blob: d296ee34db0fa3768df8463577d91de8f04a63df [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
Glenn Kasten153b9fe2013-07-15 11:23:36 -070023#include "Configuration.h"
Eric Laurent81784c32012-11-19 14:55:58 -080024#include <math.h>
25#include <fcntl.h>
Glenn Kastenad8510a2015-02-17 16:24:07 -080026#include <linux/futex.h>
Eric Laurent81784c32012-11-19 14:55:58 -080027#include <sys/stat.h>
Glenn Kastenad8510a2015-02-17 16:24:07 -080028#include <sys/syscall.h>
Eric Laurent81784c32012-11-19 14:55:58 -080029#include <cutils/properties.h>
Glenn Kasten1ab85ec2013-05-31 09:18:43 -070030#include <media/AudioParameter.h>
Andy Hungcd044842014-08-07 11:04:34 -070031#include <media/AudioResamplerPublic.h>
Eric Laurent81784c32012-11-19 14:55:58 -080032#include <utils/Log.h>
Alex Ray371eb972012-11-30 11:11:54 -080033#include <utils/Trace.h>
Eric Laurent81784c32012-11-19 14:55:58 -080034
35#include <private/media/AudioTrackShared.h>
36#include <hardware/audio.h>
37#include <audio_effects/effect_ns.h>
38#include <audio_effects/effect_aec.h>
Andy Hung2ddee192015-12-18 17:34:44 -080039#include <audio_utils/conversion.h>
Eric Laurent81784c32012-11-19 14:55:58 -080040#include <audio_utils/primitives.h>
Andy Hung98ef9782014-03-04 14:46:50 -080041#include <audio_utils/format.h>
Glenn Kastenc56f3422014-03-21 17:53:17 -070042#include <audio_utils/minifloat.h>
Eric Laurent81784c32012-11-19 14:55:58 -080043
44// NBAIO implementations
Glenn Kasten6dbb5e32014-05-13 10:38:42 -070045#include <media/nbaio/AudioStreamInSource.h>
Eric Laurent81784c32012-11-19 14:55:58 -080046#include <media/nbaio/AudioStreamOutSink.h>
47#include <media/nbaio/MonoPipe.h>
48#include <media/nbaio/MonoPipeReader.h>
49#include <media/nbaio/Pipe.h>
50#include <media/nbaio/PipeReader.h>
51#include <media/nbaio/SourceAudioBufferProvider.h>
Wei Jia3f273d12015-11-24 09:06:49 -080052#include <mediautils/BatteryNotifier.h>
Eric Laurent81784c32012-11-19 14:55:58 -080053
54#include <powermanager/PowerManager.h>
55
Eric Laurent81784c32012-11-19 14:55:58 -080056#include "AudioFlinger.h"
57#include "AudioMixer.h"
Andy Hungd330ee42015-04-20 13:23:41 -070058#include "BufferProviders.h"
Eric Laurent81784c32012-11-19 14:55:58 -080059#include "FastMixer.h"
Glenn Kasten6dbb5e32014-05-13 10:38:42 -070060#include "FastCapture.h"
Eric Laurent81784c32012-11-19 14:55:58 -080061#include "ServiceUtilities.h"
Eino-Ville Talvalaf99498e2015-09-25 16:52:55 -070062#include "mediautils/SchedulingPolicyService.h"
Eric Laurent81784c32012-11-19 14:55:58 -080063
Eric Laurent81784c32012-11-19 14:55:58 -080064#ifdef ADD_BATTERY_DATA
65#include <media/IMediaPlayerService.h>
66#include <media/IMediaDeathNotifier.h>
67#endif
68
Eric Laurent81784c32012-11-19 14:55:58 -080069#ifdef DEBUG_CPU_USAGE
70#include <cpustats/CentralTendencyStatistics.h>
71#include <cpustats/ThreadCpuUsage.h>
72#endif
73
Glenn Kastenc05b8d72016-03-24 09:48:17 -070074#include "AutoPark.h"
75
Eric Laurent81784c32012-11-19 14:55:58 -080076// ----------------------------------------------------------------------------
77
78// Note: the following macro is used for extremely verbose logging message. In
79// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
80// 0; but one side effect of this is to turn all LOGV's as well. Some messages
81// are so verbose that we want to suppress them even when we have ALOG_ASSERT
82// turned on. Do not uncomment the #def below unless you really know what you
83// are doing and want to see all of the extremely verbose messages.
84//#define VERY_VERY_VERBOSE_LOGGING
85#ifdef VERY_VERY_VERBOSE_LOGGING
86#define ALOGVV ALOGV
87#else
88#define ALOGVV(a...) do { } while(0)
89#endif
90
Andy Hung6770c6f2015-04-07 13:43:36 -070091// TODO: Move these macro/inlines to a header file.
Glenn Kasten49d00ad2014-07-21 11:22:03 -070092#define max(a, b) ((a) > (b) ? (a) : (b))
Andy Hung6770c6f2015-04-07 13:43:36 -070093template <typename T>
94static inline T min(const T& a, const T& b)
95{
96 return a < b ? a : b;
97}
Glenn Kasten49d00ad2014-07-21 11:22:03 -070098
Andy Hungd330ee42015-04-20 13:23:41 -070099#ifndef ARRAY_SIZE
100#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0]))
101#endif
102
Eric Laurent81784c32012-11-19 14:55:58 -0800103namespace android {
104
105// retry counts for buffer fill timeout
106// 50 * ~20msecs = 1 second
107static const int8_t kMaxTrackRetries = 50;
108static const int8_t kMaxTrackStartupRetries = 50;
109// allow less retry attempts on direct output thread.
110// direct outputs can be a scarce resource in audio hardware and should
111// be released as quickly as possible.
112static const int8_t kMaxTrackRetriesDirect = 2;
Eric Laurente93cc032016-05-05 10:15:10 -0700113
Eric Laurent51716182016-02-29 18:00:56 -0800114
Eric Laurent81784c32012-11-19 14:55:58 -0800115
116// don't warn about blocked writes or record buffer overflows more often than this
117static const nsecs_t kWarningThrottleNs = seconds(5);
118
119// RecordThread loop sleep time upon application overrun or audio HAL read error
120static const int kRecordThreadSleepUs = 5000;
121
Eric Laurent10351942014-05-08 18:49:52 -0700122// maximum time to wait in sendConfigEvent_l() for a status to be received
123static const nsecs_t kConfigEventTimeoutNs = seconds(2);
Eric Laurent81784c32012-11-19 14:55:58 -0800124
125// minimum sleep time for the mixer thread loop when tracks are active but in underrun
126static const uint32_t kMinThreadSleepTimeUs = 5000;
127// maximum divider applied to the active sleep time in the mixer thread loop
128static const uint32_t kMaxThreadSleepTimeShift = 2;
129
Andy Hung09a50072014-02-27 14:30:47 -0800130// minimum normal sink buffer size, expressed in milliseconds rather than frames
Glenn Kasteneb9487e2015-07-22 09:15:17 -0700131// FIXME This should be based on experimentally observed scheduling jitter
Andy Hung09a50072014-02-27 14:30:47 -0800132static const uint32_t kMinNormalSinkBufferSizeMs = 20;
133// maximum normal sink buffer size
134static const uint32_t kMaxNormalSinkBufferSizeMs = 24;
Eric Laurent81784c32012-11-19 14:55:58 -0800135
Glenn Kasteneb9487e2015-07-22 09:15:17 -0700136// minimum capture buffer size in milliseconds to _not_ need a fast capture thread
137// FIXME This should be based on experimentally observed scheduling jitter
138static const uint32_t kMinNormalCaptureBufferSizeMs = 12;
139
Eric Laurent972a1732013-09-04 09:42:59 -0700140// Offloaded output thread standby delay: allows track transition without going to standby
141static const nsecs_t kOffloadStandbyDelayNs = seconds(1);
142
Eric Laurent51716182016-02-29 18:00:56 -0800143// Direct output thread minimum sleep time in idle or active(underrun) state
144static const nsecs_t kDirectMinSleepTimeUs = 10000;
145
Eric Laurent51716182016-02-29 18:00:56 -0800146
Eric Laurent81784c32012-11-19 14:55:58 -0800147// Whether to use fast mixer
148static const enum {
149 FastMixer_Never, // never initialize or use: for debugging only
150 FastMixer_Always, // always initialize and use, even if not needed: for debugging only
151 // normal mixer multiplier is 1
152 FastMixer_Static, // initialize if needed, then use all the time if initialized,
153 // multiplier is calculated based on min & max normal mixer buffer size
154 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load,
155 // multiplier is calculated based on min & max normal mixer buffer size
156 // FIXME for FastMixer_Dynamic:
157 // Supporting this option will require fixing HALs that can't handle large writes.
158 // For example, one HAL implementation returns an error from a large write,
159 // and another HAL implementation corrupts memory, possibly in the sample rate converter.
160 // We could either fix the HAL implementations, or provide a wrapper that breaks
161 // up large writes into smaller ones, and the wrapper would need to deal with scheduler.
162} kUseFastMixer = FastMixer_Static;
163
Glenn Kasten6dbb5e32014-05-13 10:38:42 -0700164// Whether to use fast capture
165static const enum {
166 FastCapture_Never, // never initialize or use: for debugging only
167 FastCapture_Always, // always initialize and use, even if not needed: for debugging only
168 FastCapture_Static, // initialize if needed, then use all the time if initialized
169} kUseFastCapture = FastCapture_Static;
170
Eric Laurent81784c32012-11-19 14:55:58 -0800171// Priorities for requestPriority
172static const int kPriorityAudioApp = 2;
173static const int kPriorityFastMixer = 3;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -0700174static const int kPriorityFastCapture = 3;
Eric Laurent81784c32012-11-19 14:55:58 -0800175
Glenn Kastenea38ee72016-04-18 11:08:01 -0700176// IAudioFlinger::createTrack() has an in/out parameter 'pFrameCount' for the total size of the
177// track buffer in shared memory. Zero on input means to use a default value. For fast tracks,
178// AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'.
Glenn Kasten03490092014-05-27 12:30:54 -0700179
180// This is the default value, if not specified by property.
Glenn Kastenb5fed682013-12-03 09:06:43 -0800181static const int kFastTrackMultiplier = 2;
Eric Laurent81784c32012-11-19 14:55:58 -0800182
Glenn Kasten03490092014-05-27 12:30:54 -0700183// The minimum and maximum allowed values
184static const int kFastTrackMultiplierMin = 1;
185static const int kFastTrackMultiplierMax = 2;
186
187// The actual value to use, which can be specified per-device via property af.fast_track_multiplier.
188static int sFastTrackMultiplier = kFastTrackMultiplier;
189
Glenn Kastenb880f5e2014-05-07 08:43:45 -0700190// See Thread::readOnlyHeap().
191// Initially this heap is used to allocate client buffers for "fast" AudioRecord.
192// Eventually it will be the single buffer that FastCapture writes into via HAL read(),
193// and that all "fast" AudioRecord clients read from. In either case, the size can be small.
Glenn Kasten9f81de32014-07-27 15:02:23 -0700194static const size_t kRecordThreadReadOnlyHeapSize = 0x2000;
Glenn Kastenb880f5e2014-05-07 08:43:45 -0700195
Eric Laurent81784c32012-11-19 14:55:58 -0800196// ----------------------------------------------------------------------------
197
Glenn Kasten03490092014-05-27 12:30:54 -0700198static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
199
200static void sFastTrackMultiplierInit()
201{
202 char value[PROPERTY_VALUE_MAX];
203 if (property_get("af.fast_track_multiplier", value, NULL) > 0) {
204 char *endptr;
205 unsigned long ul = strtoul(value, &endptr, 0);
206 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) {
207 sFastTrackMultiplier = (int) ul;
208 }
209 }
210}
211
212// ----------------------------------------------------------------------------
213
Eric Laurent81784c32012-11-19 14:55:58 -0800214#ifdef ADD_BATTERY_DATA
215// To collect the amplifier usage
216static void addBatteryData(uint32_t params) {
217 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
218 if (service == NULL) {
219 // it already logged
220 return;
221 }
222
223 service->addBatteryData(params);
224}
225#endif
226
Andy Hung3f0c9022016-01-15 17:49:46 -0800227// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset
228struct {
229 // call when you acquire a partial wakelock
230 void acquire(const sp<IBinder> &wakeLockToken) {
231 pthread_mutex_lock(&mLock);
232 if (wakeLockToken.get() == nullptr) {
233 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
234 } else {
235 if (mCount == 0) {
236 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
237 }
238 ++mCount;
239 }
240 pthread_mutex_unlock(&mLock);
241 }
242
243 // call when you release a partial wakelock.
244 void release(const sp<IBinder> &wakeLockToken) {
245 if (wakeLockToken.get() == nullptr) {
246 return;
247 }
248 pthread_mutex_lock(&mLock);
249 if (--mCount < 0) {
250 ALOGE("negative wakelock count");
251 mCount = 0;
252 }
253 pthread_mutex_unlock(&mLock);
254 }
255
256 // retrieves the boottime timebase offset from monotonic.
257 int64_t getBoottimeOffset() {
258 pthread_mutex_lock(&mLock);
259 int64_t boottimeOffset = mBoottimeOffset;
260 pthread_mutex_unlock(&mLock);
261 return boottimeOffset;
262 }
263
264 // Adjusts the timebase offset between TIMEBASE_MONOTONIC
265 // and the selected timebase.
266 // Currently only TIMEBASE_BOOTTIME is allowed.
267 //
268 // This only needs to be called upon acquiring the first partial wakelock
269 // after all other partial wakelocks are released.
270 //
271 // We do an empirical measurement of the offset rather than parsing
272 // /proc/timer_list since the latter is not a formal kernel ABI.
273 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) {
274 int clockbase;
275 switch (timebase) {
276 case ExtendedTimestamp::TIMEBASE_BOOTTIME:
277 clockbase = SYSTEM_TIME_BOOTTIME;
278 break;
279 default:
280 LOG_ALWAYS_FATAL("invalid timebase %d", timebase);
281 break;
282 }
283 // try three times to get the clock offset, choose the one
284 // with the minimum gap in measurements.
285 const int tries = 3;
286 nsecs_t bestGap, measured;
287 for (int i = 0; i < tries; ++i) {
288 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC);
289 const nsecs_t tbase = systemTime(clockbase);
290 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC);
291 const nsecs_t gap = tmono2 - tmono;
292 if (i == 0 || gap < bestGap) {
293 bestGap = gap;
294 measured = tbase - ((tmono + tmono2) >> 1);
295 }
296 }
297
298 // to avoid micro-adjusting, we don't change the timebase
299 // unless it is significantly different.
300 //
301 // Assumption: It probably takes more than toleranceNs to
302 // suspend and resume the device.
303 static int64_t toleranceNs = 10000; // 10 us
304 if (llabs(*offset - measured) > toleranceNs) {
305 ALOGV("Adjusting timebase offset old: %lld new: %lld",
306 (long long)*offset, (long long)measured);
307 *offset = measured;
308 }
309 }
310
311 pthread_mutex_t mLock;
312 int32_t mCount;
313 int64_t mBoottimeOffset;
314} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization
Eric Laurent81784c32012-11-19 14:55:58 -0800315
316// ----------------------------------------------------------------------------
317// CPU Stats
318// ----------------------------------------------------------------------------
319
320class CpuStats {
321public:
322 CpuStats();
323 void sample(const String8 &title);
324#ifdef DEBUG_CPU_USAGE
325private:
326 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns
327 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
328
329 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
330
331 int mCpuNum; // thread's current CPU number
332 int mCpukHz; // frequency of thread's current CPU in kHz
333#endif
334};
335
336CpuStats::CpuStats()
337#ifdef DEBUG_CPU_USAGE
338 : mCpuNum(-1), mCpukHz(-1)
339#endif
340{
341}
342
Glenn Kasten0f11b512014-01-31 16:18:54 -0800343void CpuStats::sample(const String8 &title
344#ifndef DEBUG_CPU_USAGE
345 __unused
346#endif
347 ) {
Eric Laurent81784c32012-11-19 14:55:58 -0800348#ifdef DEBUG_CPU_USAGE
349 // get current thread's delta CPU time in wall clock ns
350 double wcNs;
351 bool valid = mCpuUsage.sampleAndEnable(wcNs);
352
353 // record sample for wall clock statistics
354 if (valid) {
355 mWcStats.sample(wcNs);
356 }
357
358 // get the current CPU number
359 int cpuNum = sched_getcpu();
360
361 // get the current CPU frequency in kHz
362 int cpukHz = mCpuUsage.getCpukHz(cpuNum);
363
364 // check if either CPU number or frequency changed
365 if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
366 mCpuNum = cpuNum;
367 mCpukHz = cpukHz;
368 // ignore sample for purposes of cycles
369 valid = false;
370 }
371
372 // if no change in CPU number or frequency, then record sample for cycle statistics
373 if (valid && mCpukHz > 0) {
374 double cycles = wcNs * cpukHz * 0.000001;
375 mHzStats.sample(cycles);
376 }
377
378 unsigned n = mWcStats.n();
379 // mCpuUsage.elapsed() is expensive, so don't call it every loop
380 if ((n & 127) == 1) {
381 long long elapsed = mCpuUsage.elapsed();
382 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
383 double perLoop = elapsed / (double) n;
384 double perLoop100 = perLoop * 0.01;
385 double perLoop1k = perLoop * 0.001;
386 double mean = mWcStats.mean();
387 double stddev = mWcStats.stddev();
388 double minimum = mWcStats.minimum();
389 double maximum = mWcStats.maximum();
390 double meanCycles = mHzStats.mean();
391 double stddevCycles = mHzStats.stddev();
392 double minCycles = mHzStats.minimum();
393 double maxCycles = mHzStats.maximum();
394 mCpuUsage.resetElapsed();
395 mWcStats.reset();
396 mHzStats.reset();
397 ALOGD("CPU usage for %s over past %.1f secs\n"
398 " (%u mixer loops at %.1f mean ms per loop):\n"
399 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
400 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
401 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
402 title.string(),
403 elapsed * .000000001, n, perLoop * .000001,
404 mean * .001,
405 stddev * .001,
406 minimum * .001,
407 maximum * .001,
408 mean / perLoop100,
409 stddev / perLoop100,
410 minimum / perLoop100,
411 maximum / perLoop100,
412 meanCycles / perLoop1k,
413 stddevCycles / perLoop1k,
414 minCycles / perLoop1k,
415 maxCycles / perLoop1k);
416
417 }
418 }
419#endif
420};
421
422// ----------------------------------------------------------------------------
423// ThreadBase
424// ----------------------------------------------------------------------------
425
Glenn Kasten97b7b752014-09-28 13:04:24 -0700426// static
427const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
428{
429 switch (type) {
430 case MIXER:
431 return "MIXER";
432 case DIRECT:
433 return "DIRECT";
434 case DUPLICATING:
435 return "DUPLICATING";
436 case RECORD:
437 return "RECORD";
438 case OFFLOAD:
439 return "OFFLOAD";
440 default:
441 return "unknown";
442 }
443}
444
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800445String8 devicesToString(audio_devices_t devices)
446{
447 static const struct mapping {
448 audio_devices_t mDevices;
449 const char * mString;
450 } mappingsOut[] = {
Glenn Kasten818da522015-12-02 13:53:26 -0800451 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"},
452 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"},
453 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"},
454 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"},
455 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"},
456 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"},
457 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"},
458 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"},
459 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"},
460 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"},
461 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"},
462 {AUDIO_DEVICE_OUT_HDMI, "HDMI"},
463 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"},
464 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"},
465 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"},
466 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"},
467 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"},
468 {AUDIO_DEVICE_OUT_LINE, "LINE"},
469 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"},
470 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"},
471 {AUDIO_DEVICE_OUT_FM, "FM"},
472 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"},
473 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"},
474 {AUDIO_DEVICE_OUT_IP, "IP"},
Eric Laurent58545be2016-02-22 18:54:20 -0800475 {AUDIO_DEVICE_OUT_BUS, "BUS"},
Glenn Kasten818da522015-12-02 13:53:26 -0800476 {AUDIO_DEVICE_NONE, "NONE"}, // must be last
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800477 }, mappingsIn[] = {
Glenn Kasten818da522015-12-02 13:53:26 -0800478 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"},
479 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"},
480 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"},
481 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"},
482 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"},
483 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"},
484 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"},
485 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"},
486 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"},
487 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"},
488 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"},
489 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"},
490 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"},
491 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"},
492 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"},
493 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"},
494 {AUDIO_DEVICE_IN_LINE, "LINE"},
495 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"},
496 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"},
497 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"},
498 {AUDIO_DEVICE_IN_IP, "IP"},
Eric Laurent58545be2016-02-22 18:54:20 -0800499 {AUDIO_DEVICE_IN_BUS, "BUS"},
Glenn Kasten818da522015-12-02 13:53:26 -0800500 {AUDIO_DEVICE_NONE, "NONE"}, // must be last
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800501 };
502 String8 result;
503 audio_devices_t allDevices = AUDIO_DEVICE_NONE;
504 const mapping *entry;
505 if (devices & AUDIO_DEVICE_BIT_IN) {
506 devices &= ~AUDIO_DEVICE_BIT_IN;
507 entry = mappingsIn;
508 } else {
509 entry = mappingsOut;
510 }
511 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) {
512 allDevices = (audio_devices_t) (allDevices | entry->mDevices);
513 if (devices & entry->mDevices) {
514 if (!result.isEmpty()) {
515 result.append("|");
516 }
517 result.append(entry->mString);
518 }
519 }
520 if (devices & ~allDevices) {
521 if (!result.isEmpty()) {
522 result.append("|");
523 }
524 result.appendFormat("0x%X", devices & ~allDevices);
525 }
526 if (result.isEmpty()) {
527 result.append(entry->mString);
528 }
529 return result;
530}
531
532String8 inputFlagsToString(audio_input_flags_t flags)
533{
534 static const struct mapping {
535 audio_input_flags_t mFlag;
536 const char * mString;
537 } mappings[] = {
Glenn Kasten818da522015-12-02 13:53:26 -0800538 {AUDIO_INPUT_FLAG_FAST, "FAST"},
539 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"},
540 {AUDIO_INPUT_FLAG_RAW, "RAW"},
541 {AUDIO_INPUT_FLAG_SYNC, "SYNC"},
542 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800543 };
544 String8 result;
545 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE;
546 const mapping *entry;
547 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) {
548 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag);
549 if (flags & entry->mFlag) {
550 if (!result.isEmpty()) {
551 result.append("|");
552 }
553 result.append(entry->mString);
554 }
555 }
556 if (flags & ~allFlags) {
557 if (!result.isEmpty()) {
558 result.append("|");
559 }
560 result.appendFormat("0x%X", flags & ~allFlags);
561 }
562 if (result.isEmpty()) {
563 result.append(entry->mString);
564 }
565 return result;
566}
567
568String8 outputFlagsToString(audio_output_flags_t flags)
Glenn Kasten97b7b752014-09-28 13:04:24 -0700569{
570 static const struct mapping {
571 audio_output_flags_t mFlag;
572 const char * mString;
573 } mappings[] = {
Glenn Kasten818da522015-12-02 13:53:26 -0800574 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"},
575 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"},
576 {AUDIO_OUTPUT_FLAG_FAST, "FAST"},
577 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"},
578 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"},
579 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"},
580 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"},
581 {AUDIO_OUTPUT_FLAG_RAW, "RAW"},
582 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"},
583 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"},
584 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last
Glenn Kasten97b7b752014-09-28 13:04:24 -0700585 };
586 String8 result;
587 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE;
588 const mapping *entry;
589 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) {
590 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag);
591 if (flags & entry->mFlag) {
592 if (!result.isEmpty()) {
593 result.append("|");
594 }
595 result.append(entry->mString);
596 }
597 }
598 if (flags & ~allFlags) {
599 if (!result.isEmpty()) {
600 result.append("|");
601 }
602 result.appendFormat("0x%X", flags & ~allFlags);
603 }
604 if (result.isEmpty()) {
605 result.append(entry->mString);
606 }
607 return result;
608}
609
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800610const char *sourceToString(audio_source_t source)
611{
612 switch (source) {
613 case AUDIO_SOURCE_DEFAULT: return "default";
614 case AUDIO_SOURCE_MIC: return "mic";
615 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink";
616 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink";
617 case AUDIO_SOURCE_VOICE_CALL: return "voice call";
618 case AUDIO_SOURCE_CAMCORDER: return "camcorder";
619 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition";
620 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication";
621 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix";
rago8a397d52015-12-02 11:27:57 -0800622 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed";
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800623 case AUDIO_SOURCE_FM_TUNER: return "FM tuner";
624 case AUDIO_SOURCE_HOTWORD: return "hotword";
625 default: return "unknown";
626 }
627}
628
Eric Laurent81784c32012-11-19 14:55:58 -0800629AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
Eric Laurent72e3f392015-05-20 14:43:50 -0700630 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady)
Eric Laurent81784c32012-11-19 14:55:58 -0800631 : Thread(false /*canCallJava*/),
632 mType(type),
Glenn Kasten9b58f632013-07-16 11:37:48 -0700633 mAudioFlinger(audioFlinger),
Glenn Kasten70949c42013-08-06 07:40:12 -0700634 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
Glenn Kastendeca2ae2014-02-07 10:25:56 -0800635 // are set by PlaybackThread::readOutputParameters_l() or
636 // RecordThread::readInputParameters_l()
Eric Laurentfd477972013-10-25 18:10:40 -0700637 //FIXME: mStandby should be true here. Is this some kind of hack?
Eric Laurent81784c32012-11-19 14:55:58 -0800638 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700639 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE),
640 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
Eric Laurent81784c32012-11-19 14:55:58 -0800641 // mName will be set by concrete (non-virtual) subclass
Eric Laurent72e3f392015-05-20 14:43:50 -0700642 mDeathRecipient(new PMDeathRecipient(this)),
Wei Jia3f273d12015-11-24 09:06:49 -0800643 mSystemReady(systemReady),
644 mNotifiedBatteryStart(false)
Eric Laurent81784c32012-11-19 14:55:58 -0800645{
Eric Laurent296fb132015-05-01 11:38:42 -0700646 memset(&mPatch, 0, sizeof(struct audio_patch));
Eric Laurent81784c32012-11-19 14:55:58 -0800647}
648
649AudioFlinger::ThreadBase::~ThreadBase()
650{
Glenn Kastenc6ae3c82013-07-17 09:08:51 -0700651 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns
Glenn Kastenc6ae3c82013-07-17 09:08:51 -0700652 mConfigEvents.clear();
653
Eric Laurent81784c32012-11-19 14:55:58 -0800654 // do not lock the mutex in destructor
655 releaseWakeLock_l();
656 if (mPowerManager != 0) {
Marco Nelissen06b46062014-11-14 07:58:25 -0800657 sp<IBinder> binder = IInterface::asBinder(mPowerManager);
Eric Laurent81784c32012-11-19 14:55:58 -0800658 binder->unlinkToDeath(mDeathRecipient);
659 }
660}
661
Glenn Kastencf04c2c2013-08-06 07:41:16 -0700662status_t AudioFlinger::ThreadBase::readyToRun()
663{
664 status_t status = initCheck();
665 if (status == NO_ERROR) {
666 ALOGI("AudioFlinger's thread %p ready to run", this);
667 } else {
668 ALOGE("No working audio driver found.");
669 }
670 return status;
671}
672
Eric Laurent81784c32012-11-19 14:55:58 -0800673void AudioFlinger::ThreadBase::exit()
674{
675 ALOGV("ThreadBase::exit");
676 // do any cleanup required for exit to succeed
677 preExit();
678 {
679 // This lock prevents the following race in thread (uniprocessor for illustration):
680 // if (!exitPending()) {
681 // // context switch from here to exit()
682 // // exit() calls requestExit(), what exitPending() observes
683 // // exit() calls signal(), which is dropped since no waiters
684 // // context switch back from exit() to here
685 // mWaitWorkCV.wait(...);
686 // // now thread is hung
687 // }
688 AutoMutex lock(mLock);
689 requestExit();
690 mWaitWorkCV.broadcast();
691 }
692 // When Thread::requestExitAndWait is made virtual and this method is renamed to
693 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
694 requestExitAndWait();
695}
696
697status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
698{
Eric Laurent81784c32012-11-19 14:55:58 -0800699 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
700 Mutex::Autolock _l(mLock);
701
Eric Laurent10351942014-05-08 18:49:52 -0700702 return sendSetParameterConfigEvent_l(keyValuePairs);
703}
704
705// sendConfigEvent_l() must be called with ThreadBase::mLock held
706// Can temporarily release the lock if waiting for a reply from processConfigEvents_l().
707status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
708{
709 status_t status = NO_ERROR;
710
Eric Laurent72e3f392015-05-20 14:43:50 -0700711 if (event->mRequiresSystemReady && !mSystemReady) {
712 event->mWaitStatus = false;
713 mPendingConfigEvents.add(event);
714 return status;
715 }
Eric Laurent10351942014-05-08 18:49:52 -0700716 mConfigEvents.add(event);
Glenn Kastenc42e9b42016-03-21 11:35:03 -0700717 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType);
Eric Laurent81784c32012-11-19 14:55:58 -0800718 mWaitWorkCV.signal();
Eric Laurent10351942014-05-08 18:49:52 -0700719 mLock.unlock();
720 {
721 Mutex::Autolock _l(event->mLock);
722 while (event->mWaitStatus) {
723 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) {
724 event->mStatus = TIMED_OUT;
725 event->mWaitStatus = false;
726 }
727 }
728 status = event->mStatus;
Eric Laurent81784c32012-11-19 14:55:58 -0800729 }
Eric Laurent10351942014-05-08 18:49:52 -0700730 mLock.lock();
Eric Laurent81784c32012-11-19 14:55:58 -0800731 return status;
732}
733
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700734void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid)
Eric Laurent81784c32012-11-19 14:55:58 -0800735{
736 Mutex::Autolock _l(mLock);
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700737 sendIoConfigEvent_l(event, pid);
Eric Laurent81784c32012-11-19 14:55:58 -0800738}
739
740// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700741void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid)
Eric Laurent81784c32012-11-19 14:55:58 -0800742{
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700743 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid);
Eric Laurent10351942014-05-08 18:49:52 -0700744 sendConfigEvent_l(configEvent);
Eric Laurent81784c32012-11-19 14:55:58 -0800745}
746
Eric Laurent72e3f392015-05-20 14:43:50 -0700747void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio)
748{
749 Mutex::Autolock _l(mLock);
750 sendPrioConfigEvent_l(pid, tid, prio);
751}
752
Eric Laurent81784c32012-11-19 14:55:58 -0800753// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
754void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
755{
Eric Laurent10351942014-05-08 18:49:52 -0700756 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio);
757 sendConfigEvent_l(configEvent);
Eric Laurent81784c32012-11-19 14:55:58 -0800758}
759
Eric Laurent10351942014-05-08 18:49:52 -0700760// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held
761status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
Eric Laurent81784c32012-11-19 14:55:58 -0800762{
Andy Hung2ddee192015-12-18 17:34:44 -0800763 sp<ConfigEvent> configEvent;
764 AudioParameter param(keyValuePair);
765 int value;
766 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) {
767 setMasterMono_l(value != 0);
768 if (param.size() == 1) {
769 return NO_ERROR; // should be a solo parameter - we don't pass down
770 }
771 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT));
772 configEvent = new SetParameterConfigEvent(param.toString());
773 } else {
774 configEvent = new SetParameterConfigEvent(keyValuePair);
775 }
Eric Laurent10351942014-05-08 18:49:52 -0700776 return sendConfigEvent_l(configEvent);
Glenn Kastenf7773312013-08-13 16:00:42 -0700777}
778
Eric Laurent1c333e22014-05-20 10:48:17 -0700779status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent(
780 const struct audio_patch *patch,
781 audio_patch_handle_t *handle)
782{
783 Mutex::Autolock _l(mLock);
784 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle);
785 status_t status = sendConfigEvent_l(configEvent);
786 if (status == NO_ERROR) {
787 CreateAudioPatchConfigEventData *data =
788 (CreateAudioPatchConfigEventData *)configEvent->mData.get();
789 *handle = data->mHandle;
790 }
791 return status;
792}
793
794status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent(
795 const audio_patch_handle_t handle)
796{
797 Mutex::Autolock _l(mLock);
798 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle);
799 return sendConfigEvent_l(configEvent);
800}
801
802
Glenn Kasten2cfbf882013-08-14 13:12:11 -0700803// post condition: mConfigEvents.isEmpty()
Eric Laurent021cf962014-05-13 10:18:14 -0700804void AudioFlinger::ThreadBase::processConfigEvents_l()
Glenn Kastenf7773312013-08-13 16:00:42 -0700805{
Eric Laurent10351942014-05-08 18:49:52 -0700806 bool configChanged = false;
807
Eric Laurent81784c32012-11-19 14:55:58 -0800808 while (!mConfigEvents.isEmpty()) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -0700809 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size());
Eric Laurent10351942014-05-08 18:49:52 -0700810 sp<ConfigEvent> event = mConfigEvents[0];
Eric Laurent81784c32012-11-19 14:55:58 -0800811 mConfigEvents.removeAt(0);
Eric Laurent10351942014-05-08 18:49:52 -0700812 switch (event->mType) {
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700813 case CFG_EVENT_PRIO: {
Eric Laurent10351942014-05-08 18:49:52 -0700814 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get();
815 // FIXME Need to understand why this has to be done asynchronously
816 int err = requestPriority(data->mPid, data->mTid, data->mPrio,
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700817 true /*asynchronous*/);
818 if (err != 0) {
819 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
Eric Laurent10351942014-05-08 18:49:52 -0700820 data->mPrio, data->mPid, data->mTid, err);
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700821 }
822 } break;
823 case CFG_EVENT_IO: {
Eric Laurent10351942014-05-08 18:49:52 -0700824 IoConfigEventData *data = (IoConfigEventData *)event->mData.get();
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700825 ioConfigChanged(data->mEvent, data->mPid);
Eric Laurent10351942014-05-08 18:49:52 -0700826 } break;
827 case CFG_EVENT_SET_PARAMETER: {
828 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get();
829 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) {
830 configChanged = true;
Glenn Kastend5418eb2013-08-14 13:11:06 -0700831 }
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700832 } break;
Eric Laurent1c333e22014-05-20 10:48:17 -0700833 case CFG_EVENT_CREATE_AUDIO_PATCH: {
834 CreateAudioPatchConfigEventData *data =
835 (CreateAudioPatchConfigEventData *)event->mData.get();
836 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
837 } break;
838 case CFG_EVENT_RELEASE_AUDIO_PATCH: {
839 ReleaseAudioPatchConfigEventData *data =
840 (ReleaseAudioPatchConfigEventData *)event->mData.get();
841 event->mStatus = releaseAudioPatch_l(data->mHandle);
842 } break;
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700843 default:
Eric Laurent10351942014-05-08 18:49:52 -0700844 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700845 break;
Eric Laurent81784c32012-11-19 14:55:58 -0800846 }
Eric Laurent10351942014-05-08 18:49:52 -0700847 {
848 Mutex::Autolock _l(event->mLock);
849 if (event->mWaitStatus) {
850 event->mWaitStatus = false;
851 event->mCond.signal();
852 }
853 }
854 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this);
855 }
856
857 if (configChanged) {
858 cacheParameters_l();
Eric Laurent81784c32012-11-19 14:55:58 -0800859 }
Eric Laurent81784c32012-11-19 14:55:58 -0800860}
861
Marco Nelissenb2208842014-02-07 14:00:50 -0800862String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
863 String8 s;
Glenn Kastene1635ec2015-06-08 15:46:49 -0700864 const audio_channel_representation_t representation =
865 audio_channel_mask_get_representation(mask);
Andy Hungf98ec8d2015-05-19 12:53:24 -0700866
867 switch (representation) {
868 case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
869 if (output) {
870 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
871 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
872 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
873 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
874 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
875 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
876 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
877 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
878 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
879 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
880 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
881 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
882 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
883 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
884 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
885 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
886 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
887 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
888 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, ");
889 } else {
890 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
891 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
892 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
893 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
894 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
895 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
896 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
897 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
898 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
899 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
900 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
901 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
902 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
903 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
904 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, ");
905 }
906 const int len = s.length();
907 if (len > 2) {
Glenn Kasten57c4e6f2016-03-18 14:54:07 -0700908 (void) s.lockBuffer(len); // needed?
Andy Hungf98ec8d2015-05-19 12:53:24 -0700909 s.unlockBuffer(len - 2); // remove trailing ", "
910 }
911 return s;
Marco Nelissenb2208842014-02-07 14:00:50 -0800912 }
Andy Hungf98ec8d2015-05-19 12:53:24 -0700913 case AUDIO_CHANNEL_REPRESENTATION_INDEX:
914 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask));
915 return s;
916 default:
917 s.appendFormat("unknown mask, representation:%d bits:%#x",
918 representation, audio_channel_mask_get_bits(mask));
919 return s;
Marco Nelissenb2208842014-02-07 14:00:50 -0800920 }
Marco Nelissenb2208842014-02-07 14:00:50 -0800921}
922
Glenn Kasten0f11b512014-01-31 16:18:54 -0800923void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
Eric Laurent81784c32012-11-19 14:55:58 -0800924{
925 const size_t SIZE = 256;
926 char buffer[SIZE];
927 String8 result;
928
929 bool locked = AudioFlinger::dumpTryLock(mLock);
930 if (!locked) {
Glenn Kasten97b7b752014-09-28 13:04:24 -0700931 dprintf(fd, "thread %p may be deadlocked\n", this);
Eric Laurent81784c32012-11-19 14:55:58 -0800932 }
933
Glenn Kasten0b89bc02015-03-05 16:37:47 -0800934 dprintf(fd, " Thread name: %s\n", mThreadName);
Elliott Hughes87cebad2014-05-22 10:14:43 -0700935 dprintf(fd, " I/O handle: %d\n", mId);
936 dprintf(fd, " TID: %d\n", getTid());
937 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
Glenn Kasten97b7b752014-09-28 13:04:24 -0700938 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
Elliott Hughes87cebad2014-05-22 10:14:43 -0700939 dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
Glenn Kasten97b7b752014-09-28 13:04:24 -0700940 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
Glenn Kastenc42e9b42016-03-21 11:35:03 -0700941 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize);
Glenn Kasten97b7b752014-09-28 13:04:24 -0700942 dprintf(fd, " Channel count: %u\n", mChannelCount);
943 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask,
Marco Nelissenb2208842014-02-07 14:00:50 -0800944 channelMaskToString(mChannelMask, mType != RECORD).string());
Glenn Kastenf87c2f52015-08-21 08:03:57 -0700945 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
946 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize);
Elliott Hughes87cebad2014-05-22 10:14:43 -0700947 dprintf(fd, " Pending config events:");
Marco Nelissenb2208842014-02-07 14:00:50 -0800948 size_t numConfig = mConfigEvents.size();
949 if (numConfig) {
950 for (size_t i = 0; i < numConfig; i++) {
951 mConfigEvents[i]->dump(buffer, SIZE);
Elliott Hughes87cebad2014-05-22 10:14:43 -0700952 dprintf(fd, "\n %s", buffer);
Marco Nelissenb2208842014-02-07 14:00:50 -0800953 }
Elliott Hughes87cebad2014-05-22 10:14:43 -0700954 dprintf(fd, "\n");
Marco Nelissenb2208842014-02-07 14:00:50 -0800955 } else {
Elliott Hughes87cebad2014-05-22 10:14:43 -0700956 dprintf(fd, " none\n");
Eric Laurent81784c32012-11-19 14:55:58 -0800957 }
Glenn Kasten0b89bc02015-03-05 16:37:47 -0800958 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string());
959 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string());
960 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource));
Eric Laurent81784c32012-11-19 14:55:58 -0800961
962 if (locked) {
963 mLock.unlock();
964 }
965}
966
967void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
968{
969 const size_t SIZE = 256;
970 char buffer[SIZE];
971 String8 result;
972
Marco Nelissenb2208842014-02-07 14:00:50 -0800973 size_t numEffectChains = mEffectChains.size();
Narayan Kamath1d6fa7a2014-02-11 13:47:53 +0000974 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains);
Eric Laurent81784c32012-11-19 14:55:58 -0800975 write(fd, buffer, strlen(buffer));
976
Marco Nelissenb2208842014-02-07 14:00:50 -0800977 for (size_t i = 0; i < numEffectChains; ++i) {
Eric Laurent81784c32012-11-19 14:55:58 -0800978 sp<EffectChain> chain = mEffectChains[i];
979 if (chain != 0) {
980 chain->dump(fd, args);
981 }
982 }
983}
984
Marco Nelissene14a5d62013-10-03 08:51:24 -0700985void AudioFlinger::ThreadBase::acquireWakeLock(int uid)
Eric Laurent81784c32012-11-19 14:55:58 -0800986{
987 Mutex::Autolock _l(mLock);
Marco Nelissene14a5d62013-10-03 08:51:24 -0700988 acquireWakeLock_l(uid);
Eric Laurent81784c32012-11-19 14:55:58 -0800989}
990
Narayan Kamath014e7fa2013-10-14 15:03:38 +0100991String16 AudioFlinger::ThreadBase::getWakeLockTag()
992{
993 switch (mType) {
Glenn Kastenbcb14862015-03-05 17:11:21 -0800994 case MIXER:
995 return String16("AudioMix");
996 case DIRECT:
997 return String16("AudioDirectOut");
998 case DUPLICATING:
999 return String16("AudioDup");
1000 case RECORD:
1001 return String16("AudioIn");
1002 case OFFLOAD:
1003 return String16("AudioOffload");
1004 default:
1005 ALOG_ASSERT(false);
1006 return String16("AudioUnknown");
Narayan Kamath014e7fa2013-10-14 15:03:38 +01001007 }
1008}
1009
Marco Nelissene14a5d62013-10-03 08:51:24 -07001010void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid)
Eric Laurent81784c32012-11-19 14:55:58 -08001011{
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001012 getPowerManager_l();
Eric Laurent81784c32012-11-19 14:55:58 -08001013 if (mPowerManager != 0) {
1014 sp<IBinder> binder = new BBinder();
Marco Nelissene14a5d62013-10-03 08:51:24 -07001015 status_t status;
1016 if (uid >= 0) {
Eric Laurent547789d2013-10-04 11:46:55 -07001017 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK,
Marco Nelissene14a5d62013-10-03 08:51:24 -07001018 binder,
Narayan Kamath014e7fa2013-10-14 15:03:38 +01001019 getWakeLockTag(),
Marco Nelissendcb346b2015-09-09 10:47:29 -07001020 String16("audioserver"),
Glenn Kasten3abc2de2014-09-05 16:45:52 -07001021 uid,
1022 true /* FIXME force oneway contrary to .aidl */);
Marco Nelissene14a5d62013-10-03 08:51:24 -07001023 } else {
Eric Laurent547789d2013-10-04 11:46:55 -07001024 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
Marco Nelissene14a5d62013-10-03 08:51:24 -07001025 binder,
Narayan Kamath014e7fa2013-10-14 15:03:38 +01001026 getWakeLockTag(),
Marco Nelissendcb346b2015-09-09 10:47:29 -07001027 String16("audioserver"),
Glenn Kasten3abc2de2014-09-05 16:45:52 -07001028 true /* FIXME force oneway contrary to .aidl */);
Marco Nelissene14a5d62013-10-03 08:51:24 -07001029 }
Eric Laurent81784c32012-11-19 14:55:58 -08001030 if (status == NO_ERROR) {
1031 mWakeLockToken = binder;
1032 }
Glenn Kastend7dca052015-03-05 16:05:54 -08001033 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status);
Eric Laurent81784c32012-11-19 14:55:58 -08001034 }
Wei Jia3f273d12015-11-24 09:06:49 -08001035
1036 if (!mNotifiedBatteryStart) {
1037 BatteryNotifier::getInstance().noteStartAudio();
1038 mNotifiedBatteryStart = true;
1039 }
Andy Hung3f0c9022016-01-15 17:49:46 -08001040 gBoottime.acquire(mWakeLockToken);
Andy Hung818e7a32016-02-16 18:08:07 -08001041 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] =
1042 gBoottime.getBoottimeOffset();
Eric Laurent81784c32012-11-19 14:55:58 -08001043}
1044
1045void AudioFlinger::ThreadBase::releaseWakeLock()
1046{
1047 Mutex::Autolock _l(mLock);
1048 releaseWakeLock_l();
1049}
1050
1051void AudioFlinger::ThreadBase::releaseWakeLock_l()
1052{
Andy Hung3f0c9022016-01-15 17:49:46 -08001053 gBoottime.release(mWakeLockToken);
Eric Laurent81784c32012-11-19 14:55:58 -08001054 if (mWakeLockToken != 0) {
Glenn Kastend7dca052015-03-05 16:05:54 -08001055 ALOGV("releaseWakeLock_l() %s", mThreadName);
Eric Laurent81784c32012-11-19 14:55:58 -08001056 if (mPowerManager != 0) {
Glenn Kasten3abc2de2014-09-05 16:45:52 -07001057 mPowerManager->releaseWakeLock(mWakeLockToken, 0,
1058 true /* FIXME force oneway contrary to .aidl */);
Eric Laurent81784c32012-11-19 14:55:58 -08001059 }
1060 mWakeLockToken.clear();
1061 }
Wei Jia3f273d12015-11-24 09:06:49 -08001062
1063 if (mNotifiedBatteryStart) {
1064 BatteryNotifier::getInstance().noteStopAudio();
1065 mNotifiedBatteryStart = false;
1066 }
Eric Laurent81784c32012-11-19 14:55:58 -08001067}
1068
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001069void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) {
1070 Mutex::Autolock _l(mLock);
1071 updateWakeLockUids_l(uids);
1072}
1073
1074void AudioFlinger::ThreadBase::getPowerManager_l() {
Eric Laurent72e3f392015-05-20 14:43:50 -07001075 if (mSystemReady && mPowerManager == 0) {
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001076 // use checkService() to avoid blocking if power service is not up yet
1077 sp<IBinder> binder =
1078 defaultServiceManager()->checkService(String16("power"));
1079 if (binder == 0) {
Glenn Kastend7dca052015-03-05 16:05:54 -08001080 ALOGW("Thread %s cannot connect to the power manager service", mThreadName);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001081 } else {
1082 mPowerManager = interface_cast<IPowerManager>(binder);
1083 binder->linkToDeath(mDeathRecipient);
1084 }
1085 }
1086}
1087
1088void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001089 getPowerManager_l();
Andy Hung438e7572015-12-14 15:51:17 -08001090 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called.
1091 if (mSystemReady) {
1092 ALOGE("no wake lock to update, but system ready!");
1093 } else {
1094 ALOGW("no wake lock to update, system not ready yet");
1095 }
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001096 return;
1097 }
1098 if (mPowerManager != 0) {
1099 sp<IBinder> binder = new BBinder();
1100 status_t status;
Glenn Kasten3abc2de2014-09-05 16:45:52 -07001101 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(),
1102 true /* FIXME force oneway contrary to .aidl */);
Eric Laurent4d231dc2016-03-11 18:38:23 -08001103 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001104 }
1105}
1106
Eric Laurent81784c32012-11-19 14:55:58 -08001107void AudioFlinger::ThreadBase::clearPowerManager()
1108{
1109 Mutex::Autolock _l(mLock);
1110 releaseWakeLock_l();
1111 mPowerManager.clear();
1112}
1113
Glenn Kasten0f11b512014-01-31 16:18:54 -08001114void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08001115{
1116 sp<ThreadBase> thread = mThread.promote();
1117 if (thread != 0) {
1118 thread->clearPowerManager();
1119 }
1120 ALOGW("power manager service died !!!");
1121}
1122
1123void AudioFlinger::ThreadBase::setEffectSuspended(
Glenn Kastend848eb42016-03-08 13:42:11 -08001124 const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001125{
1126 Mutex::Autolock _l(mLock);
1127 setEffectSuspended_l(type, suspend, sessionId);
1128}
1129
1130void AudioFlinger::ThreadBase::setEffectSuspended_l(
Glenn Kastend848eb42016-03-08 13:42:11 -08001131 const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001132{
1133 sp<EffectChain> chain = getEffectChain_l(sessionId);
1134 if (chain != 0) {
1135 if (type != NULL) {
1136 chain->setEffectSuspended_l(type, suspend);
1137 } else {
1138 chain->setEffectSuspendedAll_l(suspend);
1139 }
1140 }
1141
1142 updateSuspendedSessions_l(type, suspend, sessionId);
1143}
1144
1145void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1146{
1147 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
1148 if (index < 0) {
1149 return;
1150 }
1151
1152 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects =
1153 mSuspendedSessions.valueAt(index);
1154
1155 for (size_t i = 0; i < sessionEffects.size(); i++) {
1156 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
1157 for (int j = 0; j < desc->mRefCount; j++) {
1158 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1159 chain->setEffectSuspendedAll_l(true);
1160 } else {
1161 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1162 desc->mType.timeLow);
1163 chain->setEffectSuspended_l(&desc->mType, true);
1164 }
1165 }
1166 }
1167}
1168
1169void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1170 bool suspend,
Glenn Kastend848eb42016-03-08 13:42:11 -08001171 audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001172{
1173 ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
1174
1175 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1176
1177 if (suspend) {
1178 if (index >= 0) {
1179 sessionEffects = mSuspendedSessions.valueAt(index);
1180 } else {
1181 mSuspendedSessions.add(sessionId, sessionEffects);
1182 }
1183 } else {
1184 if (index < 0) {
1185 return;
1186 }
1187 sessionEffects = mSuspendedSessions.valueAt(index);
1188 }
1189
1190
1191 int key = EffectChain::kKeyForSuspendAll;
1192 if (type != NULL) {
1193 key = type->timeLow;
1194 }
1195 index = sessionEffects.indexOfKey(key);
1196
1197 sp<SuspendedSessionDesc> desc;
1198 if (suspend) {
1199 if (index >= 0) {
1200 desc = sessionEffects.valueAt(index);
1201 } else {
1202 desc = new SuspendedSessionDesc();
1203 if (type != NULL) {
1204 desc->mType = *type;
1205 }
1206 sessionEffects.add(key, desc);
1207 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1208 }
1209 desc->mRefCount++;
1210 } else {
1211 if (index < 0) {
1212 return;
1213 }
1214 desc = sessionEffects.valueAt(index);
1215 if (--desc->mRefCount == 0) {
1216 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1217 sessionEffects.removeItemsAt(index);
1218 if (sessionEffects.isEmpty()) {
1219 ALOGV("updateSuspendedSessions_l() restore removing session %d",
1220 sessionId);
1221 mSuspendedSessions.removeItem(sessionId);
1222 }
1223 }
1224 }
1225 if (!sessionEffects.isEmpty()) {
1226 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1227 }
1228}
1229
1230void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1231 bool enabled,
Glenn Kastend848eb42016-03-08 13:42:11 -08001232 audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001233{
1234 Mutex::Autolock _l(mLock);
1235 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1236}
1237
1238void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1239 bool enabled,
Glenn Kastend848eb42016-03-08 13:42:11 -08001240 audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001241{
1242 if (mType != RECORD) {
1243 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1244 // another session. This gives the priority to well behaved effect control panels
1245 // and applications not using global effects.
1246 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
1247 // global effects
1248 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
1249 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1250 }
1251 }
1252
1253 sp<EffectChain> chain = getEffectChain_l(sessionId);
1254 if (chain != 0) {
1255 chain->checkSuspendOnEffectEnabled(effect, enabled);
1256 }
1257}
1258
1259// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held
1260sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
1261 const sp<AudioFlinger::Client>& client,
1262 const sp<IEffectClient>& effectClient,
1263 int32_t priority,
Glenn Kastend848eb42016-03-08 13:42:11 -08001264 audio_session_t sessionId,
Eric Laurent81784c32012-11-19 14:55:58 -08001265 effect_descriptor_t *desc,
1266 int *enabled,
Glenn Kasten9156ef32013-08-06 15:39:08 -07001267 status_t *status)
Eric Laurent81784c32012-11-19 14:55:58 -08001268{
1269 sp<EffectModule> effect;
1270 sp<EffectHandle> handle;
1271 status_t lStatus;
1272 sp<EffectChain> chain;
1273 bool chainCreated = false;
1274 bool effectCreated = false;
1275 bool effectRegistered = false;
1276
1277 lStatus = initCheck();
1278 if (lStatus != NO_ERROR) {
1279 ALOGW("createEffect_l() Audio driver not initialized.");
1280 goto Exit;
1281 }
1282
Andy Hung98ef9782014-03-04 14:46:50 -08001283 // Reject any effect on Direct output threads for now, since the format of
1284 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo).
1285 if (mType == DIRECT) {
1286 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s",
Glenn Kastend7dca052015-03-05 16:05:54 -08001287 desc->name, mThreadName);
Andy Hung98ef9782014-03-04 14:46:50 -08001288 lStatus = BAD_VALUE;
1289 goto Exit;
1290 }
1291
Andy Hung389cfdb2014-08-07 17:49:53 -07001292 // Reject any effect on mixer or duplicating multichannel sinks.
Andy Hung9a592762014-07-21 21:56:01 -07001293 // TODO: fix both format and multichannel issues with effects.
Andy Hung389cfdb2014-08-07 17:49:53 -07001294 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) {
1295 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads",
1296 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING");
Andy Hung9a592762014-07-21 21:56:01 -07001297 lStatus = BAD_VALUE;
1298 goto Exit;
1299 }
1300
Eric Laurent5baf2af2013-09-12 17:37:00 -07001301 // Allow global effects only on offloaded and mixer threads
1302 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1303 switch (mType) {
1304 case MIXER:
1305 case OFFLOAD:
1306 break;
1307 case DIRECT:
1308 case DUPLICATING:
1309 case RECORD:
1310 default:
Glenn Kastend7dca052015-03-05 16:05:54 -08001311 ALOGW("createEffect_l() Cannot add global effect %s on thread %s",
1312 desc->name, mThreadName);
Eric Laurent5baf2af2013-09-12 17:37:00 -07001313 lStatus = BAD_VALUE;
1314 goto Exit;
1315 }
Eric Laurent81784c32012-11-19 14:55:58 -08001316 }
Eric Laurent5baf2af2013-09-12 17:37:00 -07001317
Eric Laurent81784c32012-11-19 14:55:58 -08001318 // Only Pre processor effects are allowed on input threads and only on input threads
1319 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
1320 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
1321 desc->name, desc->flags, mType);
1322 lStatus = BAD_VALUE;
1323 goto Exit;
1324 }
1325
1326 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
1327
1328 { // scope for mLock
1329 Mutex::Autolock _l(mLock);
1330
1331 // check for existing effect chain with the requested audio session
1332 chain = getEffectChain_l(sessionId);
1333 if (chain == 0) {
1334 // create a new chain for this session
1335 ALOGV("createEffect_l() new effect chain for session %d", sessionId);
1336 chain = new EffectChain(this, sessionId);
1337 addEffectChain_l(chain);
1338 chain->setStrategy(getStrategyForSession_l(sessionId));
1339 chainCreated = true;
1340 } else {
1341 effect = chain->getEffectFromDesc_l(desc);
1342 }
1343
1344 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
1345
1346 if (effect == 0) {
Glenn Kasteneeecb982016-02-26 10:44:04 -08001347 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
Eric Laurent81784c32012-11-19 14:55:58 -08001348 // Check CPU and memory usage
1349 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
1350 if (lStatus != NO_ERROR) {
1351 goto Exit;
1352 }
1353 effectRegistered = true;
1354 // create a new effect module if none present in the chain
1355 effect = new EffectModule(this, chain, desc, id, sessionId);
1356 lStatus = effect->status();
1357 if (lStatus != NO_ERROR) {
1358 goto Exit;
1359 }
Eric Laurent5baf2af2013-09-12 17:37:00 -07001360 effect->setOffloaded(mType == OFFLOAD, mId);
1361
Eric Laurent81784c32012-11-19 14:55:58 -08001362 lStatus = chain->addEffect_l(effect);
1363 if (lStatus != NO_ERROR) {
1364 goto Exit;
1365 }
1366 effectCreated = true;
1367
1368 effect->setDevice(mOutDevice);
1369 effect->setDevice(mInDevice);
1370 effect->setMode(mAudioFlinger->getMode());
1371 effect->setAudioSource(mAudioSource);
1372 }
1373 // create effect handle and connect it to effect module
1374 handle = new EffectHandle(effect, client, effectClient, priority);
Glenn Kastene75da402013-11-20 13:54:52 -08001375 lStatus = handle->initCheck();
1376 if (lStatus == OK) {
1377 lStatus = effect->addHandle(handle.get());
1378 }
Eric Laurent81784c32012-11-19 14:55:58 -08001379 if (enabled != NULL) {
1380 *enabled = (int)effect->isEnabled();
1381 }
1382 }
1383
1384Exit:
1385 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
1386 Mutex::Autolock _l(mLock);
1387 if (effectCreated) {
1388 chain->removeEffect_l(effect);
1389 }
1390 if (effectRegistered) {
1391 AudioSystem::unregisterEffect(effect->id());
1392 }
1393 if (chainCreated) {
1394 removeEffectChain_l(chain);
1395 }
1396 handle.clear();
1397 }
1398
Glenn Kasten9156ef32013-08-06 15:39:08 -07001399 *status = lStatus;
Eric Laurent81784c32012-11-19 14:55:58 -08001400 return handle;
1401}
1402
Glenn Kastend848eb42016-03-08 13:42:11 -08001403sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId,
1404 int effectId)
Eric Laurent81784c32012-11-19 14:55:58 -08001405{
1406 Mutex::Autolock _l(mLock);
1407 return getEffect_l(sessionId, effectId);
1408}
1409
Glenn Kastend848eb42016-03-08 13:42:11 -08001410sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId,
1411 int effectId)
Eric Laurent81784c32012-11-19 14:55:58 -08001412{
1413 sp<EffectChain> chain = getEffectChain_l(sessionId);
1414 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
1415}
1416
1417// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
1418// PlaybackThread::mLock held
1419status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
1420{
1421 // check for existing effect chain with the requested audio session
Glenn Kastend848eb42016-03-08 13:42:11 -08001422 audio_session_t sessionId = effect->sessionId();
Eric Laurent81784c32012-11-19 14:55:58 -08001423 sp<EffectChain> chain = getEffectChain_l(sessionId);
1424 bool chainCreated = false;
1425
Eric Laurent5baf2af2013-09-12 17:37:00 -07001426 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(),
1427 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x",
1428 this, effect->desc().name, effect->desc().flags);
1429
Eric Laurent81784c32012-11-19 14:55:58 -08001430 if (chain == 0) {
1431 // create a new chain for this session
1432 ALOGV("addEffect_l() new effect chain for session %d", sessionId);
1433 chain = new EffectChain(this, sessionId);
1434 addEffectChain_l(chain);
1435 chain->setStrategy(getStrategyForSession_l(sessionId));
1436 chainCreated = true;
1437 }
1438 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
1439
1440 if (chain->getEffectFromId_l(effect->id()) != 0) {
1441 ALOGW("addEffect_l() %p effect %s already present in chain %p",
1442 this, effect->desc().name, chain.get());
1443 return BAD_VALUE;
1444 }
1445
Eric Laurent5baf2af2013-09-12 17:37:00 -07001446 effect->setOffloaded(mType == OFFLOAD, mId);
1447
Eric Laurent81784c32012-11-19 14:55:58 -08001448 status_t status = chain->addEffect_l(effect);
1449 if (status != NO_ERROR) {
1450 if (chainCreated) {
1451 removeEffectChain_l(chain);
1452 }
1453 return status;
1454 }
1455
1456 effect->setDevice(mOutDevice);
1457 effect->setDevice(mInDevice);
1458 effect->setMode(mAudioFlinger->getMode());
1459 effect->setAudioSource(mAudioSource);
1460 return NO_ERROR;
1461}
1462
1463void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
1464
1465 ALOGV("removeEffect_l() %p effect %p", this, effect.get());
1466 effect_descriptor_t desc = effect->desc();
1467 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
1468 detachAuxEffect_l(effect->id());
1469 }
1470
1471 sp<EffectChain> chain = effect->chain().promote();
1472 if (chain != 0) {
1473 // remove effect chain if removing last effect
1474 if (chain->removeEffect_l(effect) == 0) {
1475 removeEffectChain_l(chain);
1476 }
1477 } else {
1478 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
1479 }
1480}
1481
1482void AudioFlinger::ThreadBase::lockEffectChains_l(
1483 Vector< sp<AudioFlinger::EffectChain> >& effectChains)
1484{
1485 effectChains = mEffectChains;
1486 for (size_t i = 0; i < mEffectChains.size(); i++) {
1487 mEffectChains[i]->lock();
1488 }
1489}
1490
1491void AudioFlinger::ThreadBase::unlockEffectChains(
1492 const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
1493{
1494 for (size_t i = 0; i < effectChains.size(); i++) {
1495 effectChains[i]->unlock();
1496 }
1497}
1498
Glenn Kastend848eb42016-03-08 13:42:11 -08001499sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001500{
1501 Mutex::Autolock _l(mLock);
1502 return getEffectChain_l(sessionId);
1503}
1504
Glenn Kastend848eb42016-03-08 13:42:11 -08001505sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId)
1506 const
Eric Laurent81784c32012-11-19 14:55:58 -08001507{
1508 size_t size = mEffectChains.size();
1509 for (size_t i = 0; i < size; i++) {
1510 if (mEffectChains[i]->sessionId() == sessionId) {
1511 return mEffectChains[i];
1512 }
1513 }
1514 return 0;
1515}
1516
1517void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
1518{
1519 Mutex::Autolock _l(mLock);
1520 size_t size = mEffectChains.size();
1521 for (size_t i = 0; i < size; i++) {
1522 mEffectChains[i]->setMode_l(mode);
1523 }
1524}
1525
Eric Laurent83b88082014-06-20 18:31:16 -07001526void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config)
1527{
1528 config->type = AUDIO_PORT_TYPE_MIX;
1529 config->ext.mix.handle = mId;
1530 config->sample_rate = mSampleRate;
1531 config->format = mFormat;
1532 config->channel_mask = mChannelMask;
1533 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
1534 AUDIO_PORT_CONFIG_FORMAT;
1535}
1536
Eric Laurent72e3f392015-05-20 14:43:50 -07001537void AudioFlinger::ThreadBase::systemReady()
1538{
1539 Mutex::Autolock _l(mLock);
1540 if (mSystemReady) {
1541 return;
1542 }
1543 mSystemReady = true;
1544
1545 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) {
1546 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i));
1547 }
1548 mPendingConfigEvents.clear();
1549}
1550
Eric Laurent83b88082014-06-20 18:31:16 -07001551
Eric Laurent81784c32012-11-19 14:55:58 -08001552// ----------------------------------------------------------------------------
1553// Playback
1554// ----------------------------------------------------------------------------
1555
1556AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1557 AudioStreamOut* output,
1558 audio_io_handle_t id,
1559 audio_devices_t device,
Eric Laurent72e3f392015-05-20 14:43:50 -07001560 type_t type,
Eric Laurente93cc032016-05-05 10:15:10 -07001561 bool systemReady)
Eric Laurent72e3f392015-05-20 14:43:50 -07001562 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady),
Andy Hung2098f272014-02-27 14:00:06 -08001563 mNormalFrameCount(0), mSinkBuffer(NULL),
Andy Hung6146c082014-03-18 11:56:15 -07001564 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
Andy Hung69aed5f2014-02-25 17:24:40 -08001565 mMixerBuffer(NULL),
1566 mMixerBufferSize(0),
1567 mMixerBufferFormat(AUDIO_FORMAT_INVALID),
1568 mMixerBufferValid(false),
Andy Hung6146c082014-03-18 11:56:15 -07001569 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
Andy Hung98ef9782014-03-04 14:46:50 -08001570 mEffectBuffer(NULL),
1571 mEffectBufferSize(0),
1572 mEffectBufferFormat(AUDIO_FORMAT_INVALID),
1573 mEffectBufferValid(false),
Glenn Kastenc1fac192013-08-06 07:41:36 -07001574 mSuspended(0), mBytesWritten(0),
Andy Hungc54b1ff2016-02-23 14:07:07 -08001575 mFramesWritten(0),
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001576 mActiveTracksGeneration(0),
Eric Laurent81784c32012-11-19 14:55:58 -08001577 // mStreamTypes[] initialized in constructor body
1578 mOutput(output),
1579 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
1580 mMixerStatus(MIXER_IDLE),
1581 mMixerStatusIgnoringFastTracks(MIXER_IDLE),
Eric Laurentad9cb8b2015-05-26 16:38:19 -07001582 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs),
Eric Laurentbfb1b832013-01-07 09:53:42 -08001583 mBytesRemaining(0),
1584 mCurrentWriteLength(0),
1585 mUseAsyncWrite(false),
Eric Laurent3b4529e2013-09-05 18:09:19 -07001586 mWriteAckSequence(0),
1587 mDrainSequence(0),
Eric Laurentede6c3b2013-09-19 14:37:46 -07001588 mSignalPending(false),
Eric Laurent81784c32012-11-19 14:55:58 -08001589 mScreenState(AudioFlinger::mScreenState),
1590 // index 0 is reserved for normal mixer's submix
Glenn Kastendc2c50b2016-04-21 08:13:14 -07001591 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1),
Andy Hunge10393e2015-06-12 13:59:33 -07001592 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false)
Eric Laurent81784c32012-11-19 14:55:58 -08001593{
Glenn Kastend7dca052015-03-05 16:05:54 -08001594 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
1595 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
Eric Laurent81784c32012-11-19 14:55:58 -08001596
1597 // Assumes constructor is called by AudioFlinger with it's mLock held, but
1598 // it would be safer to explicitly pass initial masterVolume/masterMute as
1599 // parameter.
1600 //
1601 // If the HAL we are using has support for master volume or master mute,
1602 // then do not attenuate or mute during mixing (just leave the volume at 1.0
1603 // and the mute set to false).
1604 mMasterVolume = audioFlinger->masterVolume_l();
1605 mMasterMute = audioFlinger->masterMute_l();
1606 if (mOutput && mOutput->audioHwDev) {
1607 if (mOutput->audioHwDev->canSetMasterVolume()) {
1608 mMasterVolume = 1.0;
1609 }
1610
1611 if (mOutput->audioHwDev->canSetMasterMute()) {
1612 mMasterMute = false;
1613 }
1614 }
1615
Glenn Kastendeca2ae2014-02-07 10:25:56 -08001616 readOutputParameters_l();
Eric Laurent81784c32012-11-19 14:55:58 -08001617
Eric Laurent223fd5c2014-11-11 13:43:36 -08001618 // ++ operator does not compile
Glenn Kasten66e46352014-01-16 17:44:23 -08001619 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT;
Eric Laurent81784c32012-11-19 14:55:58 -08001620 stream = (audio_stream_type_t) (stream + 1)) {
1621 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
1622 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
1623 }
Eric Laurent81784c32012-11-19 14:55:58 -08001624}
1625
1626AudioFlinger::PlaybackThread::~PlaybackThread()
1627{
Glenn Kasten9e58b552013-01-18 15:09:48 -08001628 mAudioFlinger->unregisterWriter(mNBLogWriter);
Andy Hung010a1a12014-03-13 13:57:33 -07001629 free(mSinkBuffer);
Andy Hung69aed5f2014-02-25 17:24:40 -08001630 free(mMixerBuffer);
Andy Hung98ef9782014-03-04 14:46:50 -08001631 free(mEffectBuffer);
Eric Laurent81784c32012-11-19 14:55:58 -08001632}
1633
1634void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1635{
1636 dumpInternals(fd, args);
1637 dumpTracks(fd, args);
1638 dumpEffectChains(fd, args);
1639}
1640
Glenn Kasten0f11b512014-01-31 16:18:54 -08001641void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08001642{
1643 const size_t SIZE = 256;
1644 char buffer[SIZE];
1645 String8 result;
1646
Marco Nelissenb2208842014-02-07 14:00:50 -08001647 result.appendFormat(" Stream volumes in dB: ");
Eric Laurent81784c32012-11-19 14:55:58 -08001648 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
1649 const stream_type_t *st = &mStreamTypes[i];
1650 if (i > 0) {
1651 result.appendFormat(", ");
1652 }
1653 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
1654 if (st->mute) {
1655 result.append("M");
1656 }
1657 }
1658 result.append("\n");
1659 write(fd, result.string(), result.length());
1660 result.clear();
1661
Eric Laurent81784c32012-11-19 14:55:58 -08001662 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way.
1663 FastTrackUnderruns underruns = getFastTrackUnderruns(0);
Elliott Hughes87cebad2014-05-22 10:14:43 -07001664 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n",
Eric Laurent81784c32012-11-19 14:55:58 -08001665 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
Marco Nelissenb2208842014-02-07 14:00:50 -08001666
1667 size_t numtracks = mTracks.size();
1668 size_t numactive = mActiveTracks.size();
Glenn Kastenc42e9b42016-03-21 11:35:03 -07001669 dprintf(fd, " %zu Tracks", numtracks);
Marco Nelissenb2208842014-02-07 14:00:50 -08001670 size_t numactiveseen = 0;
1671 if (numtracks) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07001672 dprintf(fd, " of which %zu are active\n", numactive);
Marco Nelissenb2208842014-02-07 14:00:50 -08001673 Track::appendDumpHeader(result);
1674 for (size_t i = 0; i < numtracks; ++i) {
1675 sp<Track> track = mTracks[i];
1676 if (track != 0) {
1677 bool active = mActiveTracks.indexOf(track) >= 0;
1678 if (active) {
1679 numactiveseen++;
1680 }
1681 track->dump(buffer, SIZE, active);
1682 result.append(buffer);
1683 }
1684 }
1685 } else {
1686 result.append("\n");
1687 }
1688 if (numactiveseen != numactive) {
1689 // some tracks in the active list were not in the tracks list
1690 snprintf(buffer, SIZE, " The following tracks are in the active list but"
1691 " not in the track list\n");
1692 result.append(buffer);
1693 Track::appendDumpHeader(result);
1694 for (size_t i = 0; i < numactive; ++i) {
1695 sp<Track> track = mActiveTracks[i].promote();
1696 if (track != 0 && mTracks.indexOf(track) < 0) {
1697 track->dump(buffer, SIZE, true);
1698 result.append(buffer);
1699 }
1700 }
1701 }
1702
1703 write(fd, result.string(), result.size());
Eric Laurent81784c32012-11-19 14:55:58 -08001704}
1705
1706void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1707{
Glenn Kasten97b7b752014-09-28 13:04:24 -07001708 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type()));
Glenn Kasten44182c22015-03-05 17:12:23 -08001709
1710 dumpBase(fd, args);
1711
Elliott Hughes87cebad2014-05-22 10:14:43 -07001712 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
Glenn Kastenc42e9b42016-03-21 11:35:03 -07001713 dprintf(fd, " Last write occurred (msecs): %llu\n",
1714 (unsigned long long) ns2ms(systemTime() - mLastWriteTime));
Elliott Hughes87cebad2014-05-22 10:14:43 -07001715 dprintf(fd, " Total writes: %d\n", mNumWrites);
1716 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
1717 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
1718 dprintf(fd, " Suspend count: %d\n", mSuspended);
1719 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer);
1720 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
1721 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer);
1722 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
Eric Laurent42537be2016-01-08 17:16:42 -08001723 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs);
Glenn Kasten97b7b752014-09-28 13:04:24 -07001724 AudioStreamOut *output = mOutput;
1725 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
1726 String8 flagsAsString = outputFlagsToString(flags);
1727 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string());
Eric Laurent81784c32012-11-19 14:55:58 -08001728}
1729
1730// Thread virtuals
Eric Laurent81784c32012-11-19 14:55:58 -08001731
1732void AudioFlinger::PlaybackThread::onFirstRef()
1733{
Glenn Kastend7dca052015-03-05 16:05:54 -08001734 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
Eric Laurent81784c32012-11-19 14:55:58 -08001735}
1736
1737// ThreadBase virtuals
1738void AudioFlinger::PlaybackThread::preExit()
1739{
1740 ALOGV(" preExit()");
1741 // FIXME this is using hard-coded strings but in the future, this functionality will be
1742 // converted to use audio HAL extensions required to support tunneling
1743 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1");
1744}
1745
1746// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1747sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
1748 const sp<AudioFlinger::Client>& client,
1749 audio_stream_type_t streamType,
1750 uint32_t sampleRate,
1751 audio_format_t format,
1752 audio_channel_mask_t channelMask,
Glenn Kasten74935e42013-12-19 08:56:45 -08001753 size_t *pFrameCount,
Eric Laurent81784c32012-11-19 14:55:58 -08001754 const sp<IMemory>& sharedBuffer,
Glenn Kastend848eb42016-03-08 13:42:11 -08001755 audio_session_t sessionId,
Eric Laurent81784c32012-11-19 14:55:58 -08001756 IAudioFlinger::track_flags_t *flags,
1757 pid_t tid,
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001758 int uid,
Eric Laurent81784c32012-11-19 14:55:58 -08001759 status_t *status)
1760{
Glenn Kasten74935e42013-12-19 08:56:45 -08001761 size_t frameCount = *pFrameCount;
Eric Laurent81784c32012-11-19 14:55:58 -08001762 sp<Track> track;
1763 status_t lStatus;
1764
Eric Laurent81784c32012-11-19 14:55:58 -08001765 // client expresses a preference for FAST, but we get the final say
1766 if (*flags & IAudioFlinger::TRACK_FAST) {
1767 if (
Eric Laurent81784c32012-11-19 14:55:58 -08001768 // PCM data
1769 audio_is_linear_pcm(format) &&
Andy Hung1f439e12015-05-19 12:57:41 -07001770 // TODO: extract as a data library function that checks that a computationally
1771 // expensive downmixer is not required: isFastOutputChannelConversion()
Andy Hung9a592762014-07-21 21:56:01 -07001772 (channelMask == mChannelMask ||
Andy Hung1f439e12015-05-19 12:57:41 -07001773 mChannelMask != AUDIO_CHANNEL_OUT_STEREO ||
1774 (channelMask == AUDIO_CHANNEL_OUT_MONO
1775 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) &&
Eric Laurent81784c32012-11-19 14:55:58 -08001776 // hardware sample rate
1777 (sampleRate == mSampleRate) &&
Eric Laurent81784c32012-11-19 14:55:58 -08001778 // normal mixer has an associated fast mixer
1779 hasFastMixer() &&
1780 // there are sufficient fast track slots available
1781 (mFastTrackAvailMask != 0)
1782 // FIXME test that MixerThread for this fast track has a capable output HAL
1783 // FIXME add a permission test also?
1784 ) {
Andy Hunge0a269a2016-03-23 15:13:42 -07001785 // static tracks can have any nonzero framecount, streaming tracks check against minimum.
1786 if (sharedBuffer == 0) {
Glenn Kasten03490092014-05-27 12:30:54 -07001787 // read the fast track multiplier property the first time it is needed
1788 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
1789 if (ok != 0) {
1790 ALOGE("%s pthread_once failed: %d", __func__, ok);
1791 }
Andy Hunge0a269a2016-03-23 15:13:42 -07001792 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0
Eric Laurent81784c32012-11-19 14:55:58 -08001793 }
Glenn Kastenc42e9b42016-03-21 11:35:03 -07001794 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
Eric Laurent81784c32012-11-19 14:55:58 -08001795 frameCount, mFrameCount);
1796 } else {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07001797 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu "
1798 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
Andy Hung6146c082014-03-18 11:56:15 -07001799 "sampleRate=%u mSampleRate=%u "
Eric Laurent81784c32012-11-19 14:55:58 -08001800 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
Glenn Kastend79072e2016-01-06 08:41:20 -08001801 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
Eric Laurent81784c32012-11-19 14:55:58 -08001802 audio_is_linear_pcm(format),
1803 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
1804 *flags &= ~IAudioFlinger::TRACK_FAST;
Andy Hung0e48d252015-01-26 11:43:15 -08001805 }
1806 }
1807 // For normal PCM streaming tracks, update minimum frame count.
1808 // For compatibility with AudioTrack calculation, buffer depth is forced
1809 // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
1810 // This is probably too conservative, but legacy application code may depend on it.
1811 // If you change this calculation, also review the start threshold which is related.
1812 if (!(*flags & IAudioFlinger::TRACK_FAST)
Phil Burkfdb3c072016-02-09 10:47:02 -08001813 && audio_has_proportional_frames(format) && sharedBuffer == 0) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001814 // this must match AudioTrack.cpp calculateMinFrameCount().
1815 // TODO: Move to a common library
Eric Laurent81784c32012-11-19 14:55:58 -08001816 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
1817 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
1818 if (minBufCount < 2) {
1819 minBufCount = 2;
1820 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001821 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack
1822 // or the client should compute and pass in a larger buffer request.
Andy Hung0e48d252015-01-26 11:43:15 -08001823 size_t minFrameCount =
Andy Hung8edb8dc2015-03-26 19:13:55 -07001824 minBufCount * sourceFramesNeededWithTimestretch(
1825 sampleRate, mNormalFrameCount,
1826 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/);
Andy Hung0e48d252015-01-26 11:43:15 -08001827 if (frameCount < minFrameCount) { // including frameCount == 0
Eric Laurent81784c32012-11-19 14:55:58 -08001828 frameCount = minFrameCount;
1829 }
Eric Laurent81784c32012-11-19 14:55:58 -08001830 }
Glenn Kasten74935e42013-12-19 08:56:45 -08001831 *pFrameCount = frameCount;
Eric Laurent81784c32012-11-19 14:55:58 -08001832
Glenn Kastenc3df8382014-03-13 15:05:25 -07001833 switch (mType) {
1834
1835 case DIRECT:
Phil Burkfdb3c072016-02-09 10:47:02 -08001836 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()?
Eric Laurent81784c32012-11-19 14:55:58 -08001837 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
Glenn Kastencac3daa2014-02-07 09:47:14 -08001838 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
1839 "for output %p with format %#x",
Eric Laurent81784c32012-11-19 14:55:58 -08001840 sampleRate, format, channelMask, mOutput, mFormat);
1841 lStatus = BAD_VALUE;
1842 goto Exit;
1843 }
1844 }
Glenn Kastenc3df8382014-03-13 15:05:25 -07001845 break;
1846
1847 case OFFLOAD:
Eric Laurentbfb1b832013-01-07 09:53:42 -08001848 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
Glenn Kastencac3daa2014-02-07 09:47:14 -08001849 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
1850 "for output %p with format %#x",
Eric Laurentbfb1b832013-01-07 09:53:42 -08001851 sampleRate, format, channelMask, mOutput, mFormat);
1852 lStatus = BAD_VALUE;
1853 goto Exit;
1854 }
Glenn Kastenc3df8382014-03-13 15:05:25 -07001855 break;
1856
1857 default:
Glenn Kasten993fa062014-05-02 11:14:34 -07001858 if (!audio_is_linear_pcm(format)) {
Glenn Kastencac3daa2014-02-07 09:47:14 -08001859 ALOGE("createTrack_l() Bad parameter: format %#x \""
1860 "for output %p with format %#x",
Eric Laurentbfb1b832013-01-07 09:53:42 -08001861 format, mOutput, mFormat);
1862 lStatus = BAD_VALUE;
1863 goto Exit;
1864 }
Andy Hungcd044842014-08-07 11:04:34 -07001865 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Eric Laurent81784c32012-11-19 14:55:58 -08001866 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
1867 lStatus = BAD_VALUE;
1868 goto Exit;
1869 }
Glenn Kastenc3df8382014-03-13 15:05:25 -07001870 break;
1871
Eric Laurent81784c32012-11-19 14:55:58 -08001872 }
1873
1874 lStatus = initCheck();
1875 if (lStatus != NO_ERROR) {
Glenn Kasten15e57982013-09-24 11:52:37 -07001876 ALOGE("createTrack_l() audio driver not initialized");
Eric Laurent81784c32012-11-19 14:55:58 -08001877 goto Exit;
1878 }
1879
1880 { // scope for mLock
1881 Mutex::Autolock _l(mLock);
1882
1883 // all tracks in same audio session must share the same routing strategy otherwise
1884 // conflicts will happen when tracks are moved from one output to another by audio policy
1885 // manager
1886 uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
1887 for (size_t i = 0; i < mTracks.size(); ++i) {
1888 sp<Track> t = mTracks[i];
Eric Laurent83b88082014-06-20 18:31:16 -07001889 if (t != 0 && t->isExternalTrack()) {
Eric Laurent81784c32012-11-19 14:55:58 -08001890 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
1891 if (sessionId == t->sessionId() && strategy != actual) {
1892 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
1893 strategy, actual);
1894 lStatus = BAD_VALUE;
1895 goto Exit;
1896 }
1897 }
1898 }
1899
Glenn Kastend79072e2016-01-06 08:41:20 -08001900 track = new Track(this, client, streamType, sampleRate, format,
1901 channelMask, frameCount, NULL, sharedBuffer,
1902 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT);
Glenn Kasten03003332013-08-06 15:40:54 -07001903
Glenn Kasten03003332013-08-06 15:40:54 -07001904 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
1905 if (lStatus != NO_ERROR) {
Glenn Kasten0cde0762014-01-16 15:06:36 -08001906 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
Haynes Mathew George03e9e832013-12-13 15:40:13 -08001907 // track must be cleared from the caller as the caller has the AF lock
Eric Laurent81784c32012-11-19 14:55:58 -08001908 goto Exit;
1909 }
1910 mTracks.add(track);
1911
1912 sp<EffectChain> chain = getEffectChain_l(sessionId);
1913 if (chain != 0) {
1914 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
1915 track->setMainBuffer(chain->inBuffer());
1916 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
1917 chain->incTrackCnt();
1918 }
1919
1920 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
1921 pid_t callingPid = IPCThreadState::self()->getCallingPid();
1922 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
1923 // so ask activity manager to do this on our behalf
1924 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
1925 }
1926 }
1927
1928 lStatus = NO_ERROR;
1929
1930Exit:
Glenn Kasten9156ef32013-08-06 15:39:08 -07001931 *status = lStatus;
Eric Laurent81784c32012-11-19 14:55:58 -08001932 return track;
1933}
1934
1935uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
1936{
1937 return latency;
1938}
1939
1940uint32_t AudioFlinger::PlaybackThread::latency() const
1941{
1942 Mutex::Autolock _l(mLock);
1943 return latency_l();
1944}
1945uint32_t AudioFlinger::PlaybackThread::latency_l() const
1946{
1947 if (initCheck() == NO_ERROR) {
1948 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream));
1949 } else {
1950 return 0;
1951 }
1952}
1953
1954void AudioFlinger::PlaybackThread::setMasterVolume(float value)
1955{
1956 Mutex::Autolock _l(mLock);
1957 // Don't apply master volume in SW if our HAL can do it for us.
1958 if (mOutput && mOutput->audioHwDev &&
1959 mOutput->audioHwDev->canSetMasterVolume()) {
1960 mMasterVolume = 1.0;
1961 } else {
1962 mMasterVolume = value;
1963 }
1964}
1965
1966void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
1967{
1968 Mutex::Autolock _l(mLock);
1969 // Don't apply master mute in SW if our HAL can do it for us.
1970 if (mOutput && mOutput->audioHwDev &&
1971 mOutput->audioHwDev->canSetMasterMute()) {
1972 mMasterMute = false;
1973 } else {
1974 mMasterMute = muted;
1975 }
1976}
1977
1978void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
1979{
1980 Mutex::Autolock _l(mLock);
1981 mStreamTypes[stream].volume = value;
Eric Laurentede6c3b2013-09-19 14:37:46 -07001982 broadcast_l();
Eric Laurent81784c32012-11-19 14:55:58 -08001983}
1984
1985void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
1986{
1987 Mutex::Autolock _l(mLock);
1988 mStreamTypes[stream].mute = muted;
Eric Laurentede6c3b2013-09-19 14:37:46 -07001989 broadcast_l();
Eric Laurent81784c32012-11-19 14:55:58 -08001990}
1991
1992float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
1993{
1994 Mutex::Autolock _l(mLock);
1995 return mStreamTypes[stream].volume;
1996}
1997
1998// addTrack_l() must be called with ThreadBase::mLock held
1999status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
2000{
2001 status_t status = ALREADY_EXISTS;
2002
Eric Laurent81784c32012-11-19 14:55:58 -08002003 if (mActiveTracks.indexOf(track) < 0) {
2004 // the track is newly added, make sure it fills up all its
2005 // buffers before playing. This is to ensure the client will
2006 // effectively get the latency it requested.
Eric Laurent83b88082014-06-20 18:31:16 -07002007 if (track->isExternalTrack()) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08002008 TrackBase::track_state state = track->mState;
2009 mLock.unlock();
Eric Laurente83b55d2014-11-14 10:06:21 -08002010 status = AudioSystem::startOutput(mId, track->streamType(),
Glenn Kastend848eb42016-03-08 13:42:11 -08002011 track->sessionId());
Eric Laurentbfb1b832013-01-07 09:53:42 -08002012 mLock.lock();
2013 // abort track was stopped/paused while we released the lock
2014 if (state != track->mState) {
2015 if (status == NO_ERROR) {
2016 mLock.unlock();
Eric Laurente83b55d2014-11-14 10:06:21 -08002017 AudioSystem::stopOutput(mId, track->streamType(),
Glenn Kastend848eb42016-03-08 13:42:11 -08002018 track->sessionId());
Eric Laurentbfb1b832013-01-07 09:53:42 -08002019 mLock.lock();
2020 }
2021 return INVALID_OPERATION;
2022 }
2023 // abort if start is rejected by audio policy manager
2024 if (status != NO_ERROR) {
2025 return PERMISSION_DENIED;
2026 }
2027#ifdef ADD_BATTERY_DATA
2028 // to track the speaker usage
2029 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
2030#endif
2031 }
2032
Eric Laurent51716182016-02-29 18:00:56 -08002033 // set retry count for buffer fill
2034 if (track->isOffloaded()) {
Eric Laurente93cc032016-05-05 10:15:10 -07002035 if (track->isStopping_1()) {
2036 track->mRetryCount = kMaxTrackStopRetriesOffload;
2037 } else {
2038 track->mRetryCount = kMaxTrackStartupRetriesOffload;
2039 }
2040 track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED;
Eric Laurent51716182016-02-29 18:00:56 -08002041 } else {
2042 track->mRetryCount = kMaxTrackStartupRetries;
Eric Laurente93cc032016-05-05 10:15:10 -07002043 track->mFillingUpStatus =
2044 track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
Eric Laurent51716182016-02-29 18:00:56 -08002045 }
2046
Eric Laurent81784c32012-11-19 14:55:58 -08002047 track->mResetDone = false;
2048 track->mPresentationCompleteFrames = 0;
2049 mActiveTracks.add(track);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08002050 mWakeLockUids.add(track->uid());
2051 mActiveTracksGeneration++;
Eric Laurentfd477972013-10-25 18:10:40 -07002052 mLatestActiveTrack = track;
Eric Laurentd0107bc2013-06-11 14:38:48 -07002053 sp<EffectChain> chain = getEffectChain_l(track->sessionId());
2054 if (chain != 0) {
2055 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
2056 track->sessionId());
2057 chain->incActiveTrackCnt();
Eric Laurent81784c32012-11-19 14:55:58 -08002058 }
2059
2060 status = NO_ERROR;
2061 }
2062
Haynes Mathew George4c6a4332014-01-15 12:31:39 -08002063 onAddNewTrack_l();
Eric Laurent81784c32012-11-19 14:55:58 -08002064 return status;
2065}
2066
Eric Laurentbfb1b832013-01-07 09:53:42 -08002067bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
Eric Laurent81784c32012-11-19 14:55:58 -08002068{
Eric Laurentbfb1b832013-01-07 09:53:42 -08002069 track->terminate();
Eric Laurent81784c32012-11-19 14:55:58 -08002070 // active tracks are removed by threadLoop()
Eric Laurentbfb1b832013-01-07 09:53:42 -08002071 bool trackActive = (mActiveTracks.indexOf(track) >= 0);
2072 track->mState = TrackBase::STOPPED;
2073 if (!trackActive) {
Eric Laurent81784c32012-11-19 14:55:58 -08002074 removeTrack_l(track);
Eric Laurentab5cdba2014-06-09 17:22:27 -07002075 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08002076 track->mState = TrackBase::STOPPING_1;
Eric Laurent81784c32012-11-19 14:55:58 -08002077 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08002078
2079 return trackActive;
Eric Laurent81784c32012-11-19 14:55:58 -08002080}
2081
2082void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
2083{
2084 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
2085 mTracks.remove(track);
2086 deleteTrackName_l(track->name());
2087 // redundant as track is about to be destroyed, for dumpsys only
2088 track->mName = -1;
2089 if (track->isFastTrack()) {
2090 int index = track->mFastIndex;
Glenn Kastendc2c50b2016-04-21 08:13:14 -07002091 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks);
Eric Laurent81784c32012-11-19 14:55:58 -08002092 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
2093 mFastTrackAvailMask |= 1 << index;
2094 // redundant as track is about to be destroyed, for dumpsys only
2095 track->mFastIndex = -1;
2096 }
2097 sp<EffectChain> chain = getEffectChain_l(track->sessionId());
2098 if (chain != 0) {
2099 chain->decTrackCnt();
2100 }
2101}
2102
Eric Laurentede6c3b2013-09-19 14:37:46 -07002103void AudioFlinger::PlaybackThread::broadcast_l()
Eric Laurentbfb1b832013-01-07 09:53:42 -08002104{
2105 // Thread could be blocked waiting for async
2106 // so signal it to handle state changes immediately
2107 // If threadLoop is currently unlocked a signal of mWaitWorkCV will
2108 // be lost so we also flag to prevent it blocking on mWaitWorkCV
2109 mSignalPending = true;
Eric Laurentede6c3b2013-09-19 14:37:46 -07002110 mWaitWorkCV.broadcast();
Eric Laurentbfb1b832013-01-07 09:53:42 -08002111}
2112
Eric Laurent81784c32012-11-19 14:55:58 -08002113String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
2114{
Eric Laurent81784c32012-11-19 14:55:58 -08002115 Mutex::Autolock _l(mLock);
2116 if (initCheck() != NO_ERROR) {
Glenn Kastend8ea6992013-07-16 14:17:15 -07002117 return String8();
Eric Laurent81784c32012-11-19 14:55:58 -08002118 }
2119
Glenn Kastend8ea6992013-07-16 14:17:15 -07002120 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
2121 const String8 out_s8(s);
Eric Laurent81784c32012-11-19 14:55:58 -08002122 free(s);
2123 return out_s8;
2124}
2125
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07002126void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) {
Eric Laurent73e26b62015-04-27 16:55:58 -07002127 sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
2128 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);
Eric Laurent81784c32012-11-19 14:55:58 -08002129
Eric Laurent73e26b62015-04-27 16:55:58 -07002130 desc->mIoHandle = mId;
Eric Laurent81784c32012-11-19 14:55:58 -08002131
2132 switch (event) {
Eric Laurent73e26b62015-04-27 16:55:58 -07002133 case AUDIO_OUTPUT_OPENED:
2134 case AUDIO_OUTPUT_CONFIG_CHANGED:
Eric Laurent296fb132015-05-01 11:38:42 -07002135 desc->mPatch = mPatch;
Eric Laurent73e26b62015-04-27 16:55:58 -07002136 desc->mChannelMask = mChannelMask;
2137 desc->mSamplingRate = mSampleRate;
2138 desc->mFormat = mFormat;
2139 desc->mFrameCount = mNormalFrameCount; // FIXME see
Eric Laurent81784c32012-11-19 14:55:58 -08002140 // AudioFlinger::frameCount(audio_io_handle_t)
Glenn Kasten4a8308b2016-04-18 14:10:01 -07002141 desc->mFrameCountHAL = mFrameCount;
Eric Laurent73e26b62015-04-27 16:55:58 -07002142 desc->mLatency = latency_l();
Eric Laurent81784c32012-11-19 14:55:58 -08002143 break;
2144
Eric Laurent73e26b62015-04-27 16:55:58 -07002145 case AUDIO_OUTPUT_CLOSED:
Eric Laurent81784c32012-11-19 14:55:58 -08002146 default:
2147 break;
2148 }
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07002149 mAudioFlinger->ioConfigChanged(event, desc, pid);
Eric Laurent81784c32012-11-19 14:55:58 -08002150}
2151
Eric Laurentbfb1b832013-01-07 09:53:42 -08002152void AudioFlinger::PlaybackThread::writeCallback()
2153{
2154 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002155 mCallbackThread->resetWriteBlocked();
Eric Laurentbfb1b832013-01-07 09:53:42 -08002156}
2157
2158void AudioFlinger::PlaybackThread::drainCallback()
2159{
2160 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002161 mCallbackThread->resetDraining();
Eric Laurentbfb1b832013-01-07 09:53:42 -08002162}
2163
Eric Laurent3b4529e2013-09-05 18:09:19 -07002164void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08002165{
2166 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002167 // reject out of sequence requests
2168 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) {
2169 mWriteAckSequence &= ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002170 mWaitWorkCV.signal();
2171 }
2172}
2173
Eric Laurent3b4529e2013-09-05 18:09:19 -07002174void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08002175{
2176 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002177 // reject out of sequence requests
2178 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) {
2179 mDrainSequence &= ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002180 mWaitWorkCV.signal();
2181 }
2182}
2183
2184// static
2185int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
Glenn Kasten0f11b512014-01-31 16:18:54 -08002186 void *param __unused,
Eric Laurentbfb1b832013-01-07 09:53:42 -08002187 void *cookie)
2188{
2189 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie;
2190 ALOGV("asyncCallback() event %d", event);
2191 switch (event) {
2192 case STREAM_CBK_EVENT_WRITE_READY:
2193 me->writeCallback();
2194 break;
2195 case STREAM_CBK_EVENT_DRAIN_READY:
2196 me->drainCallback();
2197 break;
2198 default:
2199 ALOGW("asyncCallback() unknown event %d", event);
2200 break;
2201 }
2202 return 0;
2203}
2204
Glenn Kastendeca2ae2014-02-07 10:25:56 -08002205void AudioFlinger::PlaybackThread::readOutputParameters_l()
Eric Laurent81784c32012-11-19 14:55:58 -08002206{
Glenn Kastenadad3d72014-02-21 14:51:43 -08002207 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
Phil Burkca5e6142015-07-14 09:42:29 -07002208 mSampleRate = mOutput->getSampleRate();
2209 mChannelMask = mOutput->getChannelMask();
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002210 if (!audio_is_output_channel(mChannelMask)) {
Glenn Kastenadad3d72014-02-21 14:51:43 -08002211 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002212 }
Andy Hung9a592762014-07-21 21:56:01 -07002213 if ((mType == MIXER || mType == DUPLICATING)
2214 && !isValidPcmSinkChannelMask(mChannelMask)) {
2215 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
2216 mChannelMask);
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002217 }
Andy Hunge5412692014-05-16 11:25:07 -07002218 mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
Phil Burkca5e6142015-07-14 09:42:29 -07002219
2220 // Get actual HAL format.
Andy Hung463be252014-07-10 16:56:07 -07002221 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
Phil Burkca5e6142015-07-14 09:42:29 -07002222 // Get format from the shim, which will be different than the HAL format
2223 // if playing compressed audio over HDMI passthrough.
2224 mFormat = mOutput->getFormat();
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002225 if (!audio_is_valid_format(mFormat)) {
Glenn Kastenadad3d72014-02-21 14:51:43 -08002226 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002227 }
Andy Hung6146c082014-03-18 11:56:15 -07002228 if ((mType == MIXER || mType == DUPLICATING)
2229 && !isValidPcmSinkFormat(mFormat)) {
2230 LOG_FATAL("HAL format %#x not supported for mixed output",
2231 mFormat);
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002232 }
Phil Burk062e67a2015-02-11 13:40:50 -08002233 mFrameSize = mOutput->getFrameSize();
Glenn Kasten70949c42013-08-06 07:40:12 -07002234 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
2235 mFrameCount = mBufferSize / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08002236 if (mFrameCount & 15) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07002237 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
Eric Laurent81784c32012-11-19 14:55:58 -08002238 mFrameCount);
2239 }
2240
Eric Laurentbfb1b832013-01-07 09:53:42 -08002241 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) &&
2242 (mOutput->stream->set_callback != NULL)) {
2243 if (mOutput->stream->set_callback(mOutput->stream,
2244 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) {
2245 mUseAsyncWrite = true;
Eric Laurent4de95592013-09-26 15:28:21 -07002246 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002247 }
2248 }
2249
Eric Laurentd1f69b02014-12-15 14:33:13 -08002250 mHwSupportsPause = false;
2251 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) {
2252 if (mOutput->stream->pause != NULL) {
2253 if (mOutput->stream->resume != NULL) {
2254 mHwSupportsPause = true;
2255 } else {
2256 ALOGW("direct output implements pause but not resume");
2257 }
2258 } else if (mOutput->stream->resume != NULL) {
2259 ALOGW("direct output implements resume but not pause");
2260 }
2261 }
Phil Burk6fc2a7c2015-04-30 16:08:10 -07002262 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) {
2263 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume");
2264 }
Eric Laurentd1f69b02014-12-15 14:33:13 -08002265
Andy Hungfbfc3952015-01-15 13:33:51 -08002266 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) {
2267 // For best precision, we use float instead of the associated output
2268 // device format (typically PCM 16 bit).
2269
2270 mFormat = AUDIO_FORMAT_PCM_FLOAT;
2271 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
2272 mBufferSize = mFrameSize * mFrameCount;
2273
2274 // TODO: We currently use the associated output device channel mask and sample rate.
2275 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads
2276 // (if a valid mask) to avoid premature downmix.
2277 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads
2278 // instead of the output device sample rate to avoid loss of high frequency information.
2279 // This may need to be updated as MixerThread/OutputTracks are added and not here.
2280 }
2281
Andy Hung09a50072014-02-27 14:30:47 -08002282 // Calculate size of normal sink buffer relative to the HAL output buffer size
Eric Laurent81784c32012-11-19 14:55:58 -08002283 double multiplier = 1.0;
2284 if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
2285 kUseFastMixer == FastMixer_Dynamic)) {
Andy Hung09a50072014-02-27 14:30:47 -08002286 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000;
2287 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000;
Eric Laurent81784c32012-11-19 14:55:58 -08002288 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
2289 minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
2290 maxNormalFrameCount = maxNormalFrameCount & ~15;
2291 if (maxNormalFrameCount < minNormalFrameCount) {
2292 maxNormalFrameCount = minNormalFrameCount;
2293 }
2294 multiplier = (double) minNormalFrameCount / (double) mFrameCount;
2295 if (multiplier <= 1.0) {
2296 multiplier = 1.0;
2297 } else if (multiplier <= 2.0) {
2298 if (2 * mFrameCount <= maxNormalFrameCount) {
2299 multiplier = 2.0;
2300 } else {
2301 multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
2302 }
2303 } else {
2304 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL
Andy Hung09a50072014-02-27 14:30:47 -08002305 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast
Eric Laurent81784c32012-11-19 14:55:58 -08002306 // track, but we sometimes have to do this to satisfy the maximum frame count
2307 // constraint)
2308 // FIXME this rounding up should not be done if no HAL SRC
2309 uint32_t truncMult = (uint32_t) multiplier;
2310 if ((truncMult & 1)) {
2311 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) {
2312 ++truncMult;
2313 }
2314 }
2315 multiplier = (double) truncMult;
2316 }
2317 }
2318 mNormalFrameCount = multiplier * mFrameCount;
2319 // round up to nearest 16 frames to satisfy AudioMixer
Eric Laurentab5cdba2014-06-09 17:22:27 -07002320 if (mType == MIXER || mType == DUPLICATING) {
2321 mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
2322 }
Glenn Kastenc42e9b42016-03-21 11:35:03 -07002323 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
Eric Laurent81784c32012-11-19 14:55:58 -08002324 mNormalFrameCount);
2325
Andy Hung08fb1742015-05-31 23:22:10 -07002326 // Check if we want to throttle the processing to no more than 2x normal rate
2327 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */);
Andy Hung40eb1a12015-06-18 13:42:02 -07002328 mThreadThrottleTimeMs = 0;
2329 mThreadThrottleEndMs = 0;
Andy Hung08fb1742015-05-31 23:22:10 -07002330 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate);
2331
Andy Hung010a1a12014-03-13 13:57:33 -07002332 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames.
2333 // Originally this was int16_t[] array, need to remove legacy implications.
2334 free(mSinkBuffer);
2335 mSinkBuffer = NULL;
Andy Hung5b10a202014-03-13 13:59:29 -07002336 // For sink buffer size, we use the frame size from the downstream sink to avoid problems
2337 // with non PCM formats for compressed music, e.g. AAC, and Offload threads.
2338 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
Andy Hung010a1a12014-03-13 13:57:33 -07002339 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
Eric Laurent81784c32012-11-19 14:55:58 -08002340
Andy Hung69aed5f2014-02-25 17:24:40 -08002341 // We resize the mMixerBuffer according to the requirements of the sink buffer which
2342 // drives the output.
2343 free(mMixerBuffer);
2344 mMixerBuffer = NULL;
2345 if (mMixerBufferEnabled) {
2346 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT.
2347 mMixerBufferSize = mNormalFrameCount * mChannelCount
2348 * audio_bytes_per_sample(mMixerBufferFormat);
2349 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
2350 }
Andy Hung98ef9782014-03-04 14:46:50 -08002351 free(mEffectBuffer);
2352 mEffectBuffer = NULL;
2353 if (mEffectBufferEnabled) {
2354 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only
2355 mEffectBufferSize = mNormalFrameCount * mChannelCount
2356 * audio_bytes_per_sample(mEffectBufferFormat);
2357 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
2358 }
Andy Hung69aed5f2014-02-25 17:24:40 -08002359
Eric Laurent81784c32012-11-19 14:55:58 -08002360 // force reconfiguration of effect chains and engines to take new buffer size and audio
2361 // parameters into account
Glenn Kastendeca2ae2014-02-07 10:25:56 -08002362 // Note that mLock is not held when readOutputParameters_l() is called from the constructor
Eric Laurent81784c32012-11-19 14:55:58 -08002363 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
2364 // matter.
2365 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
2366 Vector< sp<EffectChain> > effectChains = mEffectChains;
2367 for (size_t i = 0; i < effectChains.size(); i ++) {
2368 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
2369 }
2370}
2371
2372
Kévin PETIT377b2ec2014-02-03 12:35:36 +00002373status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
Eric Laurent81784c32012-11-19 14:55:58 -08002374{
2375 if (halFrames == NULL || dspFrames == NULL) {
2376 return BAD_VALUE;
2377 }
2378 Mutex::Autolock _l(mLock);
2379 if (initCheck() != NO_ERROR) {
2380 return INVALID_OPERATION;
2381 }
Andy Hung818e7a32016-02-16 18:08:07 -08002382 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08002383 *halFrames = framesWritten;
2384
2385 if (isSuspended()) {
2386 // return an estimation of rendered frames when the output is suspended
2387 size_t latencyFrames = (latency_l() * mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08002388 *dspFrames = (uint32_t)
2389 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0);
Eric Laurent81784c32012-11-19 14:55:58 -08002390 return NO_ERROR;
2391 } else {
Kévin PETIT377b2ec2014-02-03 12:35:36 +00002392 status_t status;
2393 uint32_t frames;
Phil Burk062e67a2015-02-11 13:40:50 -08002394 status = mOutput->getRenderPosition(&frames);
Kévin PETIT377b2ec2014-02-03 12:35:36 +00002395 *dspFrames = (size_t)frames;
2396 return status;
Eric Laurent81784c32012-11-19 14:55:58 -08002397 }
2398}
2399
Glenn Kastend848eb42016-03-08 13:42:11 -08002400uint32_t AudioFlinger::PlaybackThread::hasAudioSession(audio_session_t sessionId) const
Eric Laurent81784c32012-11-19 14:55:58 -08002401{
2402 Mutex::Autolock _l(mLock);
2403 uint32_t result = 0;
2404 if (getEffectChain_l(sessionId) != 0) {
2405 result = EFFECT_SESSION;
2406 }
2407
2408 for (size_t i = 0; i < mTracks.size(); ++i) {
2409 sp<Track> track = mTracks[i];
Glenn Kasten5736c352012-12-04 12:12:34 -08002410 if (sessionId == track->sessionId() && !track->isInvalid()) {
Eric Laurent81784c32012-11-19 14:55:58 -08002411 result |= TRACK_SESSION;
2412 break;
2413 }
2414 }
2415
2416 return result;
2417}
2418
Glenn Kastend848eb42016-03-08 13:42:11 -08002419uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08002420{
2421 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
2422 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
2423 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2424 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
2425 }
2426 for (size_t i = 0; i < mTracks.size(); i++) {
2427 sp<Track> track = mTracks[i];
Glenn Kasten5736c352012-12-04 12:12:34 -08002428 if (sessionId == track->sessionId() && !track->isInvalid()) {
Eric Laurent81784c32012-11-19 14:55:58 -08002429 return AudioSystem::getStrategyForStream(track->streamType());
2430 }
2431 }
2432 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
2433}
2434
2435
Phil Burk062e67a2015-02-11 13:40:50 -08002436AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
Eric Laurent81784c32012-11-19 14:55:58 -08002437{
2438 Mutex::Autolock _l(mLock);
2439 return mOutput;
2440}
2441
Phil Burk062e67a2015-02-11 13:40:50 -08002442AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
Eric Laurent81784c32012-11-19 14:55:58 -08002443{
2444 Mutex::Autolock _l(mLock);
2445 AudioStreamOut *output = mOutput;
2446 mOutput = NULL;
2447 // FIXME FastMixer might also have a raw ptr to mOutputSink;
2448 // must push a NULL and wait for ack
2449 mOutputSink.clear();
2450 mPipeSink.clear();
2451 mNormalSink.clear();
2452 return output;
2453}
2454
2455// this method must always be called either with ThreadBase mLock held or inside the thread loop
2456audio_stream_t* AudioFlinger::PlaybackThread::stream() const
2457{
2458 if (mOutput == NULL) {
2459 return NULL;
2460 }
2461 return &mOutput->stream->common;
2462}
2463
2464uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
2465{
2466 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
2467}
2468
2469status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
2470{
2471 if (!isValidSyncEvent(event)) {
2472 return BAD_VALUE;
2473 }
2474
2475 Mutex::Autolock _l(mLock);
2476
2477 for (size_t i = 0; i < mTracks.size(); ++i) {
2478 sp<Track> track = mTracks[i];
2479 if (event->triggerSession() == track->sessionId()) {
2480 (void) track->setSyncEvent(event);
2481 return NO_ERROR;
2482 }
2483 }
2484
2485 return NAME_NOT_FOUND;
2486}
2487
2488bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
2489{
2490 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
2491}
2492
2493void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
2494 const Vector< sp<Track> >& tracksToRemove)
2495{
2496 size_t count = tracksToRemove.size();
Glenn Kasten34fca342013-08-13 09:48:14 -07002497 if (count > 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08002498 for (size_t i = 0 ; i < count ; i++) {
2499 const sp<Track>& track = tracksToRemove.itemAt(i);
Eric Laurent83b88082014-06-20 18:31:16 -07002500 if (track->isExternalTrack()) {
Eric Laurente83b55d2014-11-14 10:06:21 -08002501 AudioSystem::stopOutput(mId, track->streamType(),
Glenn Kastend848eb42016-03-08 13:42:11 -08002502 track->sessionId());
Eric Laurentbfb1b832013-01-07 09:53:42 -08002503#ifdef ADD_BATTERY_DATA
2504 // to track the speaker usage
2505 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
2506#endif
2507 if (track->isTerminated()) {
Eric Laurente83b55d2014-11-14 10:06:21 -08002508 AudioSystem::releaseOutput(mId, track->streamType(),
Glenn Kastend848eb42016-03-08 13:42:11 -08002509 track->sessionId());
Eric Laurentbfb1b832013-01-07 09:53:42 -08002510 }
Eric Laurent81784c32012-11-19 14:55:58 -08002511 }
2512 }
2513 }
Eric Laurent81784c32012-11-19 14:55:58 -08002514}
2515
2516void AudioFlinger::PlaybackThread::checkSilentMode_l()
2517{
2518 if (!mMasterMute) {
2519 char value[PROPERTY_VALUE_MAX];
Jean-Michel Trivi32f37c22016-03-31 16:00:32 -07002520 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
2521 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX");
2522 return;
2523 }
Eric Laurent81784c32012-11-19 14:55:58 -08002524 if (property_get("ro.audio.silent", value, "0") > 0) {
2525 char *endptr;
2526 unsigned long ul = strtoul(value, &endptr, 0);
2527 if (*endptr == '\0' && ul != 0) {
2528 ALOGD("Silence is golden");
2529 // The setprop command will not allow a property to be changed after
2530 // the first time it is set, so we don't have to worry about un-muting.
2531 setMasterMute_l(true);
2532 }
2533 }
2534 }
2535}
2536
2537// shared by MIXER and DIRECT, overridden by DUPLICATING
Eric Laurentbfb1b832013-01-07 09:53:42 -08002538ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
Eric Laurent81784c32012-11-19 14:55:58 -08002539{
2540 // FIXME rewrite to reduce number of system calls
2541 mLastWriteTime = systemTime();
2542 mInWrite = true;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002543 ssize_t bytesWritten;
Andy Hung010a1a12014-03-13 13:57:33 -07002544 const size_t offset = mCurrentWriteLength - mBytesRemaining;
Eric Laurent81784c32012-11-19 14:55:58 -08002545
2546 // If an NBAIO sink is present, use it to write the normal mixer's submix
2547 if (mNormalSink != 0) {
Glenn Kasten4c053ea2014-09-28 14:41:07 -07002548
Andy Hung010a1a12014-03-13 13:57:33 -07002549 const size_t count = mBytesRemaining / mFrameSize;
2550
Simon Wilson2d590962012-11-29 15:18:50 -08002551 ATRACE_BEGIN("write");
Eric Laurent81784c32012-11-19 14:55:58 -08002552 // update the setpoint when AudioFlinger::mScreenState changes
2553 uint32_t screenState = AudioFlinger::mScreenState;
2554 if (screenState != mScreenState) {
2555 mScreenState = screenState;
2556 MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
2557 if (pipe != NULL) {
2558 pipe->setAvgFrames((mScreenState & 1) ?
2559 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
2560 }
2561 }
Andy Hung010a1a12014-03-13 13:57:33 -07002562 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
Simon Wilson2d590962012-11-29 15:18:50 -08002563 ATRACE_END();
Eric Laurent81784c32012-11-19 14:55:58 -08002564 if (framesWritten > 0) {
Andy Hung010a1a12014-03-13 13:57:33 -07002565 bytesWritten = framesWritten * mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08002566 } else {
2567 bytesWritten = framesWritten;
2568 }
2569 // otherwise use the HAL / AudioStreamOut directly
2570 } else {
Eric Laurentbfb1b832013-01-07 09:53:42 -08002571 // Direct output and offload threads
Andy Hung010a1a12014-03-13 13:57:33 -07002572
Eric Laurentbfb1b832013-01-07 09:53:42 -08002573 if (mUseAsyncWrite) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07002574 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
2575 mWriteAckSequence += 2;
2576 mWriteAckSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002577 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002578 mCallbackThread->setWriteBlocked(mWriteAckSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002579 }
Glenn Kasten767094d2013-08-23 13:51:43 -07002580 // FIXME We should have an implementation of timestamps for direct output threads.
2581 // They are used e.g for multichannel PCM playback over HDMI.
Phil Burk062e67a2015-02-11 13:40:50 -08002582 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
Eric Laurent51716182016-02-29 18:00:56 -08002583
Eric Laurentbfb1b832013-01-07 09:53:42 -08002584 if (mUseAsyncWrite &&
2585 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
2586 // do not wait for async callback in case of error of full write
Eric Laurent3b4529e2013-09-05 18:09:19 -07002587 mWriteAckSequence &= ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002588 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002589 mCallbackThread->setWriteBlocked(mWriteAckSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002590 }
Eric Laurent81784c32012-11-19 14:55:58 -08002591 }
2592
Eric Laurent81784c32012-11-19 14:55:58 -08002593 mNumWrites++;
2594 mInWrite = false;
Eric Laurentfd477972013-10-25 18:10:40 -07002595 mStandby = false;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002596 return bytesWritten;
2597}
2598
2599void AudioFlinger::PlaybackThread::threadLoop_drain()
2600{
2601 if (mOutput->stream->drain) {
2602 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full");
2603 if (mUseAsyncWrite) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07002604 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request");
2605 mDrainSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002606 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002607 mCallbackThread->setDraining(mDrainSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002608 }
2609 mOutput->stream->drain(mOutput->stream,
2610 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY
2611 : AUDIO_DRAIN_ALL);
2612 }
2613}
2614
2615void AudioFlinger::PlaybackThread::threadLoop_exit()
2616{
Eric Laurent275e8e92014-11-30 15:14:47 -08002617 {
2618 Mutex::Autolock _l(mLock);
2619 for (size_t i = 0; i < mTracks.size(); i++) {
2620 sp<Track> track = mTracks[i];
2621 track->invalidate();
2622 }
2623 }
Eric Laurent81784c32012-11-19 14:55:58 -08002624}
2625
2626/*
2627The derived values that are cached:
Andy Hung25c2dac2014-02-27 14:56:00 -08002628 - mSinkBufferSize from frame count * frame size
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002629 - mActiveSleepTimeUs from activeSleepTimeUs()
2630 - mIdleSleepTimeUs from idleSleepTimeUs()
Eric Laurent42537be2016-01-08 17:16:42 -08002631 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least
2632 kDefaultStandbyTimeInNsecs when connected to an A2DP device.
Eric Laurent81784c32012-11-19 14:55:58 -08002633 - maxPeriod from frame count and sample rate (MIXER only)
2634
2635The parameters that affect these derived values are:
2636 - frame count
2637 - frame size
2638 - sample rate
2639 - device type: A2DP or not
2640 - device latency
2641 - format: PCM or not
2642 - active sleep time
2643 - idle sleep time
2644*/
2645
2646void AudioFlinger::PlaybackThread::cacheParameters_l()
2647{
Andy Hung25c2dac2014-02-27 14:56:00 -08002648 mSinkBufferSize = mNormalFrameCount * mFrameSize;
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002649 mActiveSleepTimeUs = activeSleepTimeUs();
2650 mIdleSleepTimeUs = idleSleepTimeUs();
Eric Laurent42537be2016-01-08 17:16:42 -08002651
2652 // make sure standby delay is not too short when connected to an A2DP sink to avoid
2653 // truncating audio when going to standby.
2654 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs;
2655 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) {
2656 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) {
2657 mStandbyDelayNs = kDefaultStandbyTimeInNsecs;
2658 }
2659 }
Eric Laurent81784c32012-11-19 14:55:58 -08002660}
2661
Haynes Mathew George05317d22016-05-03 16:34:26 -07002662void AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
Eric Laurent81784c32012-11-19 14:55:58 -08002663{
Glenn Kastenc42e9b42016-03-21 11:35:03 -07002664 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu",
Eric Laurent81784c32012-11-19 14:55:58 -08002665 this, streamType, mTracks.size());
Eric Laurent81784c32012-11-19 14:55:58 -08002666
2667 size_t size = mTracks.size();
2668 for (size_t i = 0; i < size; i++) {
2669 sp<Track> t = mTracks[i];
Eric Laurentd60560a2015-04-10 11:31:20 -07002670 if (t->streamType() == streamType && t->isExternalTrack()) {
Glenn Kasten5736c352012-12-04 12:12:34 -08002671 t->invalidate();
Eric Laurent81784c32012-11-19 14:55:58 -08002672 }
2673 }
2674}
2675
Haynes Mathew George05317d22016-05-03 16:34:26 -07002676void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
2677{
2678 Mutex::Autolock _l(mLock);
2679 invalidateTracks_l(streamType);
2680}
2681
Eric Laurent81784c32012-11-19 14:55:58 -08002682status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
2683{
Glenn Kastend848eb42016-03-08 13:42:11 -08002684 audio_session_t session = chain->sessionId();
Andy Hung010a1a12014-03-13 13:57:33 -07002685 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled
2686 ? mEffectBuffer : mSinkBuffer);
Eric Laurent81784c32012-11-19 14:55:58 -08002687 bool ownsBuffer = false;
2688
2689 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
Glenn Kastend848eb42016-03-08 13:42:11 -08002690 if (session > AUDIO_SESSION_OUTPUT_MIX) {
Eric Laurent81784c32012-11-19 14:55:58 -08002691 // Only one effect chain can be present in direct output thread and it uses
Andy Hung2098f272014-02-27 14:00:06 -08002692 // the sink buffer as input
Eric Laurent81784c32012-11-19 14:55:58 -08002693 if (mType != DIRECT) {
2694 size_t numSamples = mNormalFrameCount * mChannelCount;
2695 buffer = new int16_t[numSamples];
2696 memset(buffer, 0, numSamples * sizeof(int16_t));
2697 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
2698 ownsBuffer = true;
2699 }
2700
2701 // Attach all tracks with same session ID to this chain.
2702 for (size_t i = 0; i < mTracks.size(); ++i) {
2703 sp<Track> track = mTracks[i];
2704 if (session == track->sessionId()) {
2705 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(),
2706 buffer);
2707 track->setMainBuffer(buffer);
2708 chain->incTrackCnt();
2709 }
2710 }
2711
2712 // indicate all active tracks in the chain
2713 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
2714 sp<Track> track = mActiveTracks[i].promote();
2715 if (track == 0) {
2716 continue;
2717 }
2718 if (session == track->sessionId()) {
2719 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
2720 chain->incActiveTrackCnt();
2721 }
2722 }
2723 }
Eric Laurentaaa44472014-09-12 17:41:50 -07002724 chain->setThread(this);
Eric Laurent81784c32012-11-19 14:55:58 -08002725 chain->setInBuffer(buffer, ownsBuffer);
Andy Hung010a1a12014-03-13 13:57:33 -07002726 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled
2727 ? mEffectBuffer : mSinkBuffer));
Eric Laurent81784c32012-11-19 14:55:58 -08002728 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
Glenn Kastend848eb42016-03-08 13:42:11 -08002729 // chains list in order to be processed last as it contains output stage effects.
Eric Laurent81784c32012-11-19 14:55:58 -08002730 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
2731 // session AUDIO_SESSION_OUTPUT_STAGE to be processed
Glenn Kastend848eb42016-03-08 13:42:11 -08002732 // after track specific effects and before output stage.
Eric Laurent81784c32012-11-19 14:55:58 -08002733 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
Glenn Kastend848eb42016-03-08 13:42:11 -08002734 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX.
Eric Laurent81784c32012-11-19 14:55:58 -08002735 // Effect chain for other sessions are inserted at beginning of effect
2736 // chains list to be processed before output mix effects. Relative order between other
Glenn Kastend848eb42016-03-08 13:42:11 -08002737 // sessions is not important.
2738 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 &&
2739 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX,
2740 "audio_session_t constants misdefined");
Eric Laurent81784c32012-11-19 14:55:58 -08002741 size_t size = mEffectChains.size();
2742 size_t i = 0;
2743 for (i = 0; i < size; i++) {
2744 if (mEffectChains[i]->sessionId() < session) {
2745 break;
2746 }
2747 }
2748 mEffectChains.insertAt(chain, i);
2749 checkSuspendOnAddEffectChain_l(chain);
2750
2751 return NO_ERROR;
2752}
2753
2754size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
2755{
Glenn Kastend848eb42016-03-08 13:42:11 -08002756 audio_session_t session = chain->sessionId();
Eric Laurent81784c32012-11-19 14:55:58 -08002757
2758 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
2759
2760 for (size_t i = 0; i < mEffectChains.size(); i++) {
2761 if (chain == mEffectChains[i]) {
2762 mEffectChains.removeAt(i);
2763 // detach all active tracks from the chain
2764 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
2765 sp<Track> track = mActiveTracks[i].promote();
2766 if (track == 0) {
2767 continue;
2768 }
2769 if (session == track->sessionId()) {
2770 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
2771 chain.get(), session);
2772 chain->decActiveTrackCnt();
2773 }
2774 }
2775
2776 // detach all tracks with same session ID from this chain
2777 for (size_t i = 0; i < mTracks.size(); ++i) {
2778 sp<Track> track = mTracks[i];
2779 if (session == track->sessionId()) {
Andy Hung010a1a12014-03-13 13:57:33 -07002780 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer));
Eric Laurent81784c32012-11-19 14:55:58 -08002781 chain->decTrackCnt();
2782 }
2783 }
2784 break;
2785 }
2786 }
2787 return mEffectChains.size();
2788}
2789
2790status_t AudioFlinger::PlaybackThread::attachAuxEffect(
2791 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
2792{
2793 Mutex::Autolock _l(mLock);
2794 return attachAuxEffect_l(track, EffectId);
2795}
2796
2797status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
2798 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
2799{
2800 status_t status = NO_ERROR;
2801
2802 if (EffectId == 0) {
2803 track->setAuxBuffer(0, NULL);
2804 } else {
2805 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
2806 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
2807 if (effect != 0) {
2808 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2809 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
2810 } else {
2811 status = INVALID_OPERATION;
2812 }
2813 } else {
2814 status = BAD_VALUE;
2815 }
2816 }
2817 return status;
2818}
2819
2820void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
2821{
2822 for (size_t i = 0; i < mTracks.size(); ++i) {
2823 sp<Track> track = mTracks[i];
2824 if (track->auxEffectId() == effectId) {
2825 attachAuxEffect_l(track, 0);
2826 }
2827 }
2828}
2829
2830bool AudioFlinger::PlaybackThread::threadLoop()
2831{
2832 Vector< sp<Track> > tracksToRemove;
2833
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002834 mStandbyTimeNs = systemTime();
Eric Laurent81784c32012-11-19 14:55:58 -08002835
2836 // MIXER
2837 nsecs_t lastWarning = 0;
2838
2839 // DUPLICATING
2840 // FIXME could this be made local to while loop?
2841 writeFrames = 0;
2842
Marco Nelissen462fd2f2013-01-14 14:12:05 -08002843 int lastGeneration = 0;
2844
Eric Laurent81784c32012-11-19 14:55:58 -08002845 cacheParameters_l();
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002846 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08002847
2848 if (mType == MIXER) {
2849 sleepTimeShift = 0;
2850 }
2851
2852 CpuStats cpuStats;
2853 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
2854
2855 acquireWakeLock();
2856
Glenn Kasten9e58b552013-01-18 15:09:48 -08002857 // mNBLogWriter->log can only be called while thread mutex mLock is held.
2858 // So if you need to log when mutex is unlocked, set logString to a non-NULL string,
2859 // and then that string will be logged at the next convenient opportunity.
2860 const char *logString = NULL;
2861
Eric Laurent664539d2013-09-23 18:24:31 -07002862 checkSilentMode_l();
2863
Eric Laurent81784c32012-11-19 14:55:58 -08002864 while (!exitPending())
2865 {
2866 cpuStats.sample(myName);
2867
2868 Vector< sp<EffectChain> > effectChains;
2869
Eric Laurent81784c32012-11-19 14:55:58 -08002870 { // scope for mLock
2871
2872 Mutex::Autolock _l(mLock);
2873
Eric Laurent021cf962014-05-13 10:18:14 -07002874 processConfigEvents_l();
Eric Laurent10351942014-05-08 18:49:52 -07002875
Glenn Kasten9e58b552013-01-18 15:09:48 -08002876 if (logString != NULL) {
2877 mNBLogWriter->logTimestamp();
2878 mNBLogWriter->log(logString);
2879 logString = NULL;
2880 }
2881
Glenn Kasten4c053ea2014-09-28 14:41:07 -07002882 // Gather the framesReleased counters for all active tracks,
Andy Hunge10393e2015-06-12 13:59:33 -07002883 // and associate with the sink frames written out. We need
2884 // this to convert the sink timestamp to the track timestamp.
2885 if (mNormalSink != 0) {
Andy Hungc54b1ff2016-02-23 14:07:07 -08002886 // Note: The DuplicatingThread may not have a mNormalSink.
Andy Hung818e7a32016-02-16 18:08:07 -08002887 // We always fetch the timestamp here because often the downstream
2888 // sink will block whie writing.
2889 ExtendedTimestamp timestamp; // use private copy to fetch
2890 (void) mNormalSink->getTimestamp(timestamp);
Andy Hung6d7b1192016-05-07 22:59:48 -07002891
2892 // We keep track of the last valid kernel position in case we are in underrun
2893 // and the normal mixer period is the same as the fast mixer period, or there
2894 // is some error from the HAL.
2895 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
2896 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
2897 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
2898 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
2899 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
2900
2901 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
2902 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER];
2903 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
2904 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER];
2905 } else {
2906 ALOGV("getTimestamp error - no valid kernel position");
2907 }
2908
Andy Hung818e7a32016-02-16 18:08:07 -08002909 // copy over kernel info
2910 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
2911 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
2912 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
2913 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
Andy Hungc54b1ff2016-02-23 14:07:07 -08002914 }
2915 // mFramesWritten for non-offloaded tracks are contiguous
2916 // even after standby() is called. This is useful for the track frame
2917 // to sink frame mapping.
2918 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten;
2919 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime();
2920 const size_t size = mActiveTracks.size();
2921 for (size_t i = 0; i < size; ++i) {
2922 sp<Track> t = mActiveTracks[i].promote();
2923 if (t != 0 && !t->isFastTrack()) {
2924 t->updateTrackFrameInfo(
2925 t->mAudioTrackServerProxy->framesReleased(),
2926 mFramesWritten,
2927 mTimestamp);
Andy Hunge10393e2015-06-12 13:59:33 -07002928 }
Glenn Kastenbd096fd2013-08-23 13:53:56 -07002929 }
2930
Eric Laurent81784c32012-11-19 14:55:58 -08002931 saveOutputTracks();
Eric Laurentbfb1b832013-01-07 09:53:42 -08002932 if (mSignalPending) {
2933 // A signal was raised while we were unlocked
2934 mSignalPending = false;
2935 } else if (waitingAsyncCallback_l()) {
2936 if (exitPending()) {
2937 break;
2938 }
Marco Nelissen078538c2015-05-12 09:17:57 -07002939 bool released = false;
Eric Laurent64667972016-03-30 18:19:46 -07002940 if (!keepWakeLock()) {
Marco Nelissen078538c2015-05-12 09:17:57 -07002941 releaseWakeLock_l();
2942 released = true;
2943 }
Marco Nelissen462fd2f2013-01-14 14:12:05 -08002944 mWakeLockUids.clear();
2945 mActiveTracksGeneration++;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002946 ALOGV("wait async completion");
2947 mWaitWorkCV.wait(mLock);
2948 ALOGV("async completion/wake");
Marco Nelissen078538c2015-05-12 09:17:57 -07002949 if (released) {
2950 acquireWakeLock_l();
2951 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002952 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
2953 mSleepTimeUs = 0;
Eric Laurentede6c3b2013-09-19 14:37:46 -07002954
2955 continue;
2956 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002957 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) ||
Eric Laurentbfb1b832013-01-07 09:53:42 -08002958 isSuspended()) {
2959 // put audio hardware into standby after short delay
2960 if (shouldStandby_l()) {
Eric Laurent81784c32012-11-19 14:55:58 -08002961
2962 threadLoop_standby();
2963
2964 mStandby = true;
2965 }
2966
2967 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2968 // we're about to wait, flush the binder command buffer
2969 IPCThreadState::self()->flushCommands();
2970
2971 clearOutputTracks();
2972
2973 if (exitPending()) {
2974 break;
2975 }
2976
2977 releaseWakeLock_l();
Marco Nelissen462fd2f2013-01-14 14:12:05 -08002978 mWakeLockUids.clear();
2979 mActiveTracksGeneration++;
Eric Laurent81784c32012-11-19 14:55:58 -08002980 // wait until we have something to do...
2981 ALOGV("%s going to sleep", myName.string());
2982 mWaitWorkCV.wait(mLock);
2983 ALOGV("%s waking up", myName.string());
2984 acquireWakeLock_l();
2985
2986 mMixerStatus = MIXER_IDLE;
2987 mMixerStatusIgnoringFastTracks = MIXER_IDLE;
2988 mBytesWritten = 0;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002989 mBytesRemaining = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08002990 checkSilentMode_l();
2991
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002992 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
2993 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08002994 if (mType == MIXER) {
2995 sleepTimeShift = 0;
2996 }
2997
2998 continue;
2999 }
3000 }
Eric Laurent81784c32012-11-19 14:55:58 -08003001 // mMixerStatusIgnoringFastTracks is also updated internally
3002 mMixerStatus = prepareTracks_l(&tracksToRemove);
3003
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003004 // compare with previously applied list
3005 if (lastGeneration != mActiveTracksGeneration) {
3006 // update wakelock
3007 updateWakeLockUids_l(mWakeLockUids);
3008 lastGeneration = mActiveTracksGeneration;
3009 }
3010
Eric Laurent81784c32012-11-19 14:55:58 -08003011 // prevent any changes in effect chain list and in each effect chain
3012 // during mixing and effect process as the audio buffers could be deleted
3013 // or modified if an effect is created or deleted
3014 lockEffectChains_l(effectChains);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003015 } // mLock scope ends
Eric Laurent81784c32012-11-19 14:55:58 -08003016
Eric Laurentbfb1b832013-01-07 09:53:42 -08003017 if (mBytesRemaining == 0) {
3018 mCurrentWriteLength = 0;
3019 if (mMixerStatus == MIXER_TRACKS_READY) {
3020 // threadLoop_mix() sets mCurrentWriteLength
3021 threadLoop_mix();
3022 } else if ((mMixerStatus != MIXER_DRAIN_TRACK)
3023 && (mMixerStatus != MIXER_DRAIN_ALL)) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003024 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data
Eric Laurentbfb1b832013-01-07 09:53:42 -08003025 // must be written to HAL
3026 threadLoop_sleepTime();
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003027 if (mSleepTimeUs == 0) {
Andy Hung25c2dac2014-02-27 14:56:00 -08003028 mCurrentWriteLength = mSinkBufferSize;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003029 }
3030 }
Andy Hung98ef9782014-03-04 14:46:50 -08003031 // Either threadLoop_mix() or threadLoop_sleepTime() should have set
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003032 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0.
Andy Hung98ef9782014-03-04 14:46:50 -08003033 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid)
3034 // or mSinkBuffer (if there are no effects).
3035 //
3036 // This is done pre-effects computation; if effects change to
3037 // support higher precision, this needs to move.
3038 //
3039 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003040 // TODO use mSleepTimeUs == 0 as an additional condition.
Andy Hung98ef9782014-03-04 14:46:50 -08003041 if (mMixerBufferValid) {
3042 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
3043 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
3044
Andy Hung2ddee192015-12-18 17:34:44 -08003045 // mono blend occurs for mixer threads only (not direct or offloaded)
3046 // and is handled here if we're going directly to the sink.
3047 if (requireMonoBlend() && !mEffectBufferValid) {
Glenn Kasten03c48d52016-01-27 17:25:17 -08003048 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount,
3049 true /*limit*/);
Andy Hung2ddee192015-12-18 17:34:44 -08003050 }
3051
Andy Hung98ef9782014-03-04 14:46:50 -08003052 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
3053 mNormalFrameCount * mChannelCount);
3054 }
3055
Eric Laurentbfb1b832013-01-07 09:53:42 -08003056 mBytesRemaining = mCurrentWriteLength;
3057 if (isSuspended()) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003058 mSleepTimeUs = suspendSleepTimeUs();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003059 // simulate write to HAL when suspended
Andy Hung25c2dac2014-02-27 14:56:00 -08003060 mBytesWritten += mSinkBufferSize;
Andy Hungc54b1ff2016-02-23 14:07:07 -08003061 mFramesWritten += mSinkBufferSize / mFrameSize;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003062 mBytesRemaining = 0;
3063 }
Eric Laurent81784c32012-11-19 14:55:58 -08003064
Eric Laurentbfb1b832013-01-07 09:53:42 -08003065 // only process effects if we're going to write
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003066 if (mSleepTimeUs == 0 && mType != OFFLOAD) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08003067 for (size_t i = 0; i < effectChains.size(); i ++) {
3068 effectChains[i]->process_l();
3069 }
Eric Laurent81784c32012-11-19 14:55:58 -08003070 }
3071 }
Eric Laurent59fe0102013-09-27 18:48:26 -07003072 // Process effect chains for offloaded thread even if no audio
3073 // was read from audio track: process only updates effect state
3074 // and thus does have to be synchronized with audio writes but may have
3075 // to be called while waiting for async write callback
3076 if (mType == OFFLOAD) {
3077 for (size_t i = 0; i < effectChains.size(); i ++) {
3078 effectChains[i]->process_l();
3079 }
3080 }
Eric Laurent81784c32012-11-19 14:55:58 -08003081
Andy Hung98ef9782014-03-04 14:46:50 -08003082 // Only if the Effects buffer is enabled and there is data in the
3083 // Effects buffer (buffer valid), we need to
3084 // copy into the sink buffer.
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003085 // TODO use mSleepTimeUs == 0 as an additional condition.
Andy Hung98ef9782014-03-04 14:46:50 -08003086 if (mEffectBufferValid) {
3087 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat);
Andy Hung2ddee192015-12-18 17:34:44 -08003088
3089 if (requireMonoBlend()) {
Glenn Kasten03c48d52016-01-27 17:25:17 -08003090 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount,
3091 true /*limit*/);
Andy Hung2ddee192015-12-18 17:34:44 -08003092 }
3093
Andy Hung98ef9782014-03-04 14:46:50 -08003094 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat,
3095 mNormalFrameCount * mChannelCount);
3096 }
3097
Eric Laurent81784c32012-11-19 14:55:58 -08003098 // enable changes in effect chain
3099 unlockEffectChains(effectChains);
3100
Eric Laurentbfb1b832013-01-07 09:53:42 -08003101 if (!waitingAsyncCallback()) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003102 // mSleepTimeUs == 0 means we must write to audio hardware
3103 if (mSleepTimeUs == 0) {
Andy Hung08fb1742015-05-31 23:22:10 -07003104 ssize_t ret = 0;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003105 if (mBytesRemaining) {
Andy Hung08fb1742015-05-31 23:22:10 -07003106 ret = threadLoop_write();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003107 if (ret < 0) {
3108 mBytesRemaining = 0;
3109 } else {
3110 mBytesWritten += ret;
3111 mBytesRemaining -= ret;
Andy Hungc54b1ff2016-02-23 14:07:07 -08003112 mFramesWritten += ret / mFrameSize;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003113 }
3114 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||
3115 (mMixerStatus == MIXER_DRAIN_ALL)) {
3116 threadLoop_drain();
Eric Laurent81784c32012-11-19 14:55:58 -08003117 }
Andy Hung08fb1742015-05-31 23:22:10 -07003118 if (mType == MIXER && !mStandby) {
Glenn Kasten4944acb2013-08-19 08:39:20 -07003119 // write blocked detection
3120 nsecs_t now = systemTime();
3121 nsecs_t delta = now - mLastWriteTime;
Andy Hung08fb1742015-05-31 23:22:10 -07003122 if (delta > maxPeriod) {
Glenn Kasten4944acb2013-08-19 08:39:20 -07003123 mNumDelayedWrites++;
3124 if ((now - lastWarning) > kWarningThrottleNs) {
3125 ATRACE_NAME("underrun");
3126 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
Glenn Kastenc42e9b42016-03-21 11:35:03 -07003127 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this);
Glenn Kasten4944acb2013-08-19 08:39:20 -07003128 lastWarning = now;
3129 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003130 }
Andy Hung08fb1742015-05-31 23:22:10 -07003131
3132 if (mThreadThrottle
3133 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks)
3134 && ret > 0) { // we wrote something
3135 // Limit MixerThread data processing to no more than twice the
3136 // expected processing rate.
3137 //
3138 // This helps prevent underruns with NuPlayer and other applications
3139 // which may set up buffers that are close to the minimum size, or use
3140 // deep buffers, and rely on a double-buffering sleep strategy to fill.
3141 //
3142 // The throttle smooths out sudden large data drains from the device,
3143 // e.g. when it comes out of standby, which often causes problems with
3144 // (1) mixer threads without a fast mixer (which has its own warm-up)
3145 // (2) minimum buffer sized tracks (even if the track is full,
3146 // the app won't fill fast enough to handle the sudden draw).
3147
3148 const int32_t deltaMs = delta / 1000000;
3149 const int32_t throttleMs = mHalfBufferMs - deltaMs;
3150 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) {
3151 usleep(throttleMs * 1000);
Andy Hung40eb1a12015-06-18 13:42:02 -07003152 // notify of throttle start on verbose log
3153 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs,
3154 "mixer(%p) throttle begin:"
3155 " ret(%zd) deltaMs(%d) requires sleep %d ms",
Andy Hung08fb1742015-05-31 23:22:10 -07003156 this, ret, deltaMs, throttleMs);
Andy Hung40eb1a12015-06-18 13:42:02 -07003157 mThreadThrottleTimeMs += throttleMs;
3158 } else {
3159 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs;
3160 if (diff > 0) {
3161 // notify of throttle end on debug log
Andy Hung3ea004d2016-05-05 16:48:37 -07003162 // but prevent spamming for bluetooth
3163 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()),
3164 "mixer(%p) throttle end: throttle time(%u)", this, diff);
Andy Hung40eb1a12015-06-18 13:42:02 -07003165 mThreadThrottleEndMs = mThreadThrottleTimeMs;
3166 }
Andy Hung08fb1742015-05-31 23:22:10 -07003167 }
3168 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003169 }
Eric Laurent81784c32012-11-19 14:55:58 -08003170
Eric Laurentbfb1b832013-01-07 09:53:42 -08003171 } else {
Glenn Kastene7754022014-10-31 12:11:26 -07003172 ATRACE_BEGIN("sleep");
Eric Laurente93cc032016-05-05 10:15:10 -07003173 Mutex::Autolock _l(mLock);
3174 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) {
3175 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs));
Eric Laurent51716182016-02-29 18:00:56 -08003176 }
Glenn Kastene7754022014-10-31 12:11:26 -07003177 ATRACE_END();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003178 }
Eric Laurent81784c32012-11-19 14:55:58 -08003179 }
3180
3181 // Finally let go of removed track(s), without the lock held
3182 // since we can't guarantee the destructors won't acquire that
3183 // same lock. This will also mutate and push a new fast mixer state.
3184 threadLoop_removeTracks(tracksToRemove);
3185 tracksToRemove.clear();
3186
3187 // FIXME I don't understand the need for this here;
3188 // it was in the original code but maybe the
3189 // assignment in saveOutputTracks() makes this unnecessary?
3190 clearOutputTracks();
3191
3192 // Effect chains will be actually deleted here if they were removed from
3193 // mEffectChains list during mixing or effects processing
3194 effectChains.clear();
3195
3196 // FIXME Note that the above .clear() is no longer necessary since effectChains
3197 // is now local to this block, but will keep it for now (at least until merge done).
3198 }
3199
Eric Laurentbfb1b832013-01-07 09:53:42 -08003200 threadLoop_exit();
3201
Eric Laurentcf817a22014-08-04 20:36:31 -07003202 if (!mStandby) {
3203 threadLoop_standby();
3204 mStandby = true;
Eric Laurent81784c32012-11-19 14:55:58 -08003205 }
3206
3207 releaseWakeLock();
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003208 mWakeLockUids.clear();
3209 mActiveTracksGeneration++;
Eric Laurent81784c32012-11-19 14:55:58 -08003210
3211 ALOGV("Thread %p type %d exiting", this, mType);
3212 return false;
3213}
3214
Eric Laurentbfb1b832013-01-07 09:53:42 -08003215// removeTracks_l() must be called with ThreadBase::mLock held
3216void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
3217{
3218 size_t count = tracksToRemove.size();
Glenn Kasten34fca342013-08-13 09:48:14 -07003219 if (count > 0) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08003220 for (size_t i=0 ; i<count ; i++) {
3221 const sp<Track>& track = tracksToRemove.itemAt(i);
3222 mActiveTracks.remove(track);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003223 mWakeLockUids.remove(track->uid());
3224 mActiveTracksGeneration++;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003225 ALOGV("removeTracks_l removing track on session %d", track->sessionId());
3226 sp<EffectChain> chain = getEffectChain_l(track->sessionId());
3227 if (chain != 0) {
3228 ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
3229 track->sessionId());
3230 chain->decActiveTrackCnt();
3231 }
3232 if (track->isTerminated()) {
3233 removeTrack_l(track);
3234 }
3235 }
3236 }
3237
3238}
Eric Laurent81784c32012-11-19 14:55:58 -08003239
Eric Laurentaccc1472013-09-20 09:36:34 -07003240status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
3241{
3242 if (mNormalSink != 0) {
Andy Hung818e7a32016-02-16 18:08:07 -08003243 ExtendedTimestamp ets;
3244 status_t status = mNormalSink->getTimestamp(ets);
3245 if (status == NO_ERROR) {
3246 status = ets.getBestTimestamp(&timestamp);
3247 }
3248 return status;
Eric Laurentaccc1472013-09-20 09:36:34 -07003249 }
Andy Hung9a1c8892014-12-03 11:37:42 -08003250 if ((mType == OFFLOAD || mType == DIRECT)
3251 && mOutput != NULL && mOutput->stream->get_presentation_position) {
Eric Laurentaccc1472013-09-20 09:36:34 -07003252 uint64_t position64;
Phil Burk062e67a2015-02-11 13:40:50 -08003253 int ret = mOutput->getPresentationPosition(&position64, &timestamp.mTime);
Eric Laurentaccc1472013-09-20 09:36:34 -07003254 if (ret == 0) {
3255 timestamp.mPosition = (uint32_t)position64;
3256 return NO_ERROR;
3257 }
3258 }
3259 return INVALID_OPERATION;
3260}
Eric Laurent1c333e22014-05-20 10:48:17 -07003261
Eric Laurent054d9d32015-04-24 08:48:48 -07003262status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch,
3263 audio_patch_handle_t *handle)
3264{
Glenn Kastenc05b8d72016-03-24 09:48:17 -07003265 AutoPark<FastMixer> park(mFastMixer);
Eric Laurent054d9d32015-04-24 08:48:48 -07003266
Glenn Kastenc05b8d72016-03-24 09:48:17 -07003267 status_t status = PlaybackThread::createAudioPatch_l(patch, handle);
Eric Laurent054d9d32015-04-24 08:48:48 -07003268
3269 return status;
3270}
3271
Eric Laurent1c333e22014-05-20 10:48:17 -07003272status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
3273 audio_patch_handle_t *handle)
3274{
3275 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07003276
3277 // store new device and send to effects
3278 audio_devices_t type = AUDIO_DEVICE_NONE;
3279 for (unsigned int i = 0; i < patch->num_sinks; i++) {
3280 type |= patch->sinks[i].ext.device.type;
3281 }
3282
3283#ifdef ADD_BATTERY_DATA
3284 // when changing the audio output device, call addBatteryData to notify
3285 // the change
3286 if (mOutDevice != type) {
3287 uint32_t params = 0;
3288 // check whether speaker is on
3289 if (type & AUDIO_DEVICE_OUT_SPEAKER) {
3290 params |= IMediaPlayerService::kBatteryDataSpeakerOn;
Eric Laurent1c333e22014-05-20 10:48:17 -07003291 }
3292
Eric Laurent054d9d32015-04-24 08:48:48 -07003293 audio_devices_t deviceWithoutSpeaker
3294 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
3295 // check if any other device (except speaker) is on
3296 if (type & deviceWithoutSpeaker) {
3297 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
3298 }
3299
3300 if (params != 0) {
3301 addBatteryData(params);
3302 }
3303 }
3304#endif
3305
3306 for (size_t i = 0; i < mEffectChains.size(); i++) {
3307 mEffectChains[i]->setDevice_l(type);
3308 }
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07003309
3310 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when
3311 // the thread is created so that the first patch creation triggers an ioConfigChanged callback
3312 bool configChanged = mPrevOutDevice != type;
Eric Laurent054d9d32015-04-24 08:48:48 -07003313 mOutDevice = type;
Eric Laurent296fb132015-05-01 11:38:42 -07003314 mPatch = *patch;
Eric Laurent054d9d32015-04-24 08:48:48 -07003315
3316 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
Eric Laurent1c333e22014-05-20 10:48:17 -07003317 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
3318 status = hwDevice->create_audio_patch(hwDevice,
3319 patch->num_sources,
3320 patch->sources,
3321 patch->num_sinks,
3322 patch->sinks,
3323 handle);
3324 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07003325 char *address;
3326 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) {
3327 //FIXME: we only support address on first sink with HAL version < 3.0
3328 address = audio_device_address_to_parameter(
3329 patch->sinks[0].ext.device.type,
3330 patch->sinks[0].ext.device.address);
3331 } else {
3332 address = (char *)calloc(1, 1);
3333 }
3334 AudioParameter param = AudioParameter(String8(address));
3335 free(address);
3336 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type);
3337 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3338 param.toString().string());
3339 *handle = AUDIO_PATCH_HANDLE_NONE;
Eric Laurent1c333e22014-05-20 10:48:17 -07003340 }
Eric Laurente8726fe2015-06-26 09:39:24 -07003341 if (configChanged) {
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07003342 mPrevOutDevice = type;
Eric Laurente8726fe2015-06-26 09:39:24 -07003343 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
3344 }
Eric Laurent1c333e22014-05-20 10:48:17 -07003345 return status;
3346}
3347
Eric Laurent054d9d32015-04-24 08:48:48 -07003348status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
3349{
Glenn Kastenc05b8d72016-03-24 09:48:17 -07003350 AutoPark<FastMixer> park(mFastMixer);
Eric Laurent054d9d32015-04-24 08:48:48 -07003351
3352 status_t status = PlaybackThread::releaseAudioPatch_l(handle);
3353
Eric Laurent054d9d32015-04-24 08:48:48 -07003354 return status;
3355}
3356
Eric Laurent1c333e22014-05-20 10:48:17 -07003357status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
3358{
3359 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07003360
3361 mOutDevice = AUDIO_DEVICE_NONE;
3362
Eric Laurent1c333e22014-05-20 10:48:17 -07003363 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
3364 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
3365 status = hwDevice->release_audio_patch(hwDevice, handle);
3366 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07003367 AudioParameter param;
3368 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0);
3369 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3370 param.toString().string());
Eric Laurent1c333e22014-05-20 10:48:17 -07003371 }
3372 return status;
3373}
3374
Eric Laurent83b88082014-06-20 18:31:16 -07003375void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track)
3376{
3377 Mutex::Autolock _l(mLock);
3378 mTracks.add(track);
3379}
3380
3381void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track)
3382{
3383 Mutex::Autolock _l(mLock);
3384 destroyTrack_l(track);
3385}
3386
3387void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config)
3388{
3389 ThreadBase::getAudioPortConfig(config);
3390 config->role = AUDIO_PORT_ROLE_SOURCE;
3391 config->ext.mix.hw_module = mOutput->audioHwDev->handle();
3392 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
3393}
3394
Eric Laurent81784c32012-11-19 14:55:58 -08003395// ----------------------------------------------------------------------------
3396
3397AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
Eric Laurent72e3f392015-05-20 14:43:50 -07003398 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type)
3399 : PlaybackThread(audioFlinger, output, id, device, type, systemReady),
Eric Laurent81784c32012-11-19 14:55:58 -08003400 // mAudioMixer below
3401 // mFastMixer below
Andy Hung2ddee192015-12-18 17:34:44 -08003402 mFastMixerFutex(0),
3403 mMasterMono(false)
Eric Laurent81784c32012-11-19 14:55:58 -08003404 // mOutputSink below
3405 // mPipeSink below
3406 // mNormalSink below
3407{
3408 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
Glenn Kastenc42e9b42016-03-21 11:35:03 -07003409 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, "
3410 "mFrameCount=%zu, mNormalFrameCount=%zu",
Eric Laurent81784c32012-11-19 14:55:58 -08003411 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
3412 mNormalFrameCount);
3413 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
3414
Andy Hungfbfc3952015-01-15 13:33:51 -08003415 if (type == DUPLICATING) {
3416 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks
3417 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write().
3418 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink.
3419 return;
3420 }
Eric Laurent81784c32012-11-19 14:55:58 -08003421 // create an NBAIO sink for the HAL output stream, and negotiate
3422 mOutputSink = new AudioStreamOutSink(output->stream);
3423 size_t numCounterOffers = 0;
Glenn Kastenf69f9862014-03-07 08:37:57 -08003424 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003425#if !LOG_NDEBUG
3426 ssize_t index =
3427#else
3428 (void)
3429#endif
3430 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
Eric Laurent81784c32012-11-19 14:55:58 -08003431 ALOG_ASSERT(index == 0);
3432
3433 // initialize fast mixer depending on configuration
3434 bool initFastMixer;
3435 switch (kUseFastMixer) {
3436 case FastMixer_Never:
3437 initFastMixer = false;
3438 break;
3439 case FastMixer_Always:
3440 initFastMixer = true;
3441 break;
3442 case FastMixer_Static:
3443 case FastMixer_Dynamic:
3444 initFastMixer = mFrameCount < mNormalFrameCount;
3445 break;
3446 }
3447 if (initFastMixer) {
Andy Hung1258c1a2014-05-23 21:22:17 -07003448 audio_format_t fastMixerFormat;
3449 if (mMixerBufferEnabled && mEffectBufferEnabled) {
3450 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT;
3451 } else {
3452 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
3453 }
3454 if (mFormat != fastMixerFormat) {
3455 // change our Sink format to accept our intermediate precision
3456 mFormat = fastMixerFormat;
3457 free(mSinkBuffer);
3458 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
3459 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
3460 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
3461 }
Eric Laurent81784c32012-11-19 14:55:58 -08003462
3463 // create a MonoPipe to connect our submix to FastMixer
3464 NBAIO_Format format = mOutputSink->format();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003465#ifdef TEE_SINK
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003466 NBAIO_Format origformat = format;
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003467#endif
Andy Hung1258c1a2014-05-23 21:22:17 -07003468 // adjust format to match that of the Fast Mixer
Glenn Kasten97b7b752014-09-28 13:04:24 -07003469 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat);
Andy Hung1258c1a2014-05-23 21:22:17 -07003470 format.mFormat = fastMixerFormat;
3471 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
3472
Eric Laurent81784c32012-11-19 14:55:58 -08003473 // This pipe depth compensates for scheduling latency of the normal mixer thread.
3474 // When it wakes up after a maximum latency, it runs a few cycles quickly before
3475 // finally blocking. Note the pipe implementation rounds up the request to a power of 2.
3476 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
3477 const NBAIO_Format offers[1] = {format};
3478 size_t numCounterOffers = 0;
Glenn Kastenfc302fd2016-04-11 14:11:26 -07003479#if !LOG_NDEBUG || defined(TEE_SINK)
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003480 ssize_t index =
3481#else
3482 (void)
3483#endif
3484 monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
Eric Laurent81784c32012-11-19 14:55:58 -08003485 ALOG_ASSERT(index == 0);
3486 monoPipe->setAvgFrames((mScreenState & 1) ?
3487 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
3488 mPipeSink = monoPipe;
3489
Glenn Kasten46909e72013-02-26 09:20:22 -08003490#ifdef TEE_SINK
Glenn Kastenda6ef132013-01-10 12:31:01 -08003491 if (mTeeSinkOutputEnabled) {
3492 // create a Pipe to archive a copy of FastMixer's output for dumpsys
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003493 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat);
3494 const NBAIO_Format offers2[1] = {origformat};
Glenn Kastenda6ef132013-01-10 12:31:01 -08003495 numCounterOffers = 0;
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003496 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers);
Glenn Kastenda6ef132013-01-10 12:31:01 -08003497 ALOG_ASSERT(index == 0);
3498 mTeeSink = teeSink;
3499 PipeReader *teeSource = new PipeReader(*teeSink);
3500 numCounterOffers = 0;
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003501 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers);
Glenn Kastenda6ef132013-01-10 12:31:01 -08003502 ALOG_ASSERT(index == 0);
3503 mTeeSource = teeSource;
3504 }
Glenn Kasten46909e72013-02-26 09:20:22 -08003505#endif
Eric Laurent81784c32012-11-19 14:55:58 -08003506
3507 // create fast mixer and configure it initially with just one fast track for our submix
3508 mFastMixer = new FastMixer();
3509 FastMixerStateQueue *sq = mFastMixer->sq();
3510#ifdef STATE_QUEUE_DUMP
3511 sq->setObserverDump(&mStateQueueObserverDump);
3512 sq->setMutatorDump(&mStateQueueMutatorDump);
3513#endif
3514 FastMixerState *state = sq->begin();
3515 FastTrack *fastTrack = &state->mFastTracks[0];
3516 // wrap the source side of the MonoPipe to make it an AudioBufferProvider
3517 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
3518 fastTrack->mVolumeProvider = NULL;
Andy Hunge8a1ced2014-05-09 15:02:21 -07003519 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer
3520 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
Eric Laurent81784c32012-11-19 14:55:58 -08003521 fastTrack->mGeneration++;
3522 state->mFastTracksGen++;
3523 state->mTrackMask = 1;
3524 // fast mixer will use the HAL output sink
3525 state->mOutputSink = mOutputSink.get();
3526 state->mOutputSinkGen++;
3527 state->mFrameCount = mFrameCount;
3528 state->mCommand = FastMixerState::COLD_IDLE;
3529 // already done in constructor initialization list
3530 //mFastMixerFutex = 0;
3531 state->mColdFutexAddr = &mFastMixerFutex;
3532 state->mColdGen++;
3533 state->mDumpState = &mFastMixerDumpState;
Glenn Kasten46909e72013-02-26 09:20:22 -08003534#ifdef TEE_SINK
Eric Laurent81784c32012-11-19 14:55:58 -08003535 state->mTeeSink = mTeeSink.get();
Glenn Kasten46909e72013-02-26 09:20:22 -08003536#endif
Glenn Kasten9e58b552013-01-18 15:09:48 -08003537 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
3538 state->mNBLogWriter = mFastMixerNBLogWriter.get();
Eric Laurent81784c32012-11-19 14:55:58 -08003539 sq->end();
3540 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3541
3542 // start the fast mixer
3543 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
3544 pid_t tid = mFastMixer->getTid();
Eric Laurent72e3f392015-05-20 14:43:50 -07003545 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
Eric Laurent81784c32012-11-19 14:55:58 -08003546
3547#ifdef AUDIO_WATCHDOG
3548 // create and start the watchdog
3549 mAudioWatchdog = new AudioWatchdog();
3550 mAudioWatchdog->setDump(&mAudioWatchdogDump);
3551 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
3552 tid = mAudioWatchdog->getTid();
Eric Laurent72e3f392015-05-20 14:43:50 -07003553 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
Eric Laurent81784c32012-11-19 14:55:58 -08003554#endif
3555
Eric Laurent81784c32012-11-19 14:55:58 -08003556 }
3557
3558 switch (kUseFastMixer) {
3559 case FastMixer_Never:
3560 case FastMixer_Dynamic:
3561 mNormalSink = mOutputSink;
3562 break;
3563 case FastMixer_Always:
3564 mNormalSink = mPipeSink;
3565 break;
3566 case FastMixer_Static:
3567 mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
3568 break;
3569 }
3570}
3571
3572AudioFlinger::MixerThread::~MixerThread()
3573{
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003574 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003575 FastMixerStateQueue *sq = mFastMixer->sq();
3576 FastMixerState *state = sq->begin();
3577 if (state->mCommand == FastMixerState::COLD_IDLE) {
3578 int32_t old = android_atomic_inc(&mFastMixerFutex);
3579 if (old == -1) {
Elliott Hughesee499292014-05-21 17:55:51 -07003580 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
Eric Laurent81784c32012-11-19 14:55:58 -08003581 }
3582 }
3583 state->mCommand = FastMixerState::EXIT;
3584 sq->end();
3585 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3586 mFastMixer->join();
3587 // Though the fast mixer thread has exited, it's state queue is still valid.
3588 // We'll use that extract the final state which contains one remaining fast track
3589 // corresponding to our sub-mix.
3590 state = sq->begin();
3591 ALOG_ASSERT(state->mTrackMask == 1);
3592 FastTrack *fastTrack = &state->mFastTracks[0];
3593 ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
3594 delete fastTrack->mBufferProvider;
3595 sq->end(false /*didModify*/);
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003596 mFastMixer.clear();
Eric Laurent81784c32012-11-19 14:55:58 -08003597#ifdef AUDIO_WATCHDOG
3598 if (mAudioWatchdog != 0) {
3599 mAudioWatchdog->requestExit();
3600 mAudioWatchdog->requestExitAndWait();
3601 mAudioWatchdog.clear();
3602 }
3603#endif
3604 }
Glenn Kasten9e58b552013-01-18 15:09:48 -08003605 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter);
Eric Laurent81784c32012-11-19 14:55:58 -08003606 delete mAudioMixer;
3607}
3608
3609
3610uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
3611{
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003612 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003613 MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
3614 latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
3615 }
3616 return latency;
3617}
3618
3619
3620void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
3621{
3622 PlaybackThread::threadLoop_removeTracks(tracksToRemove);
3623}
3624
Eric Laurentbfb1b832013-01-07 09:53:42 -08003625ssize_t AudioFlinger::MixerThread::threadLoop_write()
Eric Laurent81784c32012-11-19 14:55:58 -08003626{
3627 // FIXME we should only do one push per cycle; confirm this is true
3628 // Start the fast mixer if it's not already running
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003629 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003630 FastMixerStateQueue *sq = mFastMixer->sq();
3631 FastMixerState *state = sq->begin();
3632 if (state->mCommand != FastMixerState::MIX_WRITE &&
3633 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
3634 if (state->mCommand == FastMixerState::COLD_IDLE) {
Eric Laurenta2ab4502015-09-09 12:25:51 -07003635
3636 // FIXME workaround for first HAL write being CPU bound on some devices
3637 ATRACE_BEGIN("write");
3638 mOutput->write((char *)mSinkBuffer, 0);
3639 ATRACE_END();
3640
Eric Laurent81784c32012-11-19 14:55:58 -08003641 int32_t old = android_atomic_inc(&mFastMixerFutex);
3642 if (old == -1) {
Elliott Hughesee499292014-05-21 17:55:51 -07003643 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
Eric Laurent81784c32012-11-19 14:55:58 -08003644 }
3645#ifdef AUDIO_WATCHDOG
3646 if (mAudioWatchdog != 0) {
3647 mAudioWatchdog->resume();
3648 }
3649#endif
3650 }
3651 state->mCommand = FastMixerState::MIX_WRITE;
Glenn Kastend797a9d2015-03-02 14:19:25 -08003652#ifdef FAST_THREAD_STATISTICS
Glenn Kasten4182c4e2013-07-15 14:45:07 -07003653 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
Glenn Kastenfbdb2ac2015-03-02 14:47:19 -08003654 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN);
Glenn Kastend797a9d2015-03-02 14:19:25 -08003655#endif
Eric Laurent81784c32012-11-19 14:55:58 -08003656 sq->end();
3657 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3658 if (kUseFastMixer == FastMixer_Dynamic) {
3659 mNormalSink = mPipeSink;
3660 }
3661 } else {
3662 sq->end(false /*didModify*/);
3663 }
3664 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003665 return PlaybackThread::threadLoop_write();
Eric Laurent81784c32012-11-19 14:55:58 -08003666}
3667
3668void AudioFlinger::MixerThread::threadLoop_standby()
3669{
3670 // Idle the fast mixer if it's currently running
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003671 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003672 FastMixerStateQueue *sq = mFastMixer->sq();
3673 FastMixerState *state = sq->begin();
3674 if (!(state->mCommand & FastMixerState::IDLE)) {
3675 state->mCommand = FastMixerState::COLD_IDLE;
3676 state->mColdFutexAddr = &mFastMixerFutex;
3677 state->mColdGen++;
3678 mFastMixerFutex = 0;
3679 sq->end();
3680 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
3681 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
3682 if (kUseFastMixer == FastMixer_Dynamic) {
3683 mNormalSink = mOutputSink;
3684 }
3685#ifdef AUDIO_WATCHDOG
3686 if (mAudioWatchdog != 0) {
3687 mAudioWatchdog->pause();
3688 }
3689#endif
3690 } else {
3691 sq->end(false /*didModify*/);
3692 }
3693 }
3694 PlaybackThread::threadLoop_standby();
3695}
3696
Eric Laurentbfb1b832013-01-07 09:53:42 -08003697bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
3698{
3699 return false;
3700}
3701
3702bool AudioFlinger::PlaybackThread::shouldStandby_l()
3703{
3704 return !mStandby;
3705}
3706
3707bool AudioFlinger::PlaybackThread::waitingAsyncCallback()
3708{
3709 Mutex::Autolock _l(mLock);
3710 return waitingAsyncCallback_l();
3711}
3712
Eric Laurent81784c32012-11-19 14:55:58 -08003713// shared by MIXER and DIRECT, overridden by DUPLICATING
3714void AudioFlinger::PlaybackThread::threadLoop_standby()
3715{
3716 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
Phil Burk062e67a2015-02-11 13:40:50 -08003717 mOutput->standby();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003718 if (mUseAsyncWrite != 0) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07003719 // discard any pending drain or write ack by incrementing sequence
3720 mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
3721 mDrainSequence = (mDrainSequence + 2) & ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003722 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07003723 mCallbackThread->setWriteBlocked(mWriteAckSequence);
3724 mCallbackThread->setDraining(mDrainSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08003725 }
Eric Laurentd1f69b02014-12-15 14:33:13 -08003726 mHwPaused = false;
Eric Laurent81784c32012-11-19 14:55:58 -08003727}
3728
Haynes Mathew George4c6a4332014-01-15 12:31:39 -08003729void AudioFlinger::PlaybackThread::onAddNewTrack_l()
3730{
3731 ALOGV("signal playback thread");
3732 broadcast_l();
3733}
3734
Eric Laurent81784c32012-11-19 14:55:58 -08003735void AudioFlinger::MixerThread::threadLoop_mix()
3736{
Eric Laurent81784c32012-11-19 14:55:58 -08003737 // mix buffers...
Glenn Kastend79072e2016-01-06 08:41:20 -08003738 mAudioMixer->process();
Andy Hung25c2dac2014-02-27 14:56:00 -08003739 mCurrentWriteLength = mSinkBufferSize;
Eric Laurent81784c32012-11-19 14:55:58 -08003740 // increase sleep time progressively when application underrun condition clears.
3741 // Only increase sleep time if the mixer is ready for two consecutive times to avoid
3742 // that a steady state of alternating ready/not ready conditions keeps the sleep time
3743 // such that we would underrun the audio HAL.
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003744 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) {
Eric Laurent81784c32012-11-19 14:55:58 -08003745 sleepTimeShift--;
3746 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003747 mSleepTimeUs = 0;
3748 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
Eric Laurent81784c32012-11-19 14:55:58 -08003749 //TODO: delay standby when effects have a tail
Glenn Kasten4c053ea2014-09-28 14:41:07 -07003750
Eric Laurent81784c32012-11-19 14:55:58 -08003751}
3752
3753void AudioFlinger::MixerThread::threadLoop_sleepTime()
3754{
3755 // If no tracks are ready, sleep once for the duration of an output
3756 // buffer size, then write 0s to the output
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003757 if (mSleepTimeUs == 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003758 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003759 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift;
3760 if (mSleepTimeUs < kMinThreadSleepTimeUs) {
3761 mSleepTimeUs = kMinThreadSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08003762 }
3763 // reduce sleep time in case of consecutive application underruns to avoid
3764 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
3765 // duration we would end up writing less data than needed by the audio HAL if
3766 // the condition persists.
3767 if (sleepTimeShift < kMaxThreadSleepTimeShift) {
3768 sleepTimeShift++;
3769 }
3770 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003771 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08003772 }
3773 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
Andy Hung98ef9782014-03-04 14:46:50 -08003774 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared
3775 // before effects processing or output.
3776 if (mMixerBufferValid) {
3777 memset(mMixerBuffer, 0, mMixerBufferSize);
3778 } else {
3779 memset(mSinkBuffer, 0, mSinkBufferSize);
3780 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003781 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08003782 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
3783 "anticipated start");
3784 }
3785 // TODO add standby time extension fct of effect tail
3786}
3787
3788// prepareTracks_l() must be called with ThreadBase::mLock held
3789AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
3790 Vector< sp<Track> > *tracksToRemove)
3791{
3792
3793 mixer_state mixerStatus = MIXER_IDLE;
3794 // find out which tracks need to be processed
3795 size_t count = mActiveTracks.size();
3796 size_t mixedTracks = 0;
3797 size_t tracksWithEffect = 0;
3798 // counts only _active_ fast tracks
3799 size_t fastTracks = 0;
3800 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
3801
3802 float masterVolume = mMasterVolume;
3803 bool masterMute = mMasterMute;
3804
3805 if (masterMute) {
3806 masterVolume = 0;
3807 }
3808 // Delegate master volume control to effect in output mix effect chain if needed
3809 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
3810 if (chain != 0) {
3811 uint32_t v = (uint32_t)(masterVolume * (1 << 24));
3812 chain->setVolume_l(&v, &v);
3813 masterVolume = (float)((v + (1 << 23)) >> 24);
3814 chain.clear();
3815 }
3816
3817 // prepare a new state to push
3818 FastMixerStateQueue *sq = NULL;
3819 FastMixerState *state = NULL;
3820 bool didModify = false;
3821 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003822 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003823 sq = mFastMixer->sq();
3824 state = sq->begin();
3825 }
3826
Andy Hung69aed5f2014-02-25 17:24:40 -08003827 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found.
Andy Hung98ef9782014-03-04 14:46:50 -08003828 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found.
Andy Hung69aed5f2014-02-25 17:24:40 -08003829
Eric Laurent81784c32012-11-19 14:55:58 -08003830 for (size_t i=0 ; i<count ; i++) {
Glenn Kasten9fdcb0a2013-06-26 16:11:36 -07003831 const sp<Track> t = mActiveTracks[i].promote();
Eric Laurent81784c32012-11-19 14:55:58 -08003832 if (t == 0) {
3833 continue;
3834 }
3835
3836 // this const just means the local variable doesn't change
3837 Track* const track = t.get();
3838
3839 // process fast tracks
3840 if (track->isFastTrack()) {
3841
3842 // It's theoretically possible (though unlikely) for a fast track to be created
3843 // and then removed within the same normal mix cycle. This is not a problem, as
3844 // the track never becomes active so it's fast mixer slot is never touched.
3845 // The converse, of removing an (active) track and then creating a new track
3846 // at the identical fast mixer slot within the same normal mix cycle,
3847 // is impossible because the slot isn't marked available until the end of each cycle.
3848 int j = track->mFastIndex;
Glenn Kastendc2c50b2016-04-21 08:13:14 -07003849 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks);
Eric Laurent81784c32012-11-19 14:55:58 -08003850 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
3851 FastTrack *fastTrack = &state->mFastTracks[j];
3852
3853 // Determine whether the track is currently in underrun condition,
3854 // and whether it had a recent underrun.
3855 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
3856 FastTrackUnderruns underruns = ftDump->mUnderruns;
3857 uint32_t recentFull = (underruns.mBitFields.mFull -
3858 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
3859 uint32_t recentPartial = (underruns.mBitFields.mPartial -
3860 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
3861 uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
3862 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
3863 uint32_t recentUnderruns = recentPartial + recentEmpty;
3864 track->mObservedUnderruns = underruns;
3865 // don't count underruns that occur while stopping or pausing
3866 // or stopped which can occur when flush() is called while active
Glenn Kasten82aaf942013-07-17 16:05:07 -07003867 if (!(track->isStopping() || track->isPausing() || track->isStopped()) &&
3868 recentUnderruns > 0) {
3869 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun
3870 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount);
Phil Burk2812d9e2016-01-04 10:34:30 -08003871 } else {
3872 track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
Eric Laurent81784c32012-11-19 14:55:58 -08003873 }
3874
3875 // This is similar to the state machine for normal tracks,
3876 // with a few modifications for fast tracks.
3877 bool isActive = true;
3878 switch (track->mState) {
3879 case TrackBase::STOPPING_1:
3880 // track stays active in STOPPING_1 state until first underrun
Eric Laurentbfb1b832013-01-07 09:53:42 -08003881 if (recentUnderruns > 0 || track->isTerminated()) {
Eric Laurent81784c32012-11-19 14:55:58 -08003882 track->mState = TrackBase::STOPPING_2;
3883 }
3884 break;
3885 case TrackBase::PAUSING:
3886 // ramp down is not yet implemented
3887 track->setPaused();
3888 break;
3889 case TrackBase::RESUMING:
3890 // ramp up is not yet implemented
3891 track->mState = TrackBase::ACTIVE;
3892 break;
3893 case TrackBase::ACTIVE:
3894 if (recentFull > 0 || recentPartial > 0) {
3895 // track has provided at least some frames recently: reset retry count
3896 track->mRetryCount = kMaxTrackRetries;
3897 }
3898 if (recentUnderruns == 0) {
3899 // no recent underruns: stay active
3900 break;
3901 }
3902 // there has recently been an underrun of some kind
3903 if (track->sharedBuffer() == 0) {
3904 // were any of the recent underruns "empty" (no frames available)?
3905 if (recentEmpty == 0) {
3906 // no, then ignore the partial underruns as they are allowed indefinitely
3907 break;
3908 }
3909 // there has recently been an "empty" underrun: decrement the retry counter
3910 if (--(track->mRetryCount) > 0) {
3911 break;
3912 }
3913 // indicate to client process that the track was disabled because of underrun;
3914 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08003915 track->disable();
Eric Laurent81784c32012-11-19 14:55:58 -08003916 // remove from active list, but state remains ACTIVE [confusing but true]
3917 isActive = false;
3918 break;
3919 }
3920 // fall through
3921 case TrackBase::STOPPING_2:
3922 case TrackBase::PAUSED:
Eric Laurent81784c32012-11-19 14:55:58 -08003923 case TrackBase::STOPPED:
3924 case TrackBase::FLUSHED: // flush() while active
3925 // Check for presentation complete if track is inactive
3926 // We have consumed all the buffers of this track.
3927 // This would be incomplete if we auto-paused on underrun
3928 {
3929 size_t audioHALFrames =
3930 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08003931 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08003932 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
3933 // track stays in active list until presentation is complete
3934 break;
3935 }
3936 }
3937 if (track->isStopping_2()) {
3938 track->mState = TrackBase::STOPPED;
3939 }
3940 if (track->isStopped()) {
3941 // Can't reset directly, as fast mixer is still polling this track
3942 // track->reset();
3943 // So instead mark this track as needing to be reset after push with ack
3944 resetMask |= 1 << i;
3945 }
3946 isActive = false;
3947 break;
3948 case TrackBase::IDLE:
3949 default:
Glenn Kastenadad3d72014-02-21 14:51:43 -08003950 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState);
Eric Laurent81784c32012-11-19 14:55:58 -08003951 }
3952
3953 if (isActive) {
3954 // was it previously inactive?
3955 if (!(state->mTrackMask & (1 << j))) {
3956 ExtendedAudioBufferProvider *eabp = track;
3957 VolumeProvider *vp = track;
3958 fastTrack->mBufferProvider = eabp;
3959 fastTrack->mVolumeProvider = vp;
Eric Laurent81784c32012-11-19 14:55:58 -08003960 fastTrack->mChannelMask = track->mChannelMask;
Andy Hunge8a1ced2014-05-09 15:02:21 -07003961 fastTrack->mFormat = track->mFormat;
Eric Laurent81784c32012-11-19 14:55:58 -08003962 fastTrack->mGeneration++;
3963 state->mTrackMask |= 1 << j;
3964 didModify = true;
3965 // no acknowledgement required for newly active tracks
3966 }
3967 // cache the combined master volume and stream type volume for fast mixer; this
3968 // lacks any synchronization or barrier so VolumeProvider may read a stale value
Glenn Kastene4756fe2012-11-29 13:38:14 -08003969 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume;
Eric Laurent81784c32012-11-19 14:55:58 -08003970 ++fastTracks;
3971 } else {
3972 // was it previously active?
3973 if (state->mTrackMask & (1 << j)) {
3974 fastTrack->mBufferProvider = NULL;
3975 fastTrack->mGeneration++;
3976 state->mTrackMask &= ~(1 << j);
3977 didModify = true;
3978 // If any fast tracks were removed, we must wait for acknowledgement
3979 // because we're about to decrement the last sp<> on those tracks.
3980 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
3981 } else {
Glenn Kastenf7d65ee2015-12-02 13:45:01 -08003982 LOG_ALWAYS_FATAL("fast track %d should have been active; "
3983 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d",
3984 j, track->mState, state->mTrackMask, recentUnderruns,
3985 track->sharedBuffer() != 0);
Eric Laurent81784c32012-11-19 14:55:58 -08003986 }
3987 tracksToRemove->add(track);
3988 // Avoids a misleading display in dumpsys
3989 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
3990 }
3991 continue;
3992 }
3993
3994 { // local variable scope to avoid goto warning
3995
3996 audio_track_cblk_t* cblk = track->cblk();
3997
3998 // The first time a track is added we wait
3999 // for all its buffers to be filled before processing it
4000 int name = track->name();
4001 // make sure that we have enough frames to mix one full buffer.
4002 // enforce this condition only once to enable draining the buffer in case the client
4003 // app does not call stop() and relies on underrun to stop:
4004 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
4005 // during last round
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004006 size_t desiredFrames;
Andy Hung8edb8dc2015-03-26 19:13:55 -07004007 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004008 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
Andy Hung8edb8dc2015-03-26 19:13:55 -07004009
4010 desiredFrames = sourceFramesNeededWithTimestretch(
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004011 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed);
Andy Hung8edb8dc2015-03-26 19:13:55 -07004012 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed.
4013 // add frames already consumed but not yet released by the resampler
4014 // because mAudioTrackServerProxy->framesReady() will include these frames
4015 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
4016
Eric Laurent81784c32012-11-19 14:55:58 -08004017 uint32_t minFrames = 1;
4018 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
4019 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004020 minFrames = desiredFrames;
Eric Laurent81784c32012-11-19 14:55:58 -08004021 }
Eric Laurent13e4c962013-12-20 17:36:01 -08004022
4023 size_t framesReady = track->framesReady();
Glenn Kastene7754022014-10-31 12:11:26 -07004024 if (ATRACE_ENABLED()) {
4025 // I wish we had formatted trace names
4026 char traceName[16];
4027 strcpy(traceName, "nRdy");
4028 int name = track->name();
4029 if (AudioMixer::TRACK0 <= name &&
4030 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) {
4031 name -= AudioMixer::TRACK0;
4032 traceName[4] = (name / 10) + '0';
4033 traceName[5] = (name % 10) + '0';
4034 } else {
4035 traceName[4] = '?';
4036 traceName[5] = '?';
4037 }
4038 traceName[6] = '\0';
4039 ATRACE_INT(traceName, framesReady);
4040 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004041 if ((framesReady >= minFrames) && track->isReady() &&
Eric Laurent81784c32012-11-19 14:55:58 -08004042 !track->isPaused() && !track->isTerminated())
4043 {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004044 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this);
Eric Laurent81784c32012-11-19 14:55:58 -08004045
4046 mixedTracks++;
4047
Andy Hung69aed5f2014-02-25 17:24:40 -08004048 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means
4049 // there is an effect chain connected to the track
Eric Laurent81784c32012-11-19 14:55:58 -08004050 chain.clear();
Andy Hung69aed5f2014-02-25 17:24:40 -08004051 if (track->mainBuffer() != mSinkBuffer &&
4052 track->mainBuffer() != mMixerBuffer) {
Andy Hung98ef9782014-03-04 14:46:50 -08004053 if (mEffectBufferEnabled) {
4054 mEffectBufferValid = true; // Later can set directly.
4055 }
Eric Laurent81784c32012-11-19 14:55:58 -08004056 chain = getEffectChain_l(track->sessionId());
4057 // Delegate volume control to effect in track effect chain if needed
4058 if (chain != 0) {
4059 tracksWithEffect++;
4060 } else {
4061 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
4062 "session %d",
4063 name, track->sessionId());
4064 }
4065 }
4066
4067
4068 int param = AudioMixer::VOLUME;
4069 if (track->mFillingUpStatus == Track::FS_FILLED) {
4070 // no ramp for the first volume setting
4071 track->mFillingUpStatus = Track::FS_ACTIVE;
4072 if (track->mState == TrackBase::RESUMING) {
4073 track->mState = TrackBase::ACTIVE;
4074 param = AudioMixer::RAMP_VOLUME;
4075 }
4076 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004077 // FIXME should not make a decision based on mServer
4078 } else if (cblk->mServer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08004079 // If the track is stopped before the first frame was mixed,
4080 // do not apply ramp
4081 param = AudioMixer::RAMP_VOLUME;
4082 }
4083
4084 // compute volume for this track
Andy Hung6be49402014-05-30 10:42:03 -07004085 uint32_t vl, vr; // in U8.24 integer format
4086 float vlf, vrf, vaf; // in [0.0, 1.0] float format
Glenn Kastene4756fe2012-11-29 13:38:14 -08004087 if (track->isPausing() || mStreamTypes[track->streamType()].mute) {
Andy Hung6be49402014-05-30 10:42:03 -07004088 vl = vr = 0;
4089 vlf = vrf = vaf = 0.;
Eric Laurent81784c32012-11-19 14:55:58 -08004090 if (track->isPausing()) {
4091 track->setPaused();
4092 }
4093 } else {
4094
4095 // read original volumes with volume control
4096 float typeVolume = mStreamTypes[track->streamType()].volume;
4097 float v = masterVolume * typeVolume;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004098 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
Glenn Kastenc56f3422014-03-21 17:53:17 -07004099 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
Andy Hung6be49402014-05-30 10:42:03 -07004100 vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
4101 vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
Eric Laurent81784c32012-11-19 14:55:58 -08004102 // track volumes come from shared memory, so can't be trusted and must be clamped
Glenn Kastenc56f3422014-03-21 17:53:17 -07004103 if (vlf > GAIN_FLOAT_UNITY) {
4104 ALOGV("Track left volume out of range: %.3g", vlf);
4105 vlf = GAIN_FLOAT_UNITY;
Eric Laurent81784c32012-11-19 14:55:58 -08004106 }
Glenn Kastenc56f3422014-03-21 17:53:17 -07004107 if (vrf > GAIN_FLOAT_UNITY) {
4108 ALOGV("Track right volume out of range: %.3g", vrf);
4109 vrf = GAIN_FLOAT_UNITY;
Eric Laurent81784c32012-11-19 14:55:58 -08004110 }
4111 // now apply the master volume and stream type volume
Andy Hung6be49402014-05-30 10:42:03 -07004112 vlf *= v;
4113 vrf *= v;
Eric Laurent81784c32012-11-19 14:55:58 -08004114 // assuming master volume and stream type volume each go up to 1.0,
Andy Hung6be49402014-05-30 10:42:03 -07004115 // then derive vl and vr as U8.24 versions for the effect chain
4116 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;
4117 vl = (uint32_t) (scaleto8_24 * vlf);
4118 vr = (uint32_t) (scaleto8_24 * vrf);
4119 // vl and vr are now in U8.24 format
Glenn Kastene3aa6592012-12-04 12:22:46 -08004120 uint16_t sendLevel = proxy->getSendLevel_U4_12();
Eric Laurent81784c32012-11-19 14:55:58 -08004121 // send level comes from shared memory and so may be corrupt
4122 if (sendLevel > MAX_GAIN_INT) {
4123 ALOGV("Track send level out of range: %04X", sendLevel);
4124 sendLevel = MAX_GAIN_INT;
4125 }
Andy Hung6be49402014-05-30 10:42:03 -07004126 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel
4127 vaf = v * sendLevel * (1. / MAX_GAIN_INT);
Eric Laurent81784c32012-11-19 14:55:58 -08004128 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08004129
Eric Laurent81784c32012-11-19 14:55:58 -08004130 // Delegate volume control to effect in track effect chain if needed
4131 if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
4132 // Do not ramp volume if volume is controlled by effect
4133 param = AudioMixer::VOLUME;
Bryant Liub6be7f22014-06-12 22:02:41 +08004134 // Update remaining floating point volume levels
4135 vlf = (float)vl / (1 << 24);
4136 vrf = (float)vr / (1 << 24);
Eric Laurent81784c32012-11-19 14:55:58 -08004137 track->mHasVolumeController = true;
4138 } else {
4139 // force no volume ramp when volume controller was just disabled or removed
4140 // from effect chain to avoid volume spike
4141 if (track->mHasVolumeController) {
4142 param = AudioMixer::VOLUME;
4143 }
4144 track->mHasVolumeController = false;
4145 }
4146
Eric Laurent81784c32012-11-19 14:55:58 -08004147 // XXX: these things DON'T need to be done each time
4148 mAudioMixer->setBufferProvider(name, track);
4149 mAudioMixer->enable(name);
4150
Andy Hung6be49402014-05-30 10:42:03 -07004151 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);
4152 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);
4153 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);
Eric Laurent81784c32012-11-19 14:55:58 -08004154 mAudioMixer->setParameter(
4155 name,
4156 AudioMixer::TRACK,
4157 AudioMixer::FORMAT, (void *)track->format());
4158 mAudioMixer->setParameter(
4159 name,
4160 AudioMixer::TRACK,
Kévin PETIT377b2ec2014-02-03 12:35:36 +00004161 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
Andy Hung9a592762014-07-21 21:56:01 -07004162 mAudioMixer->setParameter(
4163 name,
4164 AudioMixer::TRACK,
4165 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask);
Glenn Kastene3aa6592012-12-04 12:22:46 -08004166 // limit track sample rate to 2 x output sample rate, which changes at re-configuration
Andy Hungcd044842014-08-07 11:04:34 -07004167 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004168 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
Glenn Kastene3aa6592012-12-04 12:22:46 -08004169 if (reqSampleRate == 0) {
4170 reqSampleRate = mSampleRate;
4171 } else if (reqSampleRate > maxSampleRate) {
4172 reqSampleRate = maxSampleRate;
4173 }
Eric Laurent81784c32012-11-19 14:55:58 -08004174 mAudioMixer->setParameter(
4175 name,
4176 AudioMixer::RESAMPLE,
4177 AudioMixer::SAMPLE_RATE,
Kévin PETIT377b2ec2014-02-03 12:35:36 +00004178 (void *)(uintptr_t)reqSampleRate);
Andy Hung8edb8dc2015-03-26 19:13:55 -07004179
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004180 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
Andy Hung8edb8dc2015-03-26 19:13:55 -07004181 mAudioMixer->setParameter(
4182 name,
4183 AudioMixer::TIMESTRETCH,
4184 AudioMixer::PLAYBACK_RATE,
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004185 &playbackRate);
Andy Hung8edb8dc2015-03-26 19:13:55 -07004186
Andy Hung69aed5f2014-02-25 17:24:40 -08004187 /*
4188 * Select the appropriate output buffer for the track.
4189 *
Andy Hung98ef9782014-03-04 14:46:50 -08004190 * Tracks with effects go into their own effects chain buffer
4191 * and from there into either mEffectBuffer or mSinkBuffer.
Andy Hung69aed5f2014-02-25 17:24:40 -08004192 *
4193 * Other tracks can use mMixerBuffer for higher precision
4194 * channel accumulation. If this buffer is enabled
4195 * (mMixerBufferEnabled true), then selected tracks will accumulate
4196 * into it.
4197 *
4198 */
4199 if (mMixerBufferEnabled
4200 && (track->mainBuffer() == mSinkBuffer
4201 || track->mainBuffer() == mMixerBuffer)) {
4202 mAudioMixer->setParameter(
4203 name,
4204 AudioMixer::TRACK,
Andy Hung78820702014-02-28 16:23:02 -08004205 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
Andy Hung69aed5f2014-02-25 17:24:40 -08004206 mAudioMixer->setParameter(
4207 name,
4208 AudioMixer::TRACK,
4209 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
4210 // TODO: override track->mainBuffer()?
4211 mMixerBufferValid = true;
4212 } else {
4213 mAudioMixer->setParameter(
4214 name,
4215 AudioMixer::TRACK,
Andy Hung78820702014-02-28 16:23:02 -08004216 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT);
Andy Hung69aed5f2014-02-25 17:24:40 -08004217 mAudioMixer->setParameter(
4218 name,
4219 AudioMixer::TRACK,
4220 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
4221 }
Eric Laurent81784c32012-11-19 14:55:58 -08004222 mAudioMixer->setParameter(
4223 name,
4224 AudioMixer::TRACK,
4225 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
4226
4227 // reset retry count
4228 track->mRetryCount = kMaxTrackRetries;
4229
4230 // If one track is ready, set the mixer ready if:
4231 // - the mixer was not ready during previous round OR
4232 // - no other track is not ready
4233 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
4234 mixerStatus != MIXER_TRACKS_ENABLED) {
4235 mixerStatus = MIXER_TRACKS_READY;
4236 }
4237 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004238 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
Andy Hung08fb1742015-05-31 23:22:10 -07004239 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)",
4240 track, framesReady, desiredFrames);
Glenn Kasten82aaf942013-07-17 16:05:07 -07004241 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
Phil Burk2812d9e2016-01-04 10:34:30 -08004242 } else {
4243 track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004244 }
Phil Burk2812d9e2016-01-04 10:34:30 -08004245
Eric Laurent81784c32012-11-19 14:55:58 -08004246 // clear effect chain input buffer if an active track underruns to avoid sending
4247 // previous audio buffer again to effects
4248 chain = getEffectChain_l(track->sessionId());
4249 if (chain != 0) {
4250 chain->clearInputBuffer();
4251 }
4252
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004253 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this);
Eric Laurent81784c32012-11-19 14:55:58 -08004254 if ((track->sharedBuffer() != 0) || track->isTerminated() ||
4255 track->isStopped() || track->isPaused()) {
4256 // We have consumed all the buffers of this track.
4257 // Remove it from the list of active tracks.
4258 // TODO: use actual buffer filling status instead of latency when available from
4259 // audio HAL
4260 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08004261 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08004262 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
4263 if (track->isStopped()) {
4264 track->reset();
4265 }
4266 tracksToRemove->add(track);
4267 }
4268 } else {
Eric Laurent81784c32012-11-19 14:55:58 -08004269 // No buffers for this track. Give it a few chances to
4270 // fill a buffer, then remove it from active list.
4271 if (--(track->mRetryCount) <= 0) {
Glenn Kastenc9b2e202013-02-26 11:32:32 -08004272 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
Eric Laurent81784c32012-11-19 14:55:58 -08004273 tracksToRemove->add(track);
4274 // indicate to client process that the track was disabled because of underrun;
4275 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08004276 track->disable();
Eric Laurent81784c32012-11-19 14:55:58 -08004277 // If one track is not ready, mark the mixer also not ready if:
4278 // - the mixer was ready during previous round OR
4279 // - no other track is ready
4280 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
4281 mixerStatus != MIXER_TRACKS_READY) {
4282 mixerStatus = MIXER_TRACKS_ENABLED;
4283 }
4284 }
4285 mAudioMixer->disable(name);
4286 }
4287
4288 } // local variable scope to avoid goto warning
Eric Laurent81784c32012-11-19 14:55:58 -08004289
4290 }
4291
4292 // Push the new FastMixer state if necessary
4293 bool pauseAudioWatchdog = false;
4294 if (didModify) {
4295 state->mFastTracksGen++;
4296 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
4297 if (kUseFastMixer == FastMixer_Dynamic &&
4298 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
4299 state->mCommand = FastMixerState::COLD_IDLE;
4300 state->mColdFutexAddr = &mFastMixerFutex;
4301 state->mColdGen++;
4302 mFastMixerFutex = 0;
4303 if (kUseFastMixer == FastMixer_Dynamic) {
4304 mNormalSink = mOutputSink;
4305 }
4306 // If we go into cold idle, need to wait for acknowledgement
4307 // so that fast mixer stops doing I/O.
4308 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
4309 pauseAudioWatchdog = true;
4310 }
Eric Laurent81784c32012-11-19 14:55:58 -08004311 }
4312 if (sq != NULL) {
4313 sq->end(didModify);
4314 sq->push(block);
4315 }
4316#ifdef AUDIO_WATCHDOG
4317 if (pauseAudioWatchdog && mAudioWatchdog != 0) {
4318 mAudioWatchdog->pause();
4319 }
4320#endif
4321
4322 // Now perform the deferred reset on fast tracks that have stopped
4323 while (resetMask != 0) {
4324 size_t i = __builtin_ctz(resetMask);
4325 ALOG_ASSERT(i < count);
4326 resetMask &= ~(1 << i);
4327 sp<Track> t = mActiveTracks[i].promote();
4328 if (t == 0) {
4329 continue;
4330 }
4331 Track* track = t.get();
4332 ALOG_ASSERT(track->isFastTrack() && track->isStopped());
4333 track->reset();
4334 }
4335
4336 // remove all the tracks that need to be...
Eric Laurentbfb1b832013-01-07 09:53:42 -08004337 removeTracks_l(*tracksToRemove);
Eric Laurent81784c32012-11-19 14:55:58 -08004338
Eric Laurent97d547d2014-09-02 14:45:53 -07004339 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
4340 mEffectBufferValid = true;
Marco Nelissenac302142014-10-20 13:15:38 -07004341 }
4342
4343 if (mEffectBufferValid) {
Marco Nelissen57088b52014-10-17 16:39:39 -07004344 // as long as there are effects we should clear the effects buffer, to avoid
4345 // passing a non-clean buffer to the effect chain
4346 memset(mEffectBuffer, 0, mEffectBufferSize);
Eric Laurent97d547d2014-09-02 14:45:53 -07004347 }
Andy Hung69aed5f2014-02-25 17:24:40 -08004348 // sink or mix buffer must be cleared if all tracks are connected to an
4349 // effect chain as in this case the mixer will not write to the sink or mix buffer
4350 // and track effects will accumulate into it
Eric Laurentbfb1b832013-01-07 09:53:42 -08004351 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
4352 (mixedTracks == 0 && fastTracks > 0))) {
Eric Laurent81784c32012-11-19 14:55:58 -08004353 // FIXME as a performance optimization, should remember previous zero status
Andy Hung69aed5f2014-02-25 17:24:40 -08004354 if (mMixerBufferValid) {
4355 memset(mMixerBuffer, 0, mMixerBufferSize);
4356 // TODO: In testing, mSinkBuffer below need not be cleared because
4357 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer
4358 // after mixing.
4359 //
4360 // To enforce this guarantee:
4361 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
4362 // (mixedTracks == 0 && fastTracks > 0))
4363 // must imply MIXER_TRACKS_READY.
4364 // Later, we may clear buffers regardless, and skip much of this logic.
4365 }
Andy Hung98ef9782014-03-04 14:46:50 -08004366 // FIXME as a performance optimization, should remember previous zero status
Andy Hung5567aaf2014-07-17 14:00:07 -07004367 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize);
Eric Laurent81784c32012-11-19 14:55:58 -08004368 }
4369
4370 // if any fast tracks, then status is ready
4371 mMixerStatusIgnoringFastTracks = mixerStatus;
4372 if (fastTracks > 0) {
4373 mixerStatus = MIXER_TRACKS_READY;
4374 }
4375 return mixerStatus;
4376}
4377
4378// getTrackName_l() must be called with ThreadBase::mLock held
Andy Hunge8a1ced2014-05-09 15:02:21 -07004379int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask,
Glenn Kastend848eb42016-03-08 13:42:11 -08004380 audio_format_t format, audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08004381{
Andy Hunge8a1ced2014-05-09 15:02:21 -07004382 return mAudioMixer->getTrackName(channelMask, format, sessionId);
Eric Laurent81784c32012-11-19 14:55:58 -08004383}
4384
4385// deleteTrackName_l() must be called with ThreadBase::mLock held
4386void AudioFlinger::MixerThread::deleteTrackName_l(int name)
4387{
4388 ALOGV("remove track (%d) and delete from mixer", name);
4389 mAudioMixer->deleteTrackName(name);
4390}
4391
Eric Laurent10351942014-05-08 18:49:52 -07004392// checkForNewParameter_l() must be called with ThreadBase::mLock held
4393bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair,
4394 status_t& status)
Eric Laurent81784c32012-11-19 14:55:58 -08004395{
Eric Laurent81784c32012-11-19 14:55:58 -08004396 bool reconfig = false;
Eric Laurent42537be2016-01-08 17:16:42 -08004397 bool a2dpDeviceChanged = false;
Eric Laurent81784c32012-11-19 14:55:58 -08004398
Eric Laurent10351942014-05-08 18:49:52 -07004399 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08004400
Glenn Kastenc05b8d72016-03-24 09:48:17 -07004401 AutoPark<FastMixer> park(mFastMixer);
Eric Laurent81784c32012-11-19 14:55:58 -08004402
Eric Laurent10351942014-05-08 18:49:52 -07004403 AudioParameter param = AudioParameter(keyValuePair);
4404 int value;
4405 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4406 reconfig = true;
4407 }
4408 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
Andy Hung9a592762014-07-21 21:56:01 -07004409 if (!isValidPcmSinkFormat((audio_format_t) value)) {
Eric Laurent10351942014-05-08 18:49:52 -07004410 status = BAD_VALUE;
4411 } else {
4412 // no need to save value, since it's constant
Eric Laurent81784c32012-11-19 14:55:58 -08004413 reconfig = true;
4414 }
Eric Laurent10351942014-05-08 18:49:52 -07004415 }
4416 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
Andy Hung9a592762014-07-21 21:56:01 -07004417 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) {
Eric Laurent10351942014-05-08 18:49:52 -07004418 status = BAD_VALUE;
4419 } else {
4420 // no need to save value, since it's constant
4421 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08004422 }
Eric Laurent10351942014-05-08 18:49:52 -07004423 }
4424 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4425 // do not accept frame count changes if tracks are open as the track buffer
4426 // size depends on frame count and correct behavior would not be guaranteed
4427 // if frame count is changed after track creation
4428 if (!mTracks.isEmpty()) {
4429 status = INVALID_OPERATION;
4430 } else {
4431 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08004432 }
Eric Laurent10351942014-05-08 18:49:52 -07004433 }
4434 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
Eric Laurent81784c32012-11-19 14:55:58 -08004435#ifdef ADD_BATTERY_DATA
Eric Laurent10351942014-05-08 18:49:52 -07004436 // when changing the audio output device, call addBatteryData to notify
4437 // the change
4438 if (mOutDevice != value) {
4439 uint32_t params = 0;
4440 // check whether speaker is on
4441 if (value & AUDIO_DEVICE_OUT_SPEAKER) {
4442 params |= IMediaPlayerService::kBatteryDataSpeakerOn;
Eric Laurent81784c32012-11-19 14:55:58 -08004443 }
Eric Laurent10351942014-05-08 18:49:52 -07004444
4445 audio_devices_t deviceWithoutSpeaker
4446 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
4447 // check if any other device (except speaker) is on
Eric Laurent054d9d32015-04-24 08:48:48 -07004448 if (value & deviceWithoutSpeaker) {
Eric Laurent10351942014-05-08 18:49:52 -07004449 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
4450 }
4451
4452 if (params != 0) {
4453 addBatteryData(params);
4454 }
4455 }
Eric Laurent81784c32012-11-19 14:55:58 -08004456#endif
4457
Eric Laurent10351942014-05-08 18:49:52 -07004458 // forward device change to effects that have requested to be
4459 // aware of attached audio device.
4460 if (value != AUDIO_DEVICE_NONE) {
Eric Laurent42537be2016-01-08 17:16:42 -08004461 a2dpDeviceChanged =
4462 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP);
Eric Laurent10351942014-05-08 18:49:52 -07004463 mOutDevice = value;
4464 for (size_t i = 0; i < mEffectChains.size(); i++) {
4465 mEffectChains[i]->setDevice_l(mOutDevice);
Eric Laurent81784c32012-11-19 14:55:58 -08004466 }
4467 }
Eric Laurent10351942014-05-08 18:49:52 -07004468 }
Eric Laurent81784c32012-11-19 14:55:58 -08004469
Eric Laurent10351942014-05-08 18:49:52 -07004470 if (status == NO_ERROR) {
4471 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
4472 keyValuePair.string());
4473 if (!mStandby && status == INVALID_OPERATION) {
Phil Burk062e67a2015-02-11 13:40:50 -08004474 mOutput->standby();
Eric Laurent10351942014-05-08 18:49:52 -07004475 mStandby = true;
4476 mBytesWritten = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08004477 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
Eric Laurent10351942014-05-08 18:49:52 -07004478 keyValuePair.string());
Eric Laurent81784c32012-11-19 14:55:58 -08004479 }
Eric Laurent10351942014-05-08 18:49:52 -07004480 if (status == NO_ERROR && reconfig) {
4481 readOutputParameters_l();
4482 delete mAudioMixer;
4483 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
4484 for (size_t i = 0; i < mTracks.size() ; i++) {
Andy Hunge8a1ced2014-05-09 15:02:21 -07004485 int name = getTrackName_l(mTracks[i]->mChannelMask,
4486 mTracks[i]->mFormat, mTracks[i]->mSessionId);
Eric Laurent10351942014-05-08 18:49:52 -07004487 if (name < 0) {
4488 break;
4489 }
4490 mTracks[i]->mName = name;
4491 }
Eric Laurent73e26b62015-04-27 16:55:58 -07004492 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
Eric Laurent10351942014-05-08 18:49:52 -07004493 }
Eric Laurent81784c32012-11-19 14:55:58 -08004494 }
4495
Eric Laurent42537be2016-01-08 17:16:42 -08004496 return reconfig || a2dpDeviceChanged;
Eric Laurent81784c32012-11-19 14:55:58 -08004497}
4498
4499
4500void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
4501{
Eric Laurent81784c32012-11-19 14:55:58 -08004502 PlaybackThread::dumpInternals(fd, args);
Andy Hung40eb1a12015-06-18 13:42:02 -07004503 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs);
Elliott Hughes87cebad2014-05-22 10:14:43 -07004504 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
Andy Hung2ddee192015-12-18 17:34:44 -08004505 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off");
Eric Laurent81784c32012-11-19 14:55:58 -08004506
4507 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
Glenn Kasten2f90c512015-12-02 11:40:09 -08004508 // while we are dumping it. It may be inconsistent, but it won't mutate!
4509 // This is a large object so we place it on the heap.
4510 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
4511 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState);
4512 copy->dump(fd);
4513 delete copy;
Eric Laurent81784c32012-11-19 14:55:58 -08004514
4515#ifdef STATE_QUEUE_DUMP
4516 // Similar for state queue
4517 StateQueueObserverDump observerCopy = mStateQueueObserverDump;
4518 observerCopy.dump(fd);
4519 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
4520 mutatorCopy.dump(fd);
4521#endif
4522
Glenn Kasten46909e72013-02-26 09:20:22 -08004523#ifdef TEE_SINK
Eric Laurent81784c32012-11-19 14:55:58 -08004524 // Write the tee output to a .wav file
4525 dumpTee(fd, mTeeSource, mId);
Glenn Kasten46909e72013-02-26 09:20:22 -08004526#endif
Eric Laurent81784c32012-11-19 14:55:58 -08004527
4528#ifdef AUDIO_WATCHDOG
4529 if (mAudioWatchdog != 0) {
4530 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
4531 AudioWatchdogDump wdCopy = mAudioWatchdogDump;
4532 wdCopy.dump(fd);
4533 }
4534#endif
4535}
4536
4537uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
4538{
4539 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
4540}
4541
4542uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
4543{
4544 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
4545}
4546
4547void AudioFlinger::MixerThread::cacheParameters_l()
4548{
4549 PlaybackThread::cacheParameters_l();
4550
4551 // FIXME: Relaxed timing because of a certain device that can't meet latency
4552 // Should be reduced to 2x after the vendor fixes the driver issue
4553 // increase threshold again due to low power audio mode. The way this warning
4554 // threshold is calculated and its usefulness should be reconsidered anyway.
4555 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
4556}
4557
4558// ----------------------------------------------------------------------------
4559
4560AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
Eric Laurente93cc032016-05-05 10:15:10 -07004561 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady)
4562 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady)
Eric Laurent81784c32012-11-19 14:55:58 -08004563 // mLeftVolFloat, mRightVolFloat
4564{
4565}
4566
Eric Laurentbfb1b832013-01-07 09:53:42 -08004567AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
4568 AudioStreamOut* output, audio_io_handle_t id, uint32_t device,
Eric Laurente93cc032016-05-05 10:15:10 -07004569 ThreadBase::type_t type, bool systemReady)
4570 : PlaybackThread(audioFlinger, output, id, device, type, systemReady)
Eric Laurentbfb1b832013-01-07 09:53:42 -08004571 // mLeftVolFloat, mRightVolFloat
4572{
4573}
4574
Eric Laurent81784c32012-11-19 14:55:58 -08004575AudioFlinger::DirectOutputThread::~DirectOutputThread()
4576{
4577}
4578
Eric Laurentbfb1b832013-01-07 09:53:42 -08004579void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
4580{
Eric Laurentbfb1b832013-01-07 09:53:42 -08004581 float left, right;
4582
4583 if (mMasterMute || mStreamTypes[track->streamType()].mute) {
4584 left = right = 0;
4585 } else {
4586 float typeVolume = mStreamTypes[track->streamType()].volume;
4587 float v = mMasterVolume * typeVolume;
4588 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
Glenn Kastenc56f3422014-03-21 17:53:17 -07004589 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
4590 left = float_from_gain(gain_minifloat_unpack_left(vlr));
4591 if (left > GAIN_FLOAT_UNITY) {
4592 left = GAIN_FLOAT_UNITY;
4593 }
4594 left *= v;
4595 right = float_from_gain(gain_minifloat_unpack_right(vlr));
4596 if (right > GAIN_FLOAT_UNITY) {
4597 right = GAIN_FLOAT_UNITY;
4598 }
4599 right *= v;
Eric Laurentbfb1b832013-01-07 09:53:42 -08004600 }
4601
4602 if (lastTrack) {
4603 if (left != mLeftVolFloat || right != mRightVolFloat) {
4604 mLeftVolFloat = left;
4605 mRightVolFloat = right;
4606
4607 // Convert volumes from float to 8.24
4608 uint32_t vl = (uint32_t)(left * (1 << 24));
4609 uint32_t vr = (uint32_t)(right * (1 << 24));
4610
4611 // Delegate volume control to effect in track effect chain if needed
4612 // only one effect chain can be present on DirectOutputThread, so if
4613 // there is one, the track is connected to it
4614 if (!mEffectChains.isEmpty()) {
4615 mEffectChains[0]->setVolume_l(&vl, &vr);
4616 left = (float)vl / (1 << 24);
4617 right = (float)vr / (1 << 24);
4618 }
4619 if (mOutput->stream->set_volume) {
4620 mOutput->stream->set_volume(mOutput->stream, left, right);
4621 }
4622 }
4623 }
4624}
4625
Phil Burk43b4dcc2015-06-09 16:53:44 -07004626void AudioFlinger::DirectOutputThread::onAddNewTrack_l()
4627{
4628 sp<Track> previousTrack = mPreviousTrack.promote();
4629 sp<Track> latestTrack = mLatestActiveTrack.promote();
4630
Eric Laurent0f0631e2015-07-06 18:01:25 -07004631 if (previousTrack != 0 && latestTrack != 0) {
4632 if (mType == DIRECT) {
4633 if (previousTrack.get() != latestTrack.get()) {
4634 mFlushPending = true;
4635 }
4636 } else /* mType == OFFLOAD */ {
4637 if (previousTrack->sessionId() != latestTrack->sessionId()) {
4638 mFlushPending = true;
4639 }
4640 }
Phil Burk43b4dcc2015-06-09 16:53:44 -07004641 }
4642 PlaybackThread::onAddNewTrack_l();
4643}
Eric Laurentbfb1b832013-01-07 09:53:42 -08004644
Eric Laurent81784c32012-11-19 14:55:58 -08004645AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
4646 Vector< sp<Track> > *tracksToRemove
4647)
4648{
Eric Laurentd595b7c2013-04-03 17:27:56 -07004649 size_t count = mActiveTracks.size();
Eric Laurent81784c32012-11-19 14:55:58 -08004650 mixer_state mixerStatus = MIXER_IDLE;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004651 bool doHwPause = false;
4652 bool doHwResume = false;
Eric Laurent81784c32012-11-19 14:55:58 -08004653
4654 // find out which tracks need to be processed
Eric Laurentd595b7c2013-04-03 17:27:56 -07004655 for (size_t i = 0; i < count; i++) {
4656 sp<Track> t = mActiveTracks[i].promote();
Eric Laurent81784c32012-11-19 14:55:58 -08004657 // The track died recently
4658 if (t == 0) {
Eric Laurentd595b7c2013-04-03 17:27:56 -07004659 continue;
Eric Laurent81784c32012-11-19 14:55:58 -08004660 }
4661
Phil Burk43b4dcc2015-06-09 16:53:44 -07004662 if (t->isInvalid()) {
4663 ALOGW("An invalidated track shouldn't be in active list");
4664 tracksToRemove->add(t);
4665 continue;
4666 }
4667
Eric Laurent81784c32012-11-19 14:55:58 -08004668 Track* const track = t.get();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07004669#ifdef VERY_VERY_VERBOSE_LOGGING
Eric Laurent81784c32012-11-19 14:55:58 -08004670 audio_track_cblk_t* cblk = track->cblk();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07004671#endif
Eric Laurentfd477972013-10-25 18:10:40 -07004672 // Only consider last track started for volume and mixer state control.
4673 // In theory an older track could underrun and restart after the new one starts
4674 // but as we only care about the transition phase between two tracks on a
4675 // direct output, it is not a problem to ignore the underrun case.
4676 sp<Track> l = mLatestActiveTrack.promote();
4677 bool last = l.get() == track;
Eric Laurent81784c32012-11-19 14:55:58 -08004678
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004679 if (track->isPausing()) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004680 track->setPaused();
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004681 if (mHwSupportsPause && last && !mHwPaused) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004682 doHwPause = true;
4683 mHwPaused = true;
4684 }
4685 tracksToRemove->add(track);
4686 } else if (track->isFlushPending()) {
4687 track->flushAck();
4688 if (last) {
Phil Burk43b4dcc2015-06-09 16:53:44 -07004689 mFlushPending = true;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004690 }
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004691 } else if (track->isResumePending()) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004692 track->resumeAck();
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004693 if (last && mHwPaused) {
4694 doHwResume = true;
4695 mHwPaused = false;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004696 }
4697 }
4698
Eric Laurent81784c32012-11-19 14:55:58 -08004699 // The first time a track is added we wait
Phil Burk99adee32014-12-10 16:46:30 -08004700 // for all its buffers to be filled before processing it.
4701 // Allow draining the buffer in case the client
4702 // app does not call stop() and relies on underrun to stop:
4703 // hence the test on (track->mRetryCount > 1).
4704 // If retryCount<=1 then track is about to underrun and be removed.
Phil Burkca5e6142015-07-14 09:42:29 -07004705 // Do not use a high threshold for compressed audio.
Eric Laurent81784c32012-11-19 14:55:58 -08004706 uint32_t minFrames;
Phil Burk99adee32014-12-10 16:46:30 -08004707 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()
Phil Burkfdb3c072016-02-09 10:47:02 -08004708 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08004709 minFrames = mNormalFrameCount;
4710 } else {
4711 minFrames = 1;
4712 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08004713
Eric Laurentab5cdba2014-06-09 17:22:27 -07004714 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
4715 !track->isStopping_2() && !track->isStopped())
Eric Laurent81784c32012-11-19 14:55:58 -08004716 {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004717 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer);
Eric Laurent81784c32012-11-19 14:55:58 -08004718
4719 if (track->mFillingUpStatus == Track::FS_FILLED) {
4720 track->mFillingUpStatus = Track::FS_ACTIVE;
Eric Laurent1abbdb42013-09-13 17:00:08 -07004721 // make sure processVolume_l() will apply new volume even if 0
4722 mLeftVolFloat = mRightVolFloat = -1.0;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004723 if (!mHwSupportsPause) {
4724 track->resumeAck();
Eric Laurent81784c32012-11-19 14:55:58 -08004725 }
4726 }
4727
4728 // compute volume for this track
Eric Laurentbfb1b832013-01-07 09:53:42 -08004729 processVolume_l(track, last);
4730 if (last) {
Phil Burk43b4dcc2015-06-09 16:53:44 -07004731 sp<Track> previousTrack = mPreviousTrack.promote();
4732 if (previousTrack != 0) {
4733 if (track != previousTrack.get()) {
4734 // Flush any data still being written from last track
4735 mBytesRemaining = 0;
Eric Laurent0f0631e2015-07-06 18:01:25 -07004736 // Invalidate previous track to force a seek when resuming.
4737 previousTrack->invalidate();
Phil Burk43b4dcc2015-06-09 16:53:44 -07004738 }
4739 }
4740 mPreviousTrack = track;
4741
Eric Laurentd595b7c2013-04-03 17:27:56 -07004742 // reset retry count
4743 track->mRetryCount = kMaxTrackRetriesDirect;
4744 mActiveTrack = t;
4745 mixerStatus = MIXER_TRACKS_READY;
Eric Laurent5cff4032015-05-26 13:49:58 -07004746 if (mHwPaused) {
Eric Laurent0f7b5f22014-12-19 10:43:21 -08004747 doHwResume = true;
4748 mHwPaused = false;
4749 }
Eric Laurentd595b7c2013-04-03 17:27:56 -07004750 }
Eric Laurent81784c32012-11-19 14:55:58 -08004751 } else {
Eric Laurentd595b7c2013-04-03 17:27:56 -07004752 // clear effect chain input buffer if the last active track started underruns
4753 // to avoid sending previous audio buffer again to effects
Eric Laurentfd477972013-10-25 18:10:40 -07004754 if (!mEffectChains.isEmpty() && last) {
Eric Laurent81784c32012-11-19 14:55:58 -08004755 mEffectChains[0]->clearInputBuffer();
4756 }
Eric Laurentab5cdba2014-06-09 17:22:27 -07004757 if (track->isStopping_1()) {
4758 track->mState = TrackBase::STOPPING_2;
Eric Laurentb369caf2015-03-30 20:51:47 -07004759 if (last && mHwPaused) {
4760 doHwResume = true;
4761 mHwPaused = false;
4762 }
Eric Laurentab5cdba2014-06-09 17:22:27 -07004763 }
4764 if ((track->sharedBuffer() != 0) || track->isStopped() ||
4765 track->isStopping_2() || track->isPaused()) {
Eric Laurent81784c32012-11-19 14:55:58 -08004766 // We have consumed all the buffers of this track.
4767 // Remove it from the list of active tracks.
Eric Laurentab5cdba2014-06-09 17:22:27 -07004768 size_t audioHALFrames;
Phil Burkfdb3c072016-02-09 10:47:02 -08004769 if (audio_has_proportional_frames(mFormat)) {
Eric Laurentab5cdba2014-06-09 17:22:27 -07004770 audioHALFrames = (latency_l() * mSampleRate) / 1000;
4771 } else {
4772 audioHALFrames = 0;
4773 }
4774
Andy Hung818e7a32016-02-16 18:08:07 -08004775 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurentfd477972013-10-25 18:10:40 -07004776 if (mStandby || !last ||
4777 track->presentationComplete(framesWritten, audioHALFrames)) {
Eric Laurentab5cdba2014-06-09 17:22:27 -07004778 if (track->isStopping_2()) {
4779 track->mState = TrackBase::STOPPED;
4780 }
Eric Laurent81784c32012-11-19 14:55:58 -08004781 if (track->isStopped()) {
4782 track->reset();
4783 }
Eric Laurentd595b7c2013-04-03 17:27:56 -07004784 tracksToRemove->add(track);
Eric Laurent81784c32012-11-19 14:55:58 -08004785 }
4786 } else {
4787 // No buffers for this track. Give it a few chances to
4788 // fill a buffer, then remove it from active list.
Eric Laurentd595b7c2013-04-03 17:27:56 -07004789 // Only consider last track started for mixer state control
Eric Laurent81784c32012-11-19 14:55:58 -08004790 if (--(track->mRetryCount) <= 0) {
4791 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
Eric Laurentd595b7c2013-04-03 17:27:56 -07004792 tracksToRemove->add(track);
Eric Laurenta23f17a2013-11-05 18:22:08 -08004793 // indicate to client process that the track was disabled because of underrun;
4794 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08004795 track->disable();
Eric Laurentbfb1b832013-01-07 09:53:42 -08004796 } else if (last) {
Phil Burkca5e6142015-07-14 09:42:29 -07004797 ALOGW("pause because of UNDERRUN, framesReady = %zu,"
4798 "minFrames = %u, mFormat = %#x",
4799 track->framesReady(), minFrames, mFormat);
Eric Laurent81784c32012-11-19 14:55:58 -08004800 mixerStatus = MIXER_TRACKS_ENABLED;
Eric Laurent5cff4032015-05-26 13:49:58 -07004801 if (mHwSupportsPause && !mHwPaused && !mStandby) {
Eric Laurent0f7b5f22014-12-19 10:43:21 -08004802 doHwPause = true;
4803 mHwPaused = true;
4804 }
Eric Laurent81784c32012-11-19 14:55:58 -08004805 }
4806 }
4807 }
4808 }
4809
Eric Laurentd1f69b02014-12-15 14:33:13 -08004810 // if an active track did not command a flush, check for pending flush on stopped tracks
Phil Burk43b4dcc2015-06-09 16:53:44 -07004811 if (!mFlushPending) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004812 for (size_t i = 0; i < mTracks.size(); i++) {
4813 if (mTracks[i]->isFlushPending()) {
4814 mTracks[i]->flushAck();
Phil Burk43b4dcc2015-06-09 16:53:44 -07004815 mFlushPending = true;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004816 }
4817 }
4818 }
4819
4820 // make sure the pause/flush/resume sequence is executed in the right order.
4821 // If a flush is pending and a track is active but the HW is not paused, force a HW pause
4822 // before flush and then resume HW. This can happen in case of pause/flush/resume
4823 // if resume is received before pause is executed.
4824 if (mHwSupportsPause && !mStandby &&
Phil Burk43b4dcc2015-06-09 16:53:44 -07004825 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004826 mOutput->stream->pause(mOutput->stream);
4827 }
Phil Burk43b4dcc2015-06-09 16:53:44 -07004828 if (mFlushPending) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004829 flushHw_l();
4830 }
4831 if (mHwSupportsPause && !mStandby && doHwResume) {
4832 mOutput->stream->resume(mOutput->stream);
4833 }
Eric Laurent81784c32012-11-19 14:55:58 -08004834 // remove all the tracks that need to be...
Eric Laurentbfb1b832013-01-07 09:53:42 -08004835 removeTracks_l(*tracksToRemove);
Eric Laurent81784c32012-11-19 14:55:58 -08004836
4837 return mixerStatus;
4838}
4839
4840void AudioFlinger::DirectOutputThread::threadLoop_mix()
4841{
Eric Laurent81784c32012-11-19 14:55:58 -08004842 size_t frameCount = mFrameCount;
Andy Hung2098f272014-02-27 14:00:06 -08004843 int8_t *curBuf = (int8_t *)mSinkBuffer;
Eric Laurent81784c32012-11-19 14:55:58 -08004844 // output audio to hardware
4845 while (frameCount) {
Glenn Kasten34542ac2013-06-26 11:29:02 -07004846 AudioBufferProvider::Buffer buffer;
Eric Laurent81784c32012-11-19 14:55:58 -08004847 buffer.frameCount = frameCount;
Phil Burk062e67a2015-02-11 13:40:50 -08004848 status_t status = mActiveTrack->getNextBuffer(&buffer);
4849 if (status != NO_ERROR || buffer.raw == NULL) {
Eric Laurent51716182016-02-29 18:00:56 -08004850 // no need to pad with 0 for compressed audio
4851 if (audio_has_proportional_frames(mFormat)) {
4852 memset(curBuf, 0, frameCount * mFrameSize);
4853 }
Eric Laurent81784c32012-11-19 14:55:58 -08004854 break;
4855 }
4856 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
4857 frameCount -= buffer.frameCount;
4858 curBuf += buffer.frameCount * mFrameSize;
4859 mActiveTrack->releaseBuffer(&buffer);
4860 }
Andy Hung2098f272014-02-27 14:00:06 -08004861 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004862 mSleepTimeUs = 0;
4863 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
Eric Laurent81784c32012-11-19 14:55:58 -08004864 mActiveTrack.clear();
Eric Laurent81784c32012-11-19 14:55:58 -08004865}
4866
4867void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
4868{
Eric Laurentd1f69b02014-12-15 14:33:13 -08004869 // do not write to HAL when paused
Eric Laurent0f7b5f22014-12-19 10:43:21 -08004870 if (mHwPaused || (usesHwAvSync() && mStandby)) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004871 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004872 return;
4873 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004874 if (mSleepTimeUs == 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08004875 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
Eric Laurente93cc032016-05-05 10:15:10 -07004876 mSleepTimeUs = mActiveSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08004877 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004878 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08004879 }
Phil Burkfdb3c072016-02-09 10:47:02 -08004880 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) {
Andy Hung2098f272014-02-27 14:00:06 -08004881 memset(mSinkBuffer, 0, mFrameCount * mFrameSize);
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004882 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08004883 }
4884}
4885
Eric Laurentd1f69b02014-12-15 14:33:13 -08004886void AudioFlinger::DirectOutputThread::threadLoop_exit()
4887{
4888 {
4889 Mutex::Autolock _l(mLock);
Eric Laurentd1f69b02014-12-15 14:33:13 -08004890 for (size_t i = 0; i < mTracks.size(); i++) {
4891 if (mTracks[i]->isFlushPending()) {
4892 mTracks[i]->flushAck();
Phil Burk43b4dcc2015-06-09 16:53:44 -07004893 mFlushPending = true;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004894 }
4895 }
Phil Burk43b4dcc2015-06-09 16:53:44 -07004896 if (mFlushPending) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004897 flushHw_l();
4898 }
4899 }
4900 PlaybackThread::threadLoop_exit();
4901}
4902
4903// must be called with thread mutex locked
4904bool AudioFlinger::DirectOutputThread::shouldStandby_l()
4905{
4906 bool trackPaused = false;
Eric Laurentb369caf2015-03-30 20:51:47 -07004907 bool trackStopped = false;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004908
vivek mehta9cd7ad12016-03-17 00:18:29 -07004909 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) {
4910 return !mStandby;
4911 }
4912
Eric Laurentd1f69b02014-12-15 14:33:13 -08004913 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
4914 // after a timeout and we will enter standby then.
4915 if (mTracks.size() > 0) {
4916 trackPaused = mTracks[mTracks.size() - 1]->isPaused();
Eric Laurentb369caf2015-03-30 20:51:47 -07004917 trackStopped = mTracks[mTracks.size() - 1]->isStopped() ||
4918 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004919 }
4920
Eric Laurent5cff4032015-05-26 13:49:58 -07004921 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped));
Eric Laurentd1f69b02014-12-15 14:33:13 -08004922}
4923
Eric Laurent81784c32012-11-19 14:55:58 -08004924// getTrackName_l() must be called with ThreadBase::mLock held
Glenn Kasten0f11b512014-01-31 16:18:54 -08004925int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
Glenn Kastend848eb42016-03-08 13:42:11 -08004926 audio_format_t format __unused, audio_session_t sessionId __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08004927{
4928 return 0;
4929}
4930
4931// deleteTrackName_l() must be called with ThreadBase::mLock held
Glenn Kasten0f11b512014-01-31 16:18:54 -08004932void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08004933{
4934}
4935
Eric Laurent10351942014-05-08 18:49:52 -07004936// checkForNewParameter_l() must be called with ThreadBase::mLock held
4937bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair,
4938 status_t& status)
Eric Laurent81784c32012-11-19 14:55:58 -08004939{
4940 bool reconfig = false;
Eric Laurent42537be2016-01-08 17:16:42 -08004941 bool a2dpDeviceChanged = false;
Eric Laurent81784c32012-11-19 14:55:58 -08004942
Eric Laurent10351942014-05-08 18:49:52 -07004943 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08004944
Eric Laurent10351942014-05-08 18:49:52 -07004945 AudioParameter param = AudioParameter(keyValuePair);
4946 int value;
4947 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4948 // forward device change to effects that have requested to be
4949 // aware of attached audio device.
4950 if (value != AUDIO_DEVICE_NONE) {
Eric Laurent42537be2016-01-08 17:16:42 -08004951 a2dpDeviceChanged =
4952 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP);
Eric Laurent10351942014-05-08 18:49:52 -07004953 mOutDevice = value;
4954 for (size_t i = 0; i < mEffectChains.size(); i++) {
4955 mEffectChains[i]->setDevice_l(mOutDevice);
Glenn Kastenc125f382014-04-11 18:37:33 -07004956 }
4957 }
Eric Laurent81784c32012-11-19 14:55:58 -08004958 }
Eric Laurent10351942014-05-08 18:49:52 -07004959 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4960 // do not accept frame count changes if tracks are open as the track buffer
4961 // size depends on frame count and correct behavior would not be garantied
4962 // if frame count is changed after track creation
4963 if (!mTracks.isEmpty()) {
4964 status = INVALID_OPERATION;
4965 } else {
4966 reconfig = true;
4967 }
4968 }
4969 if (status == NO_ERROR) {
4970 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
4971 keyValuePair.string());
4972 if (!mStandby && status == INVALID_OPERATION) {
Phil Burk062e67a2015-02-11 13:40:50 -08004973 mOutput->standby();
Eric Laurent10351942014-05-08 18:49:52 -07004974 mStandby = true;
4975 mBytesWritten = 0;
4976 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
4977 keyValuePair.string());
4978 }
4979 if (status == NO_ERROR && reconfig) {
4980 readOutputParameters_l();
Eric Laurent73e26b62015-04-27 16:55:58 -07004981 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
Eric Laurent10351942014-05-08 18:49:52 -07004982 }
4983 }
4984
Eric Laurent42537be2016-01-08 17:16:42 -08004985 return reconfig || a2dpDeviceChanged;
Eric Laurent81784c32012-11-19 14:55:58 -08004986}
4987
4988uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
4989{
4990 uint32_t time;
Phil Burkfdb3c072016-02-09 10:47:02 -08004991 if (audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08004992 time = PlaybackThread::activeSleepTimeUs();
4993 } else {
Eric Laurent51716182016-02-29 18:00:56 -08004994 time = kDirectMinSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08004995 }
4996 return time;
4997}
4998
4999uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
5000{
5001 uint32_t time;
Phil Burkfdb3c072016-02-09 10:47:02 -08005002 if (audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08005003 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
5004 } else {
Eric Laurent51716182016-02-29 18:00:56 -08005005 time = kDirectMinSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005006 }
5007 return time;
5008}
5009
5010uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
5011{
5012 uint32_t time;
Phil Burkfdb3c072016-02-09 10:47:02 -08005013 if (audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08005014 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
5015 } else {
Eric Laurent51716182016-02-29 18:00:56 -08005016 time = kDirectMinSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005017 }
5018 return time;
5019}
5020
5021void AudioFlinger::DirectOutputThread::cacheParameters_l()
5022{
5023 PlaybackThread::cacheParameters_l();
5024
5025 // use shorter standby delay as on normal output to release
5026 // hardware resources as soon as possible
Eric Laurentb369caf2015-03-30 20:51:47 -07005027 // no delay on outputs with HW A/V sync
5028 if (usesHwAvSync()) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005029 mStandbyDelayNs = 0;
Phil Burkfdb3c072016-02-09 10:47:02 -08005030 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005031 mStandbyDelayNs = kOffloadStandbyDelayNs;
Eric Laurent5cff4032015-05-26 13:49:58 -07005032 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005033 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2);
Eric Laurent972a1732013-09-04 09:42:59 -07005034 }
Eric Laurent81784c32012-11-19 14:55:58 -08005035}
5036
Eric Laurente659ef42014-09-29 13:06:46 -07005037void AudioFlinger::DirectOutputThread::flushHw_l()
5038{
Phil Burk062e67a2015-02-11 13:40:50 -08005039 mOutput->flush();
Eric Laurentd1f69b02014-12-15 14:33:13 -08005040 mHwPaused = false;
Phil Burk43b4dcc2015-06-09 16:53:44 -07005041 mFlushPending = false;
Eric Laurente659ef42014-09-29 13:06:46 -07005042}
5043
Eric Laurent81784c32012-11-19 14:55:58 -08005044// ----------------------------------------------------------------------------
5045
Eric Laurentbfb1b832013-01-07 09:53:42 -08005046AudioFlinger::AsyncCallbackThread::AsyncCallbackThread(
Eric Laurent4de95592013-09-26 15:28:21 -07005047 const wp<AudioFlinger::PlaybackThread>& playbackThread)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005048 : Thread(false /*canCallJava*/),
Eric Laurent4de95592013-09-26 15:28:21 -07005049 mPlaybackThread(playbackThread),
Eric Laurent3b4529e2013-09-05 18:09:19 -07005050 mWriteAckSequence(0),
5051 mDrainSequence(0)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005052{
5053}
5054
5055AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread()
5056{
5057}
5058
5059void AudioFlinger::AsyncCallbackThread::onFirstRef()
5060{
5061 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO);
5062}
5063
5064bool AudioFlinger::AsyncCallbackThread::threadLoop()
5065{
5066 while (!exitPending()) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07005067 uint32_t writeAckSequence;
5068 uint32_t drainSequence;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005069
5070 {
5071 Mutex::Autolock _l(mLock);
Haynes Mathew George24a325d2013-12-03 21:26:02 -08005072 while (!((mWriteAckSequence & 1) ||
5073 (mDrainSequence & 1) ||
5074 exitPending())) {
5075 mWaitWorkCV.wait(mLock);
5076 }
5077
Eric Laurentbfb1b832013-01-07 09:53:42 -08005078 if (exitPending()) {
5079 break;
5080 }
Eric Laurent3b4529e2013-09-05 18:09:19 -07005081 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d",
5082 mWriteAckSequence, mDrainSequence);
5083 writeAckSequence = mWriteAckSequence;
5084 mWriteAckSequence &= ~1;
5085 drainSequence = mDrainSequence;
5086 mDrainSequence &= ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005087 }
5088 {
Eric Laurent4de95592013-09-26 15:28:21 -07005089 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote();
5090 if (playbackThread != 0) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07005091 if (writeAckSequence & 1) {
Eric Laurent4de95592013-09-26 15:28:21 -07005092 playbackThread->resetWriteBlocked(writeAckSequence >> 1);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005093 }
Eric Laurent3b4529e2013-09-05 18:09:19 -07005094 if (drainSequence & 1) {
Eric Laurent4de95592013-09-26 15:28:21 -07005095 playbackThread->resetDraining(drainSequence >> 1);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005096 }
5097 }
5098 }
5099 }
5100 return false;
5101}
5102
5103void AudioFlinger::AsyncCallbackThread::exit()
5104{
5105 ALOGV("AsyncCallbackThread::exit");
5106 Mutex::Autolock _l(mLock);
5107 requestExit();
5108 mWaitWorkCV.broadcast();
5109}
5110
Eric Laurent3b4529e2013-09-05 18:09:19 -07005111void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005112{
5113 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07005114 // bit 0 is cleared
5115 mWriteAckSequence = sequence << 1;
5116}
5117
5118void AudioFlinger::AsyncCallbackThread::resetWriteBlocked()
5119{
5120 Mutex::Autolock _l(mLock);
5121 // ignore unexpected callbacks
5122 if (mWriteAckSequence & 2) {
5123 mWriteAckSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005124 mWaitWorkCV.signal();
5125 }
5126}
5127
Eric Laurent3b4529e2013-09-05 18:09:19 -07005128void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005129{
5130 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07005131 // bit 0 is cleared
5132 mDrainSequence = sequence << 1;
5133}
5134
5135void AudioFlinger::AsyncCallbackThread::resetDraining()
5136{
5137 Mutex::Autolock _l(mLock);
5138 // ignore unexpected callbacks
5139 if (mDrainSequence & 2) {
5140 mDrainSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005141 mWaitWorkCV.signal();
5142 }
5143}
5144
5145
5146// ----------------------------------------------------------------------------
5147AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
Eric Laurente93cc032016-05-05 10:15:10 -07005148 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady)
5149 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady),
Eric Laurent64667972016-03-30 18:19:46 -07005150 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005151{
Eric Laurentfd477972013-10-25 18:10:40 -07005152 //FIXME: mStandby should be set to true by ThreadBase constructor
5153 mStandby = true;
Eric Laurent64667972016-03-30 18:19:46 -07005154 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005155}
5156
Eric Laurentbfb1b832013-01-07 09:53:42 -08005157void AudioFlinger::OffloadThread::threadLoop_exit()
5158{
5159 if (mFlushPending || mHwPaused) {
5160 // If a flush is pending or track was paused, just discard buffered data
5161 flushHw_l();
5162 } else {
5163 mMixerStatus = MIXER_DRAIN_ALL;
5164 threadLoop_drain();
5165 }
Uday Gupta56604aa2014-05-13 11:19:17 -07005166 if (mUseAsyncWrite) {
5167 ALOG_ASSERT(mCallbackThread != 0);
5168 mCallbackThread->exit();
5169 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005170 PlaybackThread::threadLoop_exit();
5171}
5172
5173AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l(
5174 Vector< sp<Track> > *tracksToRemove
5175)
5176{
Eric Laurentbfb1b832013-01-07 09:53:42 -08005177 size_t count = mActiveTracks.size();
5178
5179 mixer_state mixerStatus = MIXER_IDLE;
Eric Laurent972a1732013-09-04 09:42:59 -07005180 bool doHwPause = false;
5181 bool doHwResume = false;
5182
Glenn Kastenc42e9b42016-03-21 11:35:03 -07005183 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count);
Eric Laurentede6c3b2013-09-19 14:37:46 -07005184
Eric Laurentbfb1b832013-01-07 09:53:42 -08005185 // find out which tracks need to be processed
5186 for (size_t i = 0; i < count; i++) {
5187 sp<Track> t = mActiveTracks[i].promote();
5188 // The track died recently
5189 if (t == 0) {
5190 continue;
5191 }
5192 Track* const track = t.get();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07005193#ifdef VERY_VERY_VERBOSE_LOGGING
Eric Laurentbfb1b832013-01-07 09:53:42 -08005194 audio_track_cblk_t* cblk = track->cblk();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07005195#endif
Eric Laurentfd477972013-10-25 18:10:40 -07005196 // Only consider last track started for volume and mixer state control.
5197 // In theory an older track could underrun and restart after the new one starts
5198 // but as we only care about the transition phase between two tracks on a
5199 // direct output, it is not a problem to ignore the underrun case.
5200 sp<Track> l = mLatestActiveTrack.promote();
5201 bool last = l.get() == track;
5202
Haynes Mathew George7844f672014-01-15 12:32:55 -08005203 if (track->isInvalid()) {
5204 ALOGW("An invalidated track shouldn't be in active list");
5205 tracksToRemove->add(track);
5206 continue;
5207 }
5208
5209 if (track->mState == TrackBase::IDLE) {
5210 ALOGW("An idle track shouldn't be in active list");
5211 continue;
5212 }
5213
Eric Laurentbfb1b832013-01-07 09:53:42 -08005214 if (track->isPausing()) {
5215 track->setPaused();
5216 if (last) {
Eric Laurent5cff4032015-05-26 13:49:58 -07005217 if (mHwSupportsPause && !mHwPaused) {
Eric Laurent972a1732013-09-04 09:42:59 -07005218 doHwPause = true;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005219 mHwPaused = true;
5220 }
5221 // If we were part way through writing the mixbuffer to
5222 // the HAL we must save this until we resume
5223 // BUG - this will be wrong if a different track is made active,
5224 // in that case we want to discard the pending data in the
5225 // mixbuffer and tell the client to present it again when the
5226 // track is resumed
5227 mPausedWriteLength = mCurrentWriteLength;
5228 mPausedBytesRemaining = mBytesRemaining;
5229 mBytesRemaining = 0; // stop writing
5230 }
5231 tracksToRemove->add(track);
Haynes Mathew George7844f672014-01-15 12:32:55 -08005232 } else if (track->isFlushPending()) {
Eric Laurente93cc032016-05-05 10:15:10 -07005233 if (track->isStopping_1()) {
5234 track->mRetryCount = kMaxTrackStopRetriesOffload;
5235 } else {
5236 track->mRetryCount = kMaxTrackRetriesOffload;
5237 }
Haynes Mathew George7844f672014-01-15 12:32:55 -08005238 track->flushAck();
5239 if (last) {
5240 mFlushPending = true;
5241 }
Haynes Mathew George2d3ca682014-03-07 13:43:49 -08005242 } else if (track->isResumePending()){
5243 track->resumeAck();
5244 if (last) {
5245 if (mPausedBytesRemaining) {
5246 // Need to continue write that was interrupted
5247 mCurrentWriteLength = mPausedWriteLength;
5248 mBytesRemaining = mPausedBytesRemaining;
5249 mPausedBytesRemaining = 0;
5250 }
5251 if (mHwPaused) {
5252 doHwResume = true;
5253 mHwPaused = false;
5254 // threadLoop_mix() will handle the case that we need to
5255 // resume an interrupted write
5256 }
5257 // enable write to audio HAL
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005258 mSleepTimeUs = 0;
Haynes Mathew George2d3ca682014-03-07 13:43:49 -08005259
5260 // Do not handle new data in this iteration even if track->framesReady()
5261 mixerStatus = MIXER_TRACKS_ENABLED;
5262 }
5263 } else if (track->framesReady() && track->isReady() &&
Eric Laurent3b4529e2013-09-05 18:09:19 -07005264 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07005265 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005266 if (track->mFillingUpStatus == Track::FS_FILLED) {
5267 track->mFillingUpStatus = Track::FS_ACTIVE;
Eric Laurent1abbdb42013-09-13 17:00:08 -07005268 // make sure processVolume_l() will apply new volume even if 0
5269 mLeftVolFloat = mRightVolFloat = -1.0;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005270 }
5271
5272 if (last) {
Eric Laurentd7e59222013-11-15 12:02:28 -08005273 sp<Track> previousTrack = mPreviousTrack.promote();
5274 if (previousTrack != 0) {
5275 if (track != previousTrack.get()) {
Eric Laurent9da3d952013-11-12 19:25:43 -08005276 // Flush any data still being written from last track
5277 mBytesRemaining = 0;
5278 if (mPausedBytesRemaining) {
5279 // Last track was paused so we also need to flush saved
5280 // mixbuffer state and invalidate track so that it will
5281 // re-submit that unwritten data when it is next resumed
5282 mPausedBytesRemaining = 0;
5283 // Invalidate is a bit drastic - would be more efficient
5284 // to have a flag to tell client that some of the
5285 // previously written data was lost
Eric Laurentd7e59222013-11-15 12:02:28 -08005286 previousTrack->invalidate();
Eric Laurent9da3d952013-11-12 19:25:43 -08005287 }
5288 // flush data already sent to the DSP if changing audio session as audio
5289 // comes from a different source. Also invalidate previous track to force a
5290 // seek when resuming.
Eric Laurentd7e59222013-11-15 12:02:28 -08005291 if (previousTrack->sessionId() != track->sessionId()) {
5292 previousTrack->invalidate();
Eric Laurent9da3d952013-11-12 19:25:43 -08005293 }
5294 }
5295 }
5296 mPreviousTrack = track;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005297 // reset retry count
Eric Laurente93cc032016-05-05 10:15:10 -07005298 if (track->isStopping_1()) {
5299 track->mRetryCount = kMaxTrackStopRetriesOffload;
5300 } else {
5301 track->mRetryCount = kMaxTrackRetriesOffload;
5302 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005303 mActiveTrack = t;
5304 mixerStatus = MIXER_TRACKS_READY;
5305 }
5306 } else {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07005307 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005308 if (track->isStopping_1()) {
Eric Laurente93cc032016-05-05 10:15:10 -07005309 if (--(track->mRetryCount) <= 0) {
5310 // Hardware buffer can hold a large amount of audio so we must
5311 // wait for all current track's data to drain before we say
5312 // that the track is stopped.
5313 if (mBytesRemaining == 0) {
5314 // Only start draining when all data in mixbuffer
5315 // has been written
5316 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2");
5317 track->mState = TrackBase::STOPPING_2; // so presentation completes after
5318 // drain do not drain if no data was ever sent to HAL (mStandby == true)
5319 if (last && !mStandby) {
5320 // do not modify drain sequence if we are already draining. This happens
5321 // when resuming from pause after drain.
5322 if ((mDrainSequence & 1) == 0) {
5323 mSleepTimeUs = 0;
5324 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
5325 mixerStatus = MIXER_DRAIN_TRACK;
5326 mDrainSequence += 2;
5327 }
5328 if (mHwPaused) {
5329 // It is possible to move from PAUSED to STOPPING_1 without
5330 // a resume so we must ensure hardware is running
5331 doHwResume = true;
5332 mHwPaused = false;
5333 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005334 }
5335 }
Eric Laurente93cc032016-05-05 10:15:10 -07005336 } else if (last) {
5337 ALOGV("stopping1 underrun retries left %d", track->mRetryCount);
5338 mixerStatus = MIXER_TRACKS_ENABLED;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005339 }
5340 } else if (track->isStopping_2()) {
Eric Laurent6a51d7e2013-10-17 18:59:26 -07005341 // Drain has completed or we are in standby, signal presentation complete
5342 if (!(mDrainSequence & 1) || !last || mStandby) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08005343 track->mState = TrackBase::STOPPED;
5344 size_t audioHALFrames =
5345 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08005346 int64_t framesWritten =
Phil Burk062e67a2015-02-11 13:40:50 -08005347 mBytesWritten / mOutput->getFrameSize();
Eric Laurentbfb1b832013-01-07 09:53:42 -08005348 track->presentationComplete(framesWritten, audioHALFrames);
5349 track->reset();
5350 tracksToRemove->add(track);
5351 }
5352 } else {
5353 // No buffers for this track. Give it a few chances to
5354 // fill a buffer, then remove it from active list.
5355 if (--(track->mRetryCount) <= 0) {
5356 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list",
5357 track->name());
5358 tracksToRemove->add(track);
Eric Laurenta23f17a2013-11-05 18:22:08 -08005359 // indicate to client process that the track was disabled because of underrun;
5360 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08005361 track->disable();
Eric Laurentbfb1b832013-01-07 09:53:42 -08005362 } else if (last){
5363 mixerStatus = MIXER_TRACKS_ENABLED;
5364 }
5365 }
5366 }
5367 // compute volume for this track
5368 processVolume_l(track, last);
5369 }
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005370
Eric Laurentea0fade2013-10-04 16:23:48 -07005371 // make sure the pause/flush/resume sequence is executed in the right order.
5372 // If a flush is pending and a track is active but the HW is not paused, force a HW pause
5373 // before flush and then resume HW. This can happen in case of pause/flush/resume
5374 // if resume is received before pause is executed.
Eric Laurentfd477972013-10-25 18:10:40 -07005375 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
Eric Laurent972a1732013-09-04 09:42:59 -07005376 mOutput->stream->pause(mOutput->stream);
5377 }
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005378 if (mFlushPending) {
5379 flushHw_l();
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005380 }
Eric Laurentfd477972013-10-25 18:10:40 -07005381 if (!mStandby && doHwResume) {
Eric Laurent972a1732013-09-04 09:42:59 -07005382 mOutput->stream->resume(mOutput->stream);
5383 }
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005384
Eric Laurentbfb1b832013-01-07 09:53:42 -08005385 // remove all the tracks that need to be...
5386 removeTracks_l(*tracksToRemove);
5387
5388 return mixerStatus;
5389}
5390
Eric Laurentbfb1b832013-01-07 09:53:42 -08005391// must be called with thread mutex locked
5392bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
5393{
Eric Laurent3b4529e2013-09-05 18:09:19 -07005394 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d",
5395 mWriteAckSequence, mDrainSequence);
5396 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08005397 return true;
5398 }
5399 return false;
5400}
5401
Eric Laurentbfb1b832013-01-07 09:53:42 -08005402bool AudioFlinger::OffloadThread::waitingAsyncCallback()
5403{
5404 Mutex::Autolock _l(mLock);
5405 return waitingAsyncCallback_l();
5406}
5407
5408void AudioFlinger::OffloadThread::flushHw_l()
5409{
Eric Laurente659ef42014-09-29 13:06:46 -07005410 DirectOutputThread::flushHw_l();
Eric Laurentbfb1b832013-01-07 09:53:42 -08005411 // Flush anything still waiting in the mixbuffer
5412 mCurrentWriteLength = 0;
5413 mBytesRemaining = 0;
5414 mPausedWriteLength = 0;
5415 mPausedBytesRemaining = 0;
Eric Laurent3eaf66b2016-04-01 14:44:17 -07005416 // reset bytes written count to reflect that DSP buffers are empty after flush.
5417 mBytesWritten = 0;
Haynes Mathew George0f02f262014-01-11 13:03:57 -08005418
Eric Laurentbfb1b832013-01-07 09:53:42 -08005419 if (mUseAsyncWrite) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07005420 // discard any pending drain or write ack by incrementing sequence
5421 mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
5422 mDrainSequence = (mDrainSequence + 2) & ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005423 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07005424 mCallbackThread->setWriteBlocked(mWriteAckSequence);
5425 mCallbackThread->setDraining(mDrainSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005426 }
5427}
5428
Haynes Mathew George05317d22016-05-03 16:34:26 -07005429void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType)
5430{
5431 Mutex::Autolock _l(mLock);
5432 mFlushPending = true;
5433 PlaybackThread::invalidateTracks_l(streamType);
5434}
5435
Eric Laurentbfb1b832013-01-07 09:53:42 -08005436// ----------------------------------------------------------------------------
5437
Eric Laurent81784c32012-11-19 14:55:58 -08005438AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
Eric Laurent72e3f392015-05-20 14:43:50 -07005439 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady)
Eric Laurent81784c32012-11-19 14:55:58 -08005440 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
Eric Laurent72e3f392015-05-20 14:43:50 -07005441 systemReady, DUPLICATING),
Eric Laurent81784c32012-11-19 14:55:58 -08005442 mWaitTimeMs(UINT_MAX)
5443{
5444 addOutputTrack(mainThread);
5445}
5446
5447AudioFlinger::DuplicatingThread::~DuplicatingThread()
5448{
5449 for (size_t i = 0; i < mOutputTracks.size(); i++) {
5450 mOutputTracks[i]->destroy();
5451 }
5452}
5453
5454void AudioFlinger::DuplicatingThread::threadLoop_mix()
5455{
5456 // mix buffers...
5457 if (outputsReady(outputTracks)) {
Glenn Kastend79072e2016-01-06 08:41:20 -08005458 mAudioMixer->process();
Eric Laurent81784c32012-11-19 14:55:58 -08005459 } else {
Eric Laurent02b57082014-11-07 17:28:28 -08005460 if (mMixerBufferValid) {
5461 memset(mMixerBuffer, 0, mMixerBufferSize);
5462 } else {
5463 memset(mSinkBuffer, 0, mSinkBufferSize);
5464 }
Eric Laurent81784c32012-11-19 14:55:58 -08005465 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005466 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08005467 writeFrames = mNormalFrameCount;
Andy Hung25c2dac2014-02-27 14:56:00 -08005468 mCurrentWriteLength = mSinkBufferSize;
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005469 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
Eric Laurent81784c32012-11-19 14:55:58 -08005470}
5471
5472void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
5473{
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005474 if (mSleepTimeUs == 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08005475 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005476 mSleepTimeUs = mActiveSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005477 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005478 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005479 }
5480 } else if (mBytesWritten != 0) {
5481 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
5482 writeFrames = mNormalFrameCount;
Andy Hung25c2dac2014-02-27 14:56:00 -08005483 memset(mSinkBuffer, 0, mSinkBufferSize);
Eric Laurent81784c32012-11-19 14:55:58 -08005484 } else {
5485 // flush remaining overflow buffers in output tracks
5486 writeFrames = 0;
5487 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005488 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08005489 }
5490}
5491
Eric Laurentbfb1b832013-01-07 09:53:42 -08005492ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
Eric Laurent81784c32012-11-19 14:55:58 -08005493{
5494 for (size_t i = 0; i < outputTracks.size(); i++) {
Andy Hungc25b84a2015-01-14 19:04:10 -08005495 outputTracks[i]->write(mSinkBuffer, writeFrames);
Eric Laurent81784c32012-11-19 14:55:58 -08005496 }
Eric Laurent2c3740f2013-10-30 16:57:06 -07005497 mStandby = false;
Andy Hung25c2dac2014-02-27 14:56:00 -08005498 return (ssize_t)mSinkBufferSize;
Eric Laurent81784c32012-11-19 14:55:58 -08005499}
5500
5501void AudioFlinger::DuplicatingThread::threadLoop_standby()
5502{
5503 // DuplicatingThread implements standby by stopping all tracks
5504 for (size_t i = 0; i < outputTracks.size(); i++) {
5505 outputTracks[i]->stop();
5506 }
5507}
5508
5509void AudioFlinger::DuplicatingThread::saveOutputTracks()
5510{
5511 outputTracks = mOutputTracks;
5512}
5513
5514void AudioFlinger::DuplicatingThread::clearOutputTracks()
5515{
5516 outputTracks.clear();
5517}
5518
5519void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
5520{
5521 Mutex::Autolock _l(mLock);
Andy Hungc25b84a2015-01-14 19:04:10 -08005522 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
5523 // Adjust for thread->sampleRate() to determine minimum buffer frame count.
5524 // Then triple buffer because Threads do not run synchronously and may not be clock locked.
5525 const size_t frameCount =
5526 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate());
5527 // TODO: Consider asynchronous sample rate conversion to handle clock disparity
5528 // from different OutputTracks and their associated MixerThreads (e.g. one may
5529 // nearly empty and the other may be dropping data).
5530
5531 sp<OutputTrack> outputTrack = new OutputTrack(thread,
Eric Laurent81784c32012-11-19 14:55:58 -08005532 this,
5533 mSampleRate,
Andy Hungc25b84a2015-01-14 19:04:10 -08005534 mFormat,
Eric Laurent81784c32012-11-19 14:55:58 -08005535 mChannelMask,
Marco Nelissen462fd2f2013-01-14 14:12:05 -08005536 frameCount,
5537 IPCThreadState::self()->getCallingUid());
Eric Laurent81784c32012-11-19 14:55:58 -08005538 if (outputTrack->cblk() != NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -08005539 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
Eric Laurent81784c32012-11-19 14:55:58 -08005540 mOutputTracks.add(outputTrack);
Andy Hungc25b84a2015-01-14 19:04:10 -08005541 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
Eric Laurent81784c32012-11-19 14:55:58 -08005542 updateWaitTime_l();
5543 }
5544}
5545
5546void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
5547{
5548 Mutex::Autolock _l(mLock);
5549 for (size_t i = 0; i < mOutputTracks.size(); i++) {
5550 if (mOutputTracks[i]->thread() == thread) {
5551 mOutputTracks[i]->destroy();
5552 mOutputTracks.removeAt(i);
5553 updateWaitTime_l();
Eric Laurentf6870ae2015-05-08 10:50:03 -07005554 if (thread->getOutput() == mOutput) {
5555 mOutput = NULL;
5556 }
Eric Laurent81784c32012-11-19 14:55:58 -08005557 return;
5558 }
5559 }
Eric Laurentf6870ae2015-05-08 10:50:03 -07005560 ALOGV("removeOutputTrack(): unknown thread: %p", thread);
Eric Laurent81784c32012-11-19 14:55:58 -08005561}
5562
5563// caller must hold mLock
5564void AudioFlinger::DuplicatingThread::updateWaitTime_l()
5565{
5566 mWaitTimeMs = UINT_MAX;
5567 for (size_t i = 0; i < mOutputTracks.size(); i++) {
5568 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
5569 if (strong != 0) {
5570 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
5571 if (waitTimeMs < mWaitTimeMs) {
5572 mWaitTimeMs = waitTimeMs;
5573 }
5574 }
5575 }
5576}
5577
5578
5579bool AudioFlinger::DuplicatingThread::outputsReady(
5580 const SortedVector< sp<OutputTrack> > &outputTracks)
5581{
5582 for (size_t i = 0; i < outputTracks.size(); i++) {
5583 sp<ThreadBase> thread = outputTracks[i]->thread().promote();
5584 if (thread == 0) {
5585 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
5586 outputTracks[i].get());
5587 return false;
5588 }
5589 PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
5590 // see note at standby() declaration
5591 if (playbackThread->standby() && !playbackThread->isSuspended()) {
5592 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
5593 thread.get());
5594 return false;
5595 }
5596 }
5597 return true;
5598}
5599
5600uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
5601{
5602 return (mWaitTimeMs * 1000) / 2;
5603}
5604
5605void AudioFlinger::DuplicatingThread::cacheParameters_l()
5606{
5607 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
5608 updateWaitTime_l();
5609
5610 MixerThread::cacheParameters_l();
5611}
5612
5613// ----------------------------------------------------------------------------
5614// Record
5615// ----------------------------------------------------------------------------
5616
5617AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
5618 AudioStreamIn *input,
Eric Laurent81784c32012-11-19 14:55:58 -08005619 audio_io_handle_t id,
Eric Laurentd3922f72013-02-01 17:57:04 -08005620 audio_devices_t outDevice,
Eric Laurent72e3f392015-05-20 14:43:50 -07005621 audio_devices_t inDevice,
5622 bool systemReady
Glenn Kasten46909e72013-02-26 09:20:22 -08005623#ifdef TEE_SINK
5624 , const sp<NBAIO_Sink>& teeSink
5625#endif
5626 ) :
Eric Laurent72e3f392015-05-20 14:43:50 -07005627 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady),
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005628 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL),
Glenn Kastendeca2ae2014-02-07 10:25:56 -08005629 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l()
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08005630 mRsmpInRear(0)
Glenn Kasten46909e72013-02-26 09:20:22 -08005631#ifdef TEE_SINK
5632 , mTeeSink(teeSink)
5633#endif
Glenn Kastenb880f5e2014-05-07 08:43:45 -07005634 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize,
5635 "RecordThreadRO", MemoryHeapBase::READ_ONLY))
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005636 // mFastCapture below
5637 , mFastCaptureFutex(0)
5638 // mInputSource
5639 // mPipeSink
5640 // mPipeSource
5641 , mPipeFramesP2(0)
5642 // mPipeMemory
5643 // mFastCaptureNBLogWriter
Glenn Kasten6e6704c2014-07-03 10:20:00 -07005644 , mFastTrackAvail(false)
Eric Laurent81784c32012-11-19 14:55:58 -08005645{
Glenn Kastend7dca052015-03-05 16:05:54 -08005646 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id);
5647 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
Eric Laurent81784c32012-11-19 14:55:58 -08005648
Glenn Kastendeca2ae2014-02-07 10:25:56 -08005649 readInputParameters_l();
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005650
5651 // create an NBAIO source for the HAL input stream, and negotiate
5652 mInputSource = new AudioStreamInSource(input->stream);
5653 size_t numCounterOffers = 0;
5654 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07005655#if !LOG_NDEBUG
5656 ssize_t index =
5657#else
5658 (void)
5659#endif
5660 mInputSource->negotiate(offers, 1, NULL, numCounterOffers);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005661 ALOG_ASSERT(index == 0);
5662
5663 // initialize fast capture depending on configuration
5664 bool initFastCapture;
5665 switch (kUseFastCapture) {
5666 case FastCapture_Never:
5667 initFastCapture = false;
5668 break;
5669 case FastCapture_Always:
5670 initFastCapture = true;
5671 break;
5672 case FastCapture_Static:
Glenn Kasteneb9487e2015-07-22 09:15:17 -07005673 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005674 break;
5675 // case FastCapture_Dynamic:
5676 }
5677
5678 if (initFastCapture) {
Glenn Kastend198b852015-03-16 14:55:53 -07005679 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005680 NBAIO_Format format = mInputSource->format();
Glenn Kasten49d00ad2014-07-21 11:22:03 -07005681 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005682 size_t pipeSize = pipeFramesP2 * Format_frameSize(format);
5683 void *pipeBuffer;
5684 const sp<MemoryDealer> roHeap(readOnlyHeap());
5685 sp<IMemory> pipeMemory;
5686 if ((roHeap == 0) ||
5687 (pipeMemory = roHeap->allocate(pipeSize)) == 0 ||
5688 (pipeBuffer = pipeMemory->pointer()) == NULL) {
5689 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize);
5690 goto failed;
5691 }
5692 // pipe will be shared directly with fast clients, so clear to avoid leaking old information
5693 memset(pipeBuffer, 0, pipeSize);
5694 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer);
5695 const NBAIO_Format offers[1] = {format};
5696 size_t numCounterOffers = 0;
5697 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
5698 ALOG_ASSERT(index == 0);
5699 mPipeSink = pipe;
5700 PipeReader *pipeReader = new PipeReader(*pipe);
5701 numCounterOffers = 0;
5702 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
5703 ALOG_ASSERT(index == 0);
5704 mPipeSource = pipeReader;
5705 mPipeFramesP2 = pipeFramesP2;
5706 mPipeMemory = pipeMemory;
5707
5708 // create fast capture
5709 mFastCapture = new FastCapture();
5710 FastCaptureStateQueue *sq = mFastCapture->sq();
5711#ifdef STATE_QUEUE_DUMP
5712 // FIXME
5713#endif
5714 FastCaptureState *state = sq->begin();
5715 state->mCblk = NULL;
5716 state->mInputSource = mInputSource.get();
5717 state->mInputSourceGen++;
5718 state->mPipeSink = pipe;
5719 state->mPipeSinkGen++;
5720 state->mFrameCount = mFrameCount;
5721 state->mCommand = FastCaptureState::COLD_IDLE;
5722 // already done in constructor initialization list
5723 //mFastCaptureFutex = 0;
5724 state->mColdFutexAddr = &mFastCaptureFutex;
5725 state->mColdGen++;
5726 state->mDumpState = &mFastCaptureDumpState;
5727#ifdef TEE_SINK
5728 // FIXME
5729#endif
5730 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture");
5731 state->mNBLogWriter = mFastCaptureNBLogWriter.get();
5732 sq->end();
5733 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
5734
5735 // start the fast capture
5736 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
5737 pid_t tid = mFastCapture->getTid();
Glenn Kasten8379b722016-03-18 14:54:17 -07005738 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005739#ifdef AUDIO_WATCHDOG
5740 // FIXME
5741#endif
5742
Glenn Kasten6e6704c2014-07-03 10:20:00 -07005743 mFastTrackAvail = true;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005744 }
5745failed: ;
5746
5747 // FIXME mNormalSource
Eric Laurent81784c32012-11-19 14:55:58 -08005748}
5749
Eric Laurent81784c32012-11-19 14:55:58 -08005750AudioFlinger::RecordThread::~RecordThread()
5751{
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005752 if (mFastCapture != 0) {
5753 FastCaptureStateQueue *sq = mFastCapture->sq();
5754 FastCaptureState *state = sq->begin();
5755 if (state->mCommand == FastCaptureState::COLD_IDLE) {
5756 int32_t old = android_atomic_inc(&mFastCaptureFutex);
5757 if (old == -1) {
5758 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
5759 }
5760 }
5761 state->mCommand = FastCaptureState::EXIT;
5762 sq->end();
5763 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
5764 mFastCapture->join();
5765 mFastCapture.clear();
5766 }
5767 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter);
Glenn Kasten481fb672013-09-30 14:39:28 -07005768 mAudioFlinger->unregisterWriter(mNBLogWriter);
Andy Hung57446612015-04-19 23:56:46 -07005769 free(mRsmpInBuffer);
Eric Laurent81784c32012-11-19 14:55:58 -08005770}
5771
5772void AudioFlinger::RecordThread::onFirstRef()
5773{
Glenn Kastend7dca052015-03-05 16:05:54 -08005774 run(mThreadName, PRIORITY_URGENT_AUDIO);
Eric Laurent81784c32012-11-19 14:55:58 -08005775}
5776
Eric Laurent81784c32012-11-19 14:55:58 -08005777bool AudioFlinger::RecordThread::threadLoop()
5778{
Eric Laurent81784c32012-11-19 14:55:58 -08005779 nsecs_t lastWarning = 0;
5780
5781 inputStandBy();
Eric Laurent81784c32012-11-19 14:55:58 -08005782
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005783reacquire_wakelock:
5784 sp<RecordTrack> activeTrack;
Glenn Kasten2b806402013-11-20 16:37:38 -08005785 int activeTracksGen;
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005786 {
5787 Mutex::Autolock _l(mLock);
Glenn Kasten2b806402013-11-20 16:37:38 -08005788 size_t size = mActiveTracks.size();
5789 activeTracksGen = mActiveTracksGen;
5790 if (size > 0) {
5791 // FIXME an arbitrary choice
5792 activeTrack = mActiveTracks[0];
5793 acquireWakeLock_l(activeTrack->uid());
5794 if (size > 1) {
5795 SortedVector<int> tmp;
5796 for (size_t i = 0; i < size; i++) {
5797 tmp.add(mActiveTracks[i]->uid());
5798 }
5799 updateWakeLockUids_l(tmp);
5800 }
5801 } else {
5802 acquireWakeLock_l(-1);
5803 }
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005804 }
5805
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005806 // used to request a deferred sleep, to be executed later while mutex is unlocked
5807 uint32_t sleepUs = 0;
5808
5809 // loop while there is work to do
Glenn Kasten4ef0b462013-08-14 13:52:27 -07005810 for (;;) {
Glenn Kastenc527a7c2013-08-13 15:43:49 -07005811 Vector< sp<EffectChain> > effectChains;
Glenn Kasten2cfbf882013-08-14 13:12:11 -07005812
Glenn Kasten5edadd42013-08-14 16:30:49 -07005813 // sleep with mutex unlocked
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005814 if (sleepUs > 0) {
Glenn Kastene7754022014-10-31 12:11:26 -07005815 ATRACE_BEGIN("sleep");
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005816 usleep(sleepUs);
Glenn Kastene7754022014-10-31 12:11:26 -07005817 ATRACE_END();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005818 sleepUs = 0;
Glenn Kasten5edadd42013-08-14 16:30:49 -07005819 }
5820
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005821 // activeTracks accumulates a copy of a subset of mActiveTracks
5822 Vector< sp<RecordTrack> > activeTracks;
5823
Glenn Kasten735f45f2014-08-18 15:51:59 -07005824 // reference to the (first and only) active fast track
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005825 sp<RecordTrack> fastTrack;
Eric Laurent10351942014-05-08 18:49:52 -07005826
Glenn Kasten735f45f2014-08-18 15:51:59 -07005827 // reference to a fast track which is about to be removed
5828 sp<RecordTrack> fastTrackToRemove;
5829
Eric Laurent81784c32012-11-19 14:55:58 -08005830 { // scope for mLock
5831 Mutex::Autolock _l(mLock);
Eric Laurent000a4192014-01-29 15:17:32 -08005832
Eric Laurent021cf962014-05-13 10:18:14 -07005833 processConfigEvents_l();
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005834
Eric Laurent000a4192014-01-29 15:17:32 -08005835 // check exitPending here because checkForNewParameters_l() and
5836 // checkForNewParameters_l() can temporarily release mLock
5837 if (exitPending()) {
5838 break;
5839 }
5840
Glenn Kasten2b806402013-11-20 16:37:38 -08005841 // if no active track(s), then standby and release wakelock
5842 size_t size = mActiveTracks.size();
5843 if (size == 0) {
Glenn Kasten93e471f2013-08-19 08:40:07 -07005844 standbyIfNotAlreadyInStandby();
Glenn Kasten4ef0b462013-08-14 13:52:27 -07005845 // exitPending() can't become true here
Eric Laurent81784c32012-11-19 14:55:58 -08005846 releaseWakeLock_l();
5847 ALOGV("RecordThread: loop stopping");
5848 // go to sleep
5849 mWaitWorkCV.wait(mLock);
5850 ALOGV("RecordThread: loop starting");
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005851 goto reacquire_wakelock;
5852 }
5853
Glenn Kasten2b806402013-11-20 16:37:38 -08005854 if (mActiveTracksGen != activeTracksGen) {
5855 activeTracksGen = mActiveTracksGen;
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005856 SortedVector<int> tmp;
Glenn Kasten2b806402013-11-20 16:37:38 -08005857 for (size_t i = 0; i < size; i++) {
5858 tmp.add(mActiveTracks[i]->uid());
5859 }
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005860 updateWakeLockUids_l(tmp);
Eric Laurent81784c32012-11-19 14:55:58 -08005861 }
Glenn Kasten9e982352013-08-14 14:39:50 -07005862
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005863 bool doBroadcast = false;
5864 for (size_t i = 0; i < size; ) {
Glenn Kasten9e982352013-08-14 14:39:50 -07005865
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005866 activeTrack = mActiveTracks[i];
5867 if (activeTrack->isTerminated()) {
Glenn Kasten735f45f2014-08-18 15:51:59 -07005868 if (activeTrack->isFastTrack()) {
5869 ALOG_ASSERT(fastTrackToRemove == 0);
5870 fastTrackToRemove = activeTrack;
5871 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005872 removeTrack_l(activeTrack);
Glenn Kasten2b806402013-11-20 16:37:38 -08005873 mActiveTracks.remove(activeTrack);
5874 mActiveTracksGen++;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005875 size--;
Glenn Kasten9e982352013-08-14 14:39:50 -07005876 continue;
5877 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005878
5879 TrackBase::track_state activeTrackState = activeTrack->mState;
5880 switch (activeTrackState) {
5881
5882 case TrackBase::PAUSING:
5883 mActiveTracks.remove(activeTrack);
5884 mActiveTracksGen++;
5885 doBroadcast = true;
5886 size--;
5887 continue;
5888
5889 case TrackBase::STARTING_1:
5890 sleepUs = 10000;
5891 i++;
5892 continue;
5893
5894 case TrackBase::STARTING_2:
5895 doBroadcast = true;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005896 mStandby = false;
Glenn Kasten9e982352013-08-14 14:39:50 -07005897 activeTrack->mState = TrackBase::ACTIVE;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005898 break;
5899
5900 case TrackBase::ACTIVE:
5901 break;
5902
5903 case TrackBase::IDLE:
5904 i++;
5905 continue;
5906
5907 default:
Glenn Kastenadad3d72014-02-21 14:51:43 -08005908 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState);
Glenn Kasten9e982352013-08-14 14:39:50 -07005909 }
Glenn Kasten9e982352013-08-14 14:39:50 -07005910
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005911 activeTracks.add(activeTrack);
5912 i++;
Glenn Kasten9e982352013-08-14 14:39:50 -07005913
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005914 if (activeTrack->isFastTrack()) {
5915 ALOG_ASSERT(!mFastTrackAvail);
5916 ALOG_ASSERT(fastTrack == 0);
5917 fastTrack = activeTrack;
5918 }
Glenn Kasten9e982352013-08-14 14:39:50 -07005919 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005920 if (doBroadcast) {
5921 mStartStopCond.broadcast();
5922 }
5923
5924 // sleep if there are no active tracks to process
5925 if (activeTracks.size() == 0) {
5926 if (sleepUs == 0) {
5927 sleepUs = kRecordThreadSleepUs;
5928 }
5929 continue;
5930 }
5931 sleepUs = 0;
Glenn Kasten9e982352013-08-14 14:39:50 -07005932
Eric Laurent81784c32012-11-19 14:55:58 -08005933 lockEffectChains_l(effectChains);
5934 }
5935
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005936 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0
Glenn Kasten71652682013-08-14 15:17:55 -07005937
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005938 size_t size = effectChains.size();
5939 for (size_t i = 0; i < size; i++) {
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07005940 // thread mutex is not locked, but effect chain is locked
5941 effectChains[i]->process_l();
5942 }
5943
Glenn Kasten735f45f2014-08-18 15:51:59 -07005944 // Push a new fast capture state if fast capture is not already running, or cblk change
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005945 if (mFastCapture != 0) {
5946 FastCaptureStateQueue *sq = mFastCapture->sq();
5947 FastCaptureState *state = sq->begin();
Glenn Kasten735f45f2014-08-18 15:51:59 -07005948 bool didModify = false;
5949 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005950 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME &&
5951 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) {
5952 if (state->mCommand == FastCaptureState::COLD_IDLE) {
5953 int32_t old = android_atomic_inc(&mFastCaptureFutex);
5954 if (old == -1) {
5955 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
5956 }
5957 }
5958 state->mCommand = FastCaptureState::READ_WRITE;
5959#if 0 // FIXME
5960 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
Glenn Kastenfbdb2ac2015-03-02 14:47:19 -08005961 FastThreadDumpState::kSamplingNforLowRamDevice :
5962 FastThreadDumpState::kSamplingN);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005963#endif
Glenn Kasten735f45f2014-08-18 15:51:59 -07005964 didModify = true;
5965 }
5966 audio_track_cblk_t *cblkOld = state->mCblk;
5967 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL;
5968 if (cblkNew != cblkOld) {
5969 state->mCblk = cblkNew;
5970 // block until acked if removing a fast track
5971 if (cblkOld != NULL) {
5972 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED;
5973 }
5974 didModify = true;
5975 }
5976 sq->end(didModify);
5977 if (didModify) {
5978 sq->push(block);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005979#if 0
5980 if (kUseFastCapture == FastCapture_Dynamic) {
5981 mNormalSource = mPipeSource;
5982 }
5983#endif
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005984 }
5985 }
5986
Glenn Kasten735f45f2014-08-18 15:51:59 -07005987 // now run the fast track destructor with thread mutex unlocked
5988 fastTrackToRemove.clear();
5989
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005990 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
5991 // Only the client(s) that are too slow will overrun. But if even the fastest client is too
5992 // slow, then this RecordThread will overrun by not calling HAL read often enough.
5993 // If destination is non-contiguous, first read past the nominal end of buffer, then
5994 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated.
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07005995
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005996 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005997 ssize_t framesRead;
5998
5999 // If an NBAIO source is present, use it to read the normal capture's data
6000 if (mPipeSource != 0) {
6001 size_t framesToRead = mBufferSize / mFrameSize;
Andy Hung57446612015-04-19 23:56:46 -07006002 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
Glenn Kastend79072e2016-01-06 08:41:20 -08006003 framesToRead);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006004 if (framesRead == 0) {
6005 // since pipe is non-blocking, simulate blocking input
6006 sleepUs = (framesToRead * 1000000LL) / mSampleRate;
6007 }
6008 // otherwise use the HAL / AudioStreamIn directly
6009 } else {
Glenn Kastenec6a7032016-03-14 07:40:23 -07006010 ATRACE_BEGIN("read");
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006011 ssize_t bytesRead = mInput->stream->read(mInput->stream,
Andy Hung57446612015-04-19 23:56:46 -07006012 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize);
Glenn Kastenec6a7032016-03-14 07:40:23 -07006013 ATRACE_END();
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006014 if (bytesRead < 0) {
6015 framesRead = bytesRead;
6016 } else {
6017 framesRead = bytesRead / mFrameSize;
6018 }
6019 }
6020
Andy Hung3f0c9022016-01-15 17:49:46 -08006021 // Update server timestamp with server stats
6022 // systemTime() is optional if the hardware supports timestamps.
6023 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead;
6024 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime();
6025
6026 // Update server timestamp with kernel stats
6027 if (mInput->stream->get_capture_position != nullptr) {
6028 int64_t position, time;
6029 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time);
6030 if (ret == NO_ERROR) {
6031 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position;
6032 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time;
6033 // Note: In general record buffers should tend to be empty in
6034 // a properly running pipeline.
6035 //
6036 // Also, it is not advantageous to call get_presentation_position during the read
6037 // as the read obtains a lock, preventing the timestamp call from executing.
6038 }
6039 }
6040 // Use this to track timestamp information
6041 // ALOGD("%s", mTimestamp.toString().c_str());
6042
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006043 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006044 ALOGE("read failed: framesRead=%zd", framesRead);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006045 // Force input into standby so that it tries to recover at next read attempt
6046 inputStandBy();
6047 sleepUs = kRecordThreadSleepUs;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006048 }
6049 if (framesRead <= 0) {
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006050 goto unlock;
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006051 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006052 ALOG_ASSERT(framesRead > 0);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006053
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006054 if (mTeeSink != 0) {
Andy Hung57446612015-04-19 23:56:46 -07006055 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006056 }
6057 // If destination is non-contiguous, we now correct for reading past end of buffer.
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006058 {
6059 size_t part1 = mRsmpInFramesP2 - rear;
6060 if ((size_t) framesRead > part1) {
Andy Hung57446612015-04-19 23:56:46 -07006061 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize,
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006062 (framesRead - part1) * mFrameSize);
6063 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006064 }
6065 rear = mRsmpInRear += framesRead;
6066
6067 size = activeTracks.size();
6068 // loop over each active track
6069 for (size_t i = 0; i < size; i++) {
6070 activeTrack = activeTracks[i];
6071
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006072 // skip fast tracks, as those are handled directly by FastCapture
6073 if (activeTrack->isFastTrack()) {
6074 continue;
6075 }
6076
Andy Hung73c02e42015-03-29 01:13:58 -07006077 // TODO: This code probably should be moved to RecordTrack.
Andy Hung97a893e2015-03-29 01:03:07 -07006078 // TODO: Update the activeTrack buffer converter in case of reconfigure.
6079
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006080 enum {
6081 OVERRUN_UNKNOWN,
6082 OVERRUN_TRUE,
6083 OVERRUN_FALSE
6084 } overrun = OVERRUN_UNKNOWN;
6085
6086 // loop over getNextBuffer to handle circular sink
6087 for (;;) {
6088
6089 activeTrack->mSink.frameCount = ~0;
6090 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink);
6091 size_t framesOut = activeTrack->mSink.frameCount;
6092 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
6093
Andy Hung73c02e42015-03-29 01:13:58 -07006094 // check available frames and handle overrun conditions
6095 // if the record track isn't draining fast enough.
6096 bool hasOverrun;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006097 size_t framesIn;
Andy Hung73c02e42015-03-29 01:13:58 -07006098 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun);
6099 if (hasOverrun) {
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006100 overrun = OVERRUN_TRUE;
6101 }
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08006102 if (framesOut == 0 || framesIn == 0) {
6103 break;
6104 }
6105
Andy Hung6770c6f2015-04-07 13:43:36 -07006106 // Don't allow framesOut to be larger than what is possible with resampling
6107 // from framesIn.
6108 // This isn't strictly necessary but helps limit buffer resizing in
6109 // RecordBufferConverter. TODO: remove when no longer needed.
6110 framesOut = min(framesOut,
6111 destinationFramesPossible(
6112 framesIn, mSampleRate, activeTrack->mSampleRate));
Andy Hung97a893e2015-03-29 01:03:07 -07006113 // process frames from the RecordThread buffer provider to the RecordTrack buffer
6114 framesOut = activeTrack->mRecordBufferConverter->convert(
6115 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006116
6117 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
6118 overrun = OVERRUN_FALSE;
6119 }
6120
6121 if (activeTrack->mFramesToDrop == 0) {
6122 if (framesOut > 0) {
6123 activeTrack->mSink.frameCount = framesOut;
6124 activeTrack->releaseBuffer(&activeTrack->mSink);
6125 }
6126 } else {
6127 // FIXME could do a partial drop of framesOut
6128 if (activeTrack->mFramesToDrop > 0) {
6129 activeTrack->mFramesToDrop -= framesOut;
6130 if (activeTrack->mFramesToDrop <= 0) {
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006131 activeTrack->clearSyncStartEvent();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006132 }
6133 } else {
6134 activeTrack->mFramesToDrop += framesOut;
6135 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 ||
6136 activeTrack->mSyncStartEvent->isCancelled()) {
6137 ALOGW("Synced record %s, session %d, trigger session %d",
6138 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled",
6139 activeTrack->sessionId(),
6140 (activeTrack->mSyncStartEvent != 0) ?
Glenn Kastend848eb42016-03-08 13:42:11 -08006141 activeTrack->mSyncStartEvent->triggerSession() :
6142 AUDIO_SESSION_NONE);
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006143 activeTrack->clearSyncStartEvent();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006144 }
6145 }
6146 }
6147
6148 if (framesOut == 0) {
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006149 break;
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006150 }
6151 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006152
6153 switch (overrun) {
6154 case OVERRUN_TRUE:
6155 // client isn't retrieving buffers fast enough
6156 if (!activeTrack->setOverflow()) {
6157 nsecs_t now = systemTime();
6158 // FIXME should lastWarning per track?
6159 if ((now - lastWarning) > kWarningThrottleNs) {
6160 ALOGW("RecordThread: buffer overflow");
6161 lastWarning = now;
6162 }
6163 }
6164 break;
6165 case OVERRUN_FALSE:
6166 activeTrack->clearOverflow();
6167 break;
6168 case OVERRUN_UNKNOWN:
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006169 break;
6170 }
6171
Andy Hung3f0c9022016-01-15 17:49:46 -08006172 // update frame information and push timestamp out
6173 activeTrack->updateTrackFrameInfo(
Andy Hung6ae58432016-02-16 18:32:24 -08006174 activeTrack->mServerProxy->framesReleased(),
Andy Hung3f0c9022016-01-15 17:49:46 -08006175 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER],
6176 mSampleRate, mTimestamp);
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006177 }
6178
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006179unlock:
Eric Laurent81784c32012-11-19 14:55:58 -08006180 // enable changes in effect chain
6181 unlockEffectChains(effectChains);
Glenn Kastenc527a7c2013-08-13 15:43:49 -07006182 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end
Eric Laurent81784c32012-11-19 14:55:58 -08006183 }
6184
Glenn Kasten93e471f2013-08-19 08:40:07 -07006185 standbyIfNotAlreadyInStandby();
Eric Laurent81784c32012-11-19 14:55:58 -08006186
6187 {
6188 Mutex::Autolock _l(mLock);
Eric Laurent9a54bc22013-09-09 09:08:44 -07006189 for (size_t i = 0; i < mTracks.size(); i++) {
6190 sp<RecordTrack> track = mTracks[i];
6191 track->invalidate();
6192 }
Glenn Kasten2b806402013-11-20 16:37:38 -08006193 mActiveTracks.clear();
6194 mActiveTracksGen++;
Eric Laurent81784c32012-11-19 14:55:58 -08006195 mStartStopCond.broadcast();
6196 }
6197
6198 releaseWakeLock();
6199
6200 ALOGV("RecordThread %p exiting", this);
6201 return false;
6202}
6203
Glenn Kasten93e471f2013-08-19 08:40:07 -07006204void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
Eric Laurent81784c32012-11-19 14:55:58 -08006205{
6206 if (!mStandby) {
6207 inputStandBy();
6208 mStandby = true;
6209 }
6210}
6211
6212void AudioFlinger::RecordThread::inputStandBy()
6213{
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006214 // Idle the fast capture if it's currently running
6215 if (mFastCapture != 0) {
6216 FastCaptureStateQueue *sq = mFastCapture->sq();
6217 FastCaptureState *state = sq->begin();
6218 if (!(state->mCommand & FastCaptureState::IDLE)) {
6219 state->mCommand = FastCaptureState::COLD_IDLE;
6220 state->mColdFutexAddr = &mFastCaptureFutex;
6221 state->mColdGen++;
6222 mFastCaptureFutex = 0;
6223 sq->end();
6224 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
6225 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED);
6226#if 0
6227 if (kUseFastCapture == FastCapture_Dynamic) {
6228 // FIXME
6229 }
6230#endif
6231#ifdef AUDIO_WATCHDOG
6232 // FIXME
6233#endif
6234 } else {
6235 sq->end(false /*didModify*/);
6236 }
6237 }
Eric Laurent81784c32012-11-19 14:55:58 -08006238 mInput->stream->common.standby(&mInput->stream->common);
6239}
6240
Glenn Kasten05997e22014-03-13 15:08:33 -07006241// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
Glenn Kastene198c362013-08-13 09:13:36 -07006242sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
Eric Laurent81784c32012-11-19 14:55:58 -08006243 const sp<AudioFlinger::Client>& client,
6244 uint32_t sampleRate,
6245 audio_format_t format,
6246 audio_channel_mask_t channelMask,
Glenn Kasten74935e42013-12-19 08:56:45 -08006247 size_t *pFrameCount,
Glenn Kastend848eb42016-03-08 13:42:11 -08006248 audio_session_t sessionId,
Glenn Kasten7df8c0b2014-07-03 12:23:29 -07006249 size_t *notificationFrames,
Marco Nelissen462fd2f2013-01-14 14:12:05 -08006250 int uid,
Glenn Kastenddb0ccf2013-07-31 16:14:50 -07006251 IAudioFlinger::track_flags_t *flags,
Eric Laurent81784c32012-11-19 14:55:58 -08006252 pid_t tid,
6253 status_t *status)
6254{
Glenn Kasten74935e42013-12-19 08:56:45 -08006255 size_t frameCount = *pFrameCount;
Eric Laurent81784c32012-11-19 14:55:58 -08006256 sp<RecordTrack> track;
6257 status_t lStatus;
6258
Glenn Kasten90e58b12013-07-31 16:16:02 -07006259 // client expresses a preference for FAST, but we get the final say
6260 if (*flags & IAudioFlinger::TRACK_FAST) {
6261 if (
Glenn Kastenb7fbf7e2015-03-18 12:57:28 -07006262 // we formerly checked for a callback handler (non-0 tid),
6263 // but that is no longer required for TRANSFER_OBTAIN mode
6264 //
Glenn Kasten74105912014-07-03 12:28:53 -07006265 // frame count is not specified, or is exactly the pipe depth
6266 ((frameCount == 0) || (frameCount == mPipeFramesP2)) &&
Glenn Kasten3a6c90a2014-03-13 15:07:51 -07006267 // PCM data
6268 audio_is_linear_pcm(format) &&
Glenn Kasten7fd04222016-02-02 12:38:16 -08006269 // hardware format
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006270 (format == mFormat) &&
Glenn Kasten7fd04222016-02-02 12:38:16 -08006271 // hardware channel mask
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006272 (channelMask == mChannelMask) &&
Glenn Kasten7fd04222016-02-02 12:38:16 -08006273 // hardware sample rate
Glenn Kasten90e58b12013-07-31 16:16:02 -07006274 (sampleRate == mSampleRate) &&
Glenn Kasten3a6c90a2014-03-13 15:07:51 -07006275 // record thread has an associated fast capture
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006276 hasFastCapture() &&
6277 // there are sufficient fast track slots available
6278 mFastTrackAvail
Glenn Kasten90e58b12013-07-31 16:16:02 -07006279 ) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006280 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
Glenn Kasten90e58b12013-07-31 16:16:02 -07006281 frameCount, mFrameCount);
6282 } else {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006283 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu "
Glenn Kasten74105912014-07-03 12:28:53 -07006284 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006285 "hasFastCapture=%d tid=%d mFastTrackAvail=%d",
Glenn Kasten74105912014-07-03 12:28:53 -07006286 frameCount, mFrameCount, mPipeFramesP2,
6287 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate,
6288 hasFastCapture(), tid, mFastTrackAvail);
Glenn Kasten90e58b12013-07-31 16:16:02 -07006289 *flags &= ~IAudioFlinger::TRACK_FAST;
Glenn Kasten74105912014-07-03 12:28:53 -07006290 }
6291 }
6292
6293 // compute track buffer size in frames, and suggest the notification frame count
6294 if (*flags & IAudioFlinger::TRACK_FAST) {
6295 // fast track: frame count is exactly the pipe depth
6296 frameCount = mPipeFramesP2;
6297 // ignore requested notificationFrames, and always notify exactly once every HAL buffer
6298 *notificationFrames = mFrameCount;
6299 } else {
Glenn Kasten49d00ad2014-07-21 11:22:03 -07006300 // not fast track: max notification period is resampled equivalent of one HAL buffer time
6301 // or 20 ms if there is a fast capture
6302 // TODO This could be a roundupRatio inline, and const
6303 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount)
6304 * sampleRate + mSampleRate - 1) / mSampleRate;
6305 // minimum number of notification periods is at least kMinNotifications,
6306 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs)
6307 static const size_t kMinNotifications = 3;
6308 static const uint32_t kMinMs = 30;
6309 // TODO This could be a roundupRatio inline
6310 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000;
6311 // TODO This could be a roundupRatio inline
6312 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) /
6313 maxNotificationFrames;
6314 const size_t minFrameCount = maxNotificationFrames *
6315 max(kMinNotifications, minNotificationsByMs);
6316 frameCount = max(frameCount, minFrameCount);
6317 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) {
6318 *notificationFrames = maxNotificationFrames;
Glenn Kasten74105912014-07-03 12:28:53 -07006319 }
Glenn Kasten90e58b12013-07-31 16:16:02 -07006320 }
Glenn Kasten74935e42013-12-19 08:56:45 -08006321 *pFrameCount = frameCount;
Glenn Kasten90e58b12013-07-31 16:16:02 -07006322
Glenn Kasten15e57982013-09-24 11:52:37 -07006323 lStatus = initCheck();
6324 if (lStatus != NO_ERROR) {
6325 ALOGE("createRecordTrack_l() audio driver not initialized");
6326 goto Exit;
6327 }
Eric Laurent81784c32012-11-19 14:55:58 -08006328
6329 { // scope for mLock
6330 Mutex::Autolock _l(mLock);
6331
6332 track = new RecordTrack(this, client, sampleRate,
Eric Laurent83b88082014-06-20 18:31:16 -07006333 format, channelMask, frameCount, NULL, sessionId, uid,
6334 *flags, TrackBase::TYPE_DEFAULT);
Eric Laurent81784c32012-11-19 14:55:58 -08006335
Glenn Kasten03003332013-08-06 15:40:54 -07006336 lStatus = track->initCheck();
6337 if (lStatus != NO_ERROR) {
Glenn Kasten35295072013-10-07 09:27:06 -07006338 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
Haynes Mathew George03e9e832013-12-13 15:40:13 -08006339 // track must be cleared from the caller as the caller has the AF lock
Eric Laurent81784c32012-11-19 14:55:58 -08006340 goto Exit;
6341 }
6342 mTracks.add(track);
6343
6344 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
6345 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
6346 mAudioFlinger->btNrecIsOff();
6347 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
6348 setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
Glenn Kasten90e58b12013-07-31 16:16:02 -07006349
6350 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
6351 pid_t callingPid = IPCThreadState::self()->getCallingPid();
6352 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
6353 // so ask activity manager to do this on our behalf
6354 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
6355 }
Eric Laurent81784c32012-11-19 14:55:58 -08006356 }
Glenn Kasten05997e22014-03-13 15:08:33 -07006357
Eric Laurent81784c32012-11-19 14:55:58 -08006358 lStatus = NO_ERROR;
6359
6360Exit:
Glenn Kasten9156ef32013-08-06 15:39:08 -07006361 *status = lStatus;
Eric Laurent81784c32012-11-19 14:55:58 -08006362 return track;
6363}
6364
6365status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
6366 AudioSystem::sync_event_t event,
Glenn Kastend848eb42016-03-08 13:42:11 -08006367 audio_session_t triggerSession)
Eric Laurent81784c32012-11-19 14:55:58 -08006368{
6369 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
6370 sp<ThreadBase> strongMe = this;
6371 status_t status = NO_ERROR;
6372
6373 if (event == AudioSystem::SYNC_EVENT_NONE) {
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006374 recordTrack->clearSyncStartEvent();
Eric Laurent81784c32012-11-19 14:55:58 -08006375 } else if (event != AudioSystem::SYNC_EVENT_SAME) {
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006376 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
Eric Laurent81784c32012-11-19 14:55:58 -08006377 triggerSession,
6378 recordTrack->sessionId(),
6379 syncStartEventCallback,
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006380 recordTrack);
Eric Laurent81784c32012-11-19 14:55:58 -08006381 // Sync event can be cancelled by the trigger session if the track is not in a
6382 // compatible state in which case we start record immediately
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006383 if (recordTrack->mSyncStartEvent->isCancelled()) {
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006384 recordTrack->clearSyncStartEvent();
Eric Laurent81784c32012-11-19 14:55:58 -08006385 } else {
6386 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006387 recordTrack->mFramesToDrop = -
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08006388 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000);
Eric Laurent81784c32012-11-19 14:55:58 -08006389 }
6390 }
6391
6392 {
Glenn Kasten47c20702013-08-13 15:37:35 -07006393 // This section is a rendezvous between binder thread executing start() and RecordThread
Eric Laurent81784c32012-11-19 14:55:58 -08006394 AutoMutex lock(mLock);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006395 if (mActiveTracks.indexOf(recordTrack) >= 0) {
6396 if (recordTrack->mState == TrackBase::PAUSING) {
6397 ALOGV("active record track PAUSING -> ACTIVE");
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006398 recordTrack->mState = TrackBase::ACTIVE;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006399 } else {
6400 ALOGV("active record track state %d", recordTrack->mState);
Eric Laurent81784c32012-11-19 14:55:58 -08006401 }
6402 return status;
6403 }
6404
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08006405 // TODO consider other ways of handling this, such as changing the state to :STARTING and
6406 // adding the track to mActiveTracks after returning from AudioSystem::startInput(),
6407 // or using a separate command thread
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006408 recordTrack->mState = TrackBase::STARTING_1;
Glenn Kasten2b806402013-11-20 16:37:38 -08006409 mActiveTracks.add(recordTrack);
6410 mActiveTracksGen++;
Eric Laurent83b88082014-06-20 18:31:16 -07006411 status_t status = NO_ERROR;
6412 if (recordTrack->isExternalTrack()) {
6413 mLock.unlock();
Glenn Kastend848eb42016-03-08 13:42:11 -08006414 status = AudioSystem::startInput(mId, recordTrack->sessionId());
Eric Laurent83b88082014-06-20 18:31:16 -07006415 mLock.lock();
6416 // FIXME should verify that recordTrack is still in mActiveTracks
6417 if (status != NO_ERROR) {
6418 mActiveTracks.remove(recordTrack);
6419 mActiveTracksGen++;
6420 recordTrack->clearSyncStartEvent();
6421 ALOGV("RecordThread::start error %d", status);
6422 return status;
6423 }
Eric Laurent81784c32012-11-19 14:55:58 -08006424 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006425 // Catch up with current buffer indices if thread is already running.
6426 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront
6427 // was initialized to some value closer to the thread's mRsmpInFront, then the track could
6428 // see previously buffered data before it called start(), but with greater risk of overrun.
6429
Andy Hung73c02e42015-03-29 01:13:58 -07006430 recordTrack->mResamplerBufferProvider->reset();
Andy Hung97a893e2015-03-29 01:03:07 -07006431 // clear any converter state as new data will be discontinuous
6432 recordTrack->mRecordBufferConverter->reset();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006433 recordTrack->mState = TrackBase::STARTING_2;
Eric Laurent81784c32012-11-19 14:55:58 -08006434 // signal thread to start
Eric Laurent81784c32012-11-19 14:55:58 -08006435 mWaitWorkCV.broadcast();
Glenn Kasten2b806402013-11-20 16:37:38 -08006436 if (mActiveTracks.indexOf(recordTrack) < 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08006437 ALOGV("Record failed to start");
6438 status = BAD_VALUE;
6439 goto startError;
6440 }
Eric Laurent81784c32012-11-19 14:55:58 -08006441 return status;
6442 }
Glenn Kasten7c027242012-12-26 14:43:16 -08006443
Eric Laurent81784c32012-11-19 14:55:58 -08006444startError:
Eric Laurent83b88082014-06-20 18:31:16 -07006445 if (recordTrack->isExternalTrack()) {
Glenn Kastend848eb42016-03-08 13:42:11 -08006446 AudioSystem::stopInput(mId, recordTrack->sessionId());
Eric Laurent83b88082014-06-20 18:31:16 -07006447 }
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006448 recordTrack->clearSyncStartEvent();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006449 // FIXME I wonder why we do not reset the state here?
Eric Laurent81784c32012-11-19 14:55:58 -08006450 return status;
6451}
6452
Eric Laurent81784c32012-11-19 14:55:58 -08006453void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
6454{
6455 sp<SyncEvent> strongEvent = event.promote();
6456
6457 if (strongEvent != 0) {
Eric Laurent8ea16e42014-02-20 16:26:11 -08006458 sp<RefBase> ptr = strongEvent->cookie().promote();
6459 if (ptr != 0) {
6460 RecordTrack *recordTrack = (RecordTrack *)ptr.get();
6461 recordTrack->handleSyncStartEvent(strongEvent);
6462 }
Eric Laurent81784c32012-11-19 14:55:58 -08006463 }
6464}
6465
Glenn Kastena8356f62013-07-25 14:37:52 -07006466bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
Eric Laurent81784c32012-11-19 14:55:58 -08006467 ALOGV("RecordThread::stop");
Glenn Kastena8356f62013-07-25 14:37:52 -07006468 AutoMutex _l(mLock);
Glenn Kasten2b806402013-11-20 16:37:38 -08006469 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) {
Eric Laurent81784c32012-11-19 14:55:58 -08006470 return false;
6471 }
Glenn Kasten47c20702013-08-13 15:37:35 -07006472 // note that threadLoop may still be processing the track at this point [without lock]
Eric Laurent81784c32012-11-19 14:55:58 -08006473 recordTrack->mState = TrackBase::PAUSING;
6474 // do not wait for mStartStopCond if exiting
6475 if (exitPending()) {
6476 return true;
6477 }
Glenn Kasten47c20702013-08-13 15:37:35 -07006478 // FIXME incorrect usage of wait: no explicit predicate or loop
Eric Laurent81784c32012-11-19 14:55:58 -08006479 mStartStopCond.wait(mLock);
Glenn Kasten2b806402013-11-20 16:37:38 -08006480 // if we have been restarted, recordTrack is in mActiveTracks here
6481 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08006482 ALOGV("Record stopped OK");
6483 return true;
6484 }
6485 return false;
6486}
6487
Glenn Kasten0f11b512014-01-31 16:18:54 -08006488bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const
Eric Laurent81784c32012-11-19 14:55:58 -08006489{
6490 return false;
6491}
6492
Glenn Kasten0f11b512014-01-31 16:18:54 -08006493status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08006494{
6495#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
6496 if (!isValidSyncEvent(event)) {
6497 return BAD_VALUE;
6498 }
6499
Glenn Kastend848eb42016-03-08 13:42:11 -08006500 audio_session_t eventSession = event->triggerSession();
Eric Laurent81784c32012-11-19 14:55:58 -08006501 status_t ret = NAME_NOT_FOUND;
6502
6503 Mutex::Autolock _l(mLock);
6504
6505 for (size_t i = 0; i < mTracks.size(); i++) {
6506 sp<RecordTrack> track = mTracks[i];
6507 if (eventSession == track->sessionId()) {
6508 (void) track->setSyncEvent(event);
6509 ret = NO_ERROR;
6510 }
6511 }
6512 return ret;
6513#else
6514 return BAD_VALUE;
6515#endif
6516}
6517
6518// destroyTrack_l() must be called with ThreadBase::mLock held
6519void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
6520{
Eric Laurentbfb1b832013-01-07 09:53:42 -08006521 track->terminate();
6522 track->mState = TrackBase::STOPPED;
Eric Laurent81784c32012-11-19 14:55:58 -08006523 // active tracks are removed by threadLoop()
Glenn Kasten2b806402013-11-20 16:37:38 -08006524 if (mActiveTracks.indexOf(track) < 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08006525 removeTrack_l(track);
6526 }
6527}
6528
6529void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
6530{
6531 mTracks.remove(track);
6532 // need anything related to effects here?
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006533 if (track->isFastTrack()) {
6534 ALOG_ASSERT(!mFastTrackAvail);
6535 mFastTrackAvail = true;
6536 }
Eric Laurent81784c32012-11-19 14:55:58 -08006537}
6538
6539void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
6540{
6541 dumpInternals(fd, args);
6542 dumpTracks(fd, args);
6543 dumpEffectChains(fd, args);
6544}
6545
6546void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
6547{
Elliott Hughes87cebad2014-05-22 10:14:43 -07006548 dprintf(fd, "\nInput thread %p:\n", this);
Eric Laurent81784c32012-11-19 14:55:58 -08006549
Glenn Kasten44182c22015-03-05 17:12:23 -08006550 dumpBase(fd, args);
6551
6552 if (mActiveTracks.size() == 0) {
Elliott Hughes87cebad2014-05-22 10:14:43 -07006553 dprintf(fd, " No active record clients\n");
Eric Laurent81784c32012-11-19 14:55:58 -08006554 }
Glenn Kasten6e6704c2014-07-03 10:20:00 -07006555 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no");
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006556 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
Glenn Kasten17c9c992015-03-02 15:53:01 -08006557
Glenn Kasten2f90c512015-12-02 11:40:09 -08006558 // Make a non-atomic copy of fast capture dump state so it won't change underneath us
6559 // while we are dumping it. It may be inconsistent, but it won't mutate!
6560 // This is a large object so we place it on the heap.
6561 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
6562 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState);
6563 copy->dump(fd);
6564 delete copy;
Eric Laurent81784c32012-11-19 14:55:58 -08006565}
6566
Glenn Kasten0f11b512014-01-31 16:18:54 -08006567void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08006568{
6569 const size_t SIZE = 256;
6570 char buffer[SIZE];
6571 String8 result;
6572
Marco Nelissenb2208842014-02-07 14:00:50 -08006573 size_t numtracks = mTracks.size();
6574 size_t numactive = mActiveTracks.size();
6575 size_t numactiveseen = 0;
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006576 dprintf(fd, " %zu Tracks", numtracks);
Marco Nelissenb2208842014-02-07 14:00:50 -08006577 if (numtracks) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006578 dprintf(fd, " of which %zu are active\n", numactive);
Marco Nelissenb2208842014-02-07 14:00:50 -08006579 RecordTrack::appendDumpHeader(result);
6580 for (size_t i = 0; i < numtracks ; ++i) {
6581 sp<RecordTrack> track = mTracks[i];
6582 if (track != 0) {
6583 bool active = mActiveTracks.indexOf(track) >= 0;
6584 if (active) {
6585 numactiveseen++;
6586 }
6587 track->dump(buffer, SIZE, active);
6588 result.append(buffer);
6589 }
Eric Laurent81784c32012-11-19 14:55:58 -08006590 }
Marco Nelissenb2208842014-02-07 14:00:50 -08006591 } else {
Elliott Hughes87cebad2014-05-22 10:14:43 -07006592 dprintf(fd, "\n");
Eric Laurent81784c32012-11-19 14:55:58 -08006593 }
6594
Marco Nelissenb2208842014-02-07 14:00:50 -08006595 if (numactiveseen != numactive) {
6596 snprintf(buffer, SIZE, " The following tracks are in the active list but"
6597 " not in the track list\n");
Eric Laurent81784c32012-11-19 14:55:58 -08006598 result.append(buffer);
6599 RecordTrack::appendDumpHeader(result);
Marco Nelissenb2208842014-02-07 14:00:50 -08006600 for (size_t i = 0; i < numactive; ++i) {
Glenn Kasten2b806402013-11-20 16:37:38 -08006601 sp<RecordTrack> track = mActiveTracks[i];
Marco Nelissenb2208842014-02-07 14:00:50 -08006602 if (mTracks.indexOf(track) < 0) {
6603 track->dump(buffer, SIZE, true);
6604 result.append(buffer);
6605 }
Glenn Kasten2b806402013-11-20 16:37:38 -08006606 }
Eric Laurent81784c32012-11-19 14:55:58 -08006607
6608 }
6609 write(fd, result.string(), result.size());
6610}
6611
Andy Hung73c02e42015-03-29 01:13:58 -07006612
6613void AudioFlinger::RecordThread::ResamplerBufferProvider::reset()
6614{
6615 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
6616 RecordThread *recordThread = (RecordThread *) threadBase.get();
6617 mRsmpInFront = recordThread->mRsmpInRear;
6618 mRsmpInUnrel = 0;
6619}
6620
6621void AudioFlinger::RecordThread::ResamplerBufferProvider::sync(
6622 size_t *framesAvailable, bool *hasOverrun)
6623{
6624 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
6625 RecordThread *recordThread = (RecordThread *) threadBase.get();
6626 const int32_t rear = recordThread->mRsmpInRear;
6627 const int32_t front = mRsmpInFront;
6628 const ssize_t filled = rear - front;
6629
6630 size_t framesIn;
6631 bool overrun = false;
6632 if (filled < 0) {
6633 // should not happen, but treat like a massive overrun and re-sync
6634 framesIn = 0;
6635 mRsmpInFront = rear;
6636 overrun = true;
6637 } else if ((size_t) filled <= recordThread->mRsmpInFrames) {
6638 framesIn = (size_t) filled;
6639 } else {
6640 // client is not keeping up with server, but give it latest data
6641 framesIn = recordThread->mRsmpInFrames;
6642 mRsmpInFront = /* front = */ rear - framesIn;
6643 overrun = true;
6644 }
6645 if (framesAvailable != NULL) {
6646 *framesAvailable = framesIn;
6647 }
6648 if (hasOverrun != NULL) {
6649 *hasOverrun = overrun;
6650 }
6651}
6652
Eric Laurent81784c32012-11-19 14:55:58 -08006653// AudioBufferProvider interface
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006654status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
Glenn Kastend79072e2016-01-06 08:41:20 -08006655 AudioBufferProvider::Buffer* buffer)
Eric Laurent81784c32012-11-19 14:55:58 -08006656{
Andy Hung73c02e42015-03-29 01:13:58 -07006657 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006658 if (threadBase == 0) {
6659 buffer->frameCount = 0;
Glenn Kasten607fa3e2014-02-21 14:24:58 -08006660 buffer->raw = NULL;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006661 return NOT_ENOUGH_DATA;
6662 }
6663 RecordThread *recordThread = (RecordThread *) threadBase.get();
6664 int32_t rear = recordThread->mRsmpInRear;
Andy Hung73c02e42015-03-29 01:13:58 -07006665 int32_t front = mRsmpInFront;
Glenn Kasten85948432013-08-19 12:09:05 -07006666 ssize_t filled = rear - front;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006667 // FIXME should not be P2 (don't want to increase latency)
6668 // FIXME if client not keeping up, discard
Glenn Kasten607fa3e2014-02-21 14:24:58 -08006669 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames));
Glenn Kasten85948432013-08-19 12:09:05 -07006670 // 'filled' may be non-contiguous, so return only the first contiguous chunk
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006671 front &= recordThread->mRsmpInFramesP2 - 1;
6672 size_t part1 = recordThread->mRsmpInFramesP2 - front;
Glenn Kasten85948432013-08-19 12:09:05 -07006673 if (part1 > (size_t) filled) {
6674 part1 = filled;
6675 }
6676 size_t ask = buffer->frameCount;
6677 ALOG_ASSERT(ask > 0);
6678 if (part1 > ask) {
6679 part1 = ask;
6680 }
6681 if (part1 == 0) {
Andy Hung73c02e42015-03-29 01:13:58 -07006682 // out of data is fine since the resampler will return a short-count.
Glenn Kasten85948432013-08-19 12:09:05 -07006683 buffer->raw = NULL;
6684 buffer->frameCount = 0;
Andy Hung73c02e42015-03-29 01:13:58 -07006685 mRsmpInUnrel = 0;
Glenn Kasten85948432013-08-19 12:09:05 -07006686 return NOT_ENOUGH_DATA;
Eric Laurent81784c32012-11-19 14:55:58 -08006687 }
6688
Andy Hung57446612015-04-19 23:56:46 -07006689 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize;
Glenn Kasten85948432013-08-19 12:09:05 -07006690 buffer->frameCount = part1;
Andy Hung73c02e42015-03-29 01:13:58 -07006691 mRsmpInUnrel = part1;
Eric Laurent81784c32012-11-19 14:55:58 -08006692 return NO_ERROR;
6693}
6694
6695// AudioBufferProvider interface
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006696void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
6697 AudioBufferProvider::Buffer* buffer)
Eric Laurent81784c32012-11-19 14:55:58 -08006698{
Glenn Kasten85948432013-08-19 12:09:05 -07006699 size_t stepCount = buffer->frameCount;
6700 if (stepCount == 0) {
6701 return;
6702 }
Andy Hung73c02e42015-03-29 01:13:58 -07006703 ALOG_ASSERT(stepCount <= mRsmpInUnrel);
6704 mRsmpInUnrel -= stepCount;
6705 mRsmpInFront += stepCount;
Glenn Kasten85948432013-08-19 12:09:05 -07006706 buffer->raw = NULL;
Eric Laurent81784c32012-11-19 14:55:58 -08006707 buffer->frameCount = 0;
6708}
6709
Andy Hung97a893e2015-03-29 01:03:07 -07006710AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter(
6711 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
6712 uint32_t srcSampleRate,
6713 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
6714 uint32_t dstSampleRate) :
6715 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars
6716 // mSrcFormat
6717 // mSrcSampleRate
6718 // mDstChannelMask
6719 // mDstFormat
6720 // mDstSampleRate
6721 // mSrcChannelCount
6722 // mDstChannelCount
6723 // mDstFrameSize
6724 mBuf(NULL), mBufFrames(0), mBufFrameSize(0),
Andy Hungd330ee42015-04-20 13:23:41 -07006725 mResampler(NULL),
6726 mIsLegacyDownmix(false),
6727 mIsLegacyUpmix(false),
6728 mRequiresFloat(false),
6729 mInputConverterProvider(NULL)
Andy Hung97a893e2015-03-29 01:03:07 -07006730{
6731 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate,
6732 dstChannelMask, dstFormat, dstSampleRate);
6733}
6734
6735AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() {
6736 free(mBuf);
6737 delete mResampler;
Andy Hungd330ee42015-04-20 13:23:41 -07006738 delete mInputConverterProvider;
Andy Hung97a893e2015-03-29 01:03:07 -07006739}
6740
6741size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst,
6742 AudioBufferProvider *provider, size_t frames)
6743{
Andy Hungd330ee42015-04-20 13:23:41 -07006744 if (mInputConverterProvider != NULL) {
6745 mInputConverterProvider->setBufferProvider(provider);
6746 provider = mInputConverterProvider;
6747 }
6748
6749 if (mResampler == NULL) {
Andy Hung97a893e2015-03-29 01:03:07 -07006750 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
6751 mSrcSampleRate, mSrcFormat, mDstFormat);
6752
6753 AudioBufferProvider::Buffer buffer;
6754 for (size_t i = frames; i > 0; ) {
6755 buffer.frameCount = i;
Glenn Kastend79072e2016-01-06 08:41:20 -08006756 status_t status = provider->getNextBuffer(&buffer);
Andy Hung97a893e2015-03-29 01:03:07 -07006757 if (status != OK || buffer.frameCount == 0) {
6758 frames -= i; // cannot fill request.
6759 break;
6760 }
Andy Hungd330ee42015-04-20 13:23:41 -07006761 // format convert to destination buffer
6762 convertNoResampler(dst, buffer.raw, buffer.frameCount);
Andy Hung97a893e2015-03-29 01:03:07 -07006763
6764 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize;
6765 i -= buffer.frameCount;
6766 provider->releaseBuffer(&buffer);
6767 }
6768 } else {
6769 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
6770 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat);
6771
Andy Hungd330ee42015-04-20 13:23:41 -07006772 // reallocate buffer if needed
6773 if (mBufFrameSize != 0 && mBufFrames < frames) {
6774 free(mBuf);
6775 mBufFrames = frames;
6776 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
6777 }
Andy Hung97a893e2015-03-29 01:03:07 -07006778 // resampler accumulates, but we only have one source track
Andy Hungd330ee42015-04-20 13:23:41 -07006779 memset(mBuf, 0, frames * mBufFrameSize);
6780 frames = mResampler->resample((int32_t*)mBuf, frames, provider);
6781 // format convert to destination buffer
6782 convertResampler(dst, mBuf, frames);
Andy Hung97a893e2015-03-29 01:03:07 -07006783 }
6784 return frames;
6785}
6786
6787status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters(
6788 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
6789 uint32_t srcSampleRate,
6790 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
6791 uint32_t dstSampleRate)
6792{
6793 // quick evaluation if there is any change.
6794 if (mSrcFormat == srcFormat
6795 && mSrcChannelMask == srcChannelMask
6796 && mSrcSampleRate == srcSampleRate
6797 && mDstFormat == dstFormat
6798 && mDstChannelMask == dstChannelMask
6799 && mDstSampleRate == dstSampleRate) {
6800 return NO_ERROR;
6801 }
6802
Andy Hungdb4c0312015-05-06 08:46:52 -07006803 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x"
6804 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u",
6805 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate);
Andy Hung97a893e2015-03-29 01:03:07 -07006806 const bool valid =
6807 audio_is_input_channel(srcChannelMask)
6808 && audio_is_input_channel(dstChannelMask)
6809 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat)
6810 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat)
6811 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX)
6812 ; // no upsampling checks for now
6813 if (!valid) {
6814 return BAD_VALUE;
6815 }
6816
6817 mSrcFormat = srcFormat;
6818 mSrcChannelMask = srcChannelMask;
6819 mSrcSampleRate = srcSampleRate;
6820 mDstFormat = dstFormat;
6821 mDstChannelMask = dstChannelMask;
6822 mDstSampleRate = dstSampleRate;
6823
6824 // compute derived parameters
6825 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask);
6826 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask);
6827 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat);
6828
Andy Hungd330ee42015-04-20 13:23:41 -07006829 // do we need to resample?
6830 delete mResampler;
6831 mResampler = NULL;
6832 if (mSrcSampleRate != mDstSampleRate) {
6833 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT,
6834 mSrcChannelCount, mDstSampleRate);
6835 mResampler->setSampleRate(mSrcSampleRate);
6836 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
6837 }
6838
6839 // are we running legacy channel conversion modes?
6840 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO
6841 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK)
6842 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO;
6843 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO
6844 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO
6845 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK);
6846
6847 // do we need to process in float?
6848 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix;
6849
6850 // do we need a staging buffer to convert for destination (we can still optimize this)?
6851 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity
6852 if (mResampler != NULL) {
6853 mBufFrameSize = max(mSrcChannelCount, FCC_2)
6854 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
Andy Hunga97630b2015-07-22 23:27:24 -07006855 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float
Andy Hungd330ee42015-04-20 13:23:41 -07006856 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
6857 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) {
Andy Hung97a893e2015-03-29 01:03:07 -07006858 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat);
6859 } else {
6860 mBufFrameSize = 0;
6861 }
6862 mBufFrames = 0; // force the buffer to be resized.
6863
Andy Hungd330ee42015-04-20 13:23:41 -07006864 // do we need an input converter buffer provider to give us float?
6865 delete mInputConverterProvider;
6866 mInputConverterProvider = NULL;
6867 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) {
6868 mInputConverterProvider = new ReformatBufferProvider(
6869 audio_channel_count_from_in_mask(mSrcChannelMask),
6870 mSrcFormat,
6871 AUDIO_FORMAT_PCM_FLOAT,
6872 256 /* provider buffer frame count */);
6873 }
6874
6875 // do we need a remixer to do channel mask conversion
6876 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) {
6877 (void) memcpy_by_index_array_initialization_from_channel_mask(
6878 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask);
Andy Hung97a893e2015-03-29 01:03:07 -07006879 }
6880 return NO_ERROR;
6881}
6882
Andy Hungd330ee42015-04-20 13:23:41 -07006883void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler(
6884 void *dst, const void *src, size_t frames)
Andy Hung97a893e2015-03-29 01:03:07 -07006885{
Andy Hungd330ee42015-04-20 13:23:41 -07006886 // src is native type unless there is legacy upmix or downmix, whereupon it is float.
Andy Hung97a893e2015-03-29 01:03:07 -07006887 if (mBufFrameSize != 0 && mBufFrames < frames) {
6888 free(mBuf);
6889 mBufFrames = frames;
6890 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
6891 }
Andy Hungd330ee42015-04-20 13:23:41 -07006892 // do we need to do legacy upmix and downmix?
6893 if (mIsLegacyUpmix || mIsLegacyDownmix) {
Andy Hung97a893e2015-03-29 01:03:07 -07006894 void *dstBuf = mBuf != NULL ? mBuf : dst;
Andy Hungd330ee42015-04-20 13:23:41 -07006895 if (mIsLegacyUpmix) {
6896 upmix_to_stereo_float_from_mono_float((float *)dstBuf,
6897 (const float *)src, frames);
6898 } else /*mIsLegacyDownmix */ {
6899 downmix_to_mono_float_from_stereo_float((float *)dstBuf,
6900 (const float *)src, frames);
Andy Hung97a893e2015-03-29 01:03:07 -07006901 }
Andy Hungd330ee42015-04-20 13:23:41 -07006902 if (mBuf != NULL) {
6903 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT,
6904 frames * mDstChannelCount);
6905 }
6906 return;
6907 }
6908 // do we need to do channel mask conversion?
6909 if (mSrcChannelMask != mDstChannelMask) {
Andy Hung97a893e2015-03-29 01:03:07 -07006910 void *dstBuf = mBuf != NULL ? mBuf : dst;
Andy Hungd330ee42015-04-20 13:23:41 -07006911 memcpy_by_index_array(dstBuf, mDstChannelCount,
6912 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames);
6913 if (dstBuf == dst) {
6914 return; // format is the same
6915 }
6916 }
6917 // convert to destination buffer
6918 const void *convertBuf = mBuf != NULL ? mBuf : src;
6919 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat,
6920 frames * mDstChannelCount);
6921}
6922
6923void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler(
6924 void *dst, /*not-a-const*/ void *src, size_t frames)
6925{
6926 // src buffer format is ALWAYS float when entering this routine
6927 if (mIsLegacyUpmix) {
6928 ; // mono to stereo already handled by resampler
6929 } else if (mIsLegacyDownmix
6930 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) {
6931 // the resampler outputs stereo for mono input channel (a feature?)
6932 // must convert to mono
6933 downmix_to_mono_float_from_stereo_float((float *)src,
6934 (const float *)src, frames);
6935 } else if (mSrcChannelMask != mDstChannelMask) {
6936 // convert to mono channel again for channel mask conversion (could be skipped
6937 // with further optimization).
Andy Hung97a893e2015-03-29 01:03:07 -07006938 if (mSrcChannelCount == 1) {
Andy Hungd330ee42015-04-20 13:23:41 -07006939 downmix_to_mono_float_from_stereo_float((float *)src,
6940 (const float *)src, frames);
Andy Hung97a893e2015-03-29 01:03:07 -07006941 }
Andy Hungd330ee42015-04-20 13:23:41 -07006942 // convert to destination format (in place, OK as float is larger than other types)
6943 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) {
6944 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
6945 frames * mSrcChannelCount);
6946 }
6947 // channel convert and save to dst
6948 memcpy_by_index_array(dst, mDstChannelCount,
6949 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames);
6950 return;
Andy Hung97a893e2015-03-29 01:03:07 -07006951 }
Andy Hungd330ee42015-04-20 13:23:41 -07006952 // convert to destination format and save to dst
6953 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
6954 frames * mDstChannelCount);
Andy Hung97a893e2015-03-29 01:03:07 -07006955}
6956
Eric Laurent10351942014-05-08 18:49:52 -07006957bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
6958 status_t& status)
Eric Laurent81784c32012-11-19 14:55:58 -08006959{
6960 bool reconfig = false;
6961
Eric Laurent10351942014-05-08 18:49:52 -07006962 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08006963
Eric Laurent10351942014-05-08 18:49:52 -07006964 audio_format_t reqFormat = mFormat;
6965 uint32_t samplingRate = mSampleRate;
Glenn Kastene1635ec2015-06-08 15:46:49 -07006966 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs).
Eric Laurent10351942014-05-08 18:49:52 -07006967 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount);
6968
6969 AudioParameter param = AudioParameter(keyValuePair);
6970 int value;
Haynes Mathew George9ce67b52015-09-30 11:40:47 -07006971
6972 // scope for AutoPark extends to end of method
6973 AutoPark<FastCapture> park(mFastCapture);
6974
Eric Laurent10351942014-05-08 18:49:52 -07006975 // TODO Investigate when this code runs. Check with audio policy when a sample rate and
6976 // channel count change can be requested. Do we mandate the first client defines the
6977 // HAL sampling rate and channel count or do we allow changes on the fly?
6978 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
6979 samplingRate = value;
6980 reconfig = true;
6981 }
6982 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
Andy Hung97a893e2015-03-29 01:03:07 -07006983 if (!audio_is_linear_pcm((audio_format_t) value)) {
Eric Laurent10351942014-05-08 18:49:52 -07006984 status = BAD_VALUE;
6985 } else {
6986 reqFormat = (audio_format_t) value;
Eric Laurent81784c32012-11-19 14:55:58 -08006987 reconfig = true;
6988 }
Eric Laurent10351942014-05-08 18:49:52 -07006989 }
6990 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
6991 audio_channel_mask_t mask = (audio_channel_mask_t) value;
Andy Hungd330ee42015-04-20 13:23:41 -07006992 if (!audio_is_input_channel(mask) ||
6993 audio_channel_count_from_in_mask(mask) > FCC_8) {
Eric Laurent10351942014-05-08 18:49:52 -07006994 status = BAD_VALUE;
6995 } else {
6996 channelMask = mask;
6997 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08006998 }
Eric Laurent10351942014-05-08 18:49:52 -07006999 }
7000 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
7001 // do not accept frame count changes if tracks are open as the track buffer
7002 // size depends on frame count and correct behavior would not be guaranteed
7003 // if frame count is changed after track creation
7004 if (mActiveTracks.size() > 0) {
7005 status = INVALID_OPERATION;
7006 } else {
7007 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08007008 }
Eric Laurent10351942014-05-08 18:49:52 -07007009 }
7010 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
7011 // forward device change to effects that have requested to be
7012 // aware of attached audio device.
7013 for (size_t i = 0; i < mEffectChains.size(); i++) {
7014 mEffectChains[i]->setDevice_l(value);
Eric Laurent81784c32012-11-19 14:55:58 -08007015 }
Eric Laurent81784c32012-11-19 14:55:58 -08007016
Eric Laurent10351942014-05-08 18:49:52 -07007017 // store input device and output device but do not forward output device to audio HAL.
7018 // Note that status is ignored by the caller for output device
7019 // (see AudioFlinger::setParameters()
7020 if (audio_is_output_devices(value)) {
7021 mOutDevice = value;
7022 status = BAD_VALUE;
7023 } else {
7024 mInDevice = value;
Eric Laurente8726fe2015-06-26 09:39:24 -07007025 if (value != AUDIO_DEVICE_NONE) {
7026 mPrevInDevice = value;
7027 }
Eric Laurent10351942014-05-08 18:49:52 -07007028 // disable AEC and NS if the device is a BT SCO headset supporting those
7029 // pre processings
7030 if (mTracks.size() > 0) {
7031 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
7032 mAudioFlinger->btNrecIsOff();
7033 for (size_t i = 0; i < mTracks.size(); i++) {
7034 sp<RecordTrack> track = mTracks[i];
7035 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
7036 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
Eric Laurent81784c32012-11-19 14:55:58 -08007037 }
7038 }
7039 }
Eric Laurent10351942014-05-08 18:49:52 -07007040 }
7041 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
7042 mAudioSource != (audio_source_t)value) {
7043 // forward device change to effects that have requested to be
7044 // aware of attached audio device.
7045 for (size_t i = 0; i < mEffectChains.size(); i++) {
7046 mEffectChains[i]->setAudioSource_l((audio_source_t)value);
Eric Laurent81784c32012-11-19 14:55:58 -08007047 }
Eric Laurent10351942014-05-08 18:49:52 -07007048 mAudioSource = (audio_source_t)value;
7049 }
Glenn Kastene198c362013-08-13 09:13:36 -07007050
Eric Laurent10351942014-05-08 18:49:52 -07007051 if (status == NO_ERROR) {
7052 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7053 keyValuePair.string());
7054 if (status == INVALID_OPERATION) {
7055 inputStandBy();
Eric Laurent81784c32012-11-19 14:55:58 -08007056 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7057 keyValuePair.string());
Eric Laurent10351942014-05-08 18:49:52 -07007058 }
7059 if (reconfig) {
7060 if (status == BAD_VALUE &&
Andy Hung97a893e2015-03-29 01:03:07 -07007061 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) &&
7062 audio_is_linear_pcm(reqFormat) &&
Eric Laurent10351942014-05-08 18:49:52 -07007063 (mInput->stream->common.get_sample_rate(&mInput->stream->common)
Andy Hung97a893e2015-03-29 01:03:07 -07007064 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) &&
Andy Hunge5412692014-05-16 11:25:07 -07007065 audio_channel_count_from_in_mask(
Andy Hungd1abb8f2015-05-05 23:42:34 -07007066 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) {
Eric Laurent10351942014-05-08 18:49:52 -07007067 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08007068 }
Eric Laurent10351942014-05-08 18:49:52 -07007069 if (status == NO_ERROR) {
7070 readInputParameters_l();
Eric Laurent73e26b62015-04-27 16:55:58 -07007071 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
Eric Laurent81784c32012-11-19 14:55:58 -08007072 }
7073 }
Eric Laurent81784c32012-11-19 14:55:58 -08007074 }
Eric Laurent10351942014-05-08 18:49:52 -07007075
Eric Laurent81784c32012-11-19 14:55:58 -08007076 return reconfig;
7077}
7078
7079String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
7080{
Eric Laurent81784c32012-11-19 14:55:58 -08007081 Mutex::Autolock _l(mLock);
7082 if (initCheck() != NO_ERROR) {
Glenn Kastend8ea6992013-07-16 14:17:15 -07007083 return String8();
Eric Laurent81784c32012-11-19 14:55:58 -08007084 }
7085
Glenn Kastend8ea6992013-07-16 14:17:15 -07007086 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
7087 const String8 out_s8(s);
Eric Laurent81784c32012-11-19 14:55:58 -08007088 free(s);
7089 return out_s8;
7090}
7091
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07007092void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) {
Eric Laurent73e26b62015-04-27 16:55:58 -07007093 sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
7094
7095 desc->mIoHandle = mId;
Eric Laurent81784c32012-11-19 14:55:58 -08007096
7097 switch (event) {
Eric Laurent73e26b62015-04-27 16:55:58 -07007098 case AUDIO_INPUT_OPENED:
7099 case AUDIO_INPUT_CONFIG_CHANGED:
Eric Laurent296fb132015-05-01 11:38:42 -07007100 desc->mPatch = mPatch;
Eric Laurent73e26b62015-04-27 16:55:58 -07007101 desc->mChannelMask = mChannelMask;
7102 desc->mSamplingRate = mSampleRate;
7103 desc->mFormat = mFormat;
7104 desc->mFrameCount = mFrameCount;
Glenn Kasten4a8308b2016-04-18 14:10:01 -07007105 desc->mFrameCountHAL = mFrameCount;
Eric Laurent73e26b62015-04-27 16:55:58 -07007106 desc->mLatency = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08007107 break;
7108
Eric Laurent73e26b62015-04-27 16:55:58 -07007109 case AUDIO_INPUT_CLOSED:
Eric Laurent81784c32012-11-19 14:55:58 -08007110 default:
7111 break;
7112 }
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07007113 mAudioFlinger->ioConfigChanged(event, desc, pid);
Eric Laurent81784c32012-11-19 14:55:58 -08007114}
7115
Glenn Kastendeca2ae2014-02-07 10:25:56 -08007116void AudioFlinger::RecordThread::readInputParameters_l()
Eric Laurent81784c32012-11-19 14:55:58 -08007117{
Eric Laurent81784c32012-11-19 14:55:58 -08007118 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
7119 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
Andy Hunge5412692014-05-16 11:25:07 -07007120 mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
Andy Hungd330ee42015-04-20 13:23:41 -07007121 if (mChannelCount > FCC_8) {
7122 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8);
7123 }
Andy Hung463be252014-07-10 16:56:07 -07007124 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common);
7125 mFormat = mHALFormat;
Andy Hungd330ee42015-04-20 13:23:41 -07007126 if (!audio_is_linear_pcm(mFormat)) {
7127 ALOGE("HAL format %#x is not linear pcm", mFormat);
Glenn Kasten291bb6d2013-07-16 17:23:39 -07007128 }
Eric Laurent665470b2014-07-03 16:37:08 -07007129 mFrameSize = audio_stream_in_frame_size(mInput->stream);
Glenn Kasten548efc92012-11-29 08:48:51 -08007130 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
7131 mFrameCount = mBufferSize / mFrameSize;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08007132 // This is the formula for calculating the temporary buffer size.
Glenn Kastene8426142014-02-28 16:45:03 -08007133 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to
Glenn Kasten85948432013-08-19 12:09:05 -07007134 // 1 full output buffer, regardless of the alignment of the available input.
Glenn Kastene8426142014-02-28 16:45:03 -08007135 // The value is somewhat arbitrary, and could probably be even larger.
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08007136 // A larger value should allow more old data to be read after a track calls start(),
7137 // without increasing latency.
Andy Hung97a893e2015-03-29 01:03:07 -07007138 //
7139 // Note this is independent of the maximum downsampling ratio permitted for capture.
Glenn Kastene8426142014-02-28 16:45:03 -08007140 mRsmpInFrames = mFrameCount * 7;
Glenn Kasten85948432013-08-19 12:09:05 -07007141 mRsmpInFramesP2 = roundup(mRsmpInFrames);
Andy Hung57446612015-04-19 23:56:46 -07007142 free(mRsmpInBuffer);
Andy Hung0a01c2f2015-09-21 12:44:54 -07007143 mRsmpInBuffer = NULL;
Glenn Kasten49d00ad2014-07-21 11:22:03 -07007144
7145 // TODO optimize audio capture buffer sizes ...
7146 // Here we calculate the size of the sliding buffer used as a source
7147 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7).
7148 // For current HAL frame counts, this is usually 2048 = 40 ms. It would
7149 // be better to have it derived from the pipe depth in the long term.
7150 // The current value is higher than necessary. However it should not add to latency.
7151
Glenn Kasten85948432013-08-19 12:09:05 -07007152 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
Andy Hung0a01c2f2015-09-21 12:44:54 -07007153 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize;
7154 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize);
7155 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here.
Eric Laurent81784c32012-11-19 14:55:58 -08007156
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08007157 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
7158 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
Eric Laurent81784c32012-11-19 14:55:58 -08007159}
7160
Glenn Kasten5f972c02014-01-13 09:59:31 -08007161uint32_t AudioFlinger::RecordThread::getInputFramesLost()
Eric Laurent81784c32012-11-19 14:55:58 -08007162{
7163 Mutex::Autolock _l(mLock);
7164 if (initCheck() != NO_ERROR) {
7165 return 0;
7166 }
7167
7168 return mInput->stream->get_input_frames_lost(mInput->stream);
7169}
7170
Glenn Kastend848eb42016-03-08 13:42:11 -08007171uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const
Eric Laurent81784c32012-11-19 14:55:58 -08007172{
7173 Mutex::Autolock _l(mLock);
7174 uint32_t result = 0;
7175 if (getEffectChain_l(sessionId) != 0) {
7176 result = EFFECT_SESSION;
7177 }
7178
7179 for (size_t i = 0; i < mTracks.size(); ++i) {
7180 if (sessionId == mTracks[i]->sessionId()) {
7181 result |= TRACK_SESSION;
7182 break;
7183 }
7184 }
7185
7186 return result;
7187}
7188
Glenn Kastend848eb42016-03-08 13:42:11 -08007189KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const
Eric Laurent81784c32012-11-19 14:55:58 -08007190{
Glenn Kastend848eb42016-03-08 13:42:11 -08007191 KeyedVector<audio_session_t, bool> ids;
Eric Laurent81784c32012-11-19 14:55:58 -08007192 Mutex::Autolock _l(mLock);
7193 for (size_t j = 0; j < mTracks.size(); ++j) {
7194 sp<RecordThread::RecordTrack> track = mTracks[j];
Glenn Kastend848eb42016-03-08 13:42:11 -08007195 audio_session_t sessionId = track->sessionId();
Eric Laurent81784c32012-11-19 14:55:58 -08007196 if (ids.indexOfKey(sessionId) < 0) {
7197 ids.add(sessionId, true);
7198 }
7199 }
7200 return ids;
7201}
7202
7203AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
7204{
7205 Mutex::Autolock _l(mLock);
7206 AudioStreamIn *input = mInput;
7207 mInput = NULL;
7208 return input;
7209}
7210
7211// this method must always be called either with ThreadBase mLock held or inside the thread loop
7212audio_stream_t* AudioFlinger::RecordThread::stream() const
7213{
7214 if (mInput == NULL) {
7215 return NULL;
7216 }
7217 return &mInput->stream->common;
7218}
7219
7220status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
7221{
7222 // only one chain per input thread
7223 if (mEffectChains.size() != 0) {
Eric Laurentaaa44472014-09-12 17:41:50 -07007224 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this);
Eric Laurent81784c32012-11-19 14:55:58 -08007225 return INVALID_OPERATION;
7226 }
7227 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
Eric Laurentaaa44472014-09-12 17:41:50 -07007228 chain->setThread(this);
Eric Laurent81784c32012-11-19 14:55:58 -08007229 chain->setInBuffer(NULL);
7230 chain->setOutBuffer(NULL);
7231
7232 checkSuspendOnAddEffectChain_l(chain);
7233
Eric Laurent1b928682014-10-02 19:41:47 -07007234 // make sure enabled pre processing effects state is communicated to the HAL as we
7235 // just moved them to a new input stream.
7236 chain->syncHalEffectsState();
7237
Eric Laurent81784c32012-11-19 14:55:58 -08007238 mEffectChains.add(chain);
7239
7240 return NO_ERROR;
7241}
7242
7243size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
7244{
7245 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
7246 ALOGW_IF(mEffectChains.size() != 1,
Glenn Kastenc42e9b42016-03-21 11:35:03 -07007247 "removeEffectChain_l() %p invalid chain size %zu on thread %p",
Eric Laurent81784c32012-11-19 14:55:58 -08007248 chain.get(), mEffectChains.size(), this);
7249 if (mEffectChains.size() == 1) {
7250 mEffectChains.removeAt(0);
7251 }
7252 return 0;
7253}
7254
Eric Laurent1c333e22014-05-20 10:48:17 -07007255status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch,
7256 audio_patch_handle_t *handle)
7257{
7258 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07007259
7260 // store new device and send to effects
7261 mInDevice = patch->sources[0].ext.device.type;
Eric Laurent296fb132015-05-01 11:38:42 -07007262 mPatch = *patch;
Eric Laurent054d9d32015-04-24 08:48:48 -07007263 for (size_t i = 0; i < mEffectChains.size(); i++) {
7264 mEffectChains[i]->setDevice_l(mInDevice);
7265 }
7266
7267 // disable AEC and NS if the device is a BT SCO headset supporting those
7268 // pre processings
7269 if (mTracks.size() > 0) {
7270 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
7271 mAudioFlinger->btNrecIsOff();
7272 for (size_t i = 0; i < mTracks.size(); i++) {
7273 sp<RecordTrack> track = mTracks[i];
7274 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
7275 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
7276 }
7277 }
7278
7279 // store new source and send to effects
7280 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) {
7281 mAudioSource = patch->sinks[0].ext.mix.usecase.source;
Eric Laurent1c333e22014-05-20 10:48:17 -07007282 for (size_t i = 0; i < mEffectChains.size(); i++) {
Eric Laurent054d9d32015-04-24 08:48:48 -07007283 mEffectChains[i]->setAudioSource_l(mAudioSource);
Eric Laurent1c333e22014-05-20 10:48:17 -07007284 }
Eric Laurent054d9d32015-04-24 08:48:48 -07007285 }
Eric Laurent1c333e22014-05-20 10:48:17 -07007286
Eric Laurent054d9d32015-04-24 08:48:48 -07007287 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
Eric Laurent1c333e22014-05-20 10:48:17 -07007288 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice();
7289 status = hwDevice->create_audio_patch(hwDevice,
7290 patch->num_sources,
7291 patch->sources,
7292 patch->num_sinks,
7293 patch->sinks,
7294 handle);
7295 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07007296 char *address;
7297 if (strcmp(patch->sources[0].ext.device.address, "") != 0) {
7298 address = audio_device_address_to_parameter(
7299 patch->sources[0].ext.device.type,
7300 patch->sources[0].ext.device.address);
7301 } else {
7302 address = (char *)calloc(1, 1);
7303 }
7304 AudioParameter param = AudioParameter(String8(address));
7305 free(address);
7306 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING),
7307 (int)patch->sources[0].ext.device.type);
7308 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE),
7309 (int)patch->sinks[0].ext.mix.usecase.source);
7310 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7311 param.toString().string());
7312 *handle = AUDIO_PATCH_HANDLE_NONE;
Eric Laurent1c333e22014-05-20 10:48:17 -07007313 }
Eric Laurent054d9d32015-04-24 08:48:48 -07007314
Eric Laurente8726fe2015-06-26 09:39:24 -07007315 if (mInDevice != mPrevInDevice) {
7316 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
7317 mPrevInDevice = mInDevice;
7318 }
Eric Laurent296fb132015-05-01 11:38:42 -07007319
Eric Laurent1c333e22014-05-20 10:48:17 -07007320 return status;
7321}
7322
7323status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
7324{
7325 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07007326
7327 mInDevice = AUDIO_DEVICE_NONE;
7328
Eric Laurent1c333e22014-05-20 10:48:17 -07007329 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
7330 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice();
7331 status = hwDevice->release_audio_patch(hwDevice, handle);
7332 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07007333 AudioParameter param;
7334 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0);
7335 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7336 param.toString().string());
Eric Laurent1c333e22014-05-20 10:48:17 -07007337 }
7338 return status;
7339}
7340
Eric Laurent83b88082014-06-20 18:31:16 -07007341void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record)
7342{
7343 Mutex::Autolock _l(mLock);
7344 mTracks.add(record);
7345}
7346
7347void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record)
7348{
7349 Mutex::Autolock _l(mLock);
7350 destroyTrack_l(record);
7351}
7352
7353void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config)
7354{
7355 ThreadBase::getAudioPortConfig(config);
7356 config->role = AUDIO_PORT_ROLE_SINK;
7357 config->ext.mix.hw_module = mInput->audioHwDev->handle();
7358 config->ext.mix.usecase.source = mAudioSource;
7359}
Eric Laurent1c333e22014-05-20 10:48:17 -07007360
Glenn Kasten63238ef2015-03-02 15:50:29 -08007361} // namespace android