT_CLOSED:
4319 default:
4320 break;
4321 }
4322 mAudioFlinger->audioConfigChanged_l(event, mId, param2);
4323}
4324
4325void AudioFlinger::RecordThread::readInputParameters()
4326{
4327 delete mRsmpInBuffer;
4328 // mRsmpInBuffer is always assigned a new[] below
4329 delete mRsmpOutBuffer;
4330 mRsmpOutBuffer = NULL;
4331 delete mResampler;
4332 mResampler = NULL;
4333
4334 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
4335 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
4336 mChannelCount = (uint16_t)popcount(mChannelMask);
4337 mFormat = mInput->stream->common.get_format(&mInput->stream->common);
4338 mFrameSize = audio_stream_frame_size(&mInput->stream->common);
4339 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
4340 mFrameCount = mInputBytes / mFrameSize;
4341 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects
4342 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
4343
4344 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
4345 {
4346 int channelCount;
4347 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
4348 // stereo to mono post process as the resampler always outputs stereo.
4349 if (mChannelCount == 1 && mReqChannelCount == 2) {
4350 channelCount = 1;
4351 } else {
4352 channelCount = 2;
4353 }
4354 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
4355 mResampler->setSampleRate(mSampleRate);
4356 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
4357 mRsmpOutBuffer = new int32_t[mFrameCount * 2];
4358
4359 // optmization: if mono to mono, alter input frame count as if we were inputing
4360 // stereo samples
4361 if (mChannelCount == 1 && mReqChannelCount == 1) {
4362 mFrameCount >>= 1;
4363 }
4364
4365 }
4366 mRsmpInIndex = mFrameCount;
4367}
4368
4369unsigned int AudioFlinger::RecordThread::getInputFramesLost()
4370{
4371 Mutex::Autolock _l(mLock);
4372 if (initCheck() != NO_ERROR) {
4373 return 0;
4374 }
4375
4376 return mInput->stream->get_input_frames_lost(mInput->stream);
4377}
4378
4379uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const
4380{
4381 Mutex::Autolock _l(mLock);
4382 uint32_t result = 0;
4383 if (getEffectChain_l(sessionId) != 0) {
4384 result = EFFECT_SESSION;
4385 }
4386
4387 for (size_t i = 0; i < mTracks.size(); ++i) {
4388 if (sessionId == mTracks[i]->sessionId()) {
4389 result |= TRACK_SESSION;
4390 break;
4391 }
4392 }
4393
4394 return result;
4395}
4396
4397KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const
4398{
4399 KeyedVector<int, bool> ids;
4400 Mutex::Autolock _l(mLock);
4401 for (size_t j = 0; j < mTracks.size(); ++j) {
4402 sp<RecordThread::RecordTrack> track = mTracks[j];
4403 int sessionId = track->sessionId();
4404 if (ids.indexOfKey(sessionId) < 0) {
4405 ids.add(sessionId, true);
4406 }
4407 }
4408 return ids;
4409}
4410
4411AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
4412{
4413 Mutex::Autolock _l(mLock);
4414 AudioStreamIn *input = mInput;
4415 mInput = NULL;
4416 return input;
4417}
4418
4419// this method must always be called either with ThreadBase mLock held or inside the thread loop
4420audio_stream_t* AudioFlinger::RecordThread::stream() const
4421{
4422 if (mInput == NULL) {
4423 return NULL;
4424 }
4425 return &mInput->stream->common;
4426}
4427
4428status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
4429{
4430 // only one chain per input thread
4431 if (mEffectChains.size() != 0) {
4432 return INVALID_OPERATION;
4433 }
4434 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
4435
4436 chain->setInBuffer(NULL);
4437 chain->setOutBuffer(NULL);
4438
4439 checkSuspendOnAddEffectChain_l(chain);
4440
4441 mEffectChains.add(chain);
4442
4443 return NO_ERROR;
4444}
4445
4446size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
4447{
4448 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
4449 ALOGW_IF(mEffectChains.size() != 1,
4450 "removeEffectChain_l() %p invalid chain size %d on thread %p",
4451 chain.get(), mEffectChains.size(), this);
4452 if (mEffectChains.size() == 1) {
4453 mEffectChains.removeAt(0);
4454 }
4455 return 0;
4456}
4457
4458}; // namespace android