blob: cab5517791961ddc0d53c1534da5dfb668347236 [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
Chih-Hung Hsiehbf291732016-05-17 15:16:07 -0700100#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
Andy Hungd330ee42015-04-20 13:23:41 -0700101#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),
Andy Hung69488c42016-05-16 18:43:33 -07001579 mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
Eric Laurent81784c32012-11-19 14:55:58 -08001580 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{
Eric Laurent81784c32012-11-19 14:55:58 -08002540 mInWrite = true;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002541 ssize_t bytesWritten;
Andy Hung010a1a12014-03-13 13:57:33 -07002542 const size_t offset = mCurrentWriteLength - mBytesRemaining;
Eric Laurent81784c32012-11-19 14:55:58 -08002543
2544 // If an NBAIO sink is present, use it to write the normal mixer's submix
2545 if (mNormalSink != 0) {
Glenn Kasten4c053ea2014-09-28 14:41:07 -07002546
Andy Hung010a1a12014-03-13 13:57:33 -07002547 const size_t count = mBytesRemaining / mFrameSize;
2548
Simon Wilson2d590962012-11-29 15:18:50 -08002549 ATRACE_BEGIN("write");
Eric Laurent81784c32012-11-19 14:55:58 -08002550 // update the setpoint when AudioFlinger::mScreenState changes
2551 uint32_t screenState = AudioFlinger::mScreenState;
2552 if (screenState != mScreenState) {
2553 mScreenState = screenState;
2554 MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
2555 if (pipe != NULL) {
2556 pipe->setAvgFrames((mScreenState & 1) ?
2557 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
2558 }
2559 }
Andy Hung010a1a12014-03-13 13:57:33 -07002560 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
Simon Wilson2d590962012-11-29 15:18:50 -08002561 ATRACE_END();
Eric Laurent81784c32012-11-19 14:55:58 -08002562 if (framesWritten > 0) {
Andy Hung010a1a12014-03-13 13:57:33 -07002563 bytesWritten = framesWritten * mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08002564 } else {
2565 bytesWritten = framesWritten;
2566 }
2567 // otherwise use the HAL / AudioStreamOut directly
2568 } else {
Eric Laurentbfb1b832013-01-07 09:53:42 -08002569 // Direct output and offload threads
Andy Hung010a1a12014-03-13 13:57:33 -07002570
Eric Laurentbfb1b832013-01-07 09:53:42 -08002571 if (mUseAsyncWrite) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07002572 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
2573 mWriteAckSequence += 2;
2574 mWriteAckSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002575 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002576 mCallbackThread->setWriteBlocked(mWriteAckSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002577 }
Glenn Kasten767094d2013-08-23 13:51:43 -07002578 // FIXME We should have an implementation of timestamps for direct output threads.
2579 // They are used e.g for multichannel PCM playback over HDMI.
Phil Burk062e67a2015-02-11 13:40:50 -08002580 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
Eric Laurent51716182016-02-29 18:00:56 -08002581
Eric Laurentbfb1b832013-01-07 09:53:42 -08002582 if (mUseAsyncWrite &&
2583 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
2584 // do not wait for async callback in case of error of full write
Eric Laurent3b4529e2013-09-05 18:09:19 -07002585 mWriteAckSequence &= ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002586 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002587 mCallbackThread->setWriteBlocked(mWriteAckSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002588 }
Eric Laurent81784c32012-11-19 14:55:58 -08002589 }
2590
Eric Laurent81784c32012-11-19 14:55:58 -08002591 mNumWrites++;
2592 mInWrite = false;
Eric Laurentfd477972013-10-25 18:10:40 -07002593 mStandby = false;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002594 return bytesWritten;
2595}
2596
2597void AudioFlinger::PlaybackThread::threadLoop_drain()
2598{
2599 if (mOutput->stream->drain) {
2600 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full");
2601 if (mUseAsyncWrite) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07002602 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request");
2603 mDrainSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002604 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002605 mCallbackThread->setDraining(mDrainSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002606 }
2607 mOutput->stream->drain(mOutput->stream,
2608 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY
2609 : AUDIO_DRAIN_ALL);
2610 }
2611}
2612
2613void AudioFlinger::PlaybackThread::threadLoop_exit()
2614{
Eric Laurent275e8e92014-11-30 15:14:47 -08002615 {
2616 Mutex::Autolock _l(mLock);
2617 for (size_t i = 0; i < mTracks.size(); i++) {
2618 sp<Track> track = mTracks[i];
2619 track->invalidate();
2620 }
2621 }
Eric Laurent81784c32012-11-19 14:55:58 -08002622}
2623
2624/*
2625The derived values that are cached:
Andy Hung25c2dac2014-02-27 14:56:00 -08002626 - mSinkBufferSize from frame count * frame size
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002627 - mActiveSleepTimeUs from activeSleepTimeUs()
2628 - mIdleSleepTimeUs from idleSleepTimeUs()
Eric Laurent42537be2016-01-08 17:16:42 -08002629 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least
2630 kDefaultStandbyTimeInNsecs when connected to an A2DP device.
Eric Laurent81784c32012-11-19 14:55:58 -08002631 - maxPeriod from frame count and sample rate (MIXER only)
2632
2633The parameters that affect these derived values are:
2634 - frame count
2635 - frame size
2636 - sample rate
2637 - device type: A2DP or not
2638 - device latency
2639 - format: PCM or not
2640 - active sleep time
2641 - idle sleep time
2642*/
2643
2644void AudioFlinger::PlaybackThread::cacheParameters_l()
2645{
Andy Hung25c2dac2014-02-27 14:56:00 -08002646 mSinkBufferSize = mNormalFrameCount * mFrameSize;
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002647 mActiveSleepTimeUs = activeSleepTimeUs();
2648 mIdleSleepTimeUs = idleSleepTimeUs();
Eric Laurent42537be2016-01-08 17:16:42 -08002649
2650 // make sure standby delay is not too short when connected to an A2DP sink to avoid
2651 // truncating audio when going to standby.
2652 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs;
2653 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) {
2654 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) {
2655 mStandbyDelayNs = kDefaultStandbyTimeInNsecs;
2656 }
2657 }
Eric Laurent81784c32012-11-19 14:55:58 -08002658}
2659
Eric Laurent13084622016-05-17 10:51:49 -07002660bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
Eric Laurent81784c32012-11-19 14:55:58 -08002661{
Glenn Kastenc42e9b42016-03-21 11:35:03 -07002662 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu",
Eric Laurent81784c32012-11-19 14:55:58 -08002663 this, streamType, mTracks.size());
Eric Laurent13084622016-05-17 10:51:49 -07002664 bool trackMatch = false;
Eric Laurent81784c32012-11-19 14:55:58 -08002665 size_t size = mTracks.size();
2666 for (size_t i = 0; i < size; i++) {
2667 sp<Track> t = mTracks[i];
Eric Laurentd60560a2015-04-10 11:31:20 -07002668 if (t->streamType() == streamType && t->isExternalTrack()) {
Glenn Kasten5736c352012-12-04 12:12:34 -08002669 t->invalidate();
Eric Laurent13084622016-05-17 10:51:49 -07002670 trackMatch = true;
Eric Laurent81784c32012-11-19 14:55:58 -08002671 }
2672 }
Eric Laurent13084622016-05-17 10:51:49 -07002673 return trackMatch;
Eric Laurent81784c32012-11-19 14:55:58 -08002674}
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();
Andy Hung69488c42016-05-16 18:43:33 -07002835 nsecs_t lastWriteFinished = -1; // time last server write completed
2836 int64_t lastFramesWritten = -1; // track changes in timestamp server frames written
Eric Laurent81784c32012-11-19 14:55:58 -08002837
2838 // MIXER
2839 nsecs_t lastWarning = 0;
2840
2841 // DUPLICATING
2842 // FIXME could this be made local to while loop?
2843 writeFrames = 0;
2844
Marco Nelissen462fd2f2013-01-14 14:12:05 -08002845 int lastGeneration = 0;
2846
Eric Laurent81784c32012-11-19 14:55:58 -08002847 cacheParameters_l();
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002848 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08002849
2850 if (mType == MIXER) {
2851 sleepTimeShift = 0;
2852 }
2853
2854 CpuStats cpuStats;
2855 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
2856
2857 acquireWakeLock();
2858
Glenn Kasten9e58b552013-01-18 15:09:48 -08002859 // mNBLogWriter->log can only be called while thread mutex mLock is held.
2860 // So if you need to log when mutex is unlocked, set logString to a non-NULL string,
2861 // and then that string will be logged at the next convenient opportunity.
2862 const char *logString = NULL;
2863
Eric Laurent664539d2013-09-23 18:24:31 -07002864 checkSilentMode_l();
2865
Eric Laurent81784c32012-11-19 14:55:58 -08002866 while (!exitPending())
2867 {
2868 cpuStats.sample(myName);
2869
2870 Vector< sp<EffectChain> > effectChains;
2871
Eric Laurent81784c32012-11-19 14:55:58 -08002872 { // scope for mLock
2873
2874 Mutex::Autolock _l(mLock);
2875
Eric Laurent021cf962014-05-13 10:18:14 -07002876 processConfigEvents_l();
Eric Laurent10351942014-05-08 18:49:52 -07002877
Glenn Kasten9e58b552013-01-18 15:09:48 -08002878 if (logString != NULL) {
2879 mNBLogWriter->logTimestamp();
2880 mNBLogWriter->log(logString);
2881 logString = NULL;
2882 }
2883
Glenn Kasten4c053ea2014-09-28 14:41:07 -07002884 // Gather the framesReleased counters for all active tracks,
Andy Hunge10393e2015-06-12 13:59:33 -07002885 // and associate with the sink frames written out. We need
2886 // this to convert the sink timestamp to the track timestamp.
Andy Hung69488c42016-05-16 18:43:33 -07002887 bool kernelLocationUpdate = false;
Andy Hunge10393e2015-06-12 13:59:33 -07002888 if (mNormalSink != 0) {
Andy Hungc54b1ff2016-02-23 14:07:07 -08002889 // Note: The DuplicatingThread may not have a mNormalSink.
Andy Hung818e7a32016-02-16 18:08:07 -08002890 // We always fetch the timestamp here because often the downstream
Andy Hung69488c42016-05-16 18:43:33 -07002891 // sink will block while writing.
Andy Hung818e7a32016-02-16 18:08:07 -08002892 ExtendedTimestamp timestamp; // use private copy to fetch
2893 (void) mNormalSink->getTimestamp(timestamp);
Andy Hung6d7b1192016-05-07 22:59:48 -07002894
2895 // We keep track of the last valid kernel position in case we are in underrun
2896 // and the normal mixer period is the same as the fast mixer period, or there
2897 // is some error from the HAL.
2898 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
2899 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
2900 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
2901 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
2902 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
2903
2904 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
2905 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER];
2906 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
2907 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung69488c42016-05-16 18:43:33 -07002908 }
2909
2910 if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
2911 kernelLocationUpdate = true;
Andy Hung6d7b1192016-05-07 22:59:48 -07002912 } else {
2913 ALOGV("getTimestamp error - no valid kernel position");
2914 }
2915
Andy Hung818e7a32016-02-16 18:08:07 -08002916 // copy over kernel info
2917 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
2918 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
2919 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
2920 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
Andy Hungc54b1ff2016-02-23 14:07:07 -08002921 }
2922 // mFramesWritten for non-offloaded tracks are contiguous
2923 // even after standby() is called. This is useful for the track frame
2924 // to sink frame mapping.
Andy Hung69488c42016-05-16 18:43:33 -07002925 bool serverLocationUpdate = false;
2926 if (mFramesWritten != lastFramesWritten) {
2927 serverLocationUpdate = true;
2928 lastFramesWritten = mFramesWritten;
2929 }
2930 // Only update timestamps if there is a meaningful change.
2931 // Either the kernel timestamp must be valid or we have written something.
2932 if (kernelLocationUpdate || serverLocationUpdate) {
2933 if (serverLocationUpdate) {
2934 // use the time before we called the HAL write - it is a bit more accurate
2935 // to when the server last read data than the current time here.
2936 //
2937 // If we haven't written anything, mLastWriteTime will be -1
2938 // and we use systemTime().
2939 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten;
2940 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1
2941 ? systemTime() : mLastWriteTime;
2942 }
2943 const size_t size = mActiveTracks.size();
2944 for (size_t i = 0; i < size; ++i) {
2945 sp<Track> t = mActiveTracks[i].promote();
2946 if (t != 0 && !t->isFastTrack()) {
2947 t->updateTrackFrameInfo(
2948 t->mAudioTrackServerProxy->framesReleased(),
2949 mFramesWritten,
2950 mTimestamp);
2951 }
Andy Hunge10393e2015-06-12 13:59:33 -07002952 }
Glenn Kastenbd096fd2013-08-23 13:53:56 -07002953 }
2954
Eric Laurent81784c32012-11-19 14:55:58 -08002955 saveOutputTracks();
Eric Laurentbfb1b832013-01-07 09:53:42 -08002956 if (mSignalPending) {
2957 // A signal was raised while we were unlocked
2958 mSignalPending = false;
2959 } else if (waitingAsyncCallback_l()) {
2960 if (exitPending()) {
2961 break;
2962 }
Marco Nelissen078538c2015-05-12 09:17:57 -07002963 bool released = false;
Eric Laurent64667972016-03-30 18:19:46 -07002964 if (!keepWakeLock()) {
Marco Nelissen078538c2015-05-12 09:17:57 -07002965 releaseWakeLock_l();
2966 released = true;
2967 }
Marco Nelissen462fd2f2013-01-14 14:12:05 -08002968 mWakeLockUids.clear();
2969 mActiveTracksGeneration++;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002970 ALOGV("wait async completion");
2971 mWaitWorkCV.wait(mLock);
2972 ALOGV("async completion/wake");
Marco Nelissen078538c2015-05-12 09:17:57 -07002973 if (released) {
2974 acquireWakeLock_l();
2975 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002976 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
2977 mSleepTimeUs = 0;
Eric Laurentede6c3b2013-09-19 14:37:46 -07002978
2979 continue;
2980 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002981 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) ||
Eric Laurentbfb1b832013-01-07 09:53:42 -08002982 isSuspended()) {
2983 // put audio hardware into standby after short delay
2984 if (shouldStandby_l()) {
Eric Laurent81784c32012-11-19 14:55:58 -08002985
2986 threadLoop_standby();
2987
2988 mStandby = true;
2989 }
2990
2991 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
2992 // we're about to wait, flush the binder command buffer
2993 IPCThreadState::self()->flushCommands();
2994
2995 clearOutputTracks();
2996
2997 if (exitPending()) {
2998 break;
2999 }
3000
3001 releaseWakeLock_l();
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003002 mWakeLockUids.clear();
3003 mActiveTracksGeneration++;
Eric Laurent81784c32012-11-19 14:55:58 -08003004 // wait until we have something to do...
3005 ALOGV("%s going to sleep", myName.string());
3006 mWaitWorkCV.wait(mLock);
3007 ALOGV("%s waking up", myName.string());
3008 acquireWakeLock_l();
3009
3010 mMixerStatus = MIXER_IDLE;
3011 mMixerStatusIgnoringFastTracks = MIXER_IDLE;
3012 mBytesWritten = 0;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003013 mBytesRemaining = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08003014 checkSilentMode_l();
3015
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003016 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
3017 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08003018 if (mType == MIXER) {
3019 sleepTimeShift = 0;
3020 }
3021
3022 continue;
3023 }
3024 }
Eric Laurent81784c32012-11-19 14:55:58 -08003025 // mMixerStatusIgnoringFastTracks is also updated internally
3026 mMixerStatus = prepareTracks_l(&tracksToRemove);
3027
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003028 // compare with previously applied list
3029 if (lastGeneration != mActiveTracksGeneration) {
3030 // update wakelock
3031 updateWakeLockUids_l(mWakeLockUids);
3032 lastGeneration = mActiveTracksGeneration;
3033 }
3034
Eric Laurent81784c32012-11-19 14:55:58 -08003035 // prevent any changes in effect chain list and in each effect chain
3036 // during mixing and effect process as the audio buffers could be deleted
3037 // or modified if an effect is created or deleted
3038 lockEffectChains_l(effectChains);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003039 } // mLock scope ends
Eric Laurent81784c32012-11-19 14:55:58 -08003040
Eric Laurentbfb1b832013-01-07 09:53:42 -08003041 if (mBytesRemaining == 0) {
3042 mCurrentWriteLength = 0;
3043 if (mMixerStatus == MIXER_TRACKS_READY) {
3044 // threadLoop_mix() sets mCurrentWriteLength
3045 threadLoop_mix();
3046 } else if ((mMixerStatus != MIXER_DRAIN_TRACK)
3047 && (mMixerStatus != MIXER_DRAIN_ALL)) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003048 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data
Eric Laurentbfb1b832013-01-07 09:53:42 -08003049 // must be written to HAL
3050 threadLoop_sleepTime();
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003051 if (mSleepTimeUs == 0) {
Andy Hung25c2dac2014-02-27 14:56:00 -08003052 mCurrentWriteLength = mSinkBufferSize;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003053 }
3054 }
Andy Hung98ef9782014-03-04 14:46:50 -08003055 // Either threadLoop_mix() or threadLoop_sleepTime() should have set
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003056 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0.
Andy Hung98ef9782014-03-04 14:46:50 -08003057 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid)
3058 // or mSinkBuffer (if there are no effects).
3059 //
3060 // This is done pre-effects computation; if effects change to
3061 // support higher precision, this needs to move.
3062 //
3063 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003064 // TODO use mSleepTimeUs == 0 as an additional condition.
Andy Hung98ef9782014-03-04 14:46:50 -08003065 if (mMixerBufferValid) {
3066 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
3067 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
3068
Andy Hung2ddee192015-12-18 17:34:44 -08003069 // mono blend occurs for mixer threads only (not direct or offloaded)
3070 // and is handled here if we're going directly to the sink.
3071 if (requireMonoBlend() && !mEffectBufferValid) {
Glenn Kasten03c48d52016-01-27 17:25:17 -08003072 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount,
3073 true /*limit*/);
Andy Hung2ddee192015-12-18 17:34:44 -08003074 }
3075
Andy Hung98ef9782014-03-04 14:46:50 -08003076 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
3077 mNormalFrameCount * mChannelCount);
3078 }
3079
Eric Laurentbfb1b832013-01-07 09:53:42 -08003080 mBytesRemaining = mCurrentWriteLength;
3081 if (isSuspended()) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003082 mSleepTimeUs = suspendSleepTimeUs();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003083 // simulate write to HAL when suspended
Andy Hung25c2dac2014-02-27 14:56:00 -08003084 mBytesWritten += mSinkBufferSize;
Andy Hungc54b1ff2016-02-23 14:07:07 -08003085 mFramesWritten += mSinkBufferSize / mFrameSize;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003086 mBytesRemaining = 0;
3087 }
Eric Laurent81784c32012-11-19 14:55:58 -08003088
Eric Laurentbfb1b832013-01-07 09:53:42 -08003089 // only process effects if we're going to write
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003090 if (mSleepTimeUs == 0 && mType != OFFLOAD) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08003091 for (size_t i = 0; i < effectChains.size(); i ++) {
3092 effectChains[i]->process_l();
3093 }
Eric Laurent81784c32012-11-19 14:55:58 -08003094 }
3095 }
Eric Laurent59fe0102013-09-27 18:48:26 -07003096 // Process effect chains for offloaded thread even if no audio
3097 // was read from audio track: process only updates effect state
3098 // and thus does have to be synchronized with audio writes but may have
3099 // to be called while waiting for async write callback
3100 if (mType == OFFLOAD) {
3101 for (size_t i = 0; i < effectChains.size(); i ++) {
3102 effectChains[i]->process_l();
3103 }
3104 }
Eric Laurent81784c32012-11-19 14:55:58 -08003105
Andy Hung98ef9782014-03-04 14:46:50 -08003106 // Only if the Effects buffer is enabled and there is data in the
3107 // Effects buffer (buffer valid), we need to
3108 // copy into the sink buffer.
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003109 // TODO use mSleepTimeUs == 0 as an additional condition.
Andy Hung98ef9782014-03-04 14:46:50 -08003110 if (mEffectBufferValid) {
3111 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat);
Andy Hung2ddee192015-12-18 17:34:44 -08003112
3113 if (requireMonoBlend()) {
Glenn Kasten03c48d52016-01-27 17:25:17 -08003114 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount,
3115 true /*limit*/);
Andy Hung2ddee192015-12-18 17:34:44 -08003116 }
3117
Andy Hung98ef9782014-03-04 14:46:50 -08003118 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat,
3119 mNormalFrameCount * mChannelCount);
3120 }
3121
Eric Laurent81784c32012-11-19 14:55:58 -08003122 // enable changes in effect chain
3123 unlockEffectChains(effectChains);
3124
Eric Laurentbfb1b832013-01-07 09:53:42 -08003125 if (!waitingAsyncCallback()) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003126 // mSleepTimeUs == 0 means we must write to audio hardware
3127 if (mSleepTimeUs == 0) {
Andy Hung08fb1742015-05-31 23:22:10 -07003128 ssize_t ret = 0;
Andy Hung69488c42016-05-16 18:43:33 -07003129 // We save lastWriteFinished here, as previousLastWriteFinished,
3130 // for throttling. On thread start, previousLastWriteFinished will be
3131 // set to -1, which properly results in no throttling after the first write.
3132 nsecs_t previousLastWriteFinished = lastWriteFinished;
3133 nsecs_t delta = 0;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003134 if (mBytesRemaining) {
Andy Hung69488c42016-05-16 18:43:33 -07003135 // FIXME rewrite to reduce number of system calls
3136 mLastWriteTime = systemTime(); // also used for dumpsys
Andy Hung08fb1742015-05-31 23:22:10 -07003137 ret = threadLoop_write();
Andy Hung69488c42016-05-16 18:43:33 -07003138 lastWriteFinished = systemTime();
3139 delta = lastWriteFinished - mLastWriteTime;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003140 if (ret < 0) {
3141 mBytesRemaining = 0;
3142 } else {
3143 mBytesWritten += ret;
3144 mBytesRemaining -= ret;
Andy Hungc54b1ff2016-02-23 14:07:07 -08003145 mFramesWritten += ret / mFrameSize;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003146 }
3147 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||
3148 (mMixerStatus == MIXER_DRAIN_ALL)) {
3149 threadLoop_drain();
Eric Laurent81784c32012-11-19 14:55:58 -08003150 }
Andy Hung08fb1742015-05-31 23:22:10 -07003151 if (mType == MIXER && !mStandby) {
Glenn Kasten4944acb2013-08-19 08:39:20 -07003152 // write blocked detection
Andy Hung08fb1742015-05-31 23:22:10 -07003153 if (delta > maxPeriod) {
Glenn Kasten4944acb2013-08-19 08:39:20 -07003154 mNumDelayedWrites++;
Andy Hung69488c42016-05-16 18:43:33 -07003155 if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) {
Glenn Kasten4944acb2013-08-19 08:39:20 -07003156 ATRACE_NAME("underrun");
3157 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
Glenn Kastenc42e9b42016-03-21 11:35:03 -07003158 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this);
Andy Hung69488c42016-05-16 18:43:33 -07003159 lastWarning = lastWriteFinished;
Glenn Kasten4944acb2013-08-19 08:39:20 -07003160 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003161 }
Andy Hung08fb1742015-05-31 23:22:10 -07003162
3163 if (mThreadThrottle
3164 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks)
3165 && ret > 0) { // we wrote something
3166 // Limit MixerThread data processing to no more than twice the
3167 // expected processing rate.
3168 //
3169 // This helps prevent underruns with NuPlayer and other applications
3170 // which may set up buffers that are close to the minimum size, or use
3171 // deep buffers, and rely on a double-buffering sleep strategy to fill.
3172 //
3173 // The throttle smooths out sudden large data drains from the device,
3174 // e.g. when it comes out of standby, which often causes problems with
3175 // (1) mixer threads without a fast mixer (which has its own warm-up)
3176 // (2) minimum buffer sized tracks (even if the track is full,
3177 // the app won't fill fast enough to handle the sudden draw).
3178
Andy Hung69488c42016-05-16 18:43:33 -07003179 // it's OK if deltaMs is an overestimate.
3180 const int32_t deltaMs =
3181 (lastWriteFinished - previousLastWriteFinished) / 1000000;
Andy Hung08fb1742015-05-31 23:22:10 -07003182 const int32_t throttleMs = mHalfBufferMs - deltaMs;
3183 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) {
3184 usleep(throttleMs * 1000);
Andy Hung40eb1a12015-06-18 13:42:02 -07003185 // notify of throttle start on verbose log
3186 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs,
3187 "mixer(%p) throttle begin:"
3188 " ret(%zd) deltaMs(%d) requires sleep %d ms",
Andy Hung08fb1742015-05-31 23:22:10 -07003189 this, ret, deltaMs, throttleMs);
Andy Hung40eb1a12015-06-18 13:42:02 -07003190 mThreadThrottleTimeMs += throttleMs;
3191 } else {
3192 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs;
3193 if (diff > 0) {
3194 // notify of throttle end on debug log
Andy Hung3ea004d2016-05-05 16:48:37 -07003195 // but prevent spamming for bluetooth
3196 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()),
3197 "mixer(%p) throttle end: throttle time(%u)", this, diff);
Andy Hung40eb1a12015-06-18 13:42:02 -07003198 mThreadThrottleEndMs = mThreadThrottleTimeMs;
3199 }
Andy Hung08fb1742015-05-31 23:22:10 -07003200 }
3201 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003202 }
Eric Laurent81784c32012-11-19 14:55:58 -08003203
Eric Laurentbfb1b832013-01-07 09:53:42 -08003204 } else {
Glenn Kastene7754022014-10-31 12:11:26 -07003205 ATRACE_BEGIN("sleep");
Eric Laurente93cc032016-05-05 10:15:10 -07003206 Mutex::Autolock _l(mLock);
3207 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) {
3208 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs));
Eric Laurent51716182016-02-29 18:00:56 -08003209 }
Glenn Kastene7754022014-10-31 12:11:26 -07003210 ATRACE_END();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003211 }
Eric Laurent81784c32012-11-19 14:55:58 -08003212 }
3213
3214 // Finally let go of removed track(s), without the lock held
3215 // since we can't guarantee the destructors won't acquire that
3216 // same lock. This will also mutate and push a new fast mixer state.
3217 threadLoop_removeTracks(tracksToRemove);
3218 tracksToRemove.clear();
3219
3220 // FIXME I don't understand the need for this here;
3221 // it was in the original code but maybe the
3222 // assignment in saveOutputTracks() makes this unnecessary?
3223 clearOutputTracks();
3224
3225 // Effect chains will be actually deleted here if they were removed from
3226 // mEffectChains list during mixing or effects processing
3227 effectChains.clear();
3228
3229 // FIXME Note that the above .clear() is no longer necessary since effectChains
3230 // is now local to this block, but will keep it for now (at least until merge done).
3231 }
3232
Eric Laurentbfb1b832013-01-07 09:53:42 -08003233 threadLoop_exit();
3234
Eric Laurentcf817a22014-08-04 20:36:31 -07003235 if (!mStandby) {
3236 threadLoop_standby();
3237 mStandby = true;
Eric Laurent81784c32012-11-19 14:55:58 -08003238 }
3239
3240 releaseWakeLock();
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003241 mWakeLockUids.clear();
3242 mActiveTracksGeneration++;
Eric Laurent81784c32012-11-19 14:55:58 -08003243
3244 ALOGV("Thread %p type %d exiting", this, mType);
3245 return false;
3246}
3247
Eric Laurentbfb1b832013-01-07 09:53:42 -08003248// removeTracks_l() must be called with ThreadBase::mLock held
3249void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
3250{
3251 size_t count = tracksToRemove.size();
Glenn Kasten34fca342013-08-13 09:48:14 -07003252 if (count > 0) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08003253 for (size_t i=0 ; i<count ; i++) {
3254 const sp<Track>& track = tracksToRemove.itemAt(i);
3255 mActiveTracks.remove(track);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003256 mWakeLockUids.remove(track->uid());
3257 mActiveTracksGeneration++;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003258 ALOGV("removeTracks_l removing track on session %d", track->sessionId());
3259 sp<EffectChain> chain = getEffectChain_l(track->sessionId());
3260 if (chain != 0) {
3261 ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
3262 track->sessionId());
3263 chain->decActiveTrackCnt();
3264 }
3265 if (track->isTerminated()) {
3266 removeTrack_l(track);
3267 }
3268 }
3269 }
3270
3271}
Eric Laurent81784c32012-11-19 14:55:58 -08003272
Eric Laurentaccc1472013-09-20 09:36:34 -07003273status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
3274{
3275 if (mNormalSink != 0) {
Andy Hung818e7a32016-02-16 18:08:07 -08003276 ExtendedTimestamp ets;
3277 status_t status = mNormalSink->getTimestamp(ets);
3278 if (status == NO_ERROR) {
3279 status = ets.getBestTimestamp(&timestamp);
3280 }
3281 return status;
Eric Laurentaccc1472013-09-20 09:36:34 -07003282 }
Andy Hung9a1c8892014-12-03 11:37:42 -08003283 if ((mType == OFFLOAD || mType == DIRECT)
3284 && mOutput != NULL && mOutput->stream->get_presentation_position) {
Eric Laurentaccc1472013-09-20 09:36:34 -07003285 uint64_t position64;
Phil Burk062e67a2015-02-11 13:40:50 -08003286 int ret = mOutput->getPresentationPosition(&position64, &timestamp.mTime);
Eric Laurentaccc1472013-09-20 09:36:34 -07003287 if (ret == 0) {
3288 timestamp.mPosition = (uint32_t)position64;
3289 return NO_ERROR;
3290 }
3291 }
3292 return INVALID_OPERATION;
3293}
Eric Laurent1c333e22014-05-20 10:48:17 -07003294
Eric Laurent054d9d32015-04-24 08:48:48 -07003295status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch,
3296 audio_patch_handle_t *handle)
3297{
Glenn Kastenc05b8d72016-03-24 09:48:17 -07003298 AutoPark<FastMixer> park(mFastMixer);
Eric Laurent054d9d32015-04-24 08:48:48 -07003299
Glenn Kastenc05b8d72016-03-24 09:48:17 -07003300 status_t status = PlaybackThread::createAudioPatch_l(patch, handle);
Eric Laurent054d9d32015-04-24 08:48:48 -07003301
3302 return status;
3303}
3304
Eric Laurent1c333e22014-05-20 10:48:17 -07003305status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
3306 audio_patch_handle_t *handle)
3307{
3308 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07003309
3310 // store new device and send to effects
3311 audio_devices_t type = AUDIO_DEVICE_NONE;
3312 for (unsigned int i = 0; i < patch->num_sinks; i++) {
3313 type |= patch->sinks[i].ext.device.type;
3314 }
3315
3316#ifdef ADD_BATTERY_DATA
3317 // when changing the audio output device, call addBatteryData to notify
3318 // the change
3319 if (mOutDevice != type) {
3320 uint32_t params = 0;
3321 // check whether speaker is on
3322 if (type & AUDIO_DEVICE_OUT_SPEAKER) {
3323 params |= IMediaPlayerService::kBatteryDataSpeakerOn;
Eric Laurent1c333e22014-05-20 10:48:17 -07003324 }
3325
Eric Laurent054d9d32015-04-24 08:48:48 -07003326 audio_devices_t deviceWithoutSpeaker
3327 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
3328 // check if any other device (except speaker) is on
3329 if (type & deviceWithoutSpeaker) {
3330 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
3331 }
3332
3333 if (params != 0) {
3334 addBatteryData(params);
3335 }
3336 }
3337#endif
3338
3339 for (size_t i = 0; i < mEffectChains.size(); i++) {
3340 mEffectChains[i]->setDevice_l(type);
3341 }
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07003342
3343 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when
3344 // the thread is created so that the first patch creation triggers an ioConfigChanged callback
3345 bool configChanged = mPrevOutDevice != type;
Eric Laurent054d9d32015-04-24 08:48:48 -07003346 mOutDevice = type;
Eric Laurent296fb132015-05-01 11:38:42 -07003347 mPatch = *patch;
Eric Laurent054d9d32015-04-24 08:48:48 -07003348
3349 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
Eric Laurent1c333e22014-05-20 10:48:17 -07003350 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
3351 status = hwDevice->create_audio_patch(hwDevice,
3352 patch->num_sources,
3353 patch->sources,
3354 patch->num_sinks,
3355 patch->sinks,
3356 handle);
3357 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07003358 char *address;
3359 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) {
3360 //FIXME: we only support address on first sink with HAL version < 3.0
3361 address = audio_device_address_to_parameter(
3362 patch->sinks[0].ext.device.type,
3363 patch->sinks[0].ext.device.address);
3364 } else {
3365 address = (char *)calloc(1, 1);
3366 }
3367 AudioParameter param = AudioParameter(String8(address));
3368 free(address);
3369 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type);
3370 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3371 param.toString().string());
3372 *handle = AUDIO_PATCH_HANDLE_NONE;
Eric Laurent1c333e22014-05-20 10:48:17 -07003373 }
Eric Laurente8726fe2015-06-26 09:39:24 -07003374 if (configChanged) {
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07003375 mPrevOutDevice = type;
Eric Laurente8726fe2015-06-26 09:39:24 -07003376 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
3377 }
Eric Laurent1c333e22014-05-20 10:48:17 -07003378 return status;
3379}
3380
Eric Laurent054d9d32015-04-24 08:48:48 -07003381status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
3382{
Glenn Kastenc05b8d72016-03-24 09:48:17 -07003383 AutoPark<FastMixer> park(mFastMixer);
Eric Laurent054d9d32015-04-24 08:48:48 -07003384
3385 status_t status = PlaybackThread::releaseAudioPatch_l(handle);
3386
Eric Laurent054d9d32015-04-24 08:48:48 -07003387 return status;
3388}
3389
Eric Laurent1c333e22014-05-20 10:48:17 -07003390status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
3391{
3392 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07003393
3394 mOutDevice = AUDIO_DEVICE_NONE;
3395
Eric Laurent1c333e22014-05-20 10:48:17 -07003396 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
3397 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice();
3398 status = hwDevice->release_audio_patch(hwDevice, handle);
3399 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07003400 AudioParameter param;
3401 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0);
3402 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3403 param.toString().string());
Eric Laurent1c333e22014-05-20 10:48:17 -07003404 }
3405 return status;
3406}
3407
Eric Laurent83b88082014-06-20 18:31:16 -07003408void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track)
3409{
3410 Mutex::Autolock _l(mLock);
3411 mTracks.add(track);
3412}
3413
3414void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track)
3415{
3416 Mutex::Autolock _l(mLock);
3417 destroyTrack_l(track);
3418}
3419
3420void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config)
3421{
3422 ThreadBase::getAudioPortConfig(config);
3423 config->role = AUDIO_PORT_ROLE_SOURCE;
3424 config->ext.mix.hw_module = mOutput->audioHwDev->handle();
3425 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
3426}
3427
Eric Laurent81784c32012-11-19 14:55:58 -08003428// ----------------------------------------------------------------------------
3429
3430AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
Eric Laurent72e3f392015-05-20 14:43:50 -07003431 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type)
3432 : PlaybackThread(audioFlinger, output, id, device, type, systemReady),
Eric Laurent81784c32012-11-19 14:55:58 -08003433 // mAudioMixer below
3434 // mFastMixer below
Andy Hung2ddee192015-12-18 17:34:44 -08003435 mFastMixerFutex(0),
3436 mMasterMono(false)
Eric Laurent81784c32012-11-19 14:55:58 -08003437 // mOutputSink below
3438 // mPipeSink below
3439 // mNormalSink below
3440{
3441 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
Glenn Kastenc42e9b42016-03-21 11:35:03 -07003442 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, "
3443 "mFrameCount=%zu, mNormalFrameCount=%zu",
Eric Laurent81784c32012-11-19 14:55:58 -08003444 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
3445 mNormalFrameCount);
3446 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
3447
Andy Hungfbfc3952015-01-15 13:33:51 -08003448 if (type == DUPLICATING) {
3449 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks
3450 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write().
3451 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink.
3452 return;
3453 }
Eric Laurent81784c32012-11-19 14:55:58 -08003454 // create an NBAIO sink for the HAL output stream, and negotiate
3455 mOutputSink = new AudioStreamOutSink(output->stream);
3456 size_t numCounterOffers = 0;
Glenn Kastenf69f9862014-03-07 08:37:57 -08003457 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003458#if !LOG_NDEBUG
3459 ssize_t index =
3460#else
3461 (void)
3462#endif
3463 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
Eric Laurent81784c32012-11-19 14:55:58 -08003464 ALOG_ASSERT(index == 0);
3465
3466 // initialize fast mixer depending on configuration
3467 bool initFastMixer;
3468 switch (kUseFastMixer) {
3469 case FastMixer_Never:
3470 initFastMixer = false;
3471 break;
3472 case FastMixer_Always:
3473 initFastMixer = true;
3474 break;
3475 case FastMixer_Static:
3476 case FastMixer_Dynamic:
3477 initFastMixer = mFrameCount < mNormalFrameCount;
3478 break;
3479 }
3480 if (initFastMixer) {
Andy Hung1258c1a2014-05-23 21:22:17 -07003481 audio_format_t fastMixerFormat;
3482 if (mMixerBufferEnabled && mEffectBufferEnabled) {
3483 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT;
3484 } else {
3485 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
3486 }
3487 if (mFormat != fastMixerFormat) {
3488 // change our Sink format to accept our intermediate precision
3489 mFormat = fastMixerFormat;
3490 free(mSinkBuffer);
3491 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
3492 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
3493 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
3494 }
Eric Laurent81784c32012-11-19 14:55:58 -08003495
3496 // create a MonoPipe to connect our submix to FastMixer
3497 NBAIO_Format format = mOutputSink->format();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003498#ifdef TEE_SINK
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003499 NBAIO_Format origformat = format;
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003500#endif
Andy Hung1258c1a2014-05-23 21:22:17 -07003501 // adjust format to match that of the Fast Mixer
Glenn Kasten97b7b752014-09-28 13:04:24 -07003502 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat);
Andy Hung1258c1a2014-05-23 21:22:17 -07003503 format.mFormat = fastMixerFormat;
3504 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
3505
Eric Laurent81784c32012-11-19 14:55:58 -08003506 // This pipe depth compensates for scheduling latency of the normal mixer thread.
3507 // When it wakes up after a maximum latency, it runs a few cycles quickly before
3508 // finally blocking. Note the pipe implementation rounds up the request to a power of 2.
3509 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
3510 const NBAIO_Format offers[1] = {format};
3511 size_t numCounterOffers = 0;
Glenn Kastenfc302fd2016-04-11 14:11:26 -07003512#if !LOG_NDEBUG || defined(TEE_SINK)
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003513 ssize_t index =
3514#else
3515 (void)
3516#endif
3517 monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
Eric Laurent81784c32012-11-19 14:55:58 -08003518 ALOG_ASSERT(index == 0);
3519 monoPipe->setAvgFrames((mScreenState & 1) ?
3520 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
3521 mPipeSink = monoPipe;
3522
Glenn Kasten46909e72013-02-26 09:20:22 -08003523#ifdef TEE_SINK
Glenn Kastenda6ef132013-01-10 12:31:01 -08003524 if (mTeeSinkOutputEnabled) {
3525 // create a Pipe to archive a copy of FastMixer's output for dumpsys
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003526 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat);
3527 const NBAIO_Format offers2[1] = {origformat};
Glenn Kastenda6ef132013-01-10 12:31:01 -08003528 numCounterOffers = 0;
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003529 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers);
Glenn Kastenda6ef132013-01-10 12:31:01 -08003530 ALOG_ASSERT(index == 0);
3531 mTeeSink = teeSink;
3532 PipeReader *teeSource = new PipeReader(*teeSink);
3533 numCounterOffers = 0;
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003534 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers);
Glenn Kastenda6ef132013-01-10 12:31:01 -08003535 ALOG_ASSERT(index == 0);
3536 mTeeSource = teeSource;
3537 }
Glenn Kasten46909e72013-02-26 09:20:22 -08003538#endif
Eric Laurent81784c32012-11-19 14:55:58 -08003539
3540 // create fast mixer and configure it initially with just one fast track for our submix
3541 mFastMixer = new FastMixer();
3542 FastMixerStateQueue *sq = mFastMixer->sq();
3543#ifdef STATE_QUEUE_DUMP
3544 sq->setObserverDump(&mStateQueueObserverDump);
3545 sq->setMutatorDump(&mStateQueueMutatorDump);
3546#endif
3547 FastMixerState *state = sq->begin();
3548 FastTrack *fastTrack = &state->mFastTracks[0];
3549 // wrap the source side of the MonoPipe to make it an AudioBufferProvider
3550 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
3551 fastTrack->mVolumeProvider = NULL;
Andy Hunge8a1ced2014-05-09 15:02:21 -07003552 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer
3553 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
Eric Laurent81784c32012-11-19 14:55:58 -08003554 fastTrack->mGeneration++;
3555 state->mFastTracksGen++;
3556 state->mTrackMask = 1;
3557 // fast mixer will use the HAL output sink
3558 state->mOutputSink = mOutputSink.get();
3559 state->mOutputSinkGen++;
3560 state->mFrameCount = mFrameCount;
3561 state->mCommand = FastMixerState::COLD_IDLE;
3562 // already done in constructor initialization list
3563 //mFastMixerFutex = 0;
3564 state->mColdFutexAddr = &mFastMixerFutex;
3565 state->mColdGen++;
3566 state->mDumpState = &mFastMixerDumpState;
Glenn Kasten46909e72013-02-26 09:20:22 -08003567#ifdef TEE_SINK
Eric Laurent81784c32012-11-19 14:55:58 -08003568 state->mTeeSink = mTeeSink.get();
Glenn Kasten46909e72013-02-26 09:20:22 -08003569#endif
Glenn Kasten9e58b552013-01-18 15:09:48 -08003570 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
3571 state->mNBLogWriter = mFastMixerNBLogWriter.get();
Eric Laurent81784c32012-11-19 14:55:58 -08003572 sq->end();
3573 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3574
3575 // start the fast mixer
3576 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
3577 pid_t tid = mFastMixer->getTid();
Eric Laurent72e3f392015-05-20 14:43:50 -07003578 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
Eric Laurent81784c32012-11-19 14:55:58 -08003579
3580#ifdef AUDIO_WATCHDOG
3581 // create and start the watchdog
3582 mAudioWatchdog = new AudioWatchdog();
3583 mAudioWatchdog->setDump(&mAudioWatchdogDump);
3584 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
3585 tid = mAudioWatchdog->getTid();
Eric Laurent72e3f392015-05-20 14:43:50 -07003586 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
Eric Laurent81784c32012-11-19 14:55:58 -08003587#endif
3588
Eric Laurent81784c32012-11-19 14:55:58 -08003589 }
3590
3591 switch (kUseFastMixer) {
3592 case FastMixer_Never:
3593 case FastMixer_Dynamic:
3594 mNormalSink = mOutputSink;
3595 break;
3596 case FastMixer_Always:
3597 mNormalSink = mPipeSink;
3598 break;
3599 case FastMixer_Static:
3600 mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
3601 break;
3602 }
3603}
3604
3605AudioFlinger::MixerThread::~MixerThread()
3606{
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003607 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003608 FastMixerStateQueue *sq = mFastMixer->sq();
3609 FastMixerState *state = sq->begin();
3610 if (state->mCommand == FastMixerState::COLD_IDLE) {
3611 int32_t old = android_atomic_inc(&mFastMixerFutex);
3612 if (old == -1) {
Elliott Hughesee499292014-05-21 17:55:51 -07003613 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
Eric Laurent81784c32012-11-19 14:55:58 -08003614 }
3615 }
3616 state->mCommand = FastMixerState::EXIT;
3617 sq->end();
3618 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3619 mFastMixer->join();
3620 // Though the fast mixer thread has exited, it's state queue is still valid.
3621 // We'll use that extract the final state which contains one remaining fast track
3622 // corresponding to our sub-mix.
3623 state = sq->begin();
3624 ALOG_ASSERT(state->mTrackMask == 1);
3625 FastTrack *fastTrack = &state->mFastTracks[0];
3626 ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
3627 delete fastTrack->mBufferProvider;
3628 sq->end(false /*didModify*/);
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003629 mFastMixer.clear();
Eric Laurent81784c32012-11-19 14:55:58 -08003630#ifdef AUDIO_WATCHDOG
3631 if (mAudioWatchdog != 0) {
3632 mAudioWatchdog->requestExit();
3633 mAudioWatchdog->requestExitAndWait();
3634 mAudioWatchdog.clear();
3635 }
3636#endif
3637 }
Glenn Kasten9e58b552013-01-18 15:09:48 -08003638 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter);
Eric Laurent81784c32012-11-19 14:55:58 -08003639 delete mAudioMixer;
3640}
3641
3642
3643uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
3644{
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003645 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003646 MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
3647 latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
3648 }
3649 return latency;
3650}
3651
3652
3653void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
3654{
3655 PlaybackThread::threadLoop_removeTracks(tracksToRemove);
3656}
3657
Eric Laurentbfb1b832013-01-07 09:53:42 -08003658ssize_t AudioFlinger::MixerThread::threadLoop_write()
Eric Laurent81784c32012-11-19 14:55:58 -08003659{
3660 // FIXME we should only do one push per cycle; confirm this is true
3661 // Start the fast mixer if it's not already running
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003662 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003663 FastMixerStateQueue *sq = mFastMixer->sq();
3664 FastMixerState *state = sq->begin();
3665 if (state->mCommand != FastMixerState::MIX_WRITE &&
3666 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
3667 if (state->mCommand == FastMixerState::COLD_IDLE) {
Eric Laurenta2ab4502015-09-09 12:25:51 -07003668
3669 // FIXME workaround for first HAL write being CPU bound on some devices
3670 ATRACE_BEGIN("write");
3671 mOutput->write((char *)mSinkBuffer, 0);
3672 ATRACE_END();
3673
Eric Laurent81784c32012-11-19 14:55:58 -08003674 int32_t old = android_atomic_inc(&mFastMixerFutex);
3675 if (old == -1) {
Elliott Hughesee499292014-05-21 17:55:51 -07003676 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
Eric Laurent81784c32012-11-19 14:55:58 -08003677 }
3678#ifdef AUDIO_WATCHDOG
3679 if (mAudioWatchdog != 0) {
3680 mAudioWatchdog->resume();
3681 }
3682#endif
3683 }
3684 state->mCommand = FastMixerState::MIX_WRITE;
Glenn Kastend797a9d2015-03-02 14:19:25 -08003685#ifdef FAST_THREAD_STATISTICS
Glenn Kasten4182c4e2013-07-15 14:45:07 -07003686 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
Glenn Kastenfbdb2ac2015-03-02 14:47:19 -08003687 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN);
Glenn Kastend797a9d2015-03-02 14:19:25 -08003688#endif
Eric Laurent81784c32012-11-19 14:55:58 -08003689 sq->end();
3690 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3691 if (kUseFastMixer == FastMixer_Dynamic) {
3692 mNormalSink = mPipeSink;
3693 }
3694 } else {
3695 sq->end(false /*didModify*/);
3696 }
3697 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003698 return PlaybackThread::threadLoop_write();
Eric Laurent81784c32012-11-19 14:55:58 -08003699}
3700
3701void AudioFlinger::MixerThread::threadLoop_standby()
3702{
3703 // Idle the fast mixer if it's currently running
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003704 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003705 FastMixerStateQueue *sq = mFastMixer->sq();
3706 FastMixerState *state = sq->begin();
3707 if (!(state->mCommand & FastMixerState::IDLE)) {
3708 state->mCommand = FastMixerState::COLD_IDLE;
3709 state->mColdFutexAddr = &mFastMixerFutex;
3710 state->mColdGen++;
3711 mFastMixerFutex = 0;
3712 sq->end();
3713 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
3714 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
3715 if (kUseFastMixer == FastMixer_Dynamic) {
3716 mNormalSink = mOutputSink;
3717 }
3718#ifdef AUDIO_WATCHDOG
3719 if (mAudioWatchdog != 0) {
3720 mAudioWatchdog->pause();
3721 }
3722#endif
3723 } else {
3724 sq->end(false /*didModify*/);
3725 }
3726 }
3727 PlaybackThread::threadLoop_standby();
3728}
3729
Eric Laurentbfb1b832013-01-07 09:53:42 -08003730bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
3731{
3732 return false;
3733}
3734
3735bool AudioFlinger::PlaybackThread::shouldStandby_l()
3736{
3737 return !mStandby;
3738}
3739
3740bool AudioFlinger::PlaybackThread::waitingAsyncCallback()
3741{
3742 Mutex::Autolock _l(mLock);
3743 return waitingAsyncCallback_l();
3744}
3745
Eric Laurent81784c32012-11-19 14:55:58 -08003746// shared by MIXER and DIRECT, overridden by DUPLICATING
3747void AudioFlinger::PlaybackThread::threadLoop_standby()
3748{
3749 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
Phil Burk062e67a2015-02-11 13:40:50 -08003750 mOutput->standby();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003751 if (mUseAsyncWrite != 0) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07003752 // discard any pending drain or write ack by incrementing sequence
3753 mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
3754 mDrainSequence = (mDrainSequence + 2) & ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003755 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07003756 mCallbackThread->setWriteBlocked(mWriteAckSequence);
3757 mCallbackThread->setDraining(mDrainSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08003758 }
Eric Laurentd1f69b02014-12-15 14:33:13 -08003759 mHwPaused = false;
Eric Laurent81784c32012-11-19 14:55:58 -08003760}
3761
Haynes Mathew George4c6a4332014-01-15 12:31:39 -08003762void AudioFlinger::PlaybackThread::onAddNewTrack_l()
3763{
3764 ALOGV("signal playback thread");
3765 broadcast_l();
3766}
3767
Eric Laurent81784c32012-11-19 14:55:58 -08003768void AudioFlinger::MixerThread::threadLoop_mix()
3769{
Eric Laurent81784c32012-11-19 14:55:58 -08003770 // mix buffers...
Glenn Kastend79072e2016-01-06 08:41:20 -08003771 mAudioMixer->process();
Andy Hung25c2dac2014-02-27 14:56:00 -08003772 mCurrentWriteLength = mSinkBufferSize;
Eric Laurent81784c32012-11-19 14:55:58 -08003773 // increase sleep time progressively when application underrun condition clears.
3774 // Only increase sleep time if the mixer is ready for two consecutive times to avoid
3775 // that a steady state of alternating ready/not ready conditions keeps the sleep time
3776 // such that we would underrun the audio HAL.
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003777 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) {
Eric Laurent81784c32012-11-19 14:55:58 -08003778 sleepTimeShift--;
3779 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003780 mSleepTimeUs = 0;
3781 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
Eric Laurent81784c32012-11-19 14:55:58 -08003782 //TODO: delay standby when effects have a tail
Glenn Kasten4c053ea2014-09-28 14:41:07 -07003783
Eric Laurent81784c32012-11-19 14:55:58 -08003784}
3785
3786void AudioFlinger::MixerThread::threadLoop_sleepTime()
3787{
3788 // If no tracks are ready, sleep once for the duration of an output
3789 // buffer size, then write 0s to the output
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003790 if (mSleepTimeUs == 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003791 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003792 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift;
3793 if (mSleepTimeUs < kMinThreadSleepTimeUs) {
3794 mSleepTimeUs = kMinThreadSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08003795 }
3796 // reduce sleep time in case of consecutive application underruns to avoid
3797 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
3798 // duration we would end up writing less data than needed by the audio HAL if
3799 // the condition persists.
3800 if (sleepTimeShift < kMaxThreadSleepTimeShift) {
3801 sleepTimeShift++;
3802 }
3803 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003804 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08003805 }
3806 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
Andy Hung98ef9782014-03-04 14:46:50 -08003807 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared
3808 // before effects processing or output.
3809 if (mMixerBufferValid) {
3810 memset(mMixerBuffer, 0, mMixerBufferSize);
3811 } else {
3812 memset(mSinkBuffer, 0, mSinkBufferSize);
3813 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003814 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08003815 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
3816 "anticipated start");
3817 }
3818 // TODO add standby time extension fct of effect tail
3819}
3820
3821// prepareTracks_l() must be called with ThreadBase::mLock held
3822AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
3823 Vector< sp<Track> > *tracksToRemove)
3824{
3825
3826 mixer_state mixerStatus = MIXER_IDLE;
3827 // find out which tracks need to be processed
3828 size_t count = mActiveTracks.size();
3829 size_t mixedTracks = 0;
3830 size_t tracksWithEffect = 0;
3831 // counts only _active_ fast tracks
3832 size_t fastTracks = 0;
3833 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
3834
3835 float masterVolume = mMasterVolume;
3836 bool masterMute = mMasterMute;
3837
3838 if (masterMute) {
3839 masterVolume = 0;
3840 }
3841 // Delegate master volume control to effect in output mix effect chain if needed
3842 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
3843 if (chain != 0) {
3844 uint32_t v = (uint32_t)(masterVolume * (1 << 24));
3845 chain->setVolume_l(&v, &v);
3846 masterVolume = (float)((v + (1 << 23)) >> 24);
3847 chain.clear();
3848 }
3849
3850 // prepare a new state to push
3851 FastMixerStateQueue *sq = NULL;
3852 FastMixerState *state = NULL;
3853 bool didModify = false;
3854 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003855 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003856 sq = mFastMixer->sq();
3857 state = sq->begin();
3858 }
3859
Andy Hung69aed5f2014-02-25 17:24:40 -08003860 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found.
Andy Hung98ef9782014-03-04 14:46:50 -08003861 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found.
Andy Hung69aed5f2014-02-25 17:24:40 -08003862
Eric Laurent81784c32012-11-19 14:55:58 -08003863 for (size_t i=0 ; i<count ; i++) {
Glenn Kasten9fdcb0a2013-06-26 16:11:36 -07003864 const sp<Track> t = mActiveTracks[i].promote();
Eric Laurent81784c32012-11-19 14:55:58 -08003865 if (t == 0) {
3866 continue;
3867 }
3868
3869 // this const just means the local variable doesn't change
3870 Track* const track = t.get();
3871
3872 // process fast tracks
3873 if (track->isFastTrack()) {
3874
3875 // It's theoretically possible (though unlikely) for a fast track to be created
3876 // and then removed within the same normal mix cycle. This is not a problem, as
3877 // the track never becomes active so it's fast mixer slot is never touched.
3878 // The converse, of removing an (active) track and then creating a new track
3879 // at the identical fast mixer slot within the same normal mix cycle,
3880 // is impossible because the slot isn't marked available until the end of each cycle.
3881 int j = track->mFastIndex;
Glenn Kastendc2c50b2016-04-21 08:13:14 -07003882 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks);
Eric Laurent81784c32012-11-19 14:55:58 -08003883 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
3884 FastTrack *fastTrack = &state->mFastTracks[j];
3885
3886 // Determine whether the track is currently in underrun condition,
3887 // and whether it had a recent underrun.
3888 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
3889 FastTrackUnderruns underruns = ftDump->mUnderruns;
3890 uint32_t recentFull = (underruns.mBitFields.mFull -
3891 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
3892 uint32_t recentPartial = (underruns.mBitFields.mPartial -
3893 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
3894 uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
3895 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
3896 uint32_t recentUnderruns = recentPartial + recentEmpty;
3897 track->mObservedUnderruns = underruns;
3898 // don't count underruns that occur while stopping or pausing
3899 // or stopped which can occur when flush() is called while active
Glenn Kasten82aaf942013-07-17 16:05:07 -07003900 if (!(track->isStopping() || track->isPausing() || track->isStopped()) &&
3901 recentUnderruns > 0) {
3902 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun
3903 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount);
Phil Burk2812d9e2016-01-04 10:34:30 -08003904 } else {
3905 track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
Eric Laurent81784c32012-11-19 14:55:58 -08003906 }
3907
3908 // This is similar to the state machine for normal tracks,
3909 // with a few modifications for fast tracks.
3910 bool isActive = true;
3911 switch (track->mState) {
3912 case TrackBase::STOPPING_1:
3913 // track stays active in STOPPING_1 state until first underrun
Eric Laurentbfb1b832013-01-07 09:53:42 -08003914 if (recentUnderruns > 0 || track->isTerminated()) {
Eric Laurent81784c32012-11-19 14:55:58 -08003915 track->mState = TrackBase::STOPPING_2;
3916 }
3917 break;
3918 case TrackBase::PAUSING:
3919 // ramp down is not yet implemented
3920 track->setPaused();
3921 break;
3922 case TrackBase::RESUMING:
3923 // ramp up is not yet implemented
3924 track->mState = TrackBase::ACTIVE;
3925 break;
3926 case TrackBase::ACTIVE:
3927 if (recentFull > 0 || recentPartial > 0) {
3928 // track has provided at least some frames recently: reset retry count
3929 track->mRetryCount = kMaxTrackRetries;
3930 }
3931 if (recentUnderruns == 0) {
3932 // no recent underruns: stay active
3933 break;
3934 }
3935 // there has recently been an underrun of some kind
3936 if (track->sharedBuffer() == 0) {
3937 // were any of the recent underruns "empty" (no frames available)?
3938 if (recentEmpty == 0) {
3939 // no, then ignore the partial underruns as they are allowed indefinitely
3940 break;
3941 }
3942 // there has recently been an "empty" underrun: decrement the retry counter
3943 if (--(track->mRetryCount) > 0) {
3944 break;
3945 }
3946 // indicate to client process that the track was disabled because of underrun;
3947 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08003948 track->disable();
Eric Laurent81784c32012-11-19 14:55:58 -08003949 // remove from active list, but state remains ACTIVE [confusing but true]
3950 isActive = false;
3951 break;
3952 }
3953 // fall through
3954 case TrackBase::STOPPING_2:
3955 case TrackBase::PAUSED:
Eric Laurent81784c32012-11-19 14:55:58 -08003956 case TrackBase::STOPPED:
3957 case TrackBase::FLUSHED: // flush() while active
3958 // Check for presentation complete if track is inactive
3959 // We have consumed all the buffers of this track.
3960 // This would be incomplete if we auto-paused on underrun
3961 {
3962 size_t audioHALFrames =
3963 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08003964 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08003965 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
3966 // track stays in active list until presentation is complete
3967 break;
3968 }
3969 }
3970 if (track->isStopping_2()) {
3971 track->mState = TrackBase::STOPPED;
3972 }
3973 if (track->isStopped()) {
3974 // Can't reset directly, as fast mixer is still polling this track
3975 // track->reset();
3976 // So instead mark this track as needing to be reset after push with ack
3977 resetMask |= 1 << i;
3978 }
3979 isActive = false;
3980 break;
3981 case TrackBase::IDLE:
3982 default:
Glenn Kastenadad3d72014-02-21 14:51:43 -08003983 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState);
Eric Laurent81784c32012-11-19 14:55:58 -08003984 }
3985
3986 if (isActive) {
3987 // was it previously inactive?
3988 if (!(state->mTrackMask & (1 << j))) {
3989 ExtendedAudioBufferProvider *eabp = track;
3990 VolumeProvider *vp = track;
3991 fastTrack->mBufferProvider = eabp;
3992 fastTrack->mVolumeProvider = vp;
Eric Laurent81784c32012-11-19 14:55:58 -08003993 fastTrack->mChannelMask = track->mChannelMask;
Andy Hunge8a1ced2014-05-09 15:02:21 -07003994 fastTrack->mFormat = track->mFormat;
Eric Laurent81784c32012-11-19 14:55:58 -08003995 fastTrack->mGeneration++;
3996 state->mTrackMask |= 1 << j;
3997 didModify = true;
3998 // no acknowledgement required for newly active tracks
3999 }
4000 // cache the combined master volume and stream type volume for fast mixer; this
4001 // lacks any synchronization or barrier so VolumeProvider may read a stale value
Glenn Kastene4756fe2012-11-29 13:38:14 -08004002 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume;
Eric Laurent81784c32012-11-19 14:55:58 -08004003 ++fastTracks;
4004 } else {
4005 // was it previously active?
4006 if (state->mTrackMask & (1 << j)) {
4007 fastTrack->mBufferProvider = NULL;
4008 fastTrack->mGeneration++;
4009 state->mTrackMask &= ~(1 << j);
4010 didModify = true;
4011 // If any fast tracks were removed, we must wait for acknowledgement
4012 // because we're about to decrement the last sp<> on those tracks.
4013 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
4014 } else {
Glenn Kastenf7d65ee2015-12-02 13:45:01 -08004015 LOG_ALWAYS_FATAL("fast track %d should have been active; "
4016 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d",
4017 j, track->mState, state->mTrackMask, recentUnderruns,
4018 track->sharedBuffer() != 0);
Eric Laurent81784c32012-11-19 14:55:58 -08004019 }
4020 tracksToRemove->add(track);
4021 // Avoids a misleading display in dumpsys
4022 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
4023 }
4024 continue;
4025 }
4026
4027 { // local variable scope to avoid goto warning
4028
4029 audio_track_cblk_t* cblk = track->cblk();
4030
4031 // The first time a track is added we wait
4032 // for all its buffers to be filled before processing it
4033 int name = track->name();
4034 // make sure that we have enough frames to mix one full buffer.
4035 // enforce this condition only once to enable draining the buffer in case the client
4036 // app does not call stop() and relies on underrun to stop:
4037 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
4038 // during last round
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004039 size_t desiredFrames;
Andy Hung8edb8dc2015-03-26 19:13:55 -07004040 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004041 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
Andy Hung8edb8dc2015-03-26 19:13:55 -07004042
4043 desiredFrames = sourceFramesNeededWithTimestretch(
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004044 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed);
Andy Hung8edb8dc2015-03-26 19:13:55 -07004045 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed.
4046 // add frames already consumed but not yet released by the resampler
4047 // because mAudioTrackServerProxy->framesReady() will include these frames
4048 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
4049
Eric Laurent81784c32012-11-19 14:55:58 -08004050 uint32_t minFrames = 1;
4051 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
4052 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004053 minFrames = desiredFrames;
Eric Laurent81784c32012-11-19 14:55:58 -08004054 }
Eric Laurent13e4c962013-12-20 17:36:01 -08004055
4056 size_t framesReady = track->framesReady();
Glenn Kastene7754022014-10-31 12:11:26 -07004057 if (ATRACE_ENABLED()) {
4058 // I wish we had formatted trace names
4059 char traceName[16];
4060 strcpy(traceName, "nRdy");
4061 int name = track->name();
4062 if (AudioMixer::TRACK0 <= name &&
4063 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) {
4064 name -= AudioMixer::TRACK0;
4065 traceName[4] = (name / 10) + '0';
4066 traceName[5] = (name % 10) + '0';
4067 } else {
4068 traceName[4] = '?';
4069 traceName[5] = '?';
4070 }
4071 traceName[6] = '\0';
4072 ATRACE_INT(traceName, framesReady);
4073 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004074 if ((framesReady >= minFrames) && track->isReady() &&
Eric Laurent81784c32012-11-19 14:55:58 -08004075 !track->isPaused() && !track->isTerminated())
4076 {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004077 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this);
Eric Laurent81784c32012-11-19 14:55:58 -08004078
4079 mixedTracks++;
4080
Andy Hung69aed5f2014-02-25 17:24:40 -08004081 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means
4082 // there is an effect chain connected to the track
Eric Laurent81784c32012-11-19 14:55:58 -08004083 chain.clear();
Andy Hung69aed5f2014-02-25 17:24:40 -08004084 if (track->mainBuffer() != mSinkBuffer &&
4085 track->mainBuffer() != mMixerBuffer) {
Andy Hung98ef9782014-03-04 14:46:50 -08004086 if (mEffectBufferEnabled) {
4087 mEffectBufferValid = true; // Later can set directly.
4088 }
Eric Laurent81784c32012-11-19 14:55:58 -08004089 chain = getEffectChain_l(track->sessionId());
4090 // Delegate volume control to effect in track effect chain if needed
4091 if (chain != 0) {
4092 tracksWithEffect++;
4093 } else {
4094 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
4095 "session %d",
4096 name, track->sessionId());
4097 }
4098 }
4099
4100
4101 int param = AudioMixer::VOLUME;
4102 if (track->mFillingUpStatus == Track::FS_FILLED) {
4103 // no ramp for the first volume setting
4104 track->mFillingUpStatus = Track::FS_ACTIVE;
4105 if (track->mState == TrackBase::RESUMING) {
4106 track->mState = TrackBase::ACTIVE;
4107 param = AudioMixer::RAMP_VOLUME;
4108 }
4109 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004110 // FIXME should not make a decision based on mServer
4111 } else if (cblk->mServer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08004112 // If the track is stopped before the first frame was mixed,
4113 // do not apply ramp
4114 param = AudioMixer::RAMP_VOLUME;
4115 }
4116
4117 // compute volume for this track
Andy Hung6be49402014-05-30 10:42:03 -07004118 uint32_t vl, vr; // in U8.24 integer format
4119 float vlf, vrf, vaf; // in [0.0, 1.0] float format
Glenn Kastene4756fe2012-11-29 13:38:14 -08004120 if (track->isPausing() || mStreamTypes[track->streamType()].mute) {
Andy Hung6be49402014-05-30 10:42:03 -07004121 vl = vr = 0;
4122 vlf = vrf = vaf = 0.;
Eric Laurent81784c32012-11-19 14:55:58 -08004123 if (track->isPausing()) {
4124 track->setPaused();
4125 }
4126 } else {
4127
4128 // read original volumes with volume control
4129 float typeVolume = mStreamTypes[track->streamType()].volume;
4130 float v = masterVolume * typeVolume;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004131 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
Glenn Kastenc56f3422014-03-21 17:53:17 -07004132 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
Andy Hung6be49402014-05-30 10:42:03 -07004133 vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
4134 vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
Eric Laurent81784c32012-11-19 14:55:58 -08004135 // track volumes come from shared memory, so can't be trusted and must be clamped
Glenn Kastenc56f3422014-03-21 17:53:17 -07004136 if (vlf > GAIN_FLOAT_UNITY) {
4137 ALOGV("Track left volume out of range: %.3g", vlf);
4138 vlf = GAIN_FLOAT_UNITY;
Eric Laurent81784c32012-11-19 14:55:58 -08004139 }
Glenn Kastenc56f3422014-03-21 17:53:17 -07004140 if (vrf > GAIN_FLOAT_UNITY) {
4141 ALOGV("Track right volume out of range: %.3g", vrf);
4142 vrf = GAIN_FLOAT_UNITY;
Eric Laurent81784c32012-11-19 14:55:58 -08004143 }
4144 // now apply the master volume and stream type volume
Andy Hung6be49402014-05-30 10:42:03 -07004145 vlf *= v;
4146 vrf *= v;
Eric Laurent81784c32012-11-19 14:55:58 -08004147 // assuming master volume and stream type volume each go up to 1.0,
Andy Hung6be49402014-05-30 10:42:03 -07004148 // then derive vl and vr as U8.24 versions for the effect chain
4149 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;
4150 vl = (uint32_t) (scaleto8_24 * vlf);
4151 vr = (uint32_t) (scaleto8_24 * vrf);
4152 // vl and vr are now in U8.24 format
Glenn Kastene3aa6592012-12-04 12:22:46 -08004153 uint16_t sendLevel = proxy->getSendLevel_U4_12();
Eric Laurent81784c32012-11-19 14:55:58 -08004154 // send level comes from shared memory and so may be corrupt
4155 if (sendLevel > MAX_GAIN_INT) {
4156 ALOGV("Track send level out of range: %04X", sendLevel);
4157 sendLevel = MAX_GAIN_INT;
4158 }
Andy Hung6be49402014-05-30 10:42:03 -07004159 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel
4160 vaf = v * sendLevel * (1. / MAX_GAIN_INT);
Eric Laurent81784c32012-11-19 14:55:58 -08004161 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08004162
Eric Laurent81784c32012-11-19 14:55:58 -08004163 // Delegate volume control to effect in track effect chain if needed
4164 if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
4165 // Do not ramp volume if volume is controlled by effect
4166 param = AudioMixer::VOLUME;
Bryant Liub6be7f22014-06-12 22:02:41 +08004167 // Update remaining floating point volume levels
4168 vlf = (float)vl / (1 << 24);
4169 vrf = (float)vr / (1 << 24);
Eric Laurent81784c32012-11-19 14:55:58 -08004170 track->mHasVolumeController = true;
4171 } else {
4172 // force no volume ramp when volume controller was just disabled or removed
4173 // from effect chain to avoid volume spike
4174 if (track->mHasVolumeController) {
4175 param = AudioMixer::VOLUME;
4176 }
4177 track->mHasVolumeController = false;
4178 }
4179
Eric Laurent81784c32012-11-19 14:55:58 -08004180 // XXX: these things DON'T need to be done each time
4181 mAudioMixer->setBufferProvider(name, track);
4182 mAudioMixer->enable(name);
4183
Andy Hung6be49402014-05-30 10:42:03 -07004184 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);
4185 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);
4186 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);
Eric Laurent81784c32012-11-19 14:55:58 -08004187 mAudioMixer->setParameter(
4188 name,
4189 AudioMixer::TRACK,
4190 AudioMixer::FORMAT, (void *)track->format());
4191 mAudioMixer->setParameter(
4192 name,
4193 AudioMixer::TRACK,
Kévin PETIT377b2ec2014-02-03 12:35:36 +00004194 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
Andy Hung9a592762014-07-21 21:56:01 -07004195 mAudioMixer->setParameter(
4196 name,
4197 AudioMixer::TRACK,
4198 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask);
Glenn Kastene3aa6592012-12-04 12:22:46 -08004199 // limit track sample rate to 2 x output sample rate, which changes at re-configuration
Andy Hungcd044842014-08-07 11:04:34 -07004200 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004201 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
Glenn Kastene3aa6592012-12-04 12:22:46 -08004202 if (reqSampleRate == 0) {
4203 reqSampleRate = mSampleRate;
4204 } else if (reqSampleRate > maxSampleRate) {
4205 reqSampleRate = maxSampleRate;
4206 }
Eric Laurent81784c32012-11-19 14:55:58 -08004207 mAudioMixer->setParameter(
4208 name,
4209 AudioMixer::RESAMPLE,
4210 AudioMixer::SAMPLE_RATE,
Kévin PETIT377b2ec2014-02-03 12:35:36 +00004211 (void *)(uintptr_t)reqSampleRate);
Andy Hung8edb8dc2015-03-26 19:13:55 -07004212
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004213 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
Andy Hung8edb8dc2015-03-26 19:13:55 -07004214 mAudioMixer->setParameter(
4215 name,
4216 AudioMixer::TIMESTRETCH,
4217 AudioMixer::PLAYBACK_RATE,
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004218 &playbackRate);
Andy Hung8edb8dc2015-03-26 19:13:55 -07004219
Andy Hung69aed5f2014-02-25 17:24:40 -08004220 /*
4221 * Select the appropriate output buffer for the track.
4222 *
Andy Hung98ef9782014-03-04 14:46:50 -08004223 * Tracks with effects go into their own effects chain buffer
4224 * and from there into either mEffectBuffer or mSinkBuffer.
Andy Hung69aed5f2014-02-25 17:24:40 -08004225 *
4226 * Other tracks can use mMixerBuffer for higher precision
4227 * channel accumulation. If this buffer is enabled
4228 * (mMixerBufferEnabled true), then selected tracks will accumulate
4229 * into it.
4230 *
4231 */
4232 if (mMixerBufferEnabled
4233 && (track->mainBuffer() == mSinkBuffer
4234 || track->mainBuffer() == mMixerBuffer)) {
4235 mAudioMixer->setParameter(
4236 name,
4237 AudioMixer::TRACK,
Andy Hung78820702014-02-28 16:23:02 -08004238 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
Andy Hung69aed5f2014-02-25 17:24:40 -08004239 mAudioMixer->setParameter(
4240 name,
4241 AudioMixer::TRACK,
4242 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
4243 // TODO: override track->mainBuffer()?
4244 mMixerBufferValid = true;
4245 } else {
4246 mAudioMixer->setParameter(
4247 name,
4248 AudioMixer::TRACK,
Andy Hung78820702014-02-28 16:23:02 -08004249 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT);
Andy Hung69aed5f2014-02-25 17:24:40 -08004250 mAudioMixer->setParameter(
4251 name,
4252 AudioMixer::TRACK,
4253 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
4254 }
Eric Laurent81784c32012-11-19 14:55:58 -08004255 mAudioMixer->setParameter(
4256 name,
4257 AudioMixer::TRACK,
4258 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
4259
4260 // reset retry count
4261 track->mRetryCount = kMaxTrackRetries;
4262
4263 // If one track is ready, set the mixer ready if:
4264 // - the mixer was not ready during previous round OR
4265 // - no other track is not ready
4266 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
4267 mixerStatus != MIXER_TRACKS_ENABLED) {
4268 mixerStatus = MIXER_TRACKS_READY;
4269 }
4270 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004271 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
Andy Hung08fb1742015-05-31 23:22:10 -07004272 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)",
4273 track, framesReady, desiredFrames);
Glenn Kasten82aaf942013-07-17 16:05:07 -07004274 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
Phil Burk2812d9e2016-01-04 10:34:30 -08004275 } else {
4276 track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004277 }
Phil Burk2812d9e2016-01-04 10:34:30 -08004278
Eric Laurent81784c32012-11-19 14:55:58 -08004279 // clear effect chain input buffer if an active track underruns to avoid sending
4280 // previous audio buffer again to effects
4281 chain = getEffectChain_l(track->sessionId());
4282 if (chain != 0) {
4283 chain->clearInputBuffer();
4284 }
4285
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004286 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this);
Eric Laurent81784c32012-11-19 14:55:58 -08004287 if ((track->sharedBuffer() != 0) || track->isTerminated() ||
4288 track->isStopped() || track->isPaused()) {
4289 // We have consumed all the buffers of this track.
4290 // Remove it from the list of active tracks.
4291 // TODO: use actual buffer filling status instead of latency when available from
4292 // audio HAL
4293 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08004294 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08004295 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
4296 if (track->isStopped()) {
4297 track->reset();
4298 }
4299 tracksToRemove->add(track);
4300 }
4301 } else {
Eric Laurent81784c32012-11-19 14:55:58 -08004302 // No buffers for this track. Give it a few chances to
4303 // fill a buffer, then remove it from active list.
4304 if (--(track->mRetryCount) <= 0) {
Glenn Kastenc9b2e202013-02-26 11:32:32 -08004305 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
Eric Laurent81784c32012-11-19 14:55:58 -08004306 tracksToRemove->add(track);
4307 // indicate to client process that the track was disabled because of underrun;
4308 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08004309 track->disable();
Eric Laurent81784c32012-11-19 14:55:58 -08004310 // If one track is not ready, mark the mixer also not ready if:
4311 // - the mixer was ready during previous round OR
4312 // - no other track is ready
4313 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
4314 mixerStatus != MIXER_TRACKS_READY) {
4315 mixerStatus = MIXER_TRACKS_ENABLED;
4316 }
4317 }
4318 mAudioMixer->disable(name);
4319 }
4320
4321 } // local variable scope to avoid goto warning
Eric Laurent81784c32012-11-19 14:55:58 -08004322
4323 }
4324
4325 // Push the new FastMixer state if necessary
4326 bool pauseAudioWatchdog = false;
4327 if (didModify) {
4328 state->mFastTracksGen++;
4329 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
4330 if (kUseFastMixer == FastMixer_Dynamic &&
4331 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
4332 state->mCommand = FastMixerState::COLD_IDLE;
4333 state->mColdFutexAddr = &mFastMixerFutex;
4334 state->mColdGen++;
4335 mFastMixerFutex = 0;
4336 if (kUseFastMixer == FastMixer_Dynamic) {
4337 mNormalSink = mOutputSink;
4338 }
4339 // If we go into cold idle, need to wait for acknowledgement
4340 // so that fast mixer stops doing I/O.
4341 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
4342 pauseAudioWatchdog = true;
4343 }
Eric Laurent81784c32012-11-19 14:55:58 -08004344 }
4345 if (sq != NULL) {
4346 sq->end(didModify);
4347 sq->push(block);
4348 }
4349#ifdef AUDIO_WATCHDOG
4350 if (pauseAudioWatchdog && mAudioWatchdog != 0) {
4351 mAudioWatchdog->pause();
4352 }
4353#endif
4354
4355 // Now perform the deferred reset on fast tracks that have stopped
4356 while (resetMask != 0) {
4357 size_t i = __builtin_ctz(resetMask);
4358 ALOG_ASSERT(i < count);
4359 resetMask &= ~(1 << i);
4360 sp<Track> t = mActiveTracks[i].promote();
4361 if (t == 0) {
4362 continue;
4363 }
4364 Track* track = t.get();
4365 ALOG_ASSERT(track->isFastTrack() && track->isStopped());
4366 track->reset();
4367 }
4368
4369 // remove all the tracks that need to be...
Eric Laurentbfb1b832013-01-07 09:53:42 -08004370 removeTracks_l(*tracksToRemove);
Eric Laurent81784c32012-11-19 14:55:58 -08004371
Eric Laurent97d547d2014-09-02 14:45:53 -07004372 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
4373 mEffectBufferValid = true;
Marco Nelissenac302142014-10-20 13:15:38 -07004374 }
4375
4376 if (mEffectBufferValid) {
Marco Nelissen57088b52014-10-17 16:39:39 -07004377 // as long as there are effects we should clear the effects buffer, to avoid
4378 // passing a non-clean buffer to the effect chain
4379 memset(mEffectBuffer, 0, mEffectBufferSize);
Eric Laurent97d547d2014-09-02 14:45:53 -07004380 }
Andy Hung69aed5f2014-02-25 17:24:40 -08004381 // sink or mix buffer must be cleared if all tracks are connected to an
4382 // effect chain as in this case the mixer will not write to the sink or mix buffer
4383 // and track effects will accumulate into it
Eric Laurentbfb1b832013-01-07 09:53:42 -08004384 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
4385 (mixedTracks == 0 && fastTracks > 0))) {
Eric Laurent81784c32012-11-19 14:55:58 -08004386 // FIXME as a performance optimization, should remember previous zero status
Andy Hung69aed5f2014-02-25 17:24:40 -08004387 if (mMixerBufferValid) {
4388 memset(mMixerBuffer, 0, mMixerBufferSize);
4389 // TODO: In testing, mSinkBuffer below need not be cleared because
4390 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer
4391 // after mixing.
4392 //
4393 // To enforce this guarantee:
4394 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
4395 // (mixedTracks == 0 && fastTracks > 0))
4396 // must imply MIXER_TRACKS_READY.
4397 // Later, we may clear buffers regardless, and skip much of this logic.
4398 }
Andy Hung98ef9782014-03-04 14:46:50 -08004399 // FIXME as a performance optimization, should remember previous zero status
Andy Hung5567aaf2014-07-17 14:00:07 -07004400 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize);
Eric Laurent81784c32012-11-19 14:55:58 -08004401 }
4402
4403 // if any fast tracks, then status is ready
4404 mMixerStatusIgnoringFastTracks = mixerStatus;
4405 if (fastTracks > 0) {
4406 mixerStatus = MIXER_TRACKS_READY;
4407 }
4408 return mixerStatus;
4409}
4410
4411// getTrackName_l() must be called with ThreadBase::mLock held
Andy Hunge8a1ced2014-05-09 15:02:21 -07004412int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask,
Glenn Kastend848eb42016-03-08 13:42:11 -08004413 audio_format_t format, audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08004414{
Andy Hunge8a1ced2014-05-09 15:02:21 -07004415 return mAudioMixer->getTrackName(channelMask, format, sessionId);
Eric Laurent81784c32012-11-19 14:55:58 -08004416}
4417
4418// deleteTrackName_l() must be called with ThreadBase::mLock held
4419void AudioFlinger::MixerThread::deleteTrackName_l(int name)
4420{
4421 ALOGV("remove track (%d) and delete from mixer", name);
4422 mAudioMixer->deleteTrackName(name);
4423}
4424
Eric Laurent10351942014-05-08 18:49:52 -07004425// checkForNewParameter_l() must be called with ThreadBase::mLock held
4426bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair,
4427 status_t& status)
Eric Laurent81784c32012-11-19 14:55:58 -08004428{
Eric Laurent81784c32012-11-19 14:55:58 -08004429 bool reconfig = false;
Eric Laurent42537be2016-01-08 17:16:42 -08004430 bool a2dpDeviceChanged = false;
Eric Laurent81784c32012-11-19 14:55:58 -08004431
Eric Laurent10351942014-05-08 18:49:52 -07004432 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08004433
Glenn Kastenc05b8d72016-03-24 09:48:17 -07004434 AutoPark<FastMixer> park(mFastMixer);
Eric Laurent81784c32012-11-19 14:55:58 -08004435
Eric Laurent10351942014-05-08 18:49:52 -07004436 AudioParameter param = AudioParameter(keyValuePair);
4437 int value;
4438 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4439 reconfig = true;
4440 }
4441 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
Andy Hung9a592762014-07-21 21:56:01 -07004442 if (!isValidPcmSinkFormat((audio_format_t) value)) {
Eric Laurent10351942014-05-08 18:49:52 -07004443 status = BAD_VALUE;
4444 } else {
4445 // no need to save value, since it's constant
Eric Laurent81784c32012-11-19 14:55:58 -08004446 reconfig = true;
4447 }
Eric Laurent10351942014-05-08 18:49:52 -07004448 }
4449 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
Andy Hung9a592762014-07-21 21:56:01 -07004450 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) {
Eric Laurent10351942014-05-08 18:49:52 -07004451 status = BAD_VALUE;
4452 } else {
4453 // no need to save value, since it's constant
4454 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08004455 }
Eric Laurent10351942014-05-08 18:49:52 -07004456 }
4457 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4458 // do not accept frame count changes if tracks are open as the track buffer
4459 // size depends on frame count and correct behavior would not be guaranteed
4460 // if frame count is changed after track creation
4461 if (!mTracks.isEmpty()) {
4462 status = INVALID_OPERATION;
4463 } else {
4464 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08004465 }
Eric Laurent10351942014-05-08 18:49:52 -07004466 }
4467 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
Eric Laurent81784c32012-11-19 14:55:58 -08004468#ifdef ADD_BATTERY_DATA
Eric Laurent10351942014-05-08 18:49:52 -07004469 // when changing the audio output device, call addBatteryData to notify
4470 // the change
4471 if (mOutDevice != value) {
4472 uint32_t params = 0;
4473 // check whether speaker is on
4474 if (value & AUDIO_DEVICE_OUT_SPEAKER) {
4475 params |= IMediaPlayerService::kBatteryDataSpeakerOn;
Eric Laurent81784c32012-11-19 14:55:58 -08004476 }
Eric Laurent10351942014-05-08 18:49:52 -07004477
4478 audio_devices_t deviceWithoutSpeaker
4479 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
4480 // check if any other device (except speaker) is on
Eric Laurent054d9d32015-04-24 08:48:48 -07004481 if (value & deviceWithoutSpeaker) {
Eric Laurent10351942014-05-08 18:49:52 -07004482 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
4483 }
4484
4485 if (params != 0) {
4486 addBatteryData(params);
4487 }
4488 }
Eric Laurent81784c32012-11-19 14:55:58 -08004489#endif
4490
Eric Laurent10351942014-05-08 18:49:52 -07004491 // forward device change to effects that have requested to be
4492 // aware of attached audio device.
4493 if (value != AUDIO_DEVICE_NONE) {
Eric Laurent42537be2016-01-08 17:16:42 -08004494 a2dpDeviceChanged =
4495 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP);
Eric Laurent10351942014-05-08 18:49:52 -07004496 mOutDevice = value;
4497 for (size_t i = 0; i < mEffectChains.size(); i++) {
4498 mEffectChains[i]->setDevice_l(mOutDevice);
Eric Laurent81784c32012-11-19 14:55:58 -08004499 }
4500 }
Eric Laurent10351942014-05-08 18:49:52 -07004501 }
Eric Laurent81784c32012-11-19 14:55:58 -08004502
Eric Laurent10351942014-05-08 18:49:52 -07004503 if (status == NO_ERROR) {
4504 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
4505 keyValuePair.string());
4506 if (!mStandby && status == INVALID_OPERATION) {
Phil Burk062e67a2015-02-11 13:40:50 -08004507 mOutput->standby();
Eric Laurent10351942014-05-08 18:49:52 -07004508 mStandby = true;
4509 mBytesWritten = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08004510 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
Eric Laurent10351942014-05-08 18:49:52 -07004511 keyValuePair.string());
Eric Laurent81784c32012-11-19 14:55:58 -08004512 }
Eric Laurent10351942014-05-08 18:49:52 -07004513 if (status == NO_ERROR && reconfig) {
4514 readOutputParameters_l();
4515 delete mAudioMixer;
4516 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
4517 for (size_t i = 0; i < mTracks.size() ; i++) {
Andy Hunge8a1ced2014-05-09 15:02:21 -07004518 int name = getTrackName_l(mTracks[i]->mChannelMask,
4519 mTracks[i]->mFormat, mTracks[i]->mSessionId);
Eric Laurent10351942014-05-08 18:49:52 -07004520 if (name < 0) {
4521 break;
4522 }
4523 mTracks[i]->mName = name;
4524 }
Eric Laurent73e26b62015-04-27 16:55:58 -07004525 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
Eric Laurent10351942014-05-08 18:49:52 -07004526 }
Eric Laurent81784c32012-11-19 14:55:58 -08004527 }
4528
Eric Laurent42537be2016-01-08 17:16:42 -08004529 return reconfig || a2dpDeviceChanged;
Eric Laurent81784c32012-11-19 14:55:58 -08004530}
4531
4532
4533void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
4534{
Eric Laurent81784c32012-11-19 14:55:58 -08004535 PlaybackThread::dumpInternals(fd, args);
Andy Hung40eb1a12015-06-18 13:42:02 -07004536 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs);
Elliott Hughes87cebad2014-05-22 10:14:43 -07004537 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
Andy Hung2ddee192015-12-18 17:34:44 -08004538 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off");
Eric Laurent81784c32012-11-19 14:55:58 -08004539
4540 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
Glenn Kasten2f90c512015-12-02 11:40:09 -08004541 // while we are dumping it. It may be inconsistent, but it won't mutate!
4542 // This is a large object so we place it on the heap.
4543 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
4544 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState);
4545 copy->dump(fd);
4546 delete copy;
Eric Laurent81784c32012-11-19 14:55:58 -08004547
4548#ifdef STATE_QUEUE_DUMP
4549 // Similar for state queue
4550 StateQueueObserverDump observerCopy = mStateQueueObserverDump;
4551 observerCopy.dump(fd);
4552 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
4553 mutatorCopy.dump(fd);
4554#endif
4555
Glenn Kasten46909e72013-02-26 09:20:22 -08004556#ifdef TEE_SINK
Eric Laurent81784c32012-11-19 14:55:58 -08004557 // Write the tee output to a .wav file
4558 dumpTee(fd, mTeeSource, mId);
Glenn Kasten46909e72013-02-26 09:20:22 -08004559#endif
Eric Laurent81784c32012-11-19 14:55:58 -08004560
4561#ifdef AUDIO_WATCHDOG
4562 if (mAudioWatchdog != 0) {
4563 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
4564 AudioWatchdogDump wdCopy = mAudioWatchdogDump;
4565 wdCopy.dump(fd);
4566 }
4567#endif
4568}
4569
4570uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
4571{
4572 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
4573}
4574
4575uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
4576{
4577 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
4578}
4579
4580void AudioFlinger::MixerThread::cacheParameters_l()
4581{
4582 PlaybackThread::cacheParameters_l();
4583
4584 // FIXME: Relaxed timing because of a certain device that can't meet latency
4585 // Should be reduced to 2x after the vendor fixes the driver issue
4586 // increase threshold again due to low power audio mode. The way this warning
4587 // threshold is calculated and its usefulness should be reconsidered anyway.
4588 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
4589}
4590
4591// ----------------------------------------------------------------------------
4592
4593AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
Eric Laurente93cc032016-05-05 10:15:10 -07004594 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady)
4595 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady)
Eric Laurent81784c32012-11-19 14:55:58 -08004596 // mLeftVolFloat, mRightVolFloat
4597{
4598}
4599
Eric Laurentbfb1b832013-01-07 09:53:42 -08004600AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
4601 AudioStreamOut* output, audio_io_handle_t id, uint32_t device,
Eric Laurente93cc032016-05-05 10:15:10 -07004602 ThreadBase::type_t type, bool systemReady)
4603 : PlaybackThread(audioFlinger, output, id, device, type, systemReady)
Eric Laurentbfb1b832013-01-07 09:53:42 -08004604 // mLeftVolFloat, mRightVolFloat
4605{
4606}
4607
Eric Laurent81784c32012-11-19 14:55:58 -08004608AudioFlinger::DirectOutputThread::~DirectOutputThread()
4609{
4610}
4611
Eric Laurentbfb1b832013-01-07 09:53:42 -08004612void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
4613{
Eric Laurentbfb1b832013-01-07 09:53:42 -08004614 float left, right;
4615
4616 if (mMasterMute || mStreamTypes[track->streamType()].mute) {
4617 left = right = 0;
4618 } else {
4619 float typeVolume = mStreamTypes[track->streamType()].volume;
4620 float v = mMasterVolume * typeVolume;
4621 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
Glenn Kastenc56f3422014-03-21 17:53:17 -07004622 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
4623 left = float_from_gain(gain_minifloat_unpack_left(vlr));
4624 if (left > GAIN_FLOAT_UNITY) {
4625 left = GAIN_FLOAT_UNITY;
4626 }
4627 left *= v;
4628 right = float_from_gain(gain_minifloat_unpack_right(vlr));
4629 if (right > GAIN_FLOAT_UNITY) {
4630 right = GAIN_FLOAT_UNITY;
4631 }
4632 right *= v;
Eric Laurentbfb1b832013-01-07 09:53:42 -08004633 }
4634
4635 if (lastTrack) {
4636 if (left != mLeftVolFloat || right != mRightVolFloat) {
4637 mLeftVolFloat = left;
4638 mRightVolFloat = right;
4639
4640 // Convert volumes from float to 8.24
4641 uint32_t vl = (uint32_t)(left * (1 << 24));
4642 uint32_t vr = (uint32_t)(right * (1 << 24));
4643
4644 // Delegate volume control to effect in track effect chain if needed
4645 // only one effect chain can be present on DirectOutputThread, so if
4646 // there is one, the track is connected to it
4647 if (!mEffectChains.isEmpty()) {
4648 mEffectChains[0]->setVolume_l(&vl, &vr);
4649 left = (float)vl / (1 << 24);
4650 right = (float)vr / (1 << 24);
4651 }
4652 if (mOutput->stream->set_volume) {
4653 mOutput->stream->set_volume(mOutput->stream, left, right);
4654 }
4655 }
4656 }
4657}
4658
Phil Burk43b4dcc2015-06-09 16:53:44 -07004659void AudioFlinger::DirectOutputThread::onAddNewTrack_l()
4660{
4661 sp<Track> previousTrack = mPreviousTrack.promote();
4662 sp<Track> latestTrack = mLatestActiveTrack.promote();
4663
Eric Laurent0f0631e2015-07-06 18:01:25 -07004664 if (previousTrack != 0 && latestTrack != 0) {
4665 if (mType == DIRECT) {
4666 if (previousTrack.get() != latestTrack.get()) {
4667 mFlushPending = true;
4668 }
4669 } else /* mType == OFFLOAD */ {
4670 if (previousTrack->sessionId() != latestTrack->sessionId()) {
4671 mFlushPending = true;
4672 }
4673 }
Phil Burk43b4dcc2015-06-09 16:53:44 -07004674 }
4675 PlaybackThread::onAddNewTrack_l();
4676}
Eric Laurentbfb1b832013-01-07 09:53:42 -08004677
Eric Laurent81784c32012-11-19 14:55:58 -08004678AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
4679 Vector< sp<Track> > *tracksToRemove
4680)
4681{
Eric Laurentd595b7c2013-04-03 17:27:56 -07004682 size_t count = mActiveTracks.size();
Eric Laurent81784c32012-11-19 14:55:58 -08004683 mixer_state mixerStatus = MIXER_IDLE;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004684 bool doHwPause = false;
4685 bool doHwResume = false;
Eric Laurent81784c32012-11-19 14:55:58 -08004686
4687 // find out which tracks need to be processed
Eric Laurentd595b7c2013-04-03 17:27:56 -07004688 for (size_t i = 0; i < count; i++) {
4689 sp<Track> t = mActiveTracks[i].promote();
Eric Laurent81784c32012-11-19 14:55:58 -08004690 // The track died recently
4691 if (t == 0) {
Eric Laurentd595b7c2013-04-03 17:27:56 -07004692 continue;
Eric Laurent81784c32012-11-19 14:55:58 -08004693 }
4694
Phil Burk43b4dcc2015-06-09 16:53:44 -07004695 if (t->isInvalid()) {
4696 ALOGW("An invalidated track shouldn't be in active list");
4697 tracksToRemove->add(t);
4698 continue;
4699 }
4700
Eric Laurent81784c32012-11-19 14:55:58 -08004701 Track* const track = t.get();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07004702#ifdef VERY_VERY_VERBOSE_LOGGING
Eric Laurent81784c32012-11-19 14:55:58 -08004703 audio_track_cblk_t* cblk = track->cblk();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07004704#endif
Eric Laurentfd477972013-10-25 18:10:40 -07004705 // Only consider last track started for volume and mixer state control.
4706 // In theory an older track could underrun and restart after the new one starts
4707 // but as we only care about the transition phase between two tracks on a
4708 // direct output, it is not a problem to ignore the underrun case.
4709 sp<Track> l = mLatestActiveTrack.promote();
4710 bool last = l.get() == track;
Eric Laurent81784c32012-11-19 14:55:58 -08004711
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004712 if (track->isPausing()) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004713 track->setPaused();
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004714 if (mHwSupportsPause && last && !mHwPaused) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004715 doHwPause = true;
4716 mHwPaused = true;
4717 }
4718 tracksToRemove->add(track);
4719 } else if (track->isFlushPending()) {
4720 track->flushAck();
4721 if (last) {
Phil Burk43b4dcc2015-06-09 16:53:44 -07004722 mFlushPending = true;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004723 }
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004724 } else if (track->isResumePending()) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004725 track->resumeAck();
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004726 if (last && mHwPaused) {
4727 doHwResume = true;
4728 mHwPaused = false;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004729 }
4730 }
4731
Eric Laurent81784c32012-11-19 14:55:58 -08004732 // The first time a track is added we wait
Phil Burk99adee32014-12-10 16:46:30 -08004733 // for all its buffers to be filled before processing it.
4734 // Allow draining the buffer in case the client
4735 // app does not call stop() and relies on underrun to stop:
4736 // hence the test on (track->mRetryCount > 1).
4737 // If retryCount<=1 then track is about to underrun and be removed.
Phil Burkca5e6142015-07-14 09:42:29 -07004738 // Do not use a high threshold for compressed audio.
Eric Laurent81784c32012-11-19 14:55:58 -08004739 uint32_t minFrames;
Phil Burk99adee32014-12-10 16:46:30 -08004740 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()
Phil Burkfdb3c072016-02-09 10:47:02 -08004741 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08004742 minFrames = mNormalFrameCount;
4743 } else {
4744 minFrames = 1;
4745 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08004746
Eric Laurentab5cdba2014-06-09 17:22:27 -07004747 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
4748 !track->isStopping_2() && !track->isStopped())
Eric Laurent81784c32012-11-19 14:55:58 -08004749 {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004750 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer);
Eric Laurent81784c32012-11-19 14:55:58 -08004751
4752 if (track->mFillingUpStatus == Track::FS_FILLED) {
4753 track->mFillingUpStatus = Track::FS_ACTIVE;
Eric Laurent1abbdb42013-09-13 17:00:08 -07004754 // make sure processVolume_l() will apply new volume even if 0
4755 mLeftVolFloat = mRightVolFloat = -1.0;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004756 if (!mHwSupportsPause) {
4757 track->resumeAck();
Eric Laurent81784c32012-11-19 14:55:58 -08004758 }
4759 }
4760
4761 // compute volume for this track
Eric Laurentbfb1b832013-01-07 09:53:42 -08004762 processVolume_l(track, last);
4763 if (last) {
Phil Burk43b4dcc2015-06-09 16:53:44 -07004764 sp<Track> previousTrack = mPreviousTrack.promote();
4765 if (previousTrack != 0) {
4766 if (track != previousTrack.get()) {
4767 // Flush any data still being written from last track
4768 mBytesRemaining = 0;
Eric Laurent0f0631e2015-07-06 18:01:25 -07004769 // Invalidate previous track to force a seek when resuming.
4770 previousTrack->invalidate();
Phil Burk43b4dcc2015-06-09 16:53:44 -07004771 }
4772 }
4773 mPreviousTrack = track;
4774
Eric Laurentd595b7c2013-04-03 17:27:56 -07004775 // reset retry count
4776 track->mRetryCount = kMaxTrackRetriesDirect;
4777 mActiveTrack = t;
4778 mixerStatus = MIXER_TRACKS_READY;
Eric Laurent5cff4032015-05-26 13:49:58 -07004779 if (mHwPaused) {
Eric Laurent0f7b5f22014-12-19 10:43:21 -08004780 doHwResume = true;
4781 mHwPaused = false;
4782 }
Eric Laurentd595b7c2013-04-03 17:27:56 -07004783 }
Eric Laurent81784c32012-11-19 14:55:58 -08004784 } else {
Eric Laurentd595b7c2013-04-03 17:27:56 -07004785 // clear effect chain input buffer if the last active track started underruns
4786 // to avoid sending previous audio buffer again to effects
Eric Laurentfd477972013-10-25 18:10:40 -07004787 if (!mEffectChains.isEmpty() && last) {
Eric Laurent81784c32012-11-19 14:55:58 -08004788 mEffectChains[0]->clearInputBuffer();
4789 }
Eric Laurentab5cdba2014-06-09 17:22:27 -07004790 if (track->isStopping_1()) {
4791 track->mState = TrackBase::STOPPING_2;
Eric Laurentb369caf2015-03-30 20:51:47 -07004792 if (last && mHwPaused) {
4793 doHwResume = true;
4794 mHwPaused = false;
4795 }
Eric Laurentab5cdba2014-06-09 17:22:27 -07004796 }
4797 if ((track->sharedBuffer() != 0) || track->isStopped() ||
4798 track->isStopping_2() || track->isPaused()) {
Eric Laurent81784c32012-11-19 14:55:58 -08004799 // We have consumed all the buffers of this track.
4800 // Remove it from the list of active tracks.
Eric Laurentab5cdba2014-06-09 17:22:27 -07004801 size_t audioHALFrames;
Phil Burkfdb3c072016-02-09 10:47:02 -08004802 if (audio_has_proportional_frames(mFormat)) {
Eric Laurentab5cdba2014-06-09 17:22:27 -07004803 audioHALFrames = (latency_l() * mSampleRate) / 1000;
4804 } else {
4805 audioHALFrames = 0;
4806 }
4807
Andy Hung818e7a32016-02-16 18:08:07 -08004808 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurentfd477972013-10-25 18:10:40 -07004809 if (mStandby || !last ||
4810 track->presentationComplete(framesWritten, audioHALFrames)) {
Eric Laurentab5cdba2014-06-09 17:22:27 -07004811 if (track->isStopping_2()) {
4812 track->mState = TrackBase::STOPPED;
4813 }
Eric Laurent81784c32012-11-19 14:55:58 -08004814 if (track->isStopped()) {
4815 track->reset();
4816 }
Eric Laurentd595b7c2013-04-03 17:27:56 -07004817 tracksToRemove->add(track);
Eric Laurent81784c32012-11-19 14:55:58 -08004818 }
4819 } else {
4820 // No buffers for this track. Give it a few chances to
4821 // fill a buffer, then remove it from active list.
Eric Laurentd595b7c2013-04-03 17:27:56 -07004822 // Only consider last track started for mixer state control
Eric Laurent81784c32012-11-19 14:55:58 -08004823 if (--(track->mRetryCount) <= 0) {
4824 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
Eric Laurentd595b7c2013-04-03 17:27:56 -07004825 tracksToRemove->add(track);
Eric Laurenta23f17a2013-11-05 18:22:08 -08004826 // indicate to client process that the track was disabled because of underrun;
4827 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08004828 track->disable();
Eric Laurentbfb1b832013-01-07 09:53:42 -08004829 } else if (last) {
Phil Burkca5e6142015-07-14 09:42:29 -07004830 ALOGW("pause because of UNDERRUN, framesReady = %zu,"
4831 "minFrames = %u, mFormat = %#x",
4832 track->framesReady(), minFrames, mFormat);
Eric Laurent81784c32012-11-19 14:55:58 -08004833 mixerStatus = MIXER_TRACKS_ENABLED;
Eric Laurent5cff4032015-05-26 13:49:58 -07004834 if (mHwSupportsPause && !mHwPaused && !mStandby) {
Eric Laurent0f7b5f22014-12-19 10:43:21 -08004835 doHwPause = true;
4836 mHwPaused = true;
4837 }
Eric Laurent81784c32012-11-19 14:55:58 -08004838 }
4839 }
4840 }
4841 }
4842
Eric Laurentd1f69b02014-12-15 14:33:13 -08004843 // if an active track did not command a flush, check for pending flush on stopped tracks
Phil Burk43b4dcc2015-06-09 16:53:44 -07004844 if (!mFlushPending) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004845 for (size_t i = 0; i < mTracks.size(); i++) {
4846 if (mTracks[i]->isFlushPending()) {
4847 mTracks[i]->flushAck();
Phil Burk43b4dcc2015-06-09 16:53:44 -07004848 mFlushPending = true;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004849 }
4850 }
4851 }
4852
4853 // make sure the pause/flush/resume sequence is executed in the right order.
4854 // If a flush is pending and a track is active but the HW is not paused, force a HW pause
4855 // before flush and then resume HW. This can happen in case of pause/flush/resume
4856 // if resume is received before pause is executed.
4857 if (mHwSupportsPause && !mStandby &&
Phil Burk43b4dcc2015-06-09 16:53:44 -07004858 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004859 mOutput->stream->pause(mOutput->stream);
4860 }
Phil Burk43b4dcc2015-06-09 16:53:44 -07004861 if (mFlushPending) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004862 flushHw_l();
4863 }
4864 if (mHwSupportsPause && !mStandby && doHwResume) {
4865 mOutput->stream->resume(mOutput->stream);
4866 }
Eric Laurent81784c32012-11-19 14:55:58 -08004867 // remove all the tracks that need to be...
Eric Laurentbfb1b832013-01-07 09:53:42 -08004868 removeTracks_l(*tracksToRemove);
Eric Laurent81784c32012-11-19 14:55:58 -08004869
4870 return mixerStatus;
4871}
4872
4873void AudioFlinger::DirectOutputThread::threadLoop_mix()
4874{
Eric Laurent81784c32012-11-19 14:55:58 -08004875 size_t frameCount = mFrameCount;
Andy Hung2098f272014-02-27 14:00:06 -08004876 int8_t *curBuf = (int8_t *)mSinkBuffer;
Eric Laurent81784c32012-11-19 14:55:58 -08004877 // output audio to hardware
4878 while (frameCount) {
Glenn Kasten34542ac2013-06-26 11:29:02 -07004879 AudioBufferProvider::Buffer buffer;
Eric Laurent81784c32012-11-19 14:55:58 -08004880 buffer.frameCount = frameCount;
Phil Burk062e67a2015-02-11 13:40:50 -08004881 status_t status = mActiveTrack->getNextBuffer(&buffer);
4882 if (status != NO_ERROR || buffer.raw == NULL) {
Eric Laurent51716182016-02-29 18:00:56 -08004883 // no need to pad with 0 for compressed audio
4884 if (audio_has_proportional_frames(mFormat)) {
4885 memset(curBuf, 0, frameCount * mFrameSize);
4886 }
Eric Laurent81784c32012-11-19 14:55:58 -08004887 break;
4888 }
4889 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
4890 frameCount -= buffer.frameCount;
4891 curBuf += buffer.frameCount * mFrameSize;
4892 mActiveTrack->releaseBuffer(&buffer);
4893 }
Andy Hung2098f272014-02-27 14:00:06 -08004894 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004895 mSleepTimeUs = 0;
4896 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
Eric Laurent81784c32012-11-19 14:55:58 -08004897 mActiveTrack.clear();
Eric Laurent81784c32012-11-19 14:55:58 -08004898}
4899
4900void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
4901{
Eric Laurentd1f69b02014-12-15 14:33:13 -08004902 // do not write to HAL when paused
Eric Laurent0f7b5f22014-12-19 10:43:21 -08004903 if (mHwPaused || (usesHwAvSync() && mStandby)) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004904 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004905 return;
4906 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004907 if (mSleepTimeUs == 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08004908 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
Eric Laurente93cc032016-05-05 10:15:10 -07004909 mSleepTimeUs = mActiveSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08004910 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004911 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08004912 }
Phil Burkfdb3c072016-02-09 10:47:02 -08004913 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) {
Andy Hung2098f272014-02-27 14:00:06 -08004914 memset(mSinkBuffer, 0, mFrameCount * mFrameSize);
Eric Laurentad9cb8b2015-05-26 16:38:19 -07004915 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08004916 }
4917}
4918
Eric Laurentd1f69b02014-12-15 14:33:13 -08004919void AudioFlinger::DirectOutputThread::threadLoop_exit()
4920{
4921 {
4922 Mutex::Autolock _l(mLock);
Eric Laurentd1f69b02014-12-15 14:33:13 -08004923 for (size_t i = 0; i < mTracks.size(); i++) {
4924 if (mTracks[i]->isFlushPending()) {
4925 mTracks[i]->flushAck();
Phil Burk43b4dcc2015-06-09 16:53:44 -07004926 mFlushPending = true;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004927 }
4928 }
Phil Burk43b4dcc2015-06-09 16:53:44 -07004929 if (mFlushPending) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004930 flushHw_l();
4931 }
4932 }
4933 PlaybackThread::threadLoop_exit();
4934}
4935
4936// must be called with thread mutex locked
4937bool AudioFlinger::DirectOutputThread::shouldStandby_l()
4938{
4939 bool trackPaused = false;
Eric Laurentb369caf2015-03-30 20:51:47 -07004940 bool trackStopped = false;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004941
vivek mehta9cd7ad12016-03-17 00:18:29 -07004942 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) {
4943 return !mStandby;
4944 }
4945
Eric Laurentd1f69b02014-12-15 14:33:13 -08004946 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
4947 // after a timeout and we will enter standby then.
4948 if (mTracks.size() > 0) {
4949 trackPaused = mTracks[mTracks.size() - 1]->isPaused();
Eric Laurentb369caf2015-03-30 20:51:47 -07004950 trackStopped = mTracks[mTracks.size() - 1]->isStopped() ||
4951 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004952 }
4953
Eric Laurent5cff4032015-05-26 13:49:58 -07004954 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped));
Eric Laurentd1f69b02014-12-15 14:33:13 -08004955}
4956
Eric Laurent81784c32012-11-19 14:55:58 -08004957// getTrackName_l() must be called with ThreadBase::mLock held
Glenn Kasten0f11b512014-01-31 16:18:54 -08004958int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
Glenn Kastend848eb42016-03-08 13:42:11 -08004959 audio_format_t format __unused, audio_session_t sessionId __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08004960{
4961 return 0;
4962}
4963
4964// deleteTrackName_l() must be called with ThreadBase::mLock held
Glenn Kasten0f11b512014-01-31 16:18:54 -08004965void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08004966{
4967}
4968
Eric Laurent10351942014-05-08 18:49:52 -07004969// checkForNewParameter_l() must be called with ThreadBase::mLock held
4970bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair,
4971 status_t& status)
Eric Laurent81784c32012-11-19 14:55:58 -08004972{
4973 bool reconfig = false;
Eric Laurent42537be2016-01-08 17:16:42 -08004974 bool a2dpDeviceChanged = false;
Eric Laurent81784c32012-11-19 14:55:58 -08004975
Eric Laurent10351942014-05-08 18:49:52 -07004976 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08004977
Eric Laurent10351942014-05-08 18:49:52 -07004978 AudioParameter param = AudioParameter(keyValuePair);
4979 int value;
4980 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
4981 // forward device change to effects that have requested to be
4982 // aware of attached audio device.
4983 if (value != AUDIO_DEVICE_NONE) {
Eric Laurent42537be2016-01-08 17:16:42 -08004984 a2dpDeviceChanged =
4985 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP);
Eric Laurent10351942014-05-08 18:49:52 -07004986 mOutDevice = value;
4987 for (size_t i = 0; i < mEffectChains.size(); i++) {
4988 mEffectChains[i]->setDevice_l(mOutDevice);
Glenn Kastenc125f382014-04-11 18:37:33 -07004989 }
4990 }
Eric Laurent81784c32012-11-19 14:55:58 -08004991 }
Eric Laurent10351942014-05-08 18:49:52 -07004992 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4993 // do not accept frame count changes if tracks are open as the track buffer
4994 // size depends on frame count and correct behavior would not be garantied
4995 // if frame count is changed after track creation
4996 if (!mTracks.isEmpty()) {
4997 status = INVALID_OPERATION;
4998 } else {
4999 reconfig = true;
5000 }
5001 }
5002 if (status == NO_ERROR) {
5003 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
5004 keyValuePair.string());
5005 if (!mStandby && status == INVALID_OPERATION) {
Phil Burk062e67a2015-02-11 13:40:50 -08005006 mOutput->standby();
Eric Laurent10351942014-05-08 18:49:52 -07005007 mStandby = true;
5008 mBytesWritten = 0;
5009 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
5010 keyValuePair.string());
5011 }
5012 if (status == NO_ERROR && reconfig) {
5013 readOutputParameters_l();
Eric Laurent73e26b62015-04-27 16:55:58 -07005014 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
Eric Laurent10351942014-05-08 18:49:52 -07005015 }
5016 }
5017
Eric Laurent42537be2016-01-08 17:16:42 -08005018 return reconfig || a2dpDeviceChanged;
Eric Laurent81784c32012-11-19 14:55:58 -08005019}
5020
5021uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
5022{
5023 uint32_t time;
Phil Burkfdb3c072016-02-09 10:47:02 -08005024 if (audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08005025 time = PlaybackThread::activeSleepTimeUs();
5026 } else {
Eric Laurent51716182016-02-29 18:00:56 -08005027 time = kDirectMinSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005028 }
5029 return time;
5030}
5031
5032uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
5033{
5034 uint32_t time;
Phil Burkfdb3c072016-02-09 10:47:02 -08005035 if (audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08005036 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
5037 } else {
Eric Laurent51716182016-02-29 18:00:56 -08005038 time = kDirectMinSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005039 }
5040 return time;
5041}
5042
5043uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
5044{
5045 uint32_t time;
Phil Burkfdb3c072016-02-09 10:47:02 -08005046 if (audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08005047 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
5048 } else {
Eric Laurent51716182016-02-29 18:00:56 -08005049 time = kDirectMinSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005050 }
5051 return time;
5052}
5053
5054void AudioFlinger::DirectOutputThread::cacheParameters_l()
5055{
5056 PlaybackThread::cacheParameters_l();
5057
5058 // use shorter standby delay as on normal output to release
5059 // hardware resources as soon as possible
Eric Laurentb369caf2015-03-30 20:51:47 -07005060 // no delay on outputs with HW A/V sync
5061 if (usesHwAvSync()) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005062 mStandbyDelayNs = 0;
Phil Burkfdb3c072016-02-09 10:47:02 -08005063 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005064 mStandbyDelayNs = kOffloadStandbyDelayNs;
Eric Laurent5cff4032015-05-26 13:49:58 -07005065 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005066 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2);
Eric Laurent972a1732013-09-04 09:42:59 -07005067 }
Eric Laurent81784c32012-11-19 14:55:58 -08005068}
5069
Eric Laurente659ef42014-09-29 13:06:46 -07005070void AudioFlinger::DirectOutputThread::flushHw_l()
5071{
Phil Burk062e67a2015-02-11 13:40:50 -08005072 mOutput->flush();
Eric Laurentd1f69b02014-12-15 14:33:13 -08005073 mHwPaused = false;
Phil Burk43b4dcc2015-06-09 16:53:44 -07005074 mFlushPending = false;
Eric Laurente659ef42014-09-29 13:06:46 -07005075}
5076
Eric Laurent81784c32012-11-19 14:55:58 -08005077// ----------------------------------------------------------------------------
5078
Eric Laurentbfb1b832013-01-07 09:53:42 -08005079AudioFlinger::AsyncCallbackThread::AsyncCallbackThread(
Eric Laurent4de95592013-09-26 15:28:21 -07005080 const wp<AudioFlinger::PlaybackThread>& playbackThread)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005081 : Thread(false /*canCallJava*/),
Eric Laurent4de95592013-09-26 15:28:21 -07005082 mPlaybackThread(playbackThread),
Eric Laurent3b4529e2013-09-05 18:09:19 -07005083 mWriteAckSequence(0),
5084 mDrainSequence(0)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005085{
5086}
5087
5088AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread()
5089{
5090}
5091
5092void AudioFlinger::AsyncCallbackThread::onFirstRef()
5093{
5094 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO);
5095}
5096
5097bool AudioFlinger::AsyncCallbackThread::threadLoop()
5098{
5099 while (!exitPending()) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07005100 uint32_t writeAckSequence;
5101 uint32_t drainSequence;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005102
5103 {
5104 Mutex::Autolock _l(mLock);
Haynes Mathew George24a325d2013-12-03 21:26:02 -08005105 while (!((mWriteAckSequence & 1) ||
5106 (mDrainSequence & 1) ||
5107 exitPending())) {
5108 mWaitWorkCV.wait(mLock);
5109 }
5110
Eric Laurentbfb1b832013-01-07 09:53:42 -08005111 if (exitPending()) {
5112 break;
5113 }
Eric Laurent3b4529e2013-09-05 18:09:19 -07005114 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d",
5115 mWriteAckSequence, mDrainSequence);
5116 writeAckSequence = mWriteAckSequence;
5117 mWriteAckSequence &= ~1;
5118 drainSequence = mDrainSequence;
5119 mDrainSequence &= ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005120 }
5121 {
Eric Laurent4de95592013-09-26 15:28:21 -07005122 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote();
5123 if (playbackThread != 0) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07005124 if (writeAckSequence & 1) {
Eric Laurent4de95592013-09-26 15:28:21 -07005125 playbackThread->resetWriteBlocked(writeAckSequence >> 1);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005126 }
Eric Laurent3b4529e2013-09-05 18:09:19 -07005127 if (drainSequence & 1) {
Eric Laurent4de95592013-09-26 15:28:21 -07005128 playbackThread->resetDraining(drainSequence >> 1);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005129 }
5130 }
5131 }
5132 }
5133 return false;
5134}
5135
5136void AudioFlinger::AsyncCallbackThread::exit()
5137{
5138 ALOGV("AsyncCallbackThread::exit");
5139 Mutex::Autolock _l(mLock);
5140 requestExit();
5141 mWaitWorkCV.broadcast();
5142}
5143
Eric Laurent3b4529e2013-09-05 18:09:19 -07005144void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005145{
5146 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07005147 // bit 0 is cleared
5148 mWriteAckSequence = sequence << 1;
5149}
5150
5151void AudioFlinger::AsyncCallbackThread::resetWriteBlocked()
5152{
5153 Mutex::Autolock _l(mLock);
5154 // ignore unexpected callbacks
5155 if (mWriteAckSequence & 2) {
5156 mWriteAckSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005157 mWaitWorkCV.signal();
5158 }
5159}
5160
Eric Laurent3b4529e2013-09-05 18:09:19 -07005161void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005162{
5163 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07005164 // bit 0 is cleared
5165 mDrainSequence = sequence << 1;
5166}
5167
5168void AudioFlinger::AsyncCallbackThread::resetDraining()
5169{
5170 Mutex::Autolock _l(mLock);
5171 // ignore unexpected callbacks
5172 if (mDrainSequence & 2) {
5173 mDrainSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005174 mWaitWorkCV.signal();
5175 }
5176}
5177
5178
5179// ----------------------------------------------------------------------------
5180AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
Eric Laurente93cc032016-05-05 10:15:10 -07005181 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady)
5182 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady),
Eric Laurent64667972016-03-30 18:19:46 -07005183 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005184{
Eric Laurentfd477972013-10-25 18:10:40 -07005185 //FIXME: mStandby should be set to true by ThreadBase constructor
5186 mStandby = true;
Eric Laurent64667972016-03-30 18:19:46 -07005187 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005188}
5189
Eric Laurentbfb1b832013-01-07 09:53:42 -08005190void AudioFlinger::OffloadThread::threadLoop_exit()
5191{
5192 if (mFlushPending || mHwPaused) {
5193 // If a flush is pending or track was paused, just discard buffered data
5194 flushHw_l();
5195 } else {
5196 mMixerStatus = MIXER_DRAIN_ALL;
5197 threadLoop_drain();
5198 }
Uday Gupta56604aa2014-05-13 11:19:17 -07005199 if (mUseAsyncWrite) {
5200 ALOG_ASSERT(mCallbackThread != 0);
5201 mCallbackThread->exit();
5202 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005203 PlaybackThread::threadLoop_exit();
5204}
5205
5206AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l(
5207 Vector< sp<Track> > *tracksToRemove
5208)
5209{
Eric Laurentbfb1b832013-01-07 09:53:42 -08005210 size_t count = mActiveTracks.size();
5211
5212 mixer_state mixerStatus = MIXER_IDLE;
Eric Laurent972a1732013-09-04 09:42:59 -07005213 bool doHwPause = false;
5214 bool doHwResume = false;
5215
Glenn Kastenc42e9b42016-03-21 11:35:03 -07005216 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count);
Eric Laurentede6c3b2013-09-19 14:37:46 -07005217
Eric Laurentbfb1b832013-01-07 09:53:42 -08005218 // find out which tracks need to be processed
5219 for (size_t i = 0; i < count; i++) {
5220 sp<Track> t = mActiveTracks[i].promote();
5221 // The track died recently
5222 if (t == 0) {
5223 continue;
5224 }
5225 Track* const track = t.get();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07005226#ifdef VERY_VERY_VERBOSE_LOGGING
Eric Laurentbfb1b832013-01-07 09:53:42 -08005227 audio_track_cblk_t* cblk = track->cblk();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07005228#endif
Eric Laurentfd477972013-10-25 18:10:40 -07005229 // Only consider last track started for volume and mixer state control.
5230 // In theory an older track could underrun and restart after the new one starts
5231 // but as we only care about the transition phase between two tracks on a
5232 // direct output, it is not a problem to ignore the underrun case.
5233 sp<Track> l = mLatestActiveTrack.promote();
5234 bool last = l.get() == track;
5235
Haynes Mathew George7844f672014-01-15 12:32:55 -08005236 if (track->isInvalid()) {
5237 ALOGW("An invalidated track shouldn't be in active list");
5238 tracksToRemove->add(track);
5239 continue;
5240 }
5241
5242 if (track->mState == TrackBase::IDLE) {
5243 ALOGW("An idle track shouldn't be in active list");
5244 continue;
5245 }
5246
Eric Laurentbfb1b832013-01-07 09:53:42 -08005247 if (track->isPausing()) {
5248 track->setPaused();
5249 if (last) {
Eric Laurent5cff4032015-05-26 13:49:58 -07005250 if (mHwSupportsPause && !mHwPaused) {
Eric Laurent972a1732013-09-04 09:42:59 -07005251 doHwPause = true;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005252 mHwPaused = true;
5253 }
5254 // If we were part way through writing the mixbuffer to
5255 // the HAL we must save this until we resume
5256 // BUG - this will be wrong if a different track is made active,
5257 // in that case we want to discard the pending data in the
5258 // mixbuffer and tell the client to present it again when the
5259 // track is resumed
5260 mPausedWriteLength = mCurrentWriteLength;
5261 mPausedBytesRemaining = mBytesRemaining;
5262 mBytesRemaining = 0; // stop writing
5263 }
5264 tracksToRemove->add(track);
Haynes Mathew George7844f672014-01-15 12:32:55 -08005265 } else if (track->isFlushPending()) {
Eric Laurente93cc032016-05-05 10:15:10 -07005266 if (track->isStopping_1()) {
5267 track->mRetryCount = kMaxTrackStopRetriesOffload;
5268 } else {
5269 track->mRetryCount = kMaxTrackRetriesOffload;
5270 }
Haynes Mathew George7844f672014-01-15 12:32:55 -08005271 track->flushAck();
5272 if (last) {
5273 mFlushPending = true;
5274 }
Haynes Mathew George2d3ca682014-03-07 13:43:49 -08005275 } else if (track->isResumePending()){
5276 track->resumeAck();
5277 if (last) {
5278 if (mPausedBytesRemaining) {
5279 // Need to continue write that was interrupted
5280 mCurrentWriteLength = mPausedWriteLength;
5281 mBytesRemaining = mPausedBytesRemaining;
5282 mPausedBytesRemaining = 0;
5283 }
5284 if (mHwPaused) {
5285 doHwResume = true;
5286 mHwPaused = false;
5287 // threadLoop_mix() will handle the case that we need to
5288 // resume an interrupted write
5289 }
5290 // enable write to audio HAL
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005291 mSleepTimeUs = 0;
Haynes Mathew George2d3ca682014-03-07 13:43:49 -08005292
5293 // Do not handle new data in this iteration even if track->framesReady()
5294 mixerStatus = MIXER_TRACKS_ENABLED;
5295 }
5296 } else if (track->framesReady() && track->isReady() &&
Eric Laurent3b4529e2013-09-05 18:09:19 -07005297 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07005298 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005299 if (track->mFillingUpStatus == Track::FS_FILLED) {
5300 track->mFillingUpStatus = Track::FS_ACTIVE;
Eric Laurent1abbdb42013-09-13 17:00:08 -07005301 // make sure processVolume_l() will apply new volume even if 0
5302 mLeftVolFloat = mRightVolFloat = -1.0;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005303 }
5304
5305 if (last) {
Eric Laurentd7e59222013-11-15 12:02:28 -08005306 sp<Track> previousTrack = mPreviousTrack.promote();
5307 if (previousTrack != 0) {
5308 if (track != previousTrack.get()) {
Eric Laurent9da3d952013-11-12 19:25:43 -08005309 // Flush any data still being written from last track
5310 mBytesRemaining = 0;
5311 if (mPausedBytesRemaining) {
5312 // Last track was paused so we also need to flush saved
5313 // mixbuffer state and invalidate track so that it will
5314 // re-submit that unwritten data when it is next resumed
5315 mPausedBytesRemaining = 0;
5316 // Invalidate is a bit drastic - would be more efficient
5317 // to have a flag to tell client that some of the
5318 // previously written data was lost
Eric Laurentd7e59222013-11-15 12:02:28 -08005319 previousTrack->invalidate();
Eric Laurent9da3d952013-11-12 19:25:43 -08005320 }
5321 // flush data already sent to the DSP if changing audio session as audio
5322 // comes from a different source. Also invalidate previous track to force a
5323 // seek when resuming.
Eric Laurentd7e59222013-11-15 12:02:28 -08005324 if (previousTrack->sessionId() != track->sessionId()) {
5325 previousTrack->invalidate();
Eric Laurent9da3d952013-11-12 19:25:43 -08005326 }
5327 }
5328 }
5329 mPreviousTrack = track;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005330 // reset retry count
Eric Laurente93cc032016-05-05 10:15:10 -07005331 if (track->isStopping_1()) {
5332 track->mRetryCount = kMaxTrackStopRetriesOffload;
5333 } else {
5334 track->mRetryCount = kMaxTrackRetriesOffload;
5335 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005336 mActiveTrack = t;
5337 mixerStatus = MIXER_TRACKS_READY;
5338 }
5339 } else {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07005340 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005341 if (track->isStopping_1()) {
Eric Laurente93cc032016-05-05 10:15:10 -07005342 if (--(track->mRetryCount) <= 0) {
5343 // Hardware buffer can hold a large amount of audio so we must
5344 // wait for all current track's data to drain before we say
5345 // that the track is stopped.
5346 if (mBytesRemaining == 0) {
5347 // Only start draining when all data in mixbuffer
5348 // has been written
5349 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2");
5350 track->mState = TrackBase::STOPPING_2; // so presentation completes after
5351 // drain do not drain if no data was ever sent to HAL (mStandby == true)
5352 if (last && !mStandby) {
5353 // do not modify drain sequence if we are already draining. This happens
5354 // when resuming from pause after drain.
5355 if ((mDrainSequence & 1) == 0) {
5356 mSleepTimeUs = 0;
5357 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
5358 mixerStatus = MIXER_DRAIN_TRACK;
5359 mDrainSequence += 2;
5360 }
5361 if (mHwPaused) {
5362 // It is possible to move from PAUSED to STOPPING_1 without
5363 // a resume so we must ensure hardware is running
5364 doHwResume = true;
5365 mHwPaused = false;
5366 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005367 }
5368 }
Eric Laurente93cc032016-05-05 10:15:10 -07005369 } else if (last) {
5370 ALOGV("stopping1 underrun retries left %d", track->mRetryCount);
5371 mixerStatus = MIXER_TRACKS_ENABLED;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005372 }
5373 } else if (track->isStopping_2()) {
Eric Laurent6a51d7e2013-10-17 18:59:26 -07005374 // Drain has completed or we are in standby, signal presentation complete
5375 if (!(mDrainSequence & 1) || !last || mStandby) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08005376 track->mState = TrackBase::STOPPED;
5377 size_t audioHALFrames =
5378 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08005379 int64_t framesWritten =
Phil Burk062e67a2015-02-11 13:40:50 -08005380 mBytesWritten / mOutput->getFrameSize();
Eric Laurentbfb1b832013-01-07 09:53:42 -08005381 track->presentationComplete(framesWritten, audioHALFrames);
5382 track->reset();
5383 tracksToRemove->add(track);
5384 }
5385 } else {
5386 // No buffers for this track. Give it a few chances to
5387 // fill a buffer, then remove it from active list.
5388 if (--(track->mRetryCount) <= 0) {
5389 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list",
5390 track->name());
5391 tracksToRemove->add(track);
Eric Laurenta23f17a2013-11-05 18:22:08 -08005392 // indicate to client process that the track was disabled because of underrun;
5393 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08005394 track->disable();
Eric Laurentbfb1b832013-01-07 09:53:42 -08005395 } else if (last){
5396 mixerStatus = MIXER_TRACKS_ENABLED;
5397 }
5398 }
5399 }
5400 // compute volume for this track
5401 processVolume_l(track, last);
5402 }
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005403
Eric Laurentea0fade2013-10-04 16:23:48 -07005404 // make sure the pause/flush/resume sequence is executed in the right order.
5405 // If a flush is pending and a track is active but the HW is not paused, force a HW pause
5406 // before flush and then resume HW. This can happen in case of pause/flush/resume
5407 // if resume is received before pause is executed.
Eric Laurentfd477972013-10-25 18:10:40 -07005408 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
Eric Laurent972a1732013-09-04 09:42:59 -07005409 mOutput->stream->pause(mOutput->stream);
5410 }
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005411 if (mFlushPending) {
5412 flushHw_l();
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005413 }
Eric Laurentfd477972013-10-25 18:10:40 -07005414 if (!mStandby && doHwResume) {
Eric Laurent972a1732013-09-04 09:42:59 -07005415 mOutput->stream->resume(mOutput->stream);
5416 }
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005417
Eric Laurentbfb1b832013-01-07 09:53:42 -08005418 // remove all the tracks that need to be...
5419 removeTracks_l(*tracksToRemove);
5420
5421 return mixerStatus;
5422}
5423
Eric Laurentbfb1b832013-01-07 09:53:42 -08005424// must be called with thread mutex locked
5425bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
5426{
Eric Laurent3b4529e2013-09-05 18:09:19 -07005427 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d",
5428 mWriteAckSequence, mDrainSequence);
5429 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08005430 return true;
5431 }
5432 return false;
5433}
5434
Eric Laurentbfb1b832013-01-07 09:53:42 -08005435bool AudioFlinger::OffloadThread::waitingAsyncCallback()
5436{
5437 Mutex::Autolock _l(mLock);
5438 return waitingAsyncCallback_l();
5439}
5440
5441void AudioFlinger::OffloadThread::flushHw_l()
5442{
Eric Laurente659ef42014-09-29 13:06:46 -07005443 DirectOutputThread::flushHw_l();
Eric Laurentbfb1b832013-01-07 09:53:42 -08005444 // Flush anything still waiting in the mixbuffer
5445 mCurrentWriteLength = 0;
5446 mBytesRemaining = 0;
5447 mPausedWriteLength = 0;
5448 mPausedBytesRemaining = 0;
Eric Laurent3eaf66b2016-04-01 14:44:17 -07005449 // reset bytes written count to reflect that DSP buffers are empty after flush.
5450 mBytesWritten = 0;
Haynes Mathew George0f02f262014-01-11 13:03:57 -08005451
Eric Laurentbfb1b832013-01-07 09:53:42 -08005452 if (mUseAsyncWrite) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07005453 // discard any pending drain or write ack by incrementing sequence
5454 mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
5455 mDrainSequence = (mDrainSequence + 2) & ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005456 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07005457 mCallbackThread->setWriteBlocked(mWriteAckSequence);
5458 mCallbackThread->setDraining(mDrainSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005459 }
5460}
5461
Haynes Mathew George05317d22016-05-03 16:34:26 -07005462void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType)
5463{
5464 Mutex::Autolock _l(mLock);
Eric Laurent13084622016-05-17 10:51:49 -07005465 if (PlaybackThread::invalidateTracks_l(streamType)) {
5466 mFlushPending = true;
5467 }
Haynes Mathew George05317d22016-05-03 16:34:26 -07005468}
5469
Eric Laurentbfb1b832013-01-07 09:53:42 -08005470// ----------------------------------------------------------------------------
5471
Eric Laurent81784c32012-11-19 14:55:58 -08005472AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
Eric Laurent72e3f392015-05-20 14:43:50 -07005473 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady)
Eric Laurent81784c32012-11-19 14:55:58 -08005474 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
Eric Laurent72e3f392015-05-20 14:43:50 -07005475 systemReady, DUPLICATING),
Eric Laurent81784c32012-11-19 14:55:58 -08005476 mWaitTimeMs(UINT_MAX)
5477{
5478 addOutputTrack(mainThread);
5479}
5480
5481AudioFlinger::DuplicatingThread::~DuplicatingThread()
5482{
5483 for (size_t i = 0; i < mOutputTracks.size(); i++) {
5484 mOutputTracks[i]->destroy();
5485 }
5486}
5487
5488void AudioFlinger::DuplicatingThread::threadLoop_mix()
5489{
5490 // mix buffers...
5491 if (outputsReady(outputTracks)) {
Glenn Kastend79072e2016-01-06 08:41:20 -08005492 mAudioMixer->process();
Eric Laurent81784c32012-11-19 14:55:58 -08005493 } else {
Eric Laurent02b57082014-11-07 17:28:28 -08005494 if (mMixerBufferValid) {
5495 memset(mMixerBuffer, 0, mMixerBufferSize);
5496 } else {
5497 memset(mSinkBuffer, 0, mSinkBufferSize);
5498 }
Eric Laurent81784c32012-11-19 14:55:58 -08005499 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005500 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08005501 writeFrames = mNormalFrameCount;
Andy Hung25c2dac2014-02-27 14:56:00 -08005502 mCurrentWriteLength = mSinkBufferSize;
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005503 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
Eric Laurent81784c32012-11-19 14:55:58 -08005504}
5505
5506void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
5507{
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005508 if (mSleepTimeUs == 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08005509 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005510 mSleepTimeUs = mActiveSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005511 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005512 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005513 }
5514 } else if (mBytesWritten != 0) {
5515 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
5516 writeFrames = mNormalFrameCount;
Andy Hung25c2dac2014-02-27 14:56:00 -08005517 memset(mSinkBuffer, 0, mSinkBufferSize);
Eric Laurent81784c32012-11-19 14:55:58 -08005518 } else {
5519 // flush remaining overflow buffers in output tracks
5520 writeFrames = 0;
5521 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005522 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08005523 }
5524}
5525
Eric Laurentbfb1b832013-01-07 09:53:42 -08005526ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
Eric Laurent81784c32012-11-19 14:55:58 -08005527{
5528 for (size_t i = 0; i < outputTracks.size(); i++) {
Andy Hungc25b84a2015-01-14 19:04:10 -08005529 outputTracks[i]->write(mSinkBuffer, writeFrames);
Eric Laurent81784c32012-11-19 14:55:58 -08005530 }
Eric Laurent2c3740f2013-10-30 16:57:06 -07005531 mStandby = false;
Andy Hung25c2dac2014-02-27 14:56:00 -08005532 return (ssize_t)mSinkBufferSize;
Eric Laurent81784c32012-11-19 14:55:58 -08005533}
5534
5535void AudioFlinger::DuplicatingThread::threadLoop_standby()
5536{
5537 // DuplicatingThread implements standby by stopping all tracks
5538 for (size_t i = 0; i < outputTracks.size(); i++) {
5539 outputTracks[i]->stop();
5540 }
5541}
5542
5543void AudioFlinger::DuplicatingThread::saveOutputTracks()
5544{
5545 outputTracks = mOutputTracks;
5546}
5547
5548void AudioFlinger::DuplicatingThread::clearOutputTracks()
5549{
5550 outputTracks.clear();
5551}
5552
5553void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
5554{
5555 Mutex::Autolock _l(mLock);
Andy Hungc25b84a2015-01-14 19:04:10 -08005556 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
5557 // Adjust for thread->sampleRate() to determine minimum buffer frame count.
5558 // Then triple buffer because Threads do not run synchronously and may not be clock locked.
5559 const size_t frameCount =
5560 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate());
5561 // TODO: Consider asynchronous sample rate conversion to handle clock disparity
5562 // from different OutputTracks and their associated MixerThreads (e.g. one may
5563 // nearly empty and the other may be dropping data).
5564
5565 sp<OutputTrack> outputTrack = new OutputTrack(thread,
Eric Laurent81784c32012-11-19 14:55:58 -08005566 this,
5567 mSampleRate,
Andy Hungc25b84a2015-01-14 19:04:10 -08005568 mFormat,
Eric Laurent81784c32012-11-19 14:55:58 -08005569 mChannelMask,
Marco Nelissen462fd2f2013-01-14 14:12:05 -08005570 frameCount,
5571 IPCThreadState::self()->getCallingUid());
Eric Laurent81784c32012-11-19 14:55:58 -08005572 if (outputTrack->cblk() != NULL) {
Eric Laurent223fd5c2014-11-11 13:43:36 -08005573 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
Eric Laurent81784c32012-11-19 14:55:58 -08005574 mOutputTracks.add(outputTrack);
Andy Hungc25b84a2015-01-14 19:04:10 -08005575 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
Eric Laurent81784c32012-11-19 14:55:58 -08005576 updateWaitTime_l();
5577 }
5578}
5579
5580void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
5581{
5582 Mutex::Autolock _l(mLock);
5583 for (size_t i = 0; i < mOutputTracks.size(); i++) {
5584 if (mOutputTracks[i]->thread() == thread) {
5585 mOutputTracks[i]->destroy();
5586 mOutputTracks.removeAt(i);
5587 updateWaitTime_l();
Eric Laurentf6870ae2015-05-08 10:50:03 -07005588 if (thread->getOutput() == mOutput) {
5589 mOutput = NULL;
5590 }
Eric Laurent81784c32012-11-19 14:55:58 -08005591 return;
5592 }
5593 }
Eric Laurentf6870ae2015-05-08 10:50:03 -07005594 ALOGV("removeOutputTrack(): unknown thread: %p", thread);
Eric Laurent81784c32012-11-19 14:55:58 -08005595}
5596
5597// caller must hold mLock
5598void AudioFlinger::DuplicatingThread::updateWaitTime_l()
5599{
5600 mWaitTimeMs = UINT_MAX;
5601 for (size_t i = 0; i < mOutputTracks.size(); i++) {
5602 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
5603 if (strong != 0) {
5604 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
5605 if (waitTimeMs < mWaitTimeMs) {
5606 mWaitTimeMs = waitTimeMs;
5607 }
5608 }
5609 }
5610}
5611
5612
5613bool AudioFlinger::DuplicatingThread::outputsReady(
5614 const SortedVector< sp<OutputTrack> > &outputTracks)
5615{
5616 for (size_t i = 0; i < outputTracks.size(); i++) {
5617 sp<ThreadBase> thread = outputTracks[i]->thread().promote();
5618 if (thread == 0) {
5619 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
5620 outputTracks[i].get());
5621 return false;
5622 }
5623 PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
5624 // see note at standby() declaration
5625 if (playbackThread->standby() && !playbackThread->isSuspended()) {
5626 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
5627 thread.get());
5628 return false;
5629 }
5630 }
5631 return true;
5632}
5633
5634uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
5635{
5636 return (mWaitTimeMs * 1000) / 2;
5637}
5638
5639void AudioFlinger::DuplicatingThread::cacheParameters_l()
5640{
5641 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
5642 updateWaitTime_l();
5643
5644 MixerThread::cacheParameters_l();
5645}
5646
5647// ----------------------------------------------------------------------------
5648// Record
5649// ----------------------------------------------------------------------------
5650
5651AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
5652 AudioStreamIn *input,
Eric Laurent81784c32012-11-19 14:55:58 -08005653 audio_io_handle_t id,
Eric Laurentd3922f72013-02-01 17:57:04 -08005654 audio_devices_t outDevice,
Eric Laurent72e3f392015-05-20 14:43:50 -07005655 audio_devices_t inDevice,
5656 bool systemReady
Glenn Kasten46909e72013-02-26 09:20:22 -08005657#ifdef TEE_SINK
5658 , const sp<NBAIO_Sink>& teeSink
5659#endif
5660 ) :
Eric Laurent72e3f392015-05-20 14:43:50 -07005661 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady),
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005662 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL),
Glenn Kastendeca2ae2014-02-07 10:25:56 -08005663 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l()
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08005664 mRsmpInRear(0)
Glenn Kasten46909e72013-02-26 09:20:22 -08005665#ifdef TEE_SINK
5666 , mTeeSink(teeSink)
5667#endif
Glenn Kastenb880f5e2014-05-07 08:43:45 -07005668 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize,
5669 "RecordThreadRO", MemoryHeapBase::READ_ONLY))
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005670 // mFastCapture below
5671 , mFastCaptureFutex(0)
5672 // mInputSource
5673 // mPipeSink
5674 // mPipeSource
5675 , mPipeFramesP2(0)
5676 // mPipeMemory
5677 // mFastCaptureNBLogWriter
Glenn Kasten6e6704c2014-07-03 10:20:00 -07005678 , mFastTrackAvail(false)
Eric Laurent81784c32012-11-19 14:55:58 -08005679{
Glenn Kastend7dca052015-03-05 16:05:54 -08005680 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id);
5681 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
Eric Laurent81784c32012-11-19 14:55:58 -08005682
Glenn Kastendeca2ae2014-02-07 10:25:56 -08005683 readInputParameters_l();
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005684
5685 // create an NBAIO source for the HAL input stream, and negotiate
5686 mInputSource = new AudioStreamInSource(input->stream);
5687 size_t numCounterOffers = 0;
5688 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07005689#if !LOG_NDEBUG
5690 ssize_t index =
5691#else
5692 (void)
5693#endif
5694 mInputSource->negotiate(offers, 1, NULL, numCounterOffers);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005695 ALOG_ASSERT(index == 0);
5696
5697 // initialize fast capture depending on configuration
5698 bool initFastCapture;
5699 switch (kUseFastCapture) {
5700 case FastCapture_Never:
5701 initFastCapture = false;
5702 break;
5703 case FastCapture_Always:
5704 initFastCapture = true;
5705 break;
5706 case FastCapture_Static:
Glenn Kasteneb9487e2015-07-22 09:15:17 -07005707 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005708 break;
5709 // case FastCapture_Dynamic:
5710 }
5711
5712 if (initFastCapture) {
Glenn Kastend198b852015-03-16 14:55:53 -07005713 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005714 NBAIO_Format format = mInputSource->format();
Glenn Kasten49d00ad2014-07-21 11:22:03 -07005715 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005716 size_t pipeSize = pipeFramesP2 * Format_frameSize(format);
5717 void *pipeBuffer;
5718 const sp<MemoryDealer> roHeap(readOnlyHeap());
5719 sp<IMemory> pipeMemory;
5720 if ((roHeap == 0) ||
5721 (pipeMemory = roHeap->allocate(pipeSize)) == 0 ||
5722 (pipeBuffer = pipeMemory->pointer()) == NULL) {
5723 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize);
5724 goto failed;
5725 }
5726 // pipe will be shared directly with fast clients, so clear to avoid leaking old information
5727 memset(pipeBuffer, 0, pipeSize);
5728 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer);
5729 const NBAIO_Format offers[1] = {format};
5730 size_t numCounterOffers = 0;
5731 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
5732 ALOG_ASSERT(index == 0);
5733 mPipeSink = pipe;
5734 PipeReader *pipeReader = new PipeReader(*pipe);
5735 numCounterOffers = 0;
5736 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
5737 ALOG_ASSERT(index == 0);
5738 mPipeSource = pipeReader;
5739 mPipeFramesP2 = pipeFramesP2;
5740 mPipeMemory = pipeMemory;
5741
5742 // create fast capture
5743 mFastCapture = new FastCapture();
5744 FastCaptureStateQueue *sq = mFastCapture->sq();
5745#ifdef STATE_QUEUE_DUMP
5746 // FIXME
5747#endif
5748 FastCaptureState *state = sq->begin();
5749 state->mCblk = NULL;
5750 state->mInputSource = mInputSource.get();
5751 state->mInputSourceGen++;
5752 state->mPipeSink = pipe;
5753 state->mPipeSinkGen++;
5754 state->mFrameCount = mFrameCount;
5755 state->mCommand = FastCaptureState::COLD_IDLE;
5756 // already done in constructor initialization list
5757 //mFastCaptureFutex = 0;
5758 state->mColdFutexAddr = &mFastCaptureFutex;
5759 state->mColdGen++;
5760 state->mDumpState = &mFastCaptureDumpState;
5761#ifdef TEE_SINK
5762 // FIXME
5763#endif
5764 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture");
5765 state->mNBLogWriter = mFastCaptureNBLogWriter.get();
5766 sq->end();
5767 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
5768
5769 // start the fast capture
5770 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
5771 pid_t tid = mFastCapture->getTid();
Glenn Kasten8379b722016-03-18 14:54:17 -07005772 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005773#ifdef AUDIO_WATCHDOG
5774 // FIXME
5775#endif
5776
Glenn Kasten6e6704c2014-07-03 10:20:00 -07005777 mFastTrackAvail = true;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005778 }
5779failed: ;
5780
5781 // FIXME mNormalSource
Eric Laurent81784c32012-11-19 14:55:58 -08005782}
5783
Eric Laurent81784c32012-11-19 14:55:58 -08005784AudioFlinger::RecordThread::~RecordThread()
5785{
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005786 if (mFastCapture != 0) {
5787 FastCaptureStateQueue *sq = mFastCapture->sq();
5788 FastCaptureState *state = sq->begin();
5789 if (state->mCommand == FastCaptureState::COLD_IDLE) {
5790 int32_t old = android_atomic_inc(&mFastCaptureFutex);
5791 if (old == -1) {
5792 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
5793 }
5794 }
5795 state->mCommand = FastCaptureState::EXIT;
5796 sq->end();
5797 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
5798 mFastCapture->join();
5799 mFastCapture.clear();
5800 }
5801 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter);
Glenn Kasten481fb672013-09-30 14:39:28 -07005802 mAudioFlinger->unregisterWriter(mNBLogWriter);
Andy Hung57446612015-04-19 23:56:46 -07005803 free(mRsmpInBuffer);
Eric Laurent81784c32012-11-19 14:55:58 -08005804}
5805
5806void AudioFlinger::RecordThread::onFirstRef()
5807{
Glenn Kastend7dca052015-03-05 16:05:54 -08005808 run(mThreadName, PRIORITY_URGENT_AUDIO);
Eric Laurent81784c32012-11-19 14:55:58 -08005809}
5810
Eric Laurent81784c32012-11-19 14:55:58 -08005811bool AudioFlinger::RecordThread::threadLoop()
5812{
Eric Laurent81784c32012-11-19 14:55:58 -08005813 nsecs_t lastWarning = 0;
5814
5815 inputStandBy();
Eric Laurent81784c32012-11-19 14:55:58 -08005816
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005817reacquire_wakelock:
5818 sp<RecordTrack> activeTrack;
Glenn Kasten2b806402013-11-20 16:37:38 -08005819 int activeTracksGen;
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005820 {
5821 Mutex::Autolock _l(mLock);
Glenn Kasten2b806402013-11-20 16:37:38 -08005822 size_t size = mActiveTracks.size();
5823 activeTracksGen = mActiveTracksGen;
5824 if (size > 0) {
5825 // FIXME an arbitrary choice
5826 activeTrack = mActiveTracks[0];
5827 acquireWakeLock_l(activeTrack->uid());
5828 if (size > 1) {
5829 SortedVector<int> tmp;
5830 for (size_t i = 0; i < size; i++) {
5831 tmp.add(mActiveTracks[i]->uid());
5832 }
5833 updateWakeLockUids_l(tmp);
5834 }
5835 } else {
5836 acquireWakeLock_l(-1);
5837 }
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005838 }
5839
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005840 // used to request a deferred sleep, to be executed later while mutex is unlocked
5841 uint32_t sleepUs = 0;
5842
5843 // loop while there is work to do
Glenn Kasten4ef0b462013-08-14 13:52:27 -07005844 for (;;) {
Glenn Kastenc527a7c2013-08-13 15:43:49 -07005845 Vector< sp<EffectChain> > effectChains;
Glenn Kasten2cfbf882013-08-14 13:12:11 -07005846
Glenn Kasten5edadd42013-08-14 16:30:49 -07005847 // sleep with mutex unlocked
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005848 if (sleepUs > 0) {
Glenn Kastene7754022014-10-31 12:11:26 -07005849 ATRACE_BEGIN("sleep");
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005850 usleep(sleepUs);
Glenn Kastene7754022014-10-31 12:11:26 -07005851 ATRACE_END();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005852 sleepUs = 0;
Glenn Kasten5edadd42013-08-14 16:30:49 -07005853 }
5854
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005855 // activeTracks accumulates a copy of a subset of mActiveTracks
5856 Vector< sp<RecordTrack> > activeTracks;
5857
Glenn Kasten735f45f2014-08-18 15:51:59 -07005858 // reference to the (first and only) active fast track
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005859 sp<RecordTrack> fastTrack;
Eric Laurent10351942014-05-08 18:49:52 -07005860
Glenn Kasten735f45f2014-08-18 15:51:59 -07005861 // reference to a fast track which is about to be removed
5862 sp<RecordTrack> fastTrackToRemove;
5863
Eric Laurent81784c32012-11-19 14:55:58 -08005864 { // scope for mLock
5865 Mutex::Autolock _l(mLock);
Eric Laurent000a4192014-01-29 15:17:32 -08005866
Eric Laurent021cf962014-05-13 10:18:14 -07005867 processConfigEvents_l();
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005868
Eric Laurent000a4192014-01-29 15:17:32 -08005869 // check exitPending here because checkForNewParameters_l() and
5870 // checkForNewParameters_l() can temporarily release mLock
5871 if (exitPending()) {
5872 break;
5873 }
5874
Glenn Kasten2b806402013-11-20 16:37:38 -08005875 // if no active track(s), then standby and release wakelock
5876 size_t size = mActiveTracks.size();
5877 if (size == 0) {
Glenn Kasten93e471f2013-08-19 08:40:07 -07005878 standbyIfNotAlreadyInStandby();
Glenn Kasten4ef0b462013-08-14 13:52:27 -07005879 // exitPending() can't become true here
Eric Laurent81784c32012-11-19 14:55:58 -08005880 releaseWakeLock_l();
5881 ALOGV("RecordThread: loop stopping");
5882 // go to sleep
5883 mWaitWorkCV.wait(mLock);
5884 ALOGV("RecordThread: loop starting");
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005885 goto reacquire_wakelock;
5886 }
5887
Glenn Kasten2b806402013-11-20 16:37:38 -08005888 if (mActiveTracksGen != activeTracksGen) {
5889 activeTracksGen = mActiveTracksGen;
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005890 SortedVector<int> tmp;
Glenn Kasten2b806402013-11-20 16:37:38 -08005891 for (size_t i = 0; i < size; i++) {
5892 tmp.add(mActiveTracks[i]->uid());
5893 }
Glenn Kastenf10ffec2013-11-20 16:40:08 -08005894 updateWakeLockUids_l(tmp);
Eric Laurent81784c32012-11-19 14:55:58 -08005895 }
Glenn Kasten9e982352013-08-14 14:39:50 -07005896
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005897 bool doBroadcast = false;
5898 for (size_t i = 0; i < size; ) {
Glenn Kasten9e982352013-08-14 14:39:50 -07005899
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005900 activeTrack = mActiveTracks[i];
5901 if (activeTrack->isTerminated()) {
Glenn Kasten735f45f2014-08-18 15:51:59 -07005902 if (activeTrack->isFastTrack()) {
5903 ALOG_ASSERT(fastTrackToRemove == 0);
5904 fastTrackToRemove = activeTrack;
5905 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005906 removeTrack_l(activeTrack);
Glenn Kasten2b806402013-11-20 16:37:38 -08005907 mActiveTracks.remove(activeTrack);
5908 mActiveTracksGen++;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005909 size--;
Glenn Kasten9e982352013-08-14 14:39:50 -07005910 continue;
5911 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005912
5913 TrackBase::track_state activeTrackState = activeTrack->mState;
5914 switch (activeTrackState) {
5915
5916 case TrackBase::PAUSING:
5917 mActiveTracks.remove(activeTrack);
5918 mActiveTracksGen++;
5919 doBroadcast = true;
5920 size--;
5921 continue;
5922
5923 case TrackBase::STARTING_1:
5924 sleepUs = 10000;
5925 i++;
5926 continue;
5927
5928 case TrackBase::STARTING_2:
5929 doBroadcast = true;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005930 mStandby = false;
Glenn Kasten9e982352013-08-14 14:39:50 -07005931 activeTrack->mState = TrackBase::ACTIVE;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005932 break;
5933
5934 case TrackBase::ACTIVE:
5935 break;
5936
5937 case TrackBase::IDLE:
5938 i++;
5939 continue;
5940
5941 default:
Glenn Kastenadad3d72014-02-21 14:51:43 -08005942 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState);
Glenn Kasten9e982352013-08-14 14:39:50 -07005943 }
Glenn Kasten9e982352013-08-14 14:39:50 -07005944
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005945 activeTracks.add(activeTrack);
5946 i++;
Glenn Kasten9e982352013-08-14 14:39:50 -07005947
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005948 if (activeTrack->isFastTrack()) {
5949 ALOG_ASSERT(!mFastTrackAvail);
5950 ALOG_ASSERT(fastTrack == 0);
5951 fastTrack = activeTrack;
5952 }
Glenn Kasten9e982352013-08-14 14:39:50 -07005953 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005954 if (doBroadcast) {
5955 mStartStopCond.broadcast();
5956 }
5957
5958 // sleep if there are no active tracks to process
5959 if (activeTracks.size() == 0) {
5960 if (sleepUs == 0) {
5961 sleepUs = kRecordThreadSleepUs;
5962 }
5963 continue;
5964 }
5965 sleepUs = 0;
Glenn Kasten9e982352013-08-14 14:39:50 -07005966
Eric Laurent81784c32012-11-19 14:55:58 -08005967 lockEffectChains_l(effectChains);
5968 }
5969
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005970 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0
Glenn Kasten71652682013-08-14 15:17:55 -07005971
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005972 size_t size = effectChains.size();
5973 for (size_t i = 0; i < size; i++) {
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07005974 // thread mutex is not locked, but effect chain is locked
5975 effectChains[i]->process_l();
5976 }
5977
Glenn Kasten735f45f2014-08-18 15:51:59 -07005978 // Push a new fast capture state if fast capture is not already running, or cblk change
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005979 if (mFastCapture != 0) {
5980 FastCaptureStateQueue *sq = mFastCapture->sq();
5981 FastCaptureState *state = sq->begin();
Glenn Kasten735f45f2014-08-18 15:51:59 -07005982 bool didModify = false;
5983 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005984 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME &&
5985 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) {
5986 if (state->mCommand == FastCaptureState::COLD_IDLE) {
5987 int32_t old = android_atomic_inc(&mFastCaptureFutex);
5988 if (old == -1) {
5989 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
5990 }
5991 }
5992 state->mCommand = FastCaptureState::READ_WRITE;
5993#if 0 // FIXME
5994 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
Glenn Kastenfbdb2ac2015-03-02 14:47:19 -08005995 FastThreadDumpState::kSamplingNforLowRamDevice :
5996 FastThreadDumpState::kSamplingN);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005997#endif
Glenn Kasten735f45f2014-08-18 15:51:59 -07005998 didModify = true;
5999 }
6000 audio_track_cblk_t *cblkOld = state->mCblk;
6001 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL;
6002 if (cblkNew != cblkOld) {
6003 state->mCblk = cblkNew;
6004 // block until acked if removing a fast track
6005 if (cblkOld != NULL) {
6006 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED;
6007 }
6008 didModify = true;
6009 }
6010 sq->end(didModify);
6011 if (didModify) {
6012 sq->push(block);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006013#if 0
6014 if (kUseFastCapture == FastCapture_Dynamic) {
6015 mNormalSource = mPipeSource;
6016 }
6017#endif
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006018 }
6019 }
6020
Glenn Kasten735f45f2014-08-18 15:51:59 -07006021 // now run the fast track destructor with thread mutex unlocked
6022 fastTrackToRemove.clear();
6023
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006024 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
6025 // Only the client(s) that are too slow will overrun. But if even the fastest client is too
6026 // slow, then this RecordThread will overrun by not calling HAL read often enough.
6027 // If destination is non-contiguous, first read past the nominal end of buffer, then
6028 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated.
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006029
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006030 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006031 ssize_t framesRead;
6032
6033 // If an NBAIO source is present, use it to read the normal capture's data
6034 if (mPipeSource != 0) {
6035 size_t framesToRead = mBufferSize / mFrameSize;
Andy Hung57446612015-04-19 23:56:46 -07006036 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
Glenn Kastend79072e2016-01-06 08:41:20 -08006037 framesToRead);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006038 if (framesRead == 0) {
6039 // since pipe is non-blocking, simulate blocking input
6040 sleepUs = (framesToRead * 1000000LL) / mSampleRate;
6041 }
6042 // otherwise use the HAL / AudioStreamIn directly
6043 } else {
Glenn Kastenec6a7032016-03-14 07:40:23 -07006044 ATRACE_BEGIN("read");
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006045 ssize_t bytesRead = mInput->stream->read(mInput->stream,
Andy Hung57446612015-04-19 23:56:46 -07006046 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize);
Glenn Kastenec6a7032016-03-14 07:40:23 -07006047 ATRACE_END();
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006048 if (bytesRead < 0) {
6049 framesRead = bytesRead;
6050 } else {
6051 framesRead = bytesRead / mFrameSize;
6052 }
6053 }
6054
Andy Hung3f0c9022016-01-15 17:49:46 -08006055 // Update server timestamp with server stats
6056 // systemTime() is optional if the hardware supports timestamps.
6057 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead;
6058 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime();
6059
6060 // Update server timestamp with kernel stats
6061 if (mInput->stream->get_capture_position != nullptr) {
6062 int64_t position, time;
6063 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time);
6064 if (ret == NO_ERROR) {
6065 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position;
6066 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time;
6067 // Note: In general record buffers should tend to be empty in
6068 // a properly running pipeline.
6069 //
6070 // Also, it is not advantageous to call get_presentation_position during the read
6071 // as the read obtains a lock, preventing the timestamp call from executing.
6072 }
6073 }
6074 // Use this to track timestamp information
6075 // ALOGD("%s", mTimestamp.toString().c_str());
6076
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006077 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006078 ALOGE("read failed: framesRead=%zd", framesRead);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006079 // Force input into standby so that it tries to recover at next read attempt
6080 inputStandBy();
6081 sleepUs = kRecordThreadSleepUs;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006082 }
6083 if (framesRead <= 0) {
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006084 goto unlock;
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006085 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006086 ALOG_ASSERT(framesRead > 0);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006087
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006088 if (mTeeSink != 0) {
Andy Hung57446612015-04-19 23:56:46 -07006089 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006090 }
6091 // If destination is non-contiguous, we now correct for reading past end of buffer.
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006092 {
6093 size_t part1 = mRsmpInFramesP2 - rear;
6094 if ((size_t) framesRead > part1) {
Andy Hung57446612015-04-19 23:56:46 -07006095 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize,
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006096 (framesRead - part1) * mFrameSize);
6097 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006098 }
6099 rear = mRsmpInRear += framesRead;
6100
6101 size = activeTracks.size();
6102 // loop over each active track
6103 for (size_t i = 0; i < size; i++) {
6104 activeTrack = activeTracks[i];
6105
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006106 // skip fast tracks, as those are handled directly by FastCapture
6107 if (activeTrack->isFastTrack()) {
6108 continue;
6109 }
6110
Andy Hung73c02e42015-03-29 01:13:58 -07006111 // TODO: This code probably should be moved to RecordTrack.
Andy Hung97a893e2015-03-29 01:03:07 -07006112 // TODO: Update the activeTrack buffer converter in case of reconfigure.
6113
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006114 enum {
6115 OVERRUN_UNKNOWN,
6116 OVERRUN_TRUE,
6117 OVERRUN_FALSE
6118 } overrun = OVERRUN_UNKNOWN;
6119
6120 // loop over getNextBuffer to handle circular sink
6121 for (;;) {
6122
6123 activeTrack->mSink.frameCount = ~0;
6124 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink);
6125 size_t framesOut = activeTrack->mSink.frameCount;
6126 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
6127
Andy Hung73c02e42015-03-29 01:13:58 -07006128 // check available frames and handle overrun conditions
6129 // if the record track isn't draining fast enough.
6130 bool hasOverrun;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006131 size_t framesIn;
Andy Hung73c02e42015-03-29 01:13:58 -07006132 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun);
6133 if (hasOverrun) {
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006134 overrun = OVERRUN_TRUE;
6135 }
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08006136 if (framesOut == 0 || framesIn == 0) {
6137 break;
6138 }
6139
Andy Hung6770c6f2015-04-07 13:43:36 -07006140 // Don't allow framesOut to be larger than what is possible with resampling
6141 // from framesIn.
6142 // This isn't strictly necessary but helps limit buffer resizing in
6143 // RecordBufferConverter. TODO: remove when no longer needed.
6144 framesOut = min(framesOut,
6145 destinationFramesPossible(
6146 framesIn, mSampleRate, activeTrack->mSampleRate));
Andy Hung97a893e2015-03-29 01:03:07 -07006147 // process frames from the RecordThread buffer provider to the RecordTrack buffer
6148 framesOut = activeTrack->mRecordBufferConverter->convert(
6149 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006150
6151 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
6152 overrun = OVERRUN_FALSE;
6153 }
6154
6155 if (activeTrack->mFramesToDrop == 0) {
6156 if (framesOut > 0) {
6157 activeTrack->mSink.frameCount = framesOut;
6158 activeTrack->releaseBuffer(&activeTrack->mSink);
6159 }
6160 } else {
6161 // FIXME could do a partial drop of framesOut
6162 if (activeTrack->mFramesToDrop > 0) {
6163 activeTrack->mFramesToDrop -= framesOut;
6164 if (activeTrack->mFramesToDrop <= 0) {
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006165 activeTrack->clearSyncStartEvent();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006166 }
6167 } else {
6168 activeTrack->mFramesToDrop += framesOut;
6169 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 ||
6170 activeTrack->mSyncStartEvent->isCancelled()) {
6171 ALOGW("Synced record %s, session %d, trigger session %d",
6172 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled",
6173 activeTrack->sessionId(),
6174 (activeTrack->mSyncStartEvent != 0) ?
Glenn Kastend848eb42016-03-08 13:42:11 -08006175 activeTrack->mSyncStartEvent->triggerSession() :
6176 AUDIO_SESSION_NONE);
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006177 activeTrack->clearSyncStartEvent();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006178 }
6179 }
6180 }
6181
6182 if (framesOut == 0) {
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006183 break;
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006184 }
6185 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006186
6187 switch (overrun) {
6188 case OVERRUN_TRUE:
6189 // client isn't retrieving buffers fast enough
6190 if (!activeTrack->setOverflow()) {
6191 nsecs_t now = systemTime();
6192 // FIXME should lastWarning per track?
6193 if ((now - lastWarning) > kWarningThrottleNs) {
6194 ALOGW("RecordThread: buffer overflow");
6195 lastWarning = now;
6196 }
6197 }
6198 break;
6199 case OVERRUN_FALSE:
6200 activeTrack->clearOverflow();
6201 break;
6202 case OVERRUN_UNKNOWN:
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006203 break;
6204 }
6205
Andy Hung3f0c9022016-01-15 17:49:46 -08006206 // update frame information and push timestamp out
6207 activeTrack->updateTrackFrameInfo(
Andy Hung6ae58432016-02-16 18:32:24 -08006208 activeTrack->mServerProxy->framesReleased(),
Andy Hung3f0c9022016-01-15 17:49:46 -08006209 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER],
6210 mSampleRate, mTimestamp);
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006211 }
6212
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006213unlock:
Eric Laurent81784c32012-11-19 14:55:58 -08006214 // enable changes in effect chain
6215 unlockEffectChains(effectChains);
Glenn Kastenc527a7c2013-08-13 15:43:49 -07006216 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end
Eric Laurent81784c32012-11-19 14:55:58 -08006217 }
6218
Glenn Kasten93e471f2013-08-19 08:40:07 -07006219 standbyIfNotAlreadyInStandby();
Eric Laurent81784c32012-11-19 14:55:58 -08006220
6221 {
6222 Mutex::Autolock _l(mLock);
Eric Laurent9a54bc22013-09-09 09:08:44 -07006223 for (size_t i = 0; i < mTracks.size(); i++) {
6224 sp<RecordTrack> track = mTracks[i];
6225 track->invalidate();
6226 }
Glenn Kasten2b806402013-11-20 16:37:38 -08006227 mActiveTracks.clear();
6228 mActiveTracksGen++;
Eric Laurent81784c32012-11-19 14:55:58 -08006229 mStartStopCond.broadcast();
6230 }
6231
6232 releaseWakeLock();
6233
6234 ALOGV("RecordThread %p exiting", this);
6235 return false;
6236}
6237
Glenn Kasten93e471f2013-08-19 08:40:07 -07006238void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
Eric Laurent81784c32012-11-19 14:55:58 -08006239{
6240 if (!mStandby) {
6241 inputStandBy();
6242 mStandby = true;
6243 }
6244}
6245
6246void AudioFlinger::RecordThread::inputStandBy()
6247{
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006248 // Idle the fast capture if it's currently running
6249 if (mFastCapture != 0) {
6250 FastCaptureStateQueue *sq = mFastCapture->sq();
6251 FastCaptureState *state = sq->begin();
6252 if (!(state->mCommand & FastCaptureState::IDLE)) {
6253 state->mCommand = FastCaptureState::COLD_IDLE;
6254 state->mColdFutexAddr = &mFastCaptureFutex;
6255 state->mColdGen++;
6256 mFastCaptureFutex = 0;
6257 sq->end();
6258 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
6259 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED);
6260#if 0
6261 if (kUseFastCapture == FastCapture_Dynamic) {
6262 // FIXME
6263 }
6264#endif
6265#ifdef AUDIO_WATCHDOG
6266 // FIXME
6267#endif
6268 } else {
6269 sq->end(false /*didModify*/);
6270 }
6271 }
Eric Laurent81784c32012-11-19 14:55:58 -08006272 mInput->stream->common.standby(&mInput->stream->common);
6273}
6274
Glenn Kasten05997e22014-03-13 15:08:33 -07006275// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
Glenn Kastene198c362013-08-13 09:13:36 -07006276sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
Eric Laurent81784c32012-11-19 14:55:58 -08006277 const sp<AudioFlinger::Client>& client,
6278 uint32_t sampleRate,
6279 audio_format_t format,
6280 audio_channel_mask_t channelMask,
Glenn Kasten74935e42013-12-19 08:56:45 -08006281 size_t *pFrameCount,
Glenn Kastend848eb42016-03-08 13:42:11 -08006282 audio_session_t sessionId,
Glenn Kasten7df8c0b2014-07-03 12:23:29 -07006283 size_t *notificationFrames,
Marco Nelissen462fd2f2013-01-14 14:12:05 -08006284 int uid,
Glenn Kastenddb0ccf2013-07-31 16:14:50 -07006285 IAudioFlinger::track_flags_t *flags,
Eric Laurent81784c32012-11-19 14:55:58 -08006286 pid_t tid,
6287 status_t *status)
6288{
Glenn Kasten74935e42013-12-19 08:56:45 -08006289 size_t frameCount = *pFrameCount;
Eric Laurent81784c32012-11-19 14:55:58 -08006290 sp<RecordTrack> track;
6291 status_t lStatus;
6292
Glenn Kasten90e58b12013-07-31 16:16:02 -07006293 // client expresses a preference for FAST, but we get the final say
6294 if (*flags & IAudioFlinger::TRACK_FAST) {
6295 if (
Glenn Kastenb7fbf7e2015-03-18 12:57:28 -07006296 // we formerly checked for a callback handler (non-0 tid),
6297 // but that is no longer required for TRANSFER_OBTAIN mode
6298 //
Glenn Kasten74105912014-07-03 12:28:53 -07006299 // frame count is not specified, or is exactly the pipe depth
6300 ((frameCount == 0) || (frameCount == mPipeFramesP2)) &&
Glenn Kasten3a6c90a2014-03-13 15:07:51 -07006301 // PCM data
6302 audio_is_linear_pcm(format) &&
Glenn Kasten7fd04222016-02-02 12:38:16 -08006303 // hardware format
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006304 (format == mFormat) &&
Glenn Kasten7fd04222016-02-02 12:38:16 -08006305 // hardware channel mask
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006306 (channelMask == mChannelMask) &&
Glenn Kasten7fd04222016-02-02 12:38:16 -08006307 // hardware sample rate
Glenn Kasten90e58b12013-07-31 16:16:02 -07006308 (sampleRate == mSampleRate) &&
Glenn Kasten3a6c90a2014-03-13 15:07:51 -07006309 // record thread has an associated fast capture
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006310 hasFastCapture() &&
6311 // there are sufficient fast track slots available
6312 mFastTrackAvail
Glenn Kasten90e58b12013-07-31 16:16:02 -07006313 ) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006314 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
Glenn Kasten90e58b12013-07-31 16:16:02 -07006315 frameCount, mFrameCount);
6316 } else {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006317 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu "
Glenn Kasten74105912014-07-03 12:28:53 -07006318 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006319 "hasFastCapture=%d tid=%d mFastTrackAvail=%d",
Glenn Kasten74105912014-07-03 12:28:53 -07006320 frameCount, mFrameCount, mPipeFramesP2,
6321 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate,
6322 hasFastCapture(), tid, mFastTrackAvail);
Glenn Kasten90e58b12013-07-31 16:16:02 -07006323 *flags &= ~IAudioFlinger::TRACK_FAST;
Glenn Kasten74105912014-07-03 12:28:53 -07006324 }
6325 }
6326
6327 // compute track buffer size in frames, and suggest the notification frame count
6328 if (*flags & IAudioFlinger::TRACK_FAST) {
6329 // fast track: frame count is exactly the pipe depth
6330 frameCount = mPipeFramesP2;
6331 // ignore requested notificationFrames, and always notify exactly once every HAL buffer
6332 *notificationFrames = mFrameCount;
6333 } else {
Glenn Kasten49d00ad2014-07-21 11:22:03 -07006334 // not fast track: max notification period is resampled equivalent of one HAL buffer time
6335 // or 20 ms if there is a fast capture
6336 // TODO This could be a roundupRatio inline, and const
6337 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount)
6338 * sampleRate + mSampleRate - 1) / mSampleRate;
6339 // minimum number of notification periods is at least kMinNotifications,
6340 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs)
6341 static const size_t kMinNotifications = 3;
6342 static const uint32_t kMinMs = 30;
6343 // TODO This could be a roundupRatio inline
6344 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000;
6345 // TODO This could be a roundupRatio inline
6346 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) /
6347 maxNotificationFrames;
6348 const size_t minFrameCount = maxNotificationFrames *
6349 max(kMinNotifications, minNotificationsByMs);
6350 frameCount = max(frameCount, minFrameCount);
6351 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) {
6352 *notificationFrames = maxNotificationFrames;
Glenn Kasten74105912014-07-03 12:28:53 -07006353 }
Glenn Kasten90e58b12013-07-31 16:16:02 -07006354 }
Glenn Kasten74935e42013-12-19 08:56:45 -08006355 *pFrameCount = frameCount;
Glenn Kasten90e58b12013-07-31 16:16:02 -07006356
Glenn Kasten15e57982013-09-24 11:52:37 -07006357 lStatus = initCheck();
6358 if (lStatus != NO_ERROR) {
6359 ALOGE("createRecordTrack_l() audio driver not initialized");
6360 goto Exit;
6361 }
Eric Laurent81784c32012-11-19 14:55:58 -08006362
6363 { // scope for mLock
6364 Mutex::Autolock _l(mLock);
6365
6366 track = new RecordTrack(this, client, sampleRate,
Eric Laurent83b88082014-06-20 18:31:16 -07006367 format, channelMask, frameCount, NULL, sessionId, uid,
6368 *flags, TrackBase::TYPE_DEFAULT);
Eric Laurent81784c32012-11-19 14:55:58 -08006369
Glenn Kasten03003332013-08-06 15:40:54 -07006370 lStatus = track->initCheck();
6371 if (lStatus != NO_ERROR) {
Glenn Kasten35295072013-10-07 09:27:06 -07006372 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
Haynes Mathew George03e9e832013-12-13 15:40:13 -08006373 // track must be cleared from the caller as the caller has the AF lock
Eric Laurent81784c32012-11-19 14:55:58 -08006374 goto Exit;
6375 }
6376 mTracks.add(track);
6377
6378 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
6379 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
6380 mAudioFlinger->btNrecIsOff();
6381 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
6382 setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
Glenn Kasten90e58b12013-07-31 16:16:02 -07006383
6384 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) {
6385 pid_t callingPid = IPCThreadState::self()->getCallingPid();
6386 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
6387 // so ask activity manager to do this on our behalf
6388 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
6389 }
Eric Laurent81784c32012-11-19 14:55:58 -08006390 }
Glenn Kasten05997e22014-03-13 15:08:33 -07006391
Eric Laurent81784c32012-11-19 14:55:58 -08006392 lStatus = NO_ERROR;
6393
6394Exit:
Glenn Kasten9156ef32013-08-06 15:39:08 -07006395 *status = lStatus;
Eric Laurent81784c32012-11-19 14:55:58 -08006396 return track;
6397}
6398
6399status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
6400 AudioSystem::sync_event_t event,
Glenn Kastend848eb42016-03-08 13:42:11 -08006401 audio_session_t triggerSession)
Eric Laurent81784c32012-11-19 14:55:58 -08006402{
6403 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
6404 sp<ThreadBase> strongMe = this;
6405 status_t status = NO_ERROR;
6406
6407 if (event == AudioSystem::SYNC_EVENT_NONE) {
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006408 recordTrack->clearSyncStartEvent();
Eric Laurent81784c32012-11-19 14:55:58 -08006409 } else if (event != AudioSystem::SYNC_EVENT_SAME) {
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006410 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
Eric Laurent81784c32012-11-19 14:55:58 -08006411 triggerSession,
6412 recordTrack->sessionId(),
6413 syncStartEventCallback,
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006414 recordTrack);
Eric Laurent81784c32012-11-19 14:55:58 -08006415 // Sync event can be cancelled by the trigger session if the track is not in a
6416 // compatible state in which case we start record immediately
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006417 if (recordTrack->mSyncStartEvent->isCancelled()) {
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006418 recordTrack->clearSyncStartEvent();
Eric Laurent81784c32012-11-19 14:55:58 -08006419 } else {
6420 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006421 recordTrack->mFramesToDrop = -
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08006422 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000);
Eric Laurent81784c32012-11-19 14:55:58 -08006423 }
6424 }
6425
6426 {
Glenn Kasten47c20702013-08-13 15:37:35 -07006427 // This section is a rendezvous between binder thread executing start() and RecordThread
Eric Laurent81784c32012-11-19 14:55:58 -08006428 AutoMutex lock(mLock);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006429 if (mActiveTracks.indexOf(recordTrack) >= 0) {
6430 if (recordTrack->mState == TrackBase::PAUSING) {
6431 ALOGV("active record track PAUSING -> ACTIVE");
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006432 recordTrack->mState = TrackBase::ACTIVE;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006433 } else {
6434 ALOGV("active record track state %d", recordTrack->mState);
Eric Laurent81784c32012-11-19 14:55:58 -08006435 }
6436 return status;
6437 }
6438
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08006439 // TODO consider other ways of handling this, such as changing the state to :STARTING and
6440 // adding the track to mActiveTracks after returning from AudioSystem::startInput(),
6441 // or using a separate command thread
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006442 recordTrack->mState = TrackBase::STARTING_1;
Glenn Kasten2b806402013-11-20 16:37:38 -08006443 mActiveTracks.add(recordTrack);
6444 mActiveTracksGen++;
Eric Laurent83b88082014-06-20 18:31:16 -07006445 status_t status = NO_ERROR;
6446 if (recordTrack->isExternalTrack()) {
6447 mLock.unlock();
Glenn Kastend848eb42016-03-08 13:42:11 -08006448 status = AudioSystem::startInput(mId, recordTrack->sessionId());
Eric Laurent83b88082014-06-20 18:31:16 -07006449 mLock.lock();
6450 // FIXME should verify that recordTrack is still in mActiveTracks
6451 if (status != NO_ERROR) {
6452 mActiveTracks.remove(recordTrack);
6453 mActiveTracksGen++;
6454 recordTrack->clearSyncStartEvent();
6455 ALOGV("RecordThread::start error %d", status);
6456 return status;
6457 }
Eric Laurent81784c32012-11-19 14:55:58 -08006458 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006459 // Catch up with current buffer indices if thread is already running.
6460 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront
6461 // was initialized to some value closer to the thread's mRsmpInFront, then the track could
6462 // see previously buffered data before it called start(), but with greater risk of overrun.
6463
Andy Hung73c02e42015-03-29 01:13:58 -07006464 recordTrack->mResamplerBufferProvider->reset();
Andy Hung97a893e2015-03-29 01:03:07 -07006465 // clear any converter state as new data will be discontinuous
6466 recordTrack->mRecordBufferConverter->reset();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006467 recordTrack->mState = TrackBase::STARTING_2;
Eric Laurent81784c32012-11-19 14:55:58 -08006468 // signal thread to start
Eric Laurent81784c32012-11-19 14:55:58 -08006469 mWaitWorkCV.broadcast();
Glenn Kasten2b806402013-11-20 16:37:38 -08006470 if (mActiveTracks.indexOf(recordTrack) < 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08006471 ALOGV("Record failed to start");
6472 status = BAD_VALUE;
6473 goto startError;
6474 }
Eric Laurent81784c32012-11-19 14:55:58 -08006475 return status;
6476 }
Glenn Kasten7c027242012-12-26 14:43:16 -08006477
Eric Laurent81784c32012-11-19 14:55:58 -08006478startError:
Eric Laurent83b88082014-06-20 18:31:16 -07006479 if (recordTrack->isExternalTrack()) {
Glenn Kastend848eb42016-03-08 13:42:11 -08006480 AudioSystem::stopInput(mId, recordTrack->sessionId());
Eric Laurent83b88082014-06-20 18:31:16 -07006481 }
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006482 recordTrack->clearSyncStartEvent();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006483 // FIXME I wonder why we do not reset the state here?
Eric Laurent81784c32012-11-19 14:55:58 -08006484 return status;
6485}
6486
Eric Laurent81784c32012-11-19 14:55:58 -08006487void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
6488{
6489 sp<SyncEvent> strongEvent = event.promote();
6490
6491 if (strongEvent != 0) {
Eric Laurent8ea16e42014-02-20 16:26:11 -08006492 sp<RefBase> ptr = strongEvent->cookie().promote();
6493 if (ptr != 0) {
6494 RecordTrack *recordTrack = (RecordTrack *)ptr.get();
6495 recordTrack->handleSyncStartEvent(strongEvent);
6496 }
Eric Laurent81784c32012-11-19 14:55:58 -08006497 }
6498}
6499
Glenn Kastena8356f62013-07-25 14:37:52 -07006500bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
Eric Laurent81784c32012-11-19 14:55:58 -08006501 ALOGV("RecordThread::stop");
Glenn Kastena8356f62013-07-25 14:37:52 -07006502 AutoMutex _l(mLock);
Glenn Kasten2b806402013-11-20 16:37:38 -08006503 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) {
Eric Laurent81784c32012-11-19 14:55:58 -08006504 return false;
6505 }
Glenn Kasten47c20702013-08-13 15:37:35 -07006506 // note that threadLoop may still be processing the track at this point [without lock]
Eric Laurent81784c32012-11-19 14:55:58 -08006507 recordTrack->mState = TrackBase::PAUSING;
6508 // do not wait for mStartStopCond if exiting
6509 if (exitPending()) {
6510 return true;
6511 }
Glenn Kasten47c20702013-08-13 15:37:35 -07006512 // FIXME incorrect usage of wait: no explicit predicate or loop
Eric Laurent81784c32012-11-19 14:55:58 -08006513 mStartStopCond.wait(mLock);
Glenn Kasten2b806402013-11-20 16:37:38 -08006514 // if we have been restarted, recordTrack is in mActiveTracks here
6515 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08006516 ALOGV("Record stopped OK");
6517 return true;
6518 }
6519 return false;
6520}
6521
Glenn Kasten0f11b512014-01-31 16:18:54 -08006522bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const
Eric Laurent81784c32012-11-19 14:55:58 -08006523{
6524 return false;
6525}
6526
Glenn Kasten0f11b512014-01-31 16:18:54 -08006527status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08006528{
6529#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
6530 if (!isValidSyncEvent(event)) {
6531 return BAD_VALUE;
6532 }
6533
Glenn Kastend848eb42016-03-08 13:42:11 -08006534 audio_session_t eventSession = event->triggerSession();
Eric Laurent81784c32012-11-19 14:55:58 -08006535 status_t ret = NAME_NOT_FOUND;
6536
6537 Mutex::Autolock _l(mLock);
6538
6539 for (size_t i = 0; i < mTracks.size(); i++) {
6540 sp<RecordTrack> track = mTracks[i];
6541 if (eventSession == track->sessionId()) {
6542 (void) track->setSyncEvent(event);
6543 ret = NO_ERROR;
6544 }
6545 }
6546 return ret;
6547#else
6548 return BAD_VALUE;
6549#endif
6550}
6551
6552// destroyTrack_l() must be called with ThreadBase::mLock held
6553void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
6554{
Eric Laurentbfb1b832013-01-07 09:53:42 -08006555 track->terminate();
6556 track->mState = TrackBase::STOPPED;
Eric Laurent81784c32012-11-19 14:55:58 -08006557 // active tracks are removed by threadLoop()
Glenn Kasten2b806402013-11-20 16:37:38 -08006558 if (mActiveTracks.indexOf(track) < 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08006559 removeTrack_l(track);
6560 }
6561}
6562
6563void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
6564{
6565 mTracks.remove(track);
6566 // need anything related to effects here?
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006567 if (track->isFastTrack()) {
6568 ALOG_ASSERT(!mFastTrackAvail);
6569 mFastTrackAvail = true;
6570 }
Eric Laurent81784c32012-11-19 14:55:58 -08006571}
6572
6573void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
6574{
6575 dumpInternals(fd, args);
6576 dumpTracks(fd, args);
6577 dumpEffectChains(fd, args);
6578}
6579
6580void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
6581{
Elliott Hughes87cebad2014-05-22 10:14:43 -07006582 dprintf(fd, "\nInput thread %p:\n", this);
Eric Laurent81784c32012-11-19 14:55:58 -08006583
Glenn Kasten44182c22015-03-05 17:12:23 -08006584 dumpBase(fd, args);
6585
6586 if (mActiveTracks.size() == 0) {
Elliott Hughes87cebad2014-05-22 10:14:43 -07006587 dprintf(fd, " No active record clients\n");
Eric Laurent81784c32012-11-19 14:55:58 -08006588 }
Glenn Kasten6e6704c2014-07-03 10:20:00 -07006589 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no");
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006590 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
Glenn Kasten17c9c992015-03-02 15:53:01 -08006591
Glenn Kasten2f90c512015-12-02 11:40:09 -08006592 // Make a non-atomic copy of fast capture dump state so it won't change underneath us
6593 // while we are dumping it. It may be inconsistent, but it won't mutate!
6594 // This is a large object so we place it on the heap.
6595 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
6596 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState);
6597 copy->dump(fd);
6598 delete copy;
Eric Laurent81784c32012-11-19 14:55:58 -08006599}
6600
Glenn Kasten0f11b512014-01-31 16:18:54 -08006601void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08006602{
6603 const size_t SIZE = 256;
6604 char buffer[SIZE];
6605 String8 result;
6606
Marco Nelissenb2208842014-02-07 14:00:50 -08006607 size_t numtracks = mTracks.size();
6608 size_t numactive = mActiveTracks.size();
6609 size_t numactiveseen = 0;
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006610 dprintf(fd, " %zu Tracks", numtracks);
Marco Nelissenb2208842014-02-07 14:00:50 -08006611 if (numtracks) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006612 dprintf(fd, " of which %zu are active\n", numactive);
Marco Nelissenb2208842014-02-07 14:00:50 -08006613 RecordTrack::appendDumpHeader(result);
6614 for (size_t i = 0; i < numtracks ; ++i) {
6615 sp<RecordTrack> track = mTracks[i];
6616 if (track != 0) {
6617 bool active = mActiveTracks.indexOf(track) >= 0;
6618 if (active) {
6619 numactiveseen++;
6620 }
6621 track->dump(buffer, SIZE, active);
6622 result.append(buffer);
6623 }
Eric Laurent81784c32012-11-19 14:55:58 -08006624 }
Marco Nelissenb2208842014-02-07 14:00:50 -08006625 } else {
Elliott Hughes87cebad2014-05-22 10:14:43 -07006626 dprintf(fd, "\n");
Eric Laurent81784c32012-11-19 14:55:58 -08006627 }
6628
Marco Nelissenb2208842014-02-07 14:00:50 -08006629 if (numactiveseen != numactive) {
6630 snprintf(buffer, SIZE, " The following tracks are in the active list but"
6631 " not in the track list\n");
Eric Laurent81784c32012-11-19 14:55:58 -08006632 result.append(buffer);
6633 RecordTrack::appendDumpHeader(result);
Marco Nelissenb2208842014-02-07 14:00:50 -08006634 for (size_t i = 0; i < numactive; ++i) {
Glenn Kasten2b806402013-11-20 16:37:38 -08006635 sp<RecordTrack> track = mActiveTracks[i];
Marco Nelissenb2208842014-02-07 14:00:50 -08006636 if (mTracks.indexOf(track) < 0) {
6637 track->dump(buffer, SIZE, true);
6638 result.append(buffer);
6639 }
Glenn Kasten2b806402013-11-20 16:37:38 -08006640 }
Eric Laurent81784c32012-11-19 14:55:58 -08006641
6642 }
6643 write(fd, result.string(), result.size());
6644}
6645
Andy Hung73c02e42015-03-29 01:13:58 -07006646
6647void AudioFlinger::RecordThread::ResamplerBufferProvider::reset()
6648{
6649 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
6650 RecordThread *recordThread = (RecordThread *) threadBase.get();
6651 mRsmpInFront = recordThread->mRsmpInRear;
6652 mRsmpInUnrel = 0;
6653}
6654
6655void AudioFlinger::RecordThread::ResamplerBufferProvider::sync(
6656 size_t *framesAvailable, bool *hasOverrun)
6657{
6658 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
6659 RecordThread *recordThread = (RecordThread *) threadBase.get();
6660 const int32_t rear = recordThread->mRsmpInRear;
6661 const int32_t front = mRsmpInFront;
6662 const ssize_t filled = rear - front;
6663
6664 size_t framesIn;
6665 bool overrun = false;
6666 if (filled < 0) {
6667 // should not happen, but treat like a massive overrun and re-sync
6668 framesIn = 0;
6669 mRsmpInFront = rear;
6670 overrun = true;
6671 } else if ((size_t) filled <= recordThread->mRsmpInFrames) {
6672 framesIn = (size_t) filled;
6673 } else {
6674 // client is not keeping up with server, but give it latest data
6675 framesIn = recordThread->mRsmpInFrames;
6676 mRsmpInFront = /* front = */ rear - framesIn;
6677 overrun = true;
6678 }
6679 if (framesAvailable != NULL) {
6680 *framesAvailable = framesIn;
6681 }
6682 if (hasOverrun != NULL) {
6683 *hasOverrun = overrun;
6684 }
6685}
6686
Eric Laurent81784c32012-11-19 14:55:58 -08006687// AudioBufferProvider interface
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006688status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
Glenn Kastend79072e2016-01-06 08:41:20 -08006689 AudioBufferProvider::Buffer* buffer)
Eric Laurent81784c32012-11-19 14:55:58 -08006690{
Andy Hung73c02e42015-03-29 01:13:58 -07006691 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006692 if (threadBase == 0) {
6693 buffer->frameCount = 0;
Glenn Kasten607fa3e2014-02-21 14:24:58 -08006694 buffer->raw = NULL;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006695 return NOT_ENOUGH_DATA;
6696 }
6697 RecordThread *recordThread = (RecordThread *) threadBase.get();
6698 int32_t rear = recordThread->mRsmpInRear;
Andy Hung73c02e42015-03-29 01:13:58 -07006699 int32_t front = mRsmpInFront;
Glenn Kasten85948432013-08-19 12:09:05 -07006700 ssize_t filled = rear - front;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006701 // FIXME should not be P2 (don't want to increase latency)
6702 // FIXME if client not keeping up, discard
Glenn Kasten607fa3e2014-02-21 14:24:58 -08006703 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames));
Glenn Kasten85948432013-08-19 12:09:05 -07006704 // 'filled' may be non-contiguous, so return only the first contiguous chunk
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006705 front &= recordThread->mRsmpInFramesP2 - 1;
6706 size_t part1 = recordThread->mRsmpInFramesP2 - front;
Glenn Kasten85948432013-08-19 12:09:05 -07006707 if (part1 > (size_t) filled) {
6708 part1 = filled;
6709 }
6710 size_t ask = buffer->frameCount;
6711 ALOG_ASSERT(ask > 0);
6712 if (part1 > ask) {
6713 part1 = ask;
6714 }
6715 if (part1 == 0) {
Andy Hung73c02e42015-03-29 01:13:58 -07006716 // out of data is fine since the resampler will return a short-count.
Glenn Kasten85948432013-08-19 12:09:05 -07006717 buffer->raw = NULL;
6718 buffer->frameCount = 0;
Andy Hung73c02e42015-03-29 01:13:58 -07006719 mRsmpInUnrel = 0;
Glenn Kasten85948432013-08-19 12:09:05 -07006720 return NOT_ENOUGH_DATA;
Eric Laurent81784c32012-11-19 14:55:58 -08006721 }
6722
Andy Hung57446612015-04-19 23:56:46 -07006723 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize;
Glenn Kasten85948432013-08-19 12:09:05 -07006724 buffer->frameCount = part1;
Andy Hung73c02e42015-03-29 01:13:58 -07006725 mRsmpInUnrel = part1;
Eric Laurent81784c32012-11-19 14:55:58 -08006726 return NO_ERROR;
6727}
6728
6729// AudioBufferProvider interface
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006730void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
6731 AudioBufferProvider::Buffer* buffer)
Eric Laurent81784c32012-11-19 14:55:58 -08006732{
Glenn Kasten85948432013-08-19 12:09:05 -07006733 size_t stepCount = buffer->frameCount;
6734 if (stepCount == 0) {
6735 return;
6736 }
Andy Hung73c02e42015-03-29 01:13:58 -07006737 ALOG_ASSERT(stepCount <= mRsmpInUnrel);
6738 mRsmpInUnrel -= stepCount;
6739 mRsmpInFront += stepCount;
Glenn Kasten85948432013-08-19 12:09:05 -07006740 buffer->raw = NULL;
Eric Laurent81784c32012-11-19 14:55:58 -08006741 buffer->frameCount = 0;
6742}
6743
Andy Hung97a893e2015-03-29 01:03:07 -07006744AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter(
6745 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
6746 uint32_t srcSampleRate,
6747 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
6748 uint32_t dstSampleRate) :
6749 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars
6750 // mSrcFormat
6751 // mSrcSampleRate
6752 // mDstChannelMask
6753 // mDstFormat
6754 // mDstSampleRate
6755 // mSrcChannelCount
6756 // mDstChannelCount
6757 // mDstFrameSize
6758 mBuf(NULL), mBufFrames(0), mBufFrameSize(0),
Andy Hungd330ee42015-04-20 13:23:41 -07006759 mResampler(NULL),
6760 mIsLegacyDownmix(false),
6761 mIsLegacyUpmix(false),
6762 mRequiresFloat(false),
6763 mInputConverterProvider(NULL)
Andy Hung97a893e2015-03-29 01:03:07 -07006764{
6765 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate,
6766 dstChannelMask, dstFormat, dstSampleRate);
6767}
6768
6769AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() {
6770 free(mBuf);
6771 delete mResampler;
Andy Hungd330ee42015-04-20 13:23:41 -07006772 delete mInputConverterProvider;
Andy Hung97a893e2015-03-29 01:03:07 -07006773}
6774
6775size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst,
6776 AudioBufferProvider *provider, size_t frames)
6777{
Andy Hungd330ee42015-04-20 13:23:41 -07006778 if (mInputConverterProvider != NULL) {
6779 mInputConverterProvider->setBufferProvider(provider);
6780 provider = mInputConverterProvider;
6781 }
6782
6783 if (mResampler == NULL) {
Andy Hung97a893e2015-03-29 01:03:07 -07006784 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
6785 mSrcSampleRate, mSrcFormat, mDstFormat);
6786
6787 AudioBufferProvider::Buffer buffer;
6788 for (size_t i = frames; i > 0; ) {
6789 buffer.frameCount = i;
Glenn Kastend79072e2016-01-06 08:41:20 -08006790 status_t status = provider->getNextBuffer(&buffer);
Andy Hung97a893e2015-03-29 01:03:07 -07006791 if (status != OK || buffer.frameCount == 0) {
6792 frames -= i; // cannot fill request.
6793 break;
6794 }
Andy Hungd330ee42015-04-20 13:23:41 -07006795 // format convert to destination buffer
6796 convertNoResampler(dst, buffer.raw, buffer.frameCount);
Andy Hung97a893e2015-03-29 01:03:07 -07006797
6798 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize;
6799 i -= buffer.frameCount;
6800 provider->releaseBuffer(&buffer);
6801 }
6802 } else {
6803 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
6804 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat);
6805
Andy Hungd330ee42015-04-20 13:23:41 -07006806 // reallocate buffer if needed
6807 if (mBufFrameSize != 0 && mBufFrames < frames) {
6808 free(mBuf);
6809 mBufFrames = frames;
6810 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
6811 }
Andy Hung97a893e2015-03-29 01:03:07 -07006812 // resampler accumulates, but we only have one source track
Andy Hungd330ee42015-04-20 13:23:41 -07006813 memset(mBuf, 0, frames * mBufFrameSize);
6814 frames = mResampler->resample((int32_t*)mBuf, frames, provider);
6815 // format convert to destination buffer
6816 convertResampler(dst, mBuf, frames);
Andy Hung97a893e2015-03-29 01:03:07 -07006817 }
6818 return frames;
6819}
6820
6821status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters(
6822 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
6823 uint32_t srcSampleRate,
6824 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
6825 uint32_t dstSampleRate)
6826{
6827 // quick evaluation if there is any change.
6828 if (mSrcFormat == srcFormat
6829 && mSrcChannelMask == srcChannelMask
6830 && mSrcSampleRate == srcSampleRate
6831 && mDstFormat == dstFormat
6832 && mDstChannelMask == dstChannelMask
6833 && mDstSampleRate == dstSampleRate) {
6834 return NO_ERROR;
6835 }
6836
Andy Hungdb4c0312015-05-06 08:46:52 -07006837 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x"
6838 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u",
6839 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate);
Andy Hung97a893e2015-03-29 01:03:07 -07006840 const bool valid =
6841 audio_is_input_channel(srcChannelMask)
6842 && audio_is_input_channel(dstChannelMask)
6843 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat)
6844 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat)
6845 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX)
6846 ; // no upsampling checks for now
6847 if (!valid) {
6848 return BAD_VALUE;
6849 }
6850
6851 mSrcFormat = srcFormat;
6852 mSrcChannelMask = srcChannelMask;
6853 mSrcSampleRate = srcSampleRate;
6854 mDstFormat = dstFormat;
6855 mDstChannelMask = dstChannelMask;
6856 mDstSampleRate = dstSampleRate;
6857
6858 // compute derived parameters
6859 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask);
6860 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask);
6861 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat);
6862
Andy Hungd330ee42015-04-20 13:23:41 -07006863 // do we need to resample?
6864 delete mResampler;
6865 mResampler = NULL;
6866 if (mSrcSampleRate != mDstSampleRate) {
6867 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT,
6868 mSrcChannelCount, mDstSampleRate);
6869 mResampler->setSampleRate(mSrcSampleRate);
6870 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
6871 }
6872
6873 // are we running legacy channel conversion modes?
6874 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO
6875 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK)
6876 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO;
6877 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO
6878 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO
6879 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK);
6880
6881 // do we need to process in float?
6882 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix;
6883
6884 // do we need a staging buffer to convert for destination (we can still optimize this)?
6885 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity
6886 if (mResampler != NULL) {
6887 mBufFrameSize = max(mSrcChannelCount, FCC_2)
6888 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
Andy Hunga97630b2015-07-22 23:27:24 -07006889 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float
Andy Hungd330ee42015-04-20 13:23:41 -07006890 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
6891 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) {
Andy Hung97a893e2015-03-29 01:03:07 -07006892 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat);
6893 } else {
6894 mBufFrameSize = 0;
6895 }
6896 mBufFrames = 0; // force the buffer to be resized.
6897
Andy Hungd330ee42015-04-20 13:23:41 -07006898 // do we need an input converter buffer provider to give us float?
6899 delete mInputConverterProvider;
6900 mInputConverterProvider = NULL;
6901 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) {
6902 mInputConverterProvider = new ReformatBufferProvider(
6903 audio_channel_count_from_in_mask(mSrcChannelMask),
6904 mSrcFormat,
6905 AUDIO_FORMAT_PCM_FLOAT,
6906 256 /* provider buffer frame count */);
6907 }
6908
6909 // do we need a remixer to do channel mask conversion
6910 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) {
6911 (void) memcpy_by_index_array_initialization_from_channel_mask(
6912 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask);
Andy Hung97a893e2015-03-29 01:03:07 -07006913 }
6914 return NO_ERROR;
6915}
6916
Andy Hungd330ee42015-04-20 13:23:41 -07006917void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler(
6918 void *dst, const void *src, size_t frames)
Andy Hung97a893e2015-03-29 01:03:07 -07006919{
Andy Hungd330ee42015-04-20 13:23:41 -07006920 // src is native type unless there is legacy upmix or downmix, whereupon it is float.
Andy Hung97a893e2015-03-29 01:03:07 -07006921 if (mBufFrameSize != 0 && mBufFrames < frames) {
6922 free(mBuf);
6923 mBufFrames = frames;
6924 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
6925 }
Andy Hungd330ee42015-04-20 13:23:41 -07006926 // do we need to do legacy upmix and downmix?
6927 if (mIsLegacyUpmix || mIsLegacyDownmix) {
Andy Hung97a893e2015-03-29 01:03:07 -07006928 void *dstBuf = mBuf != NULL ? mBuf : dst;
Andy Hungd330ee42015-04-20 13:23:41 -07006929 if (mIsLegacyUpmix) {
6930 upmix_to_stereo_float_from_mono_float((float *)dstBuf,
6931 (const float *)src, frames);
6932 } else /*mIsLegacyDownmix */ {
6933 downmix_to_mono_float_from_stereo_float((float *)dstBuf,
6934 (const float *)src, frames);
Andy Hung97a893e2015-03-29 01:03:07 -07006935 }
Andy Hungd330ee42015-04-20 13:23:41 -07006936 if (mBuf != NULL) {
6937 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT,
6938 frames * mDstChannelCount);
6939 }
6940 return;
6941 }
6942 // do we need to do channel mask conversion?
6943 if (mSrcChannelMask != mDstChannelMask) {
Andy Hung97a893e2015-03-29 01:03:07 -07006944 void *dstBuf = mBuf != NULL ? mBuf : dst;
Andy Hungd330ee42015-04-20 13:23:41 -07006945 memcpy_by_index_array(dstBuf, mDstChannelCount,
6946 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames);
6947 if (dstBuf == dst) {
6948 return; // format is the same
6949 }
6950 }
6951 // convert to destination buffer
6952 const void *convertBuf = mBuf != NULL ? mBuf : src;
6953 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat,
6954 frames * mDstChannelCount);
6955}
6956
6957void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler(
6958 void *dst, /*not-a-const*/ void *src, size_t frames)
6959{
6960 // src buffer format is ALWAYS float when entering this routine
6961 if (mIsLegacyUpmix) {
6962 ; // mono to stereo already handled by resampler
6963 } else if (mIsLegacyDownmix
6964 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) {
6965 // the resampler outputs stereo for mono input channel (a feature?)
6966 // must convert to mono
6967 downmix_to_mono_float_from_stereo_float((float *)src,
6968 (const float *)src, frames);
6969 } else if (mSrcChannelMask != mDstChannelMask) {
6970 // convert to mono channel again for channel mask conversion (could be skipped
6971 // with further optimization).
Andy Hung97a893e2015-03-29 01:03:07 -07006972 if (mSrcChannelCount == 1) {
Andy Hungd330ee42015-04-20 13:23:41 -07006973 downmix_to_mono_float_from_stereo_float((float *)src,
6974 (const float *)src, frames);
Andy Hung97a893e2015-03-29 01:03:07 -07006975 }
Andy Hungd330ee42015-04-20 13:23:41 -07006976 // convert to destination format (in place, OK as float is larger than other types)
6977 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) {
6978 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
6979 frames * mSrcChannelCount);
6980 }
6981 // channel convert and save to dst
6982 memcpy_by_index_array(dst, mDstChannelCount,
6983 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames);
6984 return;
Andy Hung97a893e2015-03-29 01:03:07 -07006985 }
Andy Hungd330ee42015-04-20 13:23:41 -07006986 // convert to destination format and save to dst
6987 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
6988 frames * mDstChannelCount);
Andy Hung97a893e2015-03-29 01:03:07 -07006989}
6990
Eric Laurent10351942014-05-08 18:49:52 -07006991bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
6992 status_t& status)
Eric Laurent81784c32012-11-19 14:55:58 -08006993{
6994 bool reconfig = false;
6995
Eric Laurent10351942014-05-08 18:49:52 -07006996 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08006997
Eric Laurent10351942014-05-08 18:49:52 -07006998 audio_format_t reqFormat = mFormat;
6999 uint32_t samplingRate = mSampleRate;
Glenn Kastene1635ec2015-06-08 15:46:49 -07007000 // 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 -07007001 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount);
7002
7003 AudioParameter param = AudioParameter(keyValuePair);
7004 int value;
Haynes Mathew George9ce67b52015-09-30 11:40:47 -07007005
7006 // scope for AutoPark extends to end of method
7007 AutoPark<FastCapture> park(mFastCapture);
7008
Eric Laurent10351942014-05-08 18:49:52 -07007009 // TODO Investigate when this code runs. Check with audio policy when a sample rate and
7010 // channel count change can be requested. Do we mandate the first client defines the
7011 // HAL sampling rate and channel count or do we allow changes on the fly?
7012 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
7013 samplingRate = value;
7014 reconfig = true;
7015 }
7016 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
Andy Hung97a893e2015-03-29 01:03:07 -07007017 if (!audio_is_linear_pcm((audio_format_t) value)) {
Eric Laurent10351942014-05-08 18:49:52 -07007018 status = BAD_VALUE;
7019 } else {
7020 reqFormat = (audio_format_t) value;
Eric Laurent81784c32012-11-19 14:55:58 -08007021 reconfig = true;
7022 }
Eric Laurent10351942014-05-08 18:49:52 -07007023 }
7024 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
7025 audio_channel_mask_t mask = (audio_channel_mask_t) value;
Andy Hungd330ee42015-04-20 13:23:41 -07007026 if (!audio_is_input_channel(mask) ||
7027 audio_channel_count_from_in_mask(mask) > FCC_8) {
Eric Laurent10351942014-05-08 18:49:52 -07007028 status = BAD_VALUE;
7029 } else {
7030 channelMask = mask;
7031 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08007032 }
Eric Laurent10351942014-05-08 18:49:52 -07007033 }
7034 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
7035 // do not accept frame count changes if tracks are open as the track buffer
7036 // size depends on frame count and correct behavior would not be guaranteed
7037 // if frame count is changed after track creation
7038 if (mActiveTracks.size() > 0) {
7039 status = INVALID_OPERATION;
7040 } else {
7041 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08007042 }
Eric Laurent10351942014-05-08 18:49:52 -07007043 }
7044 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
7045 // forward device change to effects that have requested to be
7046 // aware of attached audio device.
7047 for (size_t i = 0; i < mEffectChains.size(); i++) {
7048 mEffectChains[i]->setDevice_l(value);
Eric Laurent81784c32012-11-19 14:55:58 -08007049 }
Eric Laurent81784c32012-11-19 14:55:58 -08007050
Eric Laurent10351942014-05-08 18:49:52 -07007051 // store input device and output device but do not forward output device to audio HAL.
7052 // Note that status is ignored by the caller for output device
7053 // (see AudioFlinger::setParameters()
7054 if (audio_is_output_devices(value)) {
7055 mOutDevice = value;
7056 status = BAD_VALUE;
7057 } else {
7058 mInDevice = value;
Eric Laurente8726fe2015-06-26 09:39:24 -07007059 if (value != AUDIO_DEVICE_NONE) {
7060 mPrevInDevice = value;
7061 }
Eric Laurent10351942014-05-08 18:49:52 -07007062 // disable AEC and NS if the device is a BT SCO headset supporting those
7063 // pre processings
7064 if (mTracks.size() > 0) {
7065 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
7066 mAudioFlinger->btNrecIsOff();
7067 for (size_t i = 0; i < mTracks.size(); i++) {
7068 sp<RecordTrack> track = mTracks[i];
7069 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
7070 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
Eric Laurent81784c32012-11-19 14:55:58 -08007071 }
7072 }
7073 }
Eric Laurent10351942014-05-08 18:49:52 -07007074 }
7075 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
7076 mAudioSource != (audio_source_t)value) {
7077 // forward device change to effects that have requested to be
7078 // aware of attached audio device.
7079 for (size_t i = 0; i < mEffectChains.size(); i++) {
7080 mEffectChains[i]->setAudioSource_l((audio_source_t)value);
Eric Laurent81784c32012-11-19 14:55:58 -08007081 }
Eric Laurent10351942014-05-08 18:49:52 -07007082 mAudioSource = (audio_source_t)value;
7083 }
Glenn Kastene198c362013-08-13 09:13:36 -07007084
Eric Laurent10351942014-05-08 18:49:52 -07007085 if (status == NO_ERROR) {
7086 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7087 keyValuePair.string());
7088 if (status == INVALID_OPERATION) {
7089 inputStandBy();
Eric Laurent81784c32012-11-19 14:55:58 -08007090 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7091 keyValuePair.string());
Eric Laurent10351942014-05-08 18:49:52 -07007092 }
7093 if (reconfig) {
7094 if (status == BAD_VALUE &&
Andy Hung97a893e2015-03-29 01:03:07 -07007095 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) &&
7096 audio_is_linear_pcm(reqFormat) &&
Eric Laurent10351942014-05-08 18:49:52 -07007097 (mInput->stream->common.get_sample_rate(&mInput->stream->common)
Andy Hung97a893e2015-03-29 01:03:07 -07007098 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) &&
Andy Hunge5412692014-05-16 11:25:07 -07007099 audio_channel_count_from_in_mask(
Andy Hungd1abb8f2015-05-05 23:42:34 -07007100 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) {
Eric Laurent10351942014-05-08 18:49:52 -07007101 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08007102 }
Eric Laurent10351942014-05-08 18:49:52 -07007103 if (status == NO_ERROR) {
7104 readInputParameters_l();
Eric Laurent73e26b62015-04-27 16:55:58 -07007105 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
Eric Laurent81784c32012-11-19 14:55:58 -08007106 }
7107 }
Eric Laurent81784c32012-11-19 14:55:58 -08007108 }
Eric Laurent10351942014-05-08 18:49:52 -07007109
Eric Laurent81784c32012-11-19 14:55:58 -08007110 return reconfig;
7111}
7112
7113String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
7114{
Eric Laurent81784c32012-11-19 14:55:58 -08007115 Mutex::Autolock _l(mLock);
7116 if (initCheck() != NO_ERROR) {
Glenn Kastend8ea6992013-07-16 14:17:15 -07007117 return String8();
Eric Laurent81784c32012-11-19 14:55:58 -08007118 }
7119
Glenn Kastend8ea6992013-07-16 14:17:15 -07007120 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
7121 const String8 out_s8(s);
Eric Laurent81784c32012-11-19 14:55:58 -08007122 free(s);
7123 return out_s8;
7124}
7125
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07007126void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) {
Eric Laurent73e26b62015-04-27 16:55:58 -07007127 sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
7128
7129 desc->mIoHandle = mId;
Eric Laurent81784c32012-11-19 14:55:58 -08007130
7131 switch (event) {
Eric Laurent73e26b62015-04-27 16:55:58 -07007132 case AUDIO_INPUT_OPENED:
7133 case AUDIO_INPUT_CONFIG_CHANGED:
Eric Laurent296fb132015-05-01 11:38:42 -07007134 desc->mPatch = mPatch;
Eric Laurent73e26b62015-04-27 16:55:58 -07007135 desc->mChannelMask = mChannelMask;
7136 desc->mSamplingRate = mSampleRate;
7137 desc->mFormat = mFormat;
7138 desc->mFrameCount = mFrameCount;
Glenn Kasten4a8308b2016-04-18 14:10:01 -07007139 desc->mFrameCountHAL = mFrameCount;
Eric Laurent73e26b62015-04-27 16:55:58 -07007140 desc->mLatency = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08007141 break;
7142
Eric Laurent73e26b62015-04-27 16:55:58 -07007143 case AUDIO_INPUT_CLOSED:
Eric Laurent81784c32012-11-19 14:55:58 -08007144 default:
7145 break;
7146 }
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07007147 mAudioFlinger->ioConfigChanged(event, desc, pid);
Eric Laurent81784c32012-11-19 14:55:58 -08007148}
7149
Glenn Kastendeca2ae2014-02-07 10:25:56 -08007150void AudioFlinger::RecordThread::readInputParameters_l()
Eric Laurent81784c32012-11-19 14:55:58 -08007151{
Eric Laurent81784c32012-11-19 14:55:58 -08007152 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
7153 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
Andy Hunge5412692014-05-16 11:25:07 -07007154 mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
Andy Hungd330ee42015-04-20 13:23:41 -07007155 if (mChannelCount > FCC_8) {
7156 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8);
7157 }
Andy Hung463be252014-07-10 16:56:07 -07007158 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common);
7159 mFormat = mHALFormat;
Andy Hungd330ee42015-04-20 13:23:41 -07007160 if (!audio_is_linear_pcm(mFormat)) {
7161 ALOGE("HAL format %#x is not linear pcm", mFormat);
Glenn Kasten291bb6d2013-07-16 17:23:39 -07007162 }
Eric Laurent665470b2014-07-03 16:37:08 -07007163 mFrameSize = audio_stream_in_frame_size(mInput->stream);
Glenn Kasten548efc92012-11-29 08:48:51 -08007164 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
7165 mFrameCount = mBufferSize / mFrameSize;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08007166 // This is the formula for calculating the temporary buffer size.
Glenn Kastene8426142014-02-28 16:45:03 -08007167 // 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 -07007168 // 1 full output buffer, regardless of the alignment of the available input.
Glenn Kastene8426142014-02-28 16:45:03 -08007169 // The value is somewhat arbitrary, and could probably be even larger.
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08007170 // A larger value should allow more old data to be read after a track calls start(),
7171 // without increasing latency.
Andy Hung97a893e2015-03-29 01:03:07 -07007172 //
7173 // Note this is independent of the maximum downsampling ratio permitted for capture.
Glenn Kastene8426142014-02-28 16:45:03 -08007174 mRsmpInFrames = mFrameCount * 7;
Glenn Kasten85948432013-08-19 12:09:05 -07007175 mRsmpInFramesP2 = roundup(mRsmpInFrames);
Andy Hung57446612015-04-19 23:56:46 -07007176 free(mRsmpInBuffer);
Andy Hung0a01c2f2015-09-21 12:44:54 -07007177 mRsmpInBuffer = NULL;
Glenn Kasten49d00ad2014-07-21 11:22:03 -07007178
7179 // TODO optimize audio capture buffer sizes ...
7180 // Here we calculate the size of the sliding buffer used as a source
7181 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7).
7182 // For current HAL frame counts, this is usually 2048 = 40 ms. It would
7183 // be better to have it derived from the pipe depth in the long term.
7184 // The current value is higher than necessary. However it should not add to latency.
7185
Glenn Kasten85948432013-08-19 12:09:05 -07007186 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
Andy Hung0a01c2f2015-09-21 12:44:54 -07007187 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize;
7188 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize);
7189 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here.
Eric Laurent81784c32012-11-19 14:55:58 -08007190
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08007191 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
7192 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
Eric Laurent81784c32012-11-19 14:55:58 -08007193}
7194
Glenn Kasten5f972c02014-01-13 09:59:31 -08007195uint32_t AudioFlinger::RecordThread::getInputFramesLost()
Eric Laurent81784c32012-11-19 14:55:58 -08007196{
7197 Mutex::Autolock _l(mLock);
7198 if (initCheck() != NO_ERROR) {
7199 return 0;
7200 }
7201
7202 return mInput->stream->get_input_frames_lost(mInput->stream);
7203}
7204
Glenn Kastend848eb42016-03-08 13:42:11 -08007205uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const
Eric Laurent81784c32012-11-19 14:55:58 -08007206{
7207 Mutex::Autolock _l(mLock);
7208 uint32_t result = 0;
7209 if (getEffectChain_l(sessionId) != 0) {
7210 result = EFFECT_SESSION;
7211 }
7212
7213 for (size_t i = 0; i < mTracks.size(); ++i) {
7214 if (sessionId == mTracks[i]->sessionId()) {
7215 result |= TRACK_SESSION;
7216 break;
7217 }
7218 }
7219
7220 return result;
7221}
7222
Glenn Kastend848eb42016-03-08 13:42:11 -08007223KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const
Eric Laurent81784c32012-11-19 14:55:58 -08007224{
Glenn Kastend848eb42016-03-08 13:42:11 -08007225 KeyedVector<audio_session_t, bool> ids;
Eric Laurent81784c32012-11-19 14:55:58 -08007226 Mutex::Autolock _l(mLock);
7227 for (size_t j = 0; j < mTracks.size(); ++j) {
7228 sp<RecordThread::RecordTrack> track = mTracks[j];
Glenn Kastend848eb42016-03-08 13:42:11 -08007229 audio_session_t sessionId = track->sessionId();
Eric Laurent81784c32012-11-19 14:55:58 -08007230 if (ids.indexOfKey(sessionId) < 0) {
7231 ids.add(sessionId, true);
7232 }
7233 }
7234 return ids;
7235}
7236
7237AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
7238{
7239 Mutex::Autolock _l(mLock);
7240 AudioStreamIn *input = mInput;
7241 mInput = NULL;
7242 return input;
7243}
7244
7245// this method must always be called either with ThreadBase mLock held or inside the thread loop
7246audio_stream_t* AudioFlinger::RecordThread::stream() const
7247{
7248 if (mInput == NULL) {
7249 return NULL;
7250 }
7251 return &mInput->stream->common;
7252}
7253
7254status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
7255{
7256 // only one chain per input thread
7257 if (mEffectChains.size() != 0) {
Eric Laurentaaa44472014-09-12 17:41:50 -07007258 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this);
Eric Laurent81784c32012-11-19 14:55:58 -08007259 return INVALID_OPERATION;
7260 }
7261 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
Eric Laurentaaa44472014-09-12 17:41:50 -07007262 chain->setThread(this);
Eric Laurent81784c32012-11-19 14:55:58 -08007263 chain->setInBuffer(NULL);
7264 chain->setOutBuffer(NULL);
7265
7266 checkSuspendOnAddEffectChain_l(chain);
7267
Eric Laurent1b928682014-10-02 19:41:47 -07007268 // make sure enabled pre processing effects state is communicated to the HAL as we
7269 // just moved them to a new input stream.
7270 chain->syncHalEffectsState();
7271
Eric Laurent81784c32012-11-19 14:55:58 -08007272 mEffectChains.add(chain);
7273
7274 return NO_ERROR;
7275}
7276
7277size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
7278{
7279 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
7280 ALOGW_IF(mEffectChains.size() != 1,
Glenn Kastenc42e9b42016-03-21 11:35:03 -07007281 "removeEffectChain_l() %p invalid chain size %zu on thread %p",
Eric Laurent81784c32012-11-19 14:55:58 -08007282 chain.get(), mEffectChains.size(), this);
7283 if (mEffectChains.size() == 1) {
7284 mEffectChains.removeAt(0);
7285 }
7286 return 0;
7287}
7288
Eric Laurent1c333e22014-05-20 10:48:17 -07007289status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch,
7290 audio_patch_handle_t *handle)
7291{
7292 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07007293
7294 // store new device and send to effects
7295 mInDevice = patch->sources[0].ext.device.type;
Eric Laurent296fb132015-05-01 11:38:42 -07007296 mPatch = *patch;
Eric Laurent054d9d32015-04-24 08:48:48 -07007297 for (size_t i = 0; i < mEffectChains.size(); i++) {
7298 mEffectChains[i]->setDevice_l(mInDevice);
7299 }
7300
7301 // disable AEC and NS if the device is a BT SCO headset supporting those
7302 // pre processings
7303 if (mTracks.size() > 0) {
7304 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
7305 mAudioFlinger->btNrecIsOff();
7306 for (size_t i = 0; i < mTracks.size(); i++) {
7307 sp<RecordTrack> track = mTracks[i];
7308 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
7309 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
7310 }
7311 }
7312
7313 // store new source and send to effects
7314 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) {
7315 mAudioSource = patch->sinks[0].ext.mix.usecase.source;
Eric Laurent1c333e22014-05-20 10:48:17 -07007316 for (size_t i = 0; i < mEffectChains.size(); i++) {
Eric Laurent054d9d32015-04-24 08:48:48 -07007317 mEffectChains[i]->setAudioSource_l(mAudioSource);
Eric Laurent1c333e22014-05-20 10:48:17 -07007318 }
Eric Laurent054d9d32015-04-24 08:48:48 -07007319 }
Eric Laurent1c333e22014-05-20 10:48:17 -07007320
Eric Laurent054d9d32015-04-24 08:48:48 -07007321 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
Eric Laurent1c333e22014-05-20 10:48:17 -07007322 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice();
7323 status = hwDevice->create_audio_patch(hwDevice,
7324 patch->num_sources,
7325 patch->sources,
7326 patch->num_sinks,
7327 patch->sinks,
7328 handle);
7329 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07007330 char *address;
7331 if (strcmp(patch->sources[0].ext.device.address, "") != 0) {
7332 address = audio_device_address_to_parameter(
7333 patch->sources[0].ext.device.type,
7334 patch->sources[0].ext.device.address);
7335 } else {
7336 address = (char *)calloc(1, 1);
7337 }
7338 AudioParameter param = AudioParameter(String8(address));
7339 free(address);
7340 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING),
7341 (int)patch->sources[0].ext.device.type);
7342 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE),
7343 (int)patch->sinks[0].ext.mix.usecase.source);
7344 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7345 param.toString().string());
7346 *handle = AUDIO_PATCH_HANDLE_NONE;
Eric Laurent1c333e22014-05-20 10:48:17 -07007347 }
Eric Laurent054d9d32015-04-24 08:48:48 -07007348
Eric Laurente8726fe2015-06-26 09:39:24 -07007349 if (mInDevice != mPrevInDevice) {
7350 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
7351 mPrevInDevice = mInDevice;
7352 }
Eric Laurent296fb132015-05-01 11:38:42 -07007353
Eric Laurent1c333e22014-05-20 10:48:17 -07007354 return status;
7355}
7356
7357status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
7358{
7359 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07007360
7361 mInDevice = AUDIO_DEVICE_NONE;
7362
Eric Laurent1c333e22014-05-20 10:48:17 -07007363 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
7364 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice();
7365 status = hwDevice->release_audio_patch(hwDevice, handle);
7366 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07007367 AudioParameter param;
7368 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0);
7369 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7370 param.toString().string());
Eric Laurent1c333e22014-05-20 10:48:17 -07007371 }
7372 return status;
7373}
7374
Eric Laurent83b88082014-06-20 18:31:16 -07007375void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record)
7376{
7377 Mutex::Autolock _l(mLock);
7378 mTracks.add(record);
7379}
7380
7381void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record)
7382{
7383 Mutex::Autolock _l(mLock);
7384 destroyTrack_l(record);
7385}
7386
7387void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config)
7388{
7389 ThreadBase::getAudioPortConfig(config);
7390 config->role = AUDIO_PORT_ROLE_SINK;
7391 config->ext.mix.hw_module = mInput->audioHwDev->handle();
7392 config->ext.mix.usecase.source = mAudioSource;
7393}
Eric Laurent1c333e22014-05-20 10:48:17 -07007394
Glenn Kasten63238ef2015-03-02 15:50:29 -08007395} // namespace android