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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
Glenn Kasten1b291842016-07-18 14:55:21 -0700146// The universal constant for ubiquitous 20ms value. The value of 20ms seems to provide a good
147// balance between power consumption and latency, and allows threads to be scheduled reliably
148// by the CFS scheduler.
149// FIXME Express other hardcoded references to 20ms with references to this constant and move
150// it appropriately.
151#define FMS_20 20
Eric Laurent51716182016-02-29 18:00:56 -0800152
Eric Laurent81784c32012-11-19 14:55:58 -0800153// Whether to use fast mixer
154static const enum {
155 FastMixer_Never, // never initialize or use: for debugging only
156 FastMixer_Always, // always initialize and use, even if not needed: for debugging only
157 // normal mixer multiplier is 1
158 FastMixer_Static, // initialize if needed, then use all the time if initialized,
159 // multiplier is calculated based on min & max normal mixer buffer size
160 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load,
161 // multiplier is calculated based on min & max normal mixer buffer size
162 // FIXME for FastMixer_Dynamic:
163 // Supporting this option will require fixing HALs that can't handle large writes.
164 // For example, one HAL implementation returns an error from a large write,
165 // and another HAL implementation corrupts memory, possibly in the sample rate converter.
166 // We could either fix the HAL implementations, or provide a wrapper that breaks
167 // up large writes into smaller ones, and the wrapper would need to deal with scheduler.
168} kUseFastMixer = FastMixer_Static;
169
Glenn Kasten6dbb5e32014-05-13 10:38:42 -0700170// Whether to use fast capture
171static const enum {
172 FastCapture_Never, // never initialize or use: for debugging only
173 FastCapture_Always, // always initialize and use, even if not needed: for debugging only
174 FastCapture_Static, // initialize if needed, then use all the time if initialized
175} kUseFastCapture = FastCapture_Static;
176
Eric Laurent81784c32012-11-19 14:55:58 -0800177// Priorities for requestPriority
178static const int kPriorityAudioApp = 2;
179static const int kPriorityFastMixer = 3;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -0700180static const int kPriorityFastCapture = 3;
Eric Laurent81784c32012-11-19 14:55:58 -0800181
Glenn Kastenea38ee72016-04-18 11:08:01 -0700182// IAudioFlinger::createTrack() has an in/out parameter 'pFrameCount' for the total size of the
183// track buffer in shared memory. Zero on input means to use a default value. For fast tracks,
184// AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'.
Glenn Kasten03490092014-05-27 12:30:54 -0700185
186// This is the default value, if not specified by property.
Glenn Kastenb5fed682013-12-03 09:06:43 -0800187static const int kFastTrackMultiplier = 2;
Eric Laurent81784c32012-11-19 14:55:58 -0800188
Glenn Kasten03490092014-05-27 12:30:54 -0700189// The minimum and maximum allowed values
190static const int kFastTrackMultiplierMin = 1;
191static const int kFastTrackMultiplierMax = 2;
192
193// The actual value to use, which can be specified per-device via property af.fast_track_multiplier.
194static int sFastTrackMultiplier = kFastTrackMultiplier;
195
Glenn Kastenb880f5e2014-05-07 08:43:45 -0700196// See Thread::readOnlyHeap().
197// Initially this heap is used to allocate client buffers for "fast" AudioRecord.
198// Eventually it will be the single buffer that FastCapture writes into via HAL read(),
199// and that all "fast" AudioRecord clients read from. In either case, the size can be small.
Glenn Kasten9f81de32014-07-27 15:02:23 -0700200static const size_t kRecordThreadReadOnlyHeapSize = 0x2000;
Glenn Kastenb880f5e2014-05-07 08:43:45 -0700201
Eric Laurent81784c32012-11-19 14:55:58 -0800202// ----------------------------------------------------------------------------
203
Glenn Kasten03490092014-05-27 12:30:54 -0700204static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
205
206static void sFastTrackMultiplierInit()
207{
208 char value[PROPERTY_VALUE_MAX];
209 if (property_get("af.fast_track_multiplier", value, NULL) > 0) {
210 char *endptr;
211 unsigned long ul = strtoul(value, &endptr, 0);
212 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) {
213 sFastTrackMultiplier = (int) ul;
214 }
215 }
216}
217
218// ----------------------------------------------------------------------------
219
Eric Laurent81784c32012-11-19 14:55:58 -0800220#ifdef ADD_BATTERY_DATA
221// To collect the amplifier usage
222static void addBatteryData(uint32_t params) {
223 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
224 if (service == NULL) {
225 // it already logged
226 return;
227 }
228
229 service->addBatteryData(params);
230}
231#endif
232
Andy Hung3f0c9022016-01-15 17:49:46 -0800233// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset
234struct {
235 // call when you acquire a partial wakelock
236 void acquire(const sp<IBinder> &wakeLockToken) {
237 pthread_mutex_lock(&mLock);
238 if (wakeLockToken.get() == nullptr) {
239 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
240 } else {
241 if (mCount == 0) {
242 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME);
243 }
244 ++mCount;
245 }
246 pthread_mutex_unlock(&mLock);
247 }
248
249 // call when you release a partial wakelock.
250 void release(const sp<IBinder> &wakeLockToken) {
251 if (wakeLockToken.get() == nullptr) {
252 return;
253 }
254 pthread_mutex_lock(&mLock);
255 if (--mCount < 0) {
256 ALOGE("negative wakelock count");
257 mCount = 0;
258 }
259 pthread_mutex_unlock(&mLock);
260 }
261
262 // retrieves the boottime timebase offset from monotonic.
263 int64_t getBoottimeOffset() {
264 pthread_mutex_lock(&mLock);
265 int64_t boottimeOffset = mBoottimeOffset;
266 pthread_mutex_unlock(&mLock);
267 return boottimeOffset;
268 }
269
270 // Adjusts the timebase offset between TIMEBASE_MONOTONIC
271 // and the selected timebase.
272 // Currently only TIMEBASE_BOOTTIME is allowed.
273 //
274 // This only needs to be called upon acquiring the first partial wakelock
275 // after all other partial wakelocks are released.
276 //
277 // We do an empirical measurement of the offset rather than parsing
278 // /proc/timer_list since the latter is not a formal kernel ABI.
279 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) {
280 int clockbase;
281 switch (timebase) {
282 case ExtendedTimestamp::TIMEBASE_BOOTTIME:
283 clockbase = SYSTEM_TIME_BOOTTIME;
284 break;
285 default:
286 LOG_ALWAYS_FATAL("invalid timebase %d", timebase);
287 break;
288 }
289 // try three times to get the clock offset, choose the one
290 // with the minimum gap in measurements.
291 const int tries = 3;
292 nsecs_t bestGap, measured;
293 for (int i = 0; i < tries; ++i) {
294 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC);
295 const nsecs_t tbase = systemTime(clockbase);
296 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC);
297 const nsecs_t gap = tmono2 - tmono;
298 if (i == 0 || gap < bestGap) {
299 bestGap = gap;
300 measured = tbase - ((tmono + tmono2) >> 1);
301 }
302 }
303
304 // to avoid micro-adjusting, we don't change the timebase
305 // unless it is significantly different.
306 //
307 // Assumption: It probably takes more than toleranceNs to
308 // suspend and resume the device.
309 static int64_t toleranceNs = 10000; // 10 us
310 if (llabs(*offset - measured) > toleranceNs) {
311 ALOGV("Adjusting timebase offset old: %lld new: %lld",
312 (long long)*offset, (long long)measured);
313 *offset = measured;
314 }
315 }
316
317 pthread_mutex_t mLock;
318 int32_t mCount;
319 int64_t mBoottimeOffset;
320} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization
Eric Laurent81784c32012-11-19 14:55:58 -0800321
322// ----------------------------------------------------------------------------
323// CPU Stats
324// ----------------------------------------------------------------------------
325
326class CpuStats {
327public:
328 CpuStats();
329 void sample(const String8 &title);
330#ifdef DEBUG_CPU_USAGE
331private:
332 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns
333 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
334
335 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
336
337 int mCpuNum; // thread's current CPU number
338 int mCpukHz; // frequency of thread's current CPU in kHz
339#endif
340};
341
342CpuStats::CpuStats()
343#ifdef DEBUG_CPU_USAGE
344 : mCpuNum(-1), mCpukHz(-1)
345#endif
346{
347}
348
Glenn Kasten0f11b512014-01-31 16:18:54 -0800349void CpuStats::sample(const String8 &title
350#ifndef DEBUG_CPU_USAGE
351 __unused
352#endif
353 ) {
Eric Laurent81784c32012-11-19 14:55:58 -0800354#ifdef DEBUG_CPU_USAGE
355 // get current thread's delta CPU time in wall clock ns
356 double wcNs;
357 bool valid = mCpuUsage.sampleAndEnable(wcNs);
358
359 // record sample for wall clock statistics
360 if (valid) {
361 mWcStats.sample(wcNs);
362 }
363
364 // get the current CPU number
365 int cpuNum = sched_getcpu();
366
367 // get the current CPU frequency in kHz
368 int cpukHz = mCpuUsage.getCpukHz(cpuNum);
369
370 // check if either CPU number or frequency changed
371 if (cpuNum != mCpuNum || cpukHz != mCpukHz) {
372 mCpuNum = cpuNum;
373 mCpukHz = cpukHz;
374 // ignore sample for purposes of cycles
375 valid = false;
376 }
377
378 // if no change in CPU number or frequency, then record sample for cycle statistics
379 if (valid && mCpukHz > 0) {
380 double cycles = wcNs * cpukHz * 0.000001;
381 mHzStats.sample(cycles);
382 }
383
384 unsigned n = mWcStats.n();
385 // mCpuUsage.elapsed() is expensive, so don't call it every loop
386 if ((n & 127) == 1) {
387 long long elapsed = mCpuUsage.elapsed();
388 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
389 double perLoop = elapsed / (double) n;
390 double perLoop100 = perLoop * 0.01;
391 double perLoop1k = perLoop * 0.001;
392 double mean = mWcStats.mean();
393 double stddev = mWcStats.stddev();
394 double minimum = mWcStats.minimum();
395 double maximum = mWcStats.maximum();
396 double meanCycles = mHzStats.mean();
397 double stddevCycles = mHzStats.stddev();
398 double minCycles = mHzStats.minimum();
399 double maxCycles = mHzStats.maximum();
400 mCpuUsage.resetElapsed();
401 mWcStats.reset();
402 mHzStats.reset();
403 ALOGD("CPU usage for %s over past %.1f secs\n"
404 " (%u mixer loops at %.1f mean ms per loop):\n"
405 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n"
406 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n"
407 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f",
408 title.string(),
409 elapsed * .000000001, n, perLoop * .000001,
410 mean * .001,
411 stddev * .001,
412 minimum * .001,
413 maximum * .001,
414 mean / perLoop100,
415 stddev / perLoop100,
416 minimum / perLoop100,
417 maximum / perLoop100,
418 meanCycles / perLoop1k,
419 stddevCycles / perLoop1k,
420 minCycles / perLoop1k,
421 maxCycles / perLoop1k);
422
423 }
424 }
425#endif
426};
427
428// ----------------------------------------------------------------------------
429// ThreadBase
430// ----------------------------------------------------------------------------
431
Glenn Kasten97b7b752014-09-28 13:04:24 -0700432// static
433const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
434{
435 switch (type) {
436 case MIXER:
437 return "MIXER";
438 case DIRECT:
439 return "DIRECT";
440 case DUPLICATING:
441 return "DUPLICATING";
442 case RECORD:
443 return "RECORD";
444 case OFFLOAD:
445 return "OFFLOAD";
446 default:
447 return "unknown";
448 }
449}
450
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800451String8 devicesToString(audio_devices_t devices)
452{
453 static const struct mapping {
454 audio_devices_t mDevices;
455 const char * mString;
456 } mappingsOut[] = {
Glenn Kasten818da522015-12-02 13:53:26 -0800457 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"},
458 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"},
459 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"},
460 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"},
461 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"},
462 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"},
463 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"},
464 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"},
465 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"},
466 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"},
467 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"},
468 {AUDIO_DEVICE_OUT_HDMI, "HDMI"},
469 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"},
470 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"},
471 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"},
472 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"},
473 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"},
474 {AUDIO_DEVICE_OUT_LINE, "LINE"},
475 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"},
476 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"},
477 {AUDIO_DEVICE_OUT_FM, "FM"},
478 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"},
479 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"},
480 {AUDIO_DEVICE_OUT_IP, "IP"},
Eric Laurent58545be2016-02-22 18:54:20 -0800481 {AUDIO_DEVICE_OUT_BUS, "BUS"},
Glenn Kasten818da522015-12-02 13:53:26 -0800482 {AUDIO_DEVICE_NONE, "NONE"}, // must be last
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800483 }, mappingsIn[] = {
Glenn Kasten818da522015-12-02 13:53:26 -0800484 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"},
485 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"},
486 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"},
487 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"},
488 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"},
489 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"},
490 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"},
491 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"},
492 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"},
493 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"},
494 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"},
495 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"},
496 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"},
497 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"},
498 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"},
499 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"},
500 {AUDIO_DEVICE_IN_LINE, "LINE"},
501 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"},
502 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"},
503 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"},
504 {AUDIO_DEVICE_IN_IP, "IP"},
Eric Laurent58545be2016-02-22 18:54:20 -0800505 {AUDIO_DEVICE_IN_BUS, "BUS"},
Glenn Kasten818da522015-12-02 13:53:26 -0800506 {AUDIO_DEVICE_NONE, "NONE"}, // must be last
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800507 };
508 String8 result;
509 audio_devices_t allDevices = AUDIO_DEVICE_NONE;
510 const mapping *entry;
511 if (devices & AUDIO_DEVICE_BIT_IN) {
512 devices &= ~AUDIO_DEVICE_BIT_IN;
513 entry = mappingsIn;
514 } else {
515 entry = mappingsOut;
516 }
517 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) {
518 allDevices = (audio_devices_t) (allDevices | entry->mDevices);
519 if (devices & entry->mDevices) {
520 if (!result.isEmpty()) {
521 result.append("|");
522 }
523 result.append(entry->mString);
524 }
525 }
526 if (devices & ~allDevices) {
527 if (!result.isEmpty()) {
528 result.append("|");
529 }
530 result.appendFormat("0x%X", devices & ~allDevices);
531 }
532 if (result.isEmpty()) {
533 result.append(entry->mString);
534 }
535 return result;
536}
537
538String8 inputFlagsToString(audio_input_flags_t flags)
539{
540 static const struct mapping {
541 audio_input_flags_t mFlag;
542 const char * mString;
543 } mappings[] = {
Glenn Kasten818da522015-12-02 13:53:26 -0800544 {AUDIO_INPUT_FLAG_FAST, "FAST"},
545 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"},
546 {AUDIO_INPUT_FLAG_RAW, "RAW"},
547 {AUDIO_INPUT_FLAG_SYNC, "SYNC"},
548 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800549 };
550 String8 result;
551 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE;
552 const mapping *entry;
553 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) {
554 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag);
555 if (flags & entry->mFlag) {
556 if (!result.isEmpty()) {
557 result.append("|");
558 }
559 result.append(entry->mString);
560 }
561 }
562 if (flags & ~allFlags) {
563 if (!result.isEmpty()) {
564 result.append("|");
565 }
566 result.appendFormat("0x%X", flags & ~allFlags);
567 }
568 if (result.isEmpty()) {
569 result.append(entry->mString);
570 }
571 return result;
572}
573
574String8 outputFlagsToString(audio_output_flags_t flags)
Glenn Kasten97b7b752014-09-28 13:04:24 -0700575{
576 static const struct mapping {
577 audio_output_flags_t mFlag;
578 const char * mString;
579 } mappings[] = {
Glenn Kasten818da522015-12-02 13:53:26 -0800580 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"},
581 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"},
582 {AUDIO_OUTPUT_FLAG_FAST, "FAST"},
583 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"},
584 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"},
585 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"},
586 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"},
587 {AUDIO_OUTPUT_FLAG_RAW, "RAW"},
588 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"},
589 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"},
590 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last
Glenn Kasten97b7b752014-09-28 13:04:24 -0700591 };
592 String8 result;
593 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE;
594 const mapping *entry;
595 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) {
596 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag);
597 if (flags & entry->mFlag) {
598 if (!result.isEmpty()) {
599 result.append("|");
600 }
601 result.append(entry->mString);
602 }
603 }
604 if (flags & ~allFlags) {
605 if (!result.isEmpty()) {
606 result.append("|");
607 }
608 result.appendFormat("0x%X", flags & ~allFlags);
609 }
610 if (result.isEmpty()) {
611 result.append(entry->mString);
612 }
613 return result;
614}
615
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800616const char *sourceToString(audio_source_t source)
617{
618 switch (source) {
619 case AUDIO_SOURCE_DEFAULT: return "default";
620 case AUDIO_SOURCE_MIC: return "mic";
621 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink";
622 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink";
623 case AUDIO_SOURCE_VOICE_CALL: return "voice call";
624 case AUDIO_SOURCE_CAMCORDER: return "camcorder";
625 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition";
626 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication";
627 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix";
rago8a397d52015-12-02 11:27:57 -0800628 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed";
Glenn Kasten0f5b5622015-02-18 14:33:30 -0800629 case AUDIO_SOURCE_FM_TUNER: return "FM tuner";
630 case AUDIO_SOURCE_HOTWORD: return "hotword";
631 default: return "unknown";
632 }
633}
634
Eric Laurent81784c32012-11-19 14:55:58 -0800635AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
Eric Laurent72e3f392015-05-20 14:43:50 -0700636 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady)
Eric Laurent81784c32012-11-19 14:55:58 -0800637 : Thread(false /*canCallJava*/),
638 mType(type),
Glenn Kasten9b58f632013-07-16 11:37:48 -0700639 mAudioFlinger(audioFlinger),
Glenn Kasten70949c42013-08-06 07:40:12 -0700640 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
Glenn Kastendeca2ae2014-02-07 10:25:56 -0800641 // are set by PlaybackThread::readOutputParameters_l() or
642 // RecordThread::readInputParameters_l()
Eric Laurentfd477972013-10-25 18:10:40 -0700643 //FIXME: mStandby should be true here. Is this some kind of hack?
Eric Laurent81784c32012-11-19 14:55:58 -0800644 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700645 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE),
646 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
Eric Laurent81784c32012-11-19 14:55:58 -0800647 // mName will be set by concrete (non-virtual) subclass
Eric Laurent72e3f392015-05-20 14:43:50 -0700648 mDeathRecipient(new PMDeathRecipient(this)),
Wei Jia3f273d12015-11-24 09:06:49 -0800649 mSystemReady(systemReady),
650 mNotifiedBatteryStart(false)
Eric Laurent81784c32012-11-19 14:55:58 -0800651{
Eric Laurent296fb132015-05-01 11:38:42 -0700652 memset(&mPatch, 0, sizeof(struct audio_patch));
Eric Laurent81784c32012-11-19 14:55:58 -0800653}
654
655AudioFlinger::ThreadBase::~ThreadBase()
656{
Glenn Kastenc6ae3c82013-07-17 09:08:51 -0700657 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns
Glenn Kastenc6ae3c82013-07-17 09:08:51 -0700658 mConfigEvents.clear();
659
Eric Laurent81784c32012-11-19 14:55:58 -0800660 // do not lock the mutex in destructor
661 releaseWakeLock_l();
662 if (mPowerManager != 0) {
Marco Nelissen06b46062014-11-14 07:58:25 -0800663 sp<IBinder> binder = IInterface::asBinder(mPowerManager);
Eric Laurent81784c32012-11-19 14:55:58 -0800664 binder->unlinkToDeath(mDeathRecipient);
665 }
666}
667
Glenn Kastencf04c2c2013-08-06 07:41:16 -0700668status_t AudioFlinger::ThreadBase::readyToRun()
669{
670 status_t status = initCheck();
671 if (status == NO_ERROR) {
672 ALOGI("AudioFlinger's thread %p ready to run", this);
673 } else {
674 ALOGE("No working audio driver found.");
675 }
676 return status;
677}
678
Eric Laurent81784c32012-11-19 14:55:58 -0800679void AudioFlinger::ThreadBase::exit()
680{
681 ALOGV("ThreadBase::exit");
682 // do any cleanup required for exit to succeed
683 preExit();
684 {
685 // This lock prevents the following race in thread (uniprocessor for illustration):
686 // if (!exitPending()) {
687 // // context switch from here to exit()
688 // // exit() calls requestExit(), what exitPending() observes
689 // // exit() calls signal(), which is dropped since no waiters
690 // // context switch back from exit() to here
691 // mWaitWorkCV.wait(...);
692 // // now thread is hung
693 // }
694 AutoMutex lock(mLock);
695 requestExit();
696 mWaitWorkCV.broadcast();
697 }
698 // When Thread::requestExitAndWait is made virtual and this method is renamed to
699 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
700 requestExitAndWait();
701}
702
703status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
704{
Eric Laurent81784c32012-11-19 14:55:58 -0800705 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
706 Mutex::Autolock _l(mLock);
707
Eric Laurent10351942014-05-08 18:49:52 -0700708 return sendSetParameterConfigEvent_l(keyValuePairs);
709}
710
711// sendConfigEvent_l() must be called with ThreadBase::mLock held
712// Can temporarily release the lock if waiting for a reply from processConfigEvents_l().
713status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
714{
715 status_t status = NO_ERROR;
716
Eric Laurent72e3f392015-05-20 14:43:50 -0700717 if (event->mRequiresSystemReady && !mSystemReady) {
718 event->mWaitStatus = false;
719 mPendingConfigEvents.add(event);
720 return status;
721 }
Eric Laurent10351942014-05-08 18:49:52 -0700722 mConfigEvents.add(event);
Glenn Kastenc42e9b42016-03-21 11:35:03 -0700723 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType);
Eric Laurent81784c32012-11-19 14:55:58 -0800724 mWaitWorkCV.signal();
Eric Laurent10351942014-05-08 18:49:52 -0700725 mLock.unlock();
726 {
727 Mutex::Autolock _l(event->mLock);
728 while (event->mWaitStatus) {
729 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) {
730 event->mStatus = TIMED_OUT;
731 event->mWaitStatus = false;
732 }
733 }
734 status = event->mStatus;
Eric Laurent81784c32012-11-19 14:55:58 -0800735 }
Eric Laurent10351942014-05-08 18:49:52 -0700736 mLock.lock();
Eric Laurent81784c32012-11-19 14:55:58 -0800737 return status;
738}
739
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700740void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid)
Eric Laurent81784c32012-11-19 14:55:58 -0800741{
742 Mutex::Autolock _l(mLock);
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700743 sendIoConfigEvent_l(event, pid);
Eric Laurent81784c32012-11-19 14:55:58 -0800744}
745
746// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700747void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid)
Eric Laurent81784c32012-11-19 14:55:58 -0800748{
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700749 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid);
Eric Laurent10351942014-05-08 18:49:52 -0700750 sendConfigEvent_l(configEvent);
Eric Laurent81784c32012-11-19 14:55:58 -0800751}
752
Eric Laurent72e3f392015-05-20 14:43:50 -0700753void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio)
754{
755 Mutex::Autolock _l(mLock);
756 sendPrioConfigEvent_l(pid, tid, prio);
757}
758
Eric Laurent81784c32012-11-19 14:55:58 -0800759// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
760void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
761{
Eric Laurent10351942014-05-08 18:49:52 -0700762 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio);
763 sendConfigEvent_l(configEvent);
Eric Laurent81784c32012-11-19 14:55:58 -0800764}
765
Eric Laurent10351942014-05-08 18:49:52 -0700766// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held
767status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
Eric Laurent81784c32012-11-19 14:55:58 -0800768{
Andy Hung2ddee192015-12-18 17:34:44 -0800769 sp<ConfigEvent> configEvent;
770 AudioParameter param(keyValuePair);
771 int value;
772 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) {
773 setMasterMono_l(value != 0);
774 if (param.size() == 1) {
775 return NO_ERROR; // should be a solo parameter - we don't pass down
776 }
777 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT));
778 configEvent = new SetParameterConfigEvent(param.toString());
779 } else {
780 configEvent = new SetParameterConfigEvent(keyValuePair);
781 }
Eric Laurent10351942014-05-08 18:49:52 -0700782 return sendConfigEvent_l(configEvent);
Glenn Kastenf7773312013-08-13 16:00:42 -0700783}
784
Eric Laurent1c333e22014-05-20 10:48:17 -0700785status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent(
786 const struct audio_patch *patch,
787 audio_patch_handle_t *handle)
788{
789 Mutex::Autolock _l(mLock);
790 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle);
791 status_t status = sendConfigEvent_l(configEvent);
792 if (status == NO_ERROR) {
793 CreateAudioPatchConfigEventData *data =
794 (CreateAudioPatchConfigEventData *)configEvent->mData.get();
795 *handle = data->mHandle;
796 }
797 return status;
798}
799
800status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent(
801 const audio_patch_handle_t handle)
802{
803 Mutex::Autolock _l(mLock);
804 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle);
805 return sendConfigEvent_l(configEvent);
806}
807
808
Glenn Kasten2cfbf882013-08-14 13:12:11 -0700809// post condition: mConfigEvents.isEmpty()
Eric Laurent021cf962014-05-13 10:18:14 -0700810void AudioFlinger::ThreadBase::processConfigEvents_l()
Glenn Kastenf7773312013-08-13 16:00:42 -0700811{
Eric Laurent10351942014-05-08 18:49:52 -0700812 bool configChanged = false;
813
Eric Laurent81784c32012-11-19 14:55:58 -0800814 while (!mConfigEvents.isEmpty()) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -0700815 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size());
Eric Laurent10351942014-05-08 18:49:52 -0700816 sp<ConfigEvent> event = mConfigEvents[0];
Eric Laurent81784c32012-11-19 14:55:58 -0800817 mConfigEvents.removeAt(0);
Eric Laurent10351942014-05-08 18:49:52 -0700818 switch (event->mType) {
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700819 case CFG_EVENT_PRIO: {
Eric Laurent10351942014-05-08 18:49:52 -0700820 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get();
821 // FIXME Need to understand why this has to be done asynchronously
822 int err = requestPriority(data->mPid, data->mTid, data->mPrio,
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700823 true /*asynchronous*/);
824 if (err != 0) {
825 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
Eric Laurent10351942014-05-08 18:49:52 -0700826 data->mPrio, data->mPid, data->mTid, err);
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700827 }
828 } break;
829 case CFG_EVENT_IO: {
Eric Laurent10351942014-05-08 18:49:52 -0700830 IoConfigEventData *data = (IoConfigEventData *)event->mData.get();
Eric Laurent7c1ec5f2015-07-09 14:52:47 -0700831 ioConfigChanged(data->mEvent, data->mPid);
Eric Laurent10351942014-05-08 18:49:52 -0700832 } break;
833 case CFG_EVENT_SET_PARAMETER: {
834 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get();
835 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) {
836 configChanged = true;
Glenn Kastend5418eb2013-08-14 13:11:06 -0700837 }
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700838 } break;
Eric Laurent1c333e22014-05-20 10:48:17 -0700839 case CFG_EVENT_CREATE_AUDIO_PATCH: {
840 CreateAudioPatchConfigEventData *data =
841 (CreateAudioPatchConfigEventData *)event->mData.get();
842 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
843 } break;
844 case CFG_EVENT_RELEASE_AUDIO_PATCH: {
845 ReleaseAudioPatchConfigEventData *data =
846 (ReleaseAudioPatchConfigEventData *)event->mData.get();
847 event->mStatus = releaseAudioPatch_l(data->mHandle);
848 } break;
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700849 default:
Eric Laurent10351942014-05-08 18:49:52 -0700850 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
Glenn Kasten3468e8a2013-08-13 16:01:22 -0700851 break;
Eric Laurent81784c32012-11-19 14:55:58 -0800852 }
Eric Laurent10351942014-05-08 18:49:52 -0700853 {
854 Mutex::Autolock _l(event->mLock);
855 if (event->mWaitStatus) {
856 event->mWaitStatus = false;
857 event->mCond.signal();
858 }
859 }
860 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this);
861 }
862
863 if (configChanged) {
864 cacheParameters_l();
Eric Laurent81784c32012-11-19 14:55:58 -0800865 }
Eric Laurent81784c32012-11-19 14:55:58 -0800866}
867
Marco Nelissenb2208842014-02-07 14:00:50 -0800868String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
869 String8 s;
Glenn Kastene1635ec2015-06-08 15:46:49 -0700870 const audio_channel_representation_t representation =
871 audio_channel_mask_get_representation(mask);
Andy Hungf98ec8d2015-05-19 12:53:24 -0700872
873 switch (representation) {
874 case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
875 if (output) {
876 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
877 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
878 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
879 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
880 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
881 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
882 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
883 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
884 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
885 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
886 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
887 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
888 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
889 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
890 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
891 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
892 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
893 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
894 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, ");
895 } else {
896 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
897 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
898 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
899 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
900 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
901 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
902 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
903 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
904 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
905 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
906 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
907 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
908 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
909 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
910 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, ");
911 }
912 const int len = s.length();
913 if (len > 2) {
Glenn Kasten57c4e6f2016-03-18 14:54:07 -0700914 (void) s.lockBuffer(len); // needed?
Andy Hungf98ec8d2015-05-19 12:53:24 -0700915 s.unlockBuffer(len - 2); // remove trailing ", "
916 }
917 return s;
Marco Nelissenb2208842014-02-07 14:00:50 -0800918 }
Andy Hungf98ec8d2015-05-19 12:53:24 -0700919 case AUDIO_CHANNEL_REPRESENTATION_INDEX:
920 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask));
921 return s;
922 default:
923 s.appendFormat("unknown mask, representation:%d bits:%#x",
924 representation, audio_channel_mask_get_bits(mask));
925 return s;
Marco Nelissenb2208842014-02-07 14:00:50 -0800926 }
Marco Nelissenb2208842014-02-07 14:00:50 -0800927}
928
Glenn Kasten0f11b512014-01-31 16:18:54 -0800929void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
Eric Laurent81784c32012-11-19 14:55:58 -0800930{
931 const size_t SIZE = 256;
932 char buffer[SIZE];
933 String8 result;
934
935 bool locked = AudioFlinger::dumpTryLock(mLock);
936 if (!locked) {
Glenn Kasten97b7b752014-09-28 13:04:24 -0700937 dprintf(fd, "thread %p may be deadlocked\n", this);
Eric Laurent81784c32012-11-19 14:55:58 -0800938 }
939
Glenn Kasten0b89bc02015-03-05 16:37:47 -0800940 dprintf(fd, " Thread name: %s\n", mThreadName);
Elliott Hughes87cebad2014-05-22 10:14:43 -0700941 dprintf(fd, " I/O handle: %d\n", mId);
942 dprintf(fd, " TID: %d\n", getTid());
943 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
Glenn Kasten97b7b752014-09-28 13:04:24 -0700944 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
Elliott Hughes87cebad2014-05-22 10:14:43 -0700945 dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
Glenn Kasten97b7b752014-09-28 13:04:24 -0700946 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
Glenn Kastenc42e9b42016-03-21 11:35:03 -0700947 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize);
Glenn Kasten97b7b752014-09-28 13:04:24 -0700948 dprintf(fd, " Channel count: %u\n", mChannelCount);
949 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask,
Marco Nelissenb2208842014-02-07 14:00:50 -0800950 channelMaskToString(mChannelMask, mType != RECORD).string());
Glenn Kastenf87c2f52015-08-21 08:03:57 -0700951 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
952 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize);
Elliott Hughes87cebad2014-05-22 10:14:43 -0700953 dprintf(fd, " Pending config events:");
Marco Nelissenb2208842014-02-07 14:00:50 -0800954 size_t numConfig = mConfigEvents.size();
955 if (numConfig) {
956 for (size_t i = 0; i < numConfig; i++) {
957 mConfigEvents[i]->dump(buffer, SIZE);
Elliott Hughes87cebad2014-05-22 10:14:43 -0700958 dprintf(fd, "\n %s", buffer);
Marco Nelissenb2208842014-02-07 14:00:50 -0800959 }
Elliott Hughes87cebad2014-05-22 10:14:43 -0700960 dprintf(fd, "\n");
Marco Nelissenb2208842014-02-07 14:00:50 -0800961 } else {
Elliott Hughes87cebad2014-05-22 10:14:43 -0700962 dprintf(fd, " none\n");
Eric Laurent81784c32012-11-19 14:55:58 -0800963 }
Glenn Kasten0b89bc02015-03-05 16:37:47 -0800964 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string());
965 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string());
966 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource));
Eric Laurent81784c32012-11-19 14:55:58 -0800967
968 if (locked) {
969 mLock.unlock();
970 }
971}
972
973void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
974{
975 const size_t SIZE = 256;
976 char buffer[SIZE];
977 String8 result;
978
Marco Nelissenb2208842014-02-07 14:00:50 -0800979 size_t numEffectChains = mEffectChains.size();
Narayan Kamath1d6fa7a2014-02-11 13:47:53 +0000980 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains);
Eric Laurent81784c32012-11-19 14:55:58 -0800981 write(fd, buffer, strlen(buffer));
982
Marco Nelissenb2208842014-02-07 14:00:50 -0800983 for (size_t i = 0; i < numEffectChains; ++i) {
Eric Laurent81784c32012-11-19 14:55:58 -0800984 sp<EffectChain> chain = mEffectChains[i];
985 if (chain != 0) {
986 chain->dump(fd, args);
987 }
988 }
989}
990
Marco Nelissene14a5d62013-10-03 08:51:24 -0700991void AudioFlinger::ThreadBase::acquireWakeLock(int uid)
Eric Laurent81784c32012-11-19 14:55:58 -0800992{
993 Mutex::Autolock _l(mLock);
Marco Nelissene14a5d62013-10-03 08:51:24 -0700994 acquireWakeLock_l(uid);
Eric Laurent81784c32012-11-19 14:55:58 -0800995}
996
Narayan Kamath014e7fa2013-10-14 15:03:38 +0100997String16 AudioFlinger::ThreadBase::getWakeLockTag()
998{
999 switch (mType) {
Glenn Kastenbcb14862015-03-05 17:11:21 -08001000 case MIXER:
1001 return String16("AudioMix");
1002 case DIRECT:
1003 return String16("AudioDirectOut");
1004 case DUPLICATING:
1005 return String16("AudioDup");
1006 case RECORD:
1007 return String16("AudioIn");
1008 case OFFLOAD:
1009 return String16("AudioOffload");
1010 default:
1011 ALOG_ASSERT(false);
1012 return String16("AudioUnknown");
Narayan Kamath014e7fa2013-10-14 15:03:38 +01001013 }
1014}
1015
Marco Nelissene14a5d62013-10-03 08:51:24 -07001016void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid)
Eric Laurent81784c32012-11-19 14:55:58 -08001017{
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001018 getPowerManager_l();
Eric Laurent81784c32012-11-19 14:55:58 -08001019 if (mPowerManager != 0) {
1020 sp<IBinder> binder = new BBinder();
Marco Nelissene14a5d62013-10-03 08:51:24 -07001021 status_t status;
1022 if (uid >= 0) {
Eric Laurent547789d2013-10-04 11:46:55 -07001023 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK,
Marco Nelissene14a5d62013-10-03 08:51:24 -07001024 binder,
Narayan Kamath014e7fa2013-10-14 15:03:38 +01001025 getWakeLockTag(),
Marco Nelissendcb346b2015-09-09 10:47:29 -07001026 String16("audioserver"),
Glenn Kasten3abc2de2014-09-05 16:45:52 -07001027 uid,
1028 true /* FIXME force oneway contrary to .aidl */);
Marco Nelissene14a5d62013-10-03 08:51:24 -07001029 } else {
Eric Laurent547789d2013-10-04 11:46:55 -07001030 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
Marco Nelissene14a5d62013-10-03 08:51:24 -07001031 binder,
Narayan Kamath014e7fa2013-10-14 15:03:38 +01001032 getWakeLockTag(),
Marco Nelissendcb346b2015-09-09 10:47:29 -07001033 String16("audioserver"),
Glenn Kasten3abc2de2014-09-05 16:45:52 -07001034 true /* FIXME force oneway contrary to .aidl */);
Marco Nelissene14a5d62013-10-03 08:51:24 -07001035 }
Eric Laurent81784c32012-11-19 14:55:58 -08001036 if (status == NO_ERROR) {
1037 mWakeLockToken = binder;
1038 }
Glenn Kastend7dca052015-03-05 16:05:54 -08001039 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status);
Eric Laurent81784c32012-11-19 14:55:58 -08001040 }
Wei Jia3f273d12015-11-24 09:06:49 -08001041
1042 if (!mNotifiedBatteryStart) {
1043 BatteryNotifier::getInstance().noteStartAudio();
1044 mNotifiedBatteryStart = true;
1045 }
Andy Hung3f0c9022016-01-15 17:49:46 -08001046 gBoottime.acquire(mWakeLockToken);
Andy Hung818e7a32016-02-16 18:08:07 -08001047 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] =
1048 gBoottime.getBoottimeOffset();
Eric Laurent81784c32012-11-19 14:55:58 -08001049}
1050
1051void AudioFlinger::ThreadBase::releaseWakeLock()
1052{
1053 Mutex::Autolock _l(mLock);
1054 releaseWakeLock_l();
1055}
1056
1057void AudioFlinger::ThreadBase::releaseWakeLock_l()
1058{
Andy Hung3f0c9022016-01-15 17:49:46 -08001059 gBoottime.release(mWakeLockToken);
Eric Laurent81784c32012-11-19 14:55:58 -08001060 if (mWakeLockToken != 0) {
Glenn Kastend7dca052015-03-05 16:05:54 -08001061 ALOGV("releaseWakeLock_l() %s", mThreadName);
Eric Laurent81784c32012-11-19 14:55:58 -08001062 if (mPowerManager != 0) {
Glenn Kasten3abc2de2014-09-05 16:45:52 -07001063 mPowerManager->releaseWakeLock(mWakeLockToken, 0,
1064 true /* FIXME force oneway contrary to .aidl */);
Eric Laurent81784c32012-11-19 14:55:58 -08001065 }
1066 mWakeLockToken.clear();
1067 }
Wei Jia3f273d12015-11-24 09:06:49 -08001068
1069 if (mNotifiedBatteryStart) {
1070 BatteryNotifier::getInstance().noteStopAudio();
1071 mNotifiedBatteryStart = false;
1072 }
Eric Laurent81784c32012-11-19 14:55:58 -08001073}
1074
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001075void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) {
1076 Mutex::Autolock _l(mLock);
1077 updateWakeLockUids_l(uids);
1078}
1079
1080void AudioFlinger::ThreadBase::getPowerManager_l() {
Eric Laurent72e3f392015-05-20 14:43:50 -07001081 if (mSystemReady && mPowerManager == 0) {
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001082 // use checkService() to avoid blocking if power service is not up yet
1083 sp<IBinder> binder =
1084 defaultServiceManager()->checkService(String16("power"));
1085 if (binder == 0) {
Glenn Kastend7dca052015-03-05 16:05:54 -08001086 ALOGW("Thread %s cannot connect to the power manager service", mThreadName);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001087 } else {
1088 mPowerManager = interface_cast<IPowerManager>(binder);
1089 binder->linkToDeath(mDeathRecipient);
1090 }
1091 }
1092}
1093
1094void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001095 getPowerManager_l();
Andy Hung438e7572015-12-14 15:51:17 -08001096 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called.
1097 if (mSystemReady) {
1098 ALOGE("no wake lock to update, but system ready!");
1099 } else {
1100 ALOGW("no wake lock to update, system not ready yet");
1101 }
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001102 return;
1103 }
1104 if (mPowerManager != 0) {
1105 sp<IBinder> binder = new BBinder();
1106 status_t status;
Glenn Kasten3abc2de2014-09-05 16:45:52 -07001107 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(),
1108 true /* FIXME force oneway contrary to .aidl */);
Eric Laurent4d231dc2016-03-11 18:38:23 -08001109 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001110 }
1111}
1112
Eric Laurent81784c32012-11-19 14:55:58 -08001113void AudioFlinger::ThreadBase::clearPowerManager()
1114{
1115 Mutex::Autolock _l(mLock);
1116 releaseWakeLock_l();
1117 mPowerManager.clear();
1118}
1119
Glenn Kasten0f11b512014-01-31 16:18:54 -08001120void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08001121{
1122 sp<ThreadBase> thread = mThread.promote();
1123 if (thread != 0) {
1124 thread->clearPowerManager();
1125 }
1126 ALOGW("power manager service died !!!");
1127}
1128
1129void AudioFlinger::ThreadBase::setEffectSuspended(
Glenn Kastend848eb42016-03-08 13:42:11 -08001130 const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001131{
1132 Mutex::Autolock _l(mLock);
1133 setEffectSuspended_l(type, suspend, sessionId);
1134}
1135
1136void AudioFlinger::ThreadBase::setEffectSuspended_l(
Glenn Kastend848eb42016-03-08 13:42:11 -08001137 const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001138{
1139 sp<EffectChain> chain = getEffectChain_l(sessionId);
1140 if (chain != 0) {
1141 if (type != NULL) {
1142 chain->setEffectSuspended_l(type, suspend);
1143 } else {
1144 chain->setEffectSuspendedAll_l(suspend);
1145 }
1146 }
1147
1148 updateSuspendedSessions_l(type, suspend, sessionId);
1149}
1150
1151void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
1152{
1153 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
1154 if (index < 0) {
1155 return;
1156 }
1157
1158 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects =
1159 mSuspendedSessions.valueAt(index);
1160
1161 for (size_t i = 0; i < sessionEffects.size(); i++) {
Chih-Hung Hsiehe964d4e2016-08-09 14:31:32 -07001162 const sp<SuspendedSessionDesc>& desc = sessionEffects.valueAt(i);
Eric Laurent81784c32012-11-19 14:55:58 -08001163 for (int j = 0; j < desc->mRefCount; j++) {
1164 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
1165 chain->setEffectSuspendedAll_l(true);
1166 } else {
1167 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
1168 desc->mType.timeLow);
1169 chain->setEffectSuspended_l(&desc->mType, true);
1170 }
1171 }
1172 }
1173}
1174
1175void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
1176 bool suspend,
Glenn Kastend848eb42016-03-08 13:42:11 -08001177 audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001178{
1179 ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
1180
1181 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
1182
1183 if (suspend) {
1184 if (index >= 0) {
1185 sessionEffects = mSuspendedSessions.valueAt(index);
1186 } else {
1187 mSuspendedSessions.add(sessionId, sessionEffects);
1188 }
1189 } else {
1190 if (index < 0) {
1191 return;
1192 }
1193 sessionEffects = mSuspendedSessions.valueAt(index);
1194 }
1195
1196
1197 int key = EffectChain::kKeyForSuspendAll;
1198 if (type != NULL) {
1199 key = type->timeLow;
1200 }
1201 index = sessionEffects.indexOfKey(key);
1202
1203 sp<SuspendedSessionDesc> desc;
1204 if (suspend) {
1205 if (index >= 0) {
1206 desc = sessionEffects.valueAt(index);
1207 } else {
1208 desc = new SuspendedSessionDesc();
1209 if (type != NULL) {
1210 desc->mType = *type;
1211 }
1212 sessionEffects.add(key, desc);
1213 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
1214 }
1215 desc->mRefCount++;
1216 } else {
1217 if (index < 0) {
1218 return;
1219 }
1220 desc = sessionEffects.valueAt(index);
1221 if (--desc->mRefCount == 0) {
1222 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
1223 sessionEffects.removeItemsAt(index);
1224 if (sessionEffects.isEmpty()) {
1225 ALOGV("updateSuspendedSessions_l() restore removing session %d",
1226 sessionId);
1227 mSuspendedSessions.removeItem(sessionId);
1228 }
1229 }
1230 }
1231 if (!sessionEffects.isEmpty()) {
1232 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
1233 }
1234}
1235
1236void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
1237 bool enabled,
Glenn Kastend848eb42016-03-08 13:42:11 -08001238 audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001239{
1240 Mutex::Autolock _l(mLock);
1241 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
1242}
1243
1244void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
1245 bool enabled,
Glenn Kastend848eb42016-03-08 13:42:11 -08001246 audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001247{
1248 if (mType != RECORD) {
1249 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
1250 // another session. This gives the priority to well behaved effect control panels
1251 // and applications not using global effects.
1252 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
1253 // global effects
1254 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
1255 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
1256 }
1257 }
1258
1259 sp<EffectChain> chain = getEffectChain_l(sessionId);
1260 if (chain != 0) {
1261 chain->checkSuspendOnEffectEnabled(effect, enabled);
1262 }
1263}
1264
Eric Laurent4c415062016-06-17 16:14:16 -07001265// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
1266status_t AudioFlinger::RecordThread::checkEffectCompatibility_l(
1267 const effect_descriptor_t *desc, audio_session_t sessionId)
1268{
1269 // No global effect sessions on record threads
1270 if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
1271 ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s",
1272 desc->name, mThreadName);
1273 return BAD_VALUE;
1274 }
1275 // only pre processing effects on record thread
1276 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) {
1277 ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on record thread %s",
1278 desc->name, mThreadName);
1279 return BAD_VALUE;
1280 }
Eric Laurent6dd0fd92016-09-15 12:44:53 -07001281
1282 // always allow effects without processing load or latency
1283 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) {
1284 return NO_ERROR;
1285 }
1286
Eric Laurent4c415062016-06-17 16:14:16 -07001287 audio_input_flags_t flags = mInput->flags;
1288 if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) {
1289 if (flags & AUDIO_INPUT_FLAG_RAW) {
1290 ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode",
1291 desc->name, mThreadName);
1292 return BAD_VALUE;
1293 }
1294 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
1295 ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode",
1296 desc->name, mThreadName);
1297 return BAD_VALUE;
1298 }
1299 }
1300 return NO_ERROR;
1301}
1302
1303// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
1304status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l(
1305 const effect_descriptor_t *desc, audio_session_t sessionId)
1306{
1307 // no preprocessing on playback threads
1308 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) {
1309 ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback"
1310 " thread %s", desc->name, mThreadName);
1311 return BAD_VALUE;
1312 }
1313
1314 switch (mType) {
1315 case MIXER: {
1316 // Reject any effect on mixer multichannel sinks.
1317 // TODO: fix both format and multichannel issues with effects.
1318 if (mChannelCount != FCC_2) {
1319 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d) on MIXER"
1320 " thread %s", desc->name, mChannelCount, mThreadName);
1321 return BAD_VALUE;
1322 }
1323 audio_output_flags_t flags = mOutput->flags;
1324 if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) {
1325 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
1326 // global effects are applied only to non fast tracks if they are SW
1327 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
1328 break;
1329 }
1330 } else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
1331 // only post processing on output stage session
1332 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
1333 ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed"
1334 " on output stage session", desc->name);
1335 return BAD_VALUE;
1336 }
1337 } else {
1338 // no restriction on effects applied on non fast tracks
1339 if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) {
1340 break;
1341 }
1342 }
Eric Laurent6dd0fd92016-09-15 12:44:53 -07001343
1344 // always allow effects without processing load or latency
1345 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) {
1346 break;
1347 }
Eric Laurent4c415062016-06-17 16:14:16 -07001348 if (flags & AUDIO_OUTPUT_FLAG_RAW) {
1349 ALOGW("checkEffectCompatibility_l(): effect %s on playback thread in raw mode",
1350 desc->name);
1351 return BAD_VALUE;
1352 }
1353 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) {
1354 ALOGW("checkEffectCompatibility_l(): non HW effect %s on playback thread"
1355 " in fast mode", desc->name);
1356 return BAD_VALUE;
1357 }
1358 }
1359 } break;
1360 case OFFLOAD:
Jean-Michel Trivi773ee952016-07-11 16:53:18 -07001361 // nothing actionable on offload threads, if the effect:
1362 // - is offloadable: the effect can be created
1363 // - is NOT offloadable: the effect should still be created, but EffectHandle::enable()
1364 // will take care of invalidating the tracks of the thread
Eric Laurent4c415062016-06-17 16:14:16 -07001365 break;
1366 case DIRECT:
1367 // Reject any effect on Direct output threads for now, since the format of
1368 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo).
1369 ALOGW("checkEffectCompatibility_l(): effect %s on DIRECT output thread %s",
1370 desc->name, mThreadName);
1371 return BAD_VALUE;
1372 case DUPLICATING:
1373 // Reject any effect on mixer multichannel sinks.
1374 // TODO: fix both format and multichannel issues with effects.
1375 if (mChannelCount != FCC_2) {
1376 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d)"
1377 " on DUPLICATING thread %s", desc->name, mChannelCount, mThreadName);
1378 return BAD_VALUE;
1379 }
1380 if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) {
1381 ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING"
1382 " thread %s", desc->name, mThreadName);
1383 return BAD_VALUE;
1384 }
1385 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
1386 ALOGW("checkEffectCompatibility_l(): post processing effect %s on"
1387 " DUPLICATING thread %s", desc->name, mThreadName);
1388 return BAD_VALUE;
1389 }
1390 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) {
1391 ALOGW("checkEffectCompatibility_l(): HW tunneled effect %s on"
1392 " DUPLICATING thread %s", desc->name, mThreadName);
1393 return BAD_VALUE;
1394 }
1395 break;
1396 default:
1397 LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType);
1398 }
1399
1400 return NO_ERROR;
1401}
1402
Eric Laurent81784c32012-11-19 14:55:58 -08001403// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held
1404sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l(
1405 const sp<AudioFlinger::Client>& client,
1406 const sp<IEffectClient>& effectClient,
1407 int32_t priority,
Glenn Kastend848eb42016-03-08 13:42:11 -08001408 audio_session_t sessionId,
Eric Laurent81784c32012-11-19 14:55:58 -08001409 effect_descriptor_t *desc,
1410 int *enabled,
Glenn Kasten9156ef32013-08-06 15:39:08 -07001411 status_t *status)
Eric Laurent81784c32012-11-19 14:55:58 -08001412{
1413 sp<EffectModule> effect;
1414 sp<EffectHandle> handle;
1415 status_t lStatus;
1416 sp<EffectChain> chain;
1417 bool chainCreated = false;
1418 bool effectCreated = false;
1419 bool effectRegistered = false;
1420
1421 lStatus = initCheck();
1422 if (lStatus != NO_ERROR) {
1423 ALOGW("createEffect_l() Audio driver not initialized.");
1424 goto Exit;
1425 }
1426
Eric Laurent81784c32012-11-19 14:55:58 -08001427 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
1428
1429 { // scope for mLock
1430 Mutex::Autolock _l(mLock);
1431
Eric Laurent4c415062016-06-17 16:14:16 -07001432 lStatus = checkEffectCompatibility_l(desc, sessionId);
1433 if (lStatus != NO_ERROR) {
1434 goto Exit;
1435 }
1436
Eric Laurent81784c32012-11-19 14:55:58 -08001437 // check for existing effect chain with the requested audio session
1438 chain = getEffectChain_l(sessionId);
1439 if (chain == 0) {
1440 // create a new chain for this session
1441 ALOGV("createEffect_l() new effect chain for session %d", sessionId);
1442 chain = new EffectChain(this, sessionId);
1443 addEffectChain_l(chain);
1444 chain->setStrategy(getStrategyForSession_l(sessionId));
1445 chainCreated = true;
1446 } else {
1447 effect = chain->getEffectFromDesc_l(desc);
1448 }
1449
1450 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
1451
1452 if (effect == 0) {
Glenn Kasteneeecb982016-02-26 10:44:04 -08001453 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
Eric Laurent81784c32012-11-19 14:55:58 -08001454 // Check CPU and memory usage
1455 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
1456 if (lStatus != NO_ERROR) {
1457 goto Exit;
1458 }
1459 effectRegistered = true;
1460 // create a new effect module if none present in the chain
1461 effect = new EffectModule(this, chain, desc, id, sessionId);
1462 lStatus = effect->status();
1463 if (lStatus != NO_ERROR) {
1464 goto Exit;
1465 }
Eric Laurent5baf2af2013-09-12 17:37:00 -07001466 effect->setOffloaded(mType == OFFLOAD, mId);
1467
Eric Laurent81784c32012-11-19 14:55:58 -08001468 lStatus = chain->addEffect_l(effect);
1469 if (lStatus != NO_ERROR) {
1470 goto Exit;
1471 }
1472 effectCreated = true;
1473
1474 effect->setDevice(mOutDevice);
1475 effect->setDevice(mInDevice);
1476 effect->setMode(mAudioFlinger->getMode());
1477 effect->setAudioSource(mAudioSource);
1478 }
1479 // create effect handle and connect it to effect module
1480 handle = new EffectHandle(effect, client, effectClient, priority);
Glenn Kastene75da402013-11-20 13:54:52 -08001481 lStatus = handle->initCheck();
1482 if (lStatus == OK) {
1483 lStatus = effect->addHandle(handle.get());
1484 }
Eric Laurent81784c32012-11-19 14:55:58 -08001485 if (enabled != NULL) {
1486 *enabled = (int)effect->isEnabled();
1487 }
1488 }
1489
1490Exit:
1491 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
1492 Mutex::Autolock _l(mLock);
1493 if (effectCreated) {
1494 chain->removeEffect_l(effect);
1495 }
1496 if (effectRegistered) {
1497 AudioSystem::unregisterEffect(effect->id());
1498 }
1499 if (chainCreated) {
1500 removeEffectChain_l(chain);
1501 }
1502 handle.clear();
1503 }
1504
Glenn Kasten9156ef32013-08-06 15:39:08 -07001505 *status = lStatus;
Eric Laurent81784c32012-11-19 14:55:58 -08001506 return handle;
1507}
1508
Glenn Kastend848eb42016-03-08 13:42:11 -08001509sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId,
1510 int effectId)
Eric Laurent81784c32012-11-19 14:55:58 -08001511{
1512 Mutex::Autolock _l(mLock);
1513 return getEffect_l(sessionId, effectId);
1514}
1515
Glenn Kastend848eb42016-03-08 13:42:11 -08001516sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId,
1517 int effectId)
Eric Laurent81784c32012-11-19 14:55:58 -08001518{
1519 sp<EffectChain> chain = getEffectChain_l(sessionId);
1520 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
1521}
1522
1523// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
1524// PlaybackThread::mLock held
1525status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect)
1526{
1527 // check for existing effect chain with the requested audio session
Glenn Kastend848eb42016-03-08 13:42:11 -08001528 audio_session_t sessionId = effect->sessionId();
Eric Laurent81784c32012-11-19 14:55:58 -08001529 sp<EffectChain> chain = getEffectChain_l(sessionId);
1530 bool chainCreated = false;
1531
Eric Laurent5baf2af2013-09-12 17:37:00 -07001532 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(),
1533 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x",
1534 this, effect->desc().name, effect->desc().flags);
1535
Eric Laurent81784c32012-11-19 14:55:58 -08001536 if (chain == 0) {
1537 // create a new chain for this session
1538 ALOGV("addEffect_l() new effect chain for session %d", sessionId);
1539 chain = new EffectChain(this, sessionId);
1540 addEffectChain_l(chain);
1541 chain->setStrategy(getStrategyForSession_l(sessionId));
1542 chainCreated = true;
1543 }
1544 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
1545
1546 if (chain->getEffectFromId_l(effect->id()) != 0) {
1547 ALOGW("addEffect_l() %p effect %s already present in chain %p",
1548 this, effect->desc().name, chain.get());
1549 return BAD_VALUE;
1550 }
1551
Eric Laurent5baf2af2013-09-12 17:37:00 -07001552 effect->setOffloaded(mType == OFFLOAD, mId);
1553
Eric Laurent81784c32012-11-19 14:55:58 -08001554 status_t status = chain->addEffect_l(effect);
1555 if (status != NO_ERROR) {
1556 if (chainCreated) {
1557 removeEffectChain_l(chain);
1558 }
1559 return status;
1560 }
1561
1562 effect->setDevice(mOutDevice);
1563 effect->setDevice(mInDevice);
1564 effect->setMode(mAudioFlinger->getMode());
1565 effect->setAudioSource(mAudioSource);
1566 return NO_ERROR;
1567}
1568
1569void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
1570
1571 ALOGV("removeEffect_l() %p effect %p", this, effect.get());
1572 effect_descriptor_t desc = effect->desc();
1573 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
1574 detachAuxEffect_l(effect->id());
1575 }
1576
1577 sp<EffectChain> chain = effect->chain().promote();
1578 if (chain != 0) {
1579 // remove effect chain if removing last effect
1580 if (chain->removeEffect_l(effect) == 0) {
1581 removeEffectChain_l(chain);
1582 }
1583 } else {
1584 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
1585 }
1586}
1587
1588void AudioFlinger::ThreadBase::lockEffectChains_l(
1589 Vector< sp<AudioFlinger::EffectChain> >& effectChains)
1590{
1591 effectChains = mEffectChains;
1592 for (size_t i = 0; i < mEffectChains.size(); i++) {
1593 mEffectChains[i]->lock();
1594 }
1595}
1596
1597void AudioFlinger::ThreadBase::unlockEffectChains(
1598 const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
1599{
1600 for (size_t i = 0; i < effectChains.size(); i++) {
1601 effectChains[i]->unlock();
1602 }
1603}
1604
Glenn Kastend848eb42016-03-08 13:42:11 -08001605sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08001606{
1607 Mutex::Autolock _l(mLock);
1608 return getEffectChain_l(sessionId);
1609}
1610
Glenn Kastend848eb42016-03-08 13:42:11 -08001611sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId)
1612 const
Eric Laurent81784c32012-11-19 14:55:58 -08001613{
1614 size_t size = mEffectChains.size();
1615 for (size_t i = 0; i < size; i++) {
1616 if (mEffectChains[i]->sessionId() == sessionId) {
1617 return mEffectChains[i];
1618 }
1619 }
1620 return 0;
1621}
1622
1623void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
1624{
1625 Mutex::Autolock _l(mLock);
1626 size_t size = mEffectChains.size();
1627 for (size_t i = 0; i < size; i++) {
1628 mEffectChains[i]->setMode_l(mode);
1629 }
1630}
1631
Eric Laurent83b88082014-06-20 18:31:16 -07001632void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config)
1633{
1634 config->type = AUDIO_PORT_TYPE_MIX;
1635 config->ext.mix.handle = mId;
1636 config->sample_rate = mSampleRate;
1637 config->format = mFormat;
1638 config->channel_mask = mChannelMask;
1639 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
1640 AUDIO_PORT_CONFIG_FORMAT;
1641}
1642
Eric Laurent72e3f392015-05-20 14:43:50 -07001643void AudioFlinger::ThreadBase::systemReady()
1644{
1645 Mutex::Autolock _l(mLock);
1646 if (mSystemReady) {
1647 return;
1648 }
1649 mSystemReady = true;
1650
1651 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) {
1652 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i));
1653 }
1654 mPendingConfigEvents.clear();
1655}
1656
Eric Laurent83b88082014-06-20 18:31:16 -07001657
Eric Laurent81784c32012-11-19 14:55:58 -08001658// ----------------------------------------------------------------------------
1659// Playback
1660// ----------------------------------------------------------------------------
1661
1662AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
1663 AudioStreamOut* output,
1664 audio_io_handle_t id,
1665 audio_devices_t device,
Eric Laurent72e3f392015-05-20 14:43:50 -07001666 type_t type,
Eric Laurente93cc032016-05-05 10:15:10 -07001667 bool systemReady)
Eric Laurent72e3f392015-05-20 14:43:50 -07001668 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady),
Andy Hung2098f272014-02-27 14:00:06 -08001669 mNormalFrameCount(0), mSinkBuffer(NULL),
Andy Hung6146c082014-03-18 11:56:15 -07001670 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
Andy Hung69aed5f2014-02-25 17:24:40 -08001671 mMixerBuffer(NULL),
1672 mMixerBufferSize(0),
1673 mMixerBufferFormat(AUDIO_FORMAT_INVALID),
1674 mMixerBufferValid(false),
Andy Hung6146c082014-03-18 11:56:15 -07001675 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
Andy Hung98ef9782014-03-04 14:46:50 -08001676 mEffectBuffer(NULL),
1677 mEffectBufferSize(0),
1678 mEffectBufferFormat(AUDIO_FORMAT_INVALID),
1679 mEffectBufferValid(false),
Glenn Kastenc1fac192013-08-06 07:41:36 -07001680 mSuspended(0), mBytesWritten(0),
Andy Hungc54b1ff2016-02-23 14:07:07 -08001681 mFramesWritten(0),
Andy Hung238fa3d2016-07-28 10:53:22 -07001682 mSuspendedFrames(0),
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001683 mActiveTracksGeneration(0),
Eric Laurent81784c32012-11-19 14:55:58 -08001684 // mStreamTypes[] initialized in constructor body
1685 mOutput(output),
Andy Hung69488c42016-05-16 18:43:33 -07001686 mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
Eric Laurent81784c32012-11-19 14:55:58 -08001687 mMixerStatus(MIXER_IDLE),
1688 mMixerStatusIgnoringFastTracks(MIXER_IDLE),
Eric Laurentad9cb8b2015-05-26 16:38:19 -07001689 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs),
Eric Laurentbfb1b832013-01-07 09:53:42 -08001690 mBytesRemaining(0),
1691 mCurrentWriteLength(0),
1692 mUseAsyncWrite(false),
Eric Laurent3b4529e2013-09-05 18:09:19 -07001693 mWriteAckSequence(0),
1694 mDrainSequence(0),
Eric Laurentede6c3b2013-09-19 14:37:46 -07001695 mSignalPending(false),
Eric Laurent81784c32012-11-19 14:55:58 -08001696 mScreenState(AudioFlinger::mScreenState),
1697 // index 0 is reserved for normal mixer's submix
Glenn Kastendc2c50b2016-04-21 08:13:14 -07001698 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1),
Andy Hunge10393e2015-06-12 13:59:33 -07001699 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false)
Eric Laurent81784c32012-11-19 14:55:58 -08001700{
Glenn Kastend7dca052015-03-05 16:05:54 -08001701 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id);
1702 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
Eric Laurent81784c32012-11-19 14:55:58 -08001703
1704 // Assumes constructor is called by AudioFlinger with it's mLock held, but
1705 // it would be safer to explicitly pass initial masterVolume/masterMute as
1706 // parameter.
1707 //
1708 // If the HAL we are using has support for master volume or master mute,
1709 // then do not attenuate or mute during mixing (just leave the volume at 1.0
1710 // and the mute set to false).
1711 mMasterVolume = audioFlinger->masterVolume_l();
1712 mMasterMute = audioFlinger->masterMute_l();
1713 if (mOutput && mOutput->audioHwDev) {
1714 if (mOutput->audioHwDev->canSetMasterVolume()) {
1715 mMasterVolume = 1.0;
1716 }
1717
1718 if (mOutput->audioHwDev->canSetMasterMute()) {
1719 mMasterMute = false;
1720 }
1721 }
1722
Glenn Kastendeca2ae2014-02-07 10:25:56 -08001723 readOutputParameters_l();
Eric Laurent81784c32012-11-19 14:55:58 -08001724
Eric Laurent223fd5c2014-11-11 13:43:36 -08001725 // ++ operator does not compile
Glenn Kasten66e46352014-01-16 17:44:23 -08001726 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT;
Eric Laurent81784c32012-11-19 14:55:58 -08001727 stream = (audio_stream_type_t) (stream + 1)) {
1728 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
1729 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
1730 }
Eric Laurent81784c32012-11-19 14:55:58 -08001731}
1732
1733AudioFlinger::PlaybackThread::~PlaybackThread()
1734{
Glenn Kasten9e58b552013-01-18 15:09:48 -08001735 mAudioFlinger->unregisterWriter(mNBLogWriter);
Andy Hung010a1a12014-03-13 13:57:33 -07001736 free(mSinkBuffer);
Andy Hung69aed5f2014-02-25 17:24:40 -08001737 free(mMixerBuffer);
Andy Hung98ef9782014-03-04 14:46:50 -08001738 free(mEffectBuffer);
Eric Laurent81784c32012-11-19 14:55:58 -08001739}
1740
1741void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
1742{
1743 dumpInternals(fd, args);
1744 dumpTracks(fd, args);
1745 dumpEffectChains(fd, args);
1746}
1747
Glenn Kasten0f11b512014-01-31 16:18:54 -08001748void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08001749{
1750 const size_t SIZE = 256;
1751 char buffer[SIZE];
1752 String8 result;
1753
Marco Nelissenb2208842014-02-07 14:00:50 -08001754 result.appendFormat(" Stream volumes in dB: ");
Eric Laurent81784c32012-11-19 14:55:58 -08001755 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) {
1756 const stream_type_t *st = &mStreamTypes[i];
1757 if (i > 0) {
1758 result.appendFormat(", ");
1759 }
1760 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume));
1761 if (st->mute) {
1762 result.append("M");
1763 }
1764 }
1765 result.append("\n");
1766 write(fd, result.string(), result.length());
1767 result.clear();
1768
Eric Laurent81784c32012-11-19 14:55:58 -08001769 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way.
1770 FastTrackUnderruns underruns = getFastTrackUnderruns(0);
Elliott Hughes87cebad2014-05-22 10:14:43 -07001771 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n",
Eric Laurent81784c32012-11-19 14:55:58 -08001772 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
Marco Nelissenb2208842014-02-07 14:00:50 -08001773
1774 size_t numtracks = mTracks.size();
1775 size_t numactive = mActiveTracks.size();
Glenn Kastenc42e9b42016-03-21 11:35:03 -07001776 dprintf(fd, " %zu Tracks", numtracks);
Marco Nelissenb2208842014-02-07 14:00:50 -08001777 size_t numactiveseen = 0;
1778 if (numtracks) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07001779 dprintf(fd, " of which %zu are active\n", numactive);
Marco Nelissenb2208842014-02-07 14:00:50 -08001780 Track::appendDumpHeader(result);
1781 for (size_t i = 0; i < numtracks; ++i) {
1782 sp<Track> track = mTracks[i];
1783 if (track != 0) {
1784 bool active = mActiveTracks.indexOf(track) >= 0;
1785 if (active) {
1786 numactiveseen++;
1787 }
1788 track->dump(buffer, SIZE, active);
1789 result.append(buffer);
1790 }
1791 }
1792 } else {
1793 result.append("\n");
1794 }
1795 if (numactiveseen != numactive) {
1796 // some tracks in the active list were not in the tracks list
1797 snprintf(buffer, SIZE, " The following tracks are in the active list but"
1798 " not in the track list\n");
1799 result.append(buffer);
1800 Track::appendDumpHeader(result);
1801 for (size_t i = 0; i < numactive; ++i) {
1802 sp<Track> track = mActiveTracks[i].promote();
1803 if (track != 0 && mTracks.indexOf(track) < 0) {
1804 track->dump(buffer, SIZE, true);
1805 result.append(buffer);
1806 }
1807 }
1808 }
1809
1810 write(fd, result.string(), result.size());
Eric Laurent81784c32012-11-19 14:55:58 -08001811}
1812
1813void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
1814{
Glenn Kasten97b7b752014-09-28 13:04:24 -07001815 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type()));
Glenn Kasten44182c22015-03-05 17:12:23 -08001816
1817 dumpBase(fd, args);
1818
Elliott Hughes87cebad2014-05-22 10:14:43 -07001819 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
Glenn Kastenc42e9b42016-03-21 11:35:03 -07001820 dprintf(fd, " Last write occurred (msecs): %llu\n",
1821 (unsigned long long) ns2ms(systemTime() - mLastWriteTime));
Elliott Hughes87cebad2014-05-22 10:14:43 -07001822 dprintf(fd, " Total writes: %d\n", mNumWrites);
1823 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
1824 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
1825 dprintf(fd, " Suspend count: %d\n", mSuspended);
1826 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer);
1827 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
1828 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer);
1829 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
Eric Laurent42537be2016-01-08 17:16:42 -08001830 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs);
Glenn Kasten97b7b752014-09-28 13:04:24 -07001831 AudioStreamOut *output = mOutput;
1832 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
1833 String8 flagsAsString = outputFlagsToString(flags);
1834 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string());
Eric Laurent81784c32012-11-19 14:55:58 -08001835}
1836
1837// Thread virtuals
Eric Laurent81784c32012-11-19 14:55:58 -08001838
1839void AudioFlinger::PlaybackThread::onFirstRef()
1840{
Glenn Kastend7dca052015-03-05 16:05:54 -08001841 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
Eric Laurent81784c32012-11-19 14:55:58 -08001842}
1843
1844// ThreadBase virtuals
1845void AudioFlinger::PlaybackThread::preExit()
1846{
1847 ALOGV(" preExit()");
1848 // FIXME this is using hard-coded strings but in the future, this functionality will be
1849 // converted to use audio HAL extensions required to support tunneling
1850 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1");
1851}
1852
1853// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
1854sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
1855 const sp<AudioFlinger::Client>& client,
1856 audio_stream_type_t streamType,
1857 uint32_t sampleRate,
1858 audio_format_t format,
1859 audio_channel_mask_t channelMask,
Glenn Kasten74935e42013-12-19 08:56:45 -08001860 size_t *pFrameCount,
Eric Laurent81784c32012-11-19 14:55:58 -08001861 const sp<IMemory>& sharedBuffer,
Glenn Kastend848eb42016-03-08 13:42:11 -08001862 audio_session_t sessionId,
Eric Laurent05067782016-06-01 18:27:28 -07001863 audio_output_flags_t *flags,
Eric Laurent81784c32012-11-19 14:55:58 -08001864 pid_t tid,
Marco Nelissen462fd2f2013-01-14 14:12:05 -08001865 int uid,
Eric Laurent81784c32012-11-19 14:55:58 -08001866 status_t *status)
1867{
Glenn Kasten74935e42013-12-19 08:56:45 -08001868 size_t frameCount = *pFrameCount;
Eric Laurent81784c32012-11-19 14:55:58 -08001869 sp<Track> track;
1870 status_t lStatus;
Eric Laurent05067782016-06-01 18:27:28 -07001871 audio_output_flags_t outputFlags = mOutput->flags;
1872
1873 // special case for FAST flag considered OK if fast mixer is present
1874 if (hasFastMixer()) {
1875 outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST);
1876 }
1877
1878 // Check if requested flags are compatible with output stream flags
1879 if ((*flags & outputFlags) != *flags) {
1880 ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)",
1881 *flags, outputFlags);
1882 *flags = (audio_output_flags_t)(*flags & outputFlags);
1883 }
Eric Laurent81784c32012-11-19 14:55:58 -08001884
Eric Laurent81784c32012-11-19 14:55:58 -08001885 // client expresses a preference for FAST, but we get the final say
Eric Laurent05067782016-06-01 18:27:28 -07001886 if (*flags & AUDIO_OUTPUT_FLAG_FAST) {
Eric Laurent81784c32012-11-19 14:55:58 -08001887 if (
Eric Laurent81784c32012-11-19 14:55:58 -08001888 // PCM data
1889 audio_is_linear_pcm(format) &&
Andy Hung1f439e12015-05-19 12:57:41 -07001890 // TODO: extract as a data library function that checks that a computationally
1891 // expensive downmixer is not required: isFastOutputChannelConversion()
Andy Hung9a592762014-07-21 21:56:01 -07001892 (channelMask == mChannelMask ||
Andy Hung1f439e12015-05-19 12:57:41 -07001893 mChannelMask != AUDIO_CHANNEL_OUT_STEREO ||
1894 (channelMask == AUDIO_CHANNEL_OUT_MONO
1895 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) &&
Eric Laurent81784c32012-11-19 14:55:58 -08001896 // hardware sample rate
1897 (sampleRate == mSampleRate) &&
Eric Laurent81784c32012-11-19 14:55:58 -08001898 // normal mixer has an associated fast mixer
1899 hasFastMixer() &&
1900 // there are sufficient fast track slots available
1901 (mFastTrackAvailMask != 0)
1902 // FIXME test that MixerThread for this fast track has a capable output HAL
1903 // FIXME add a permission test also?
1904 ) {
Andy Hunge0a269a2016-03-23 15:13:42 -07001905 // static tracks can have any nonzero framecount, streaming tracks check against minimum.
1906 if (sharedBuffer == 0) {
Glenn Kasten03490092014-05-27 12:30:54 -07001907 // read the fast track multiplier property the first time it is needed
1908 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit);
1909 if (ok != 0) {
1910 ALOGE("%s pthread_once failed: %d", __func__, ok);
1911 }
Andy Hunge0a269a2016-03-23 15:13:42 -07001912 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0
Eric Laurent81784c32012-11-19 14:55:58 -08001913 }
Eric Laurent4c415062016-06-17 16:14:16 -07001914
1915 // check compatibility with audio effects.
1916 { // scope for mLock
1917 Mutex::Autolock _l(mLock);
1918 // do not accept RAW flag if post processing are present. Note that post processing on
1919 // a fast mixer are necessarily hardware
1920 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE);
1921 if (chain != 0) {
Eric Laurent122f7e72016-06-29 11:53:29 -07001922 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0,
Eric Laurent4c415062016-06-17 16:14:16 -07001923 "AUDIO_OUTPUT_FLAG_RAW denied: post processing effect present");
1924 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW);
1925 }
1926 // Do not accept FAST flag if software global effects are present
1927 chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
1928 if (chain != 0) {
Eric Laurent122f7e72016-06-29 11:53:29 -07001929 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0,
Eric Laurent4c415062016-06-17 16:14:16 -07001930 "AUDIO_OUTPUT_FLAG_RAW denied: global effect present");
1931 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW);
1932 if (chain->hasSoftwareEffect()) {
1933 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software global effect present");
1934 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
1935 }
1936 }
1937 // Do not accept FAST flag if the session has software effects
1938 chain = getEffectChain_l(sessionId);
1939 if (chain != 0) {
Eric Laurent122f7e72016-06-29 11:53:29 -07001940 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0,
Eric Laurent4c415062016-06-17 16:14:16 -07001941 "AUDIO_OUTPUT_FLAG_RAW denied: effect present on session");
1942 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW);
1943 if (chain->hasSoftwareEffect()) {
1944 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software effect present on session");
1945 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
1946 }
1947 }
1948 }
Eric Laurent122f7e72016-06-29 11:53:29 -07001949 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0,
Eric Laurent4c415062016-06-17 16:14:16 -07001950 "AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
1951 frameCount, mFrameCount);
Eric Laurent81784c32012-11-19 14:55:58 -08001952 } else {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07001953 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu "
1954 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
Andy Hung6146c082014-03-18 11:56:15 -07001955 "sampleRate=%u mSampleRate=%u "
Eric Laurent81784c32012-11-19 14:55:58 -08001956 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
Glenn Kastend79072e2016-01-06 08:41:20 -08001957 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
Eric Laurent81784c32012-11-19 14:55:58 -08001958 audio_is_linear_pcm(format),
1959 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
Eric Laurent4c415062016-06-17 16:14:16 -07001960 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST);
Andy Hung0e48d252015-01-26 11:43:15 -08001961 }
1962 }
1963 // For normal PCM streaming tracks, update minimum frame count.
1964 // For compatibility with AudioTrack calculation, buffer depth is forced
1965 // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
1966 // This is probably too conservative, but legacy application code may depend on it.
1967 // If you change this calculation, also review the start threshold which is related.
Eric Laurent05067782016-06-01 18:27:28 -07001968 if (!(*flags & AUDIO_OUTPUT_FLAG_FAST)
Phil Burkfdb3c072016-02-09 10:47:02 -08001969 && audio_has_proportional_frames(format) && sharedBuffer == 0) {
Andy Hung8edb8dc2015-03-26 19:13:55 -07001970 // this must match AudioTrack.cpp calculateMinFrameCount().
1971 // TODO: Move to a common library
Eric Laurent81784c32012-11-19 14:55:58 -08001972 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
1973 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
1974 if (minBufCount < 2) {
1975 minBufCount = 2;
1976 }
Andy Hung8edb8dc2015-03-26 19:13:55 -07001977 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack
1978 // or the client should compute and pass in a larger buffer request.
Andy Hung0e48d252015-01-26 11:43:15 -08001979 size_t minFrameCount =
Andy Hung8edb8dc2015-03-26 19:13:55 -07001980 minBufCount * sourceFramesNeededWithTimestretch(
1981 sampleRate, mNormalFrameCount,
1982 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/);
Andy Hung0e48d252015-01-26 11:43:15 -08001983 if (frameCount < minFrameCount) { // including frameCount == 0
Eric Laurent81784c32012-11-19 14:55:58 -08001984 frameCount = minFrameCount;
1985 }
Eric Laurent81784c32012-11-19 14:55:58 -08001986 }
Glenn Kasten74935e42013-12-19 08:56:45 -08001987 *pFrameCount = frameCount;
Eric Laurent81784c32012-11-19 14:55:58 -08001988
Glenn Kastenc3df8382014-03-13 15:05:25 -07001989 switch (mType) {
1990
1991 case DIRECT:
Phil Burkfdb3c072016-02-09 10:47:02 -08001992 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()?
Eric Laurent81784c32012-11-19 14:55:58 -08001993 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
Glenn Kastencac3daa2014-02-07 09:47:14 -08001994 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x "
1995 "for output %p with format %#x",
Eric Laurent81784c32012-11-19 14:55:58 -08001996 sampleRate, format, channelMask, mOutput, mFormat);
1997 lStatus = BAD_VALUE;
1998 goto Exit;
1999 }
2000 }
Glenn Kastenc3df8382014-03-13 15:05:25 -07002001 break;
2002
2003 case OFFLOAD:
Eric Laurentbfb1b832013-01-07 09:53:42 -08002004 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
Glenn Kastencac3daa2014-02-07 09:47:14 -08002005 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \""
2006 "for output %p with format %#x",
Eric Laurentbfb1b832013-01-07 09:53:42 -08002007 sampleRate, format, channelMask, mOutput, mFormat);
2008 lStatus = BAD_VALUE;
2009 goto Exit;
2010 }
Glenn Kastenc3df8382014-03-13 15:05:25 -07002011 break;
2012
2013 default:
Glenn Kasten993fa062014-05-02 11:14:34 -07002014 if (!audio_is_linear_pcm(format)) {
Glenn Kastencac3daa2014-02-07 09:47:14 -08002015 ALOGE("createTrack_l() Bad parameter: format %#x \""
2016 "for output %p with format %#x",
Eric Laurentbfb1b832013-01-07 09:53:42 -08002017 format, mOutput, mFormat);
2018 lStatus = BAD_VALUE;
2019 goto Exit;
2020 }
Andy Hungcd044842014-08-07 11:04:34 -07002021 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
Eric Laurent81784c32012-11-19 14:55:58 -08002022 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate);
2023 lStatus = BAD_VALUE;
2024 goto Exit;
2025 }
Glenn Kastenc3df8382014-03-13 15:05:25 -07002026 break;
2027
Eric Laurent81784c32012-11-19 14:55:58 -08002028 }
2029
2030 lStatus = initCheck();
2031 if (lStatus != NO_ERROR) {
Glenn Kasten15e57982013-09-24 11:52:37 -07002032 ALOGE("createTrack_l() audio driver not initialized");
Eric Laurent81784c32012-11-19 14:55:58 -08002033 goto Exit;
2034 }
2035
2036 { // scope for mLock
2037 Mutex::Autolock _l(mLock);
2038
2039 // all tracks in same audio session must share the same routing strategy otherwise
2040 // conflicts will happen when tracks are moved from one output to another by audio policy
2041 // manager
2042 uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
2043 for (size_t i = 0; i < mTracks.size(); ++i) {
2044 sp<Track> t = mTracks[i];
Eric Laurent83b88082014-06-20 18:31:16 -07002045 if (t != 0 && t->isExternalTrack()) {
Eric Laurent81784c32012-11-19 14:55:58 -08002046 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
2047 if (sessionId == t->sessionId() && strategy != actual) {
2048 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
2049 strategy, actual);
2050 lStatus = BAD_VALUE;
2051 goto Exit;
2052 }
2053 }
2054 }
2055
Glenn Kastend79072e2016-01-06 08:41:20 -08002056 track = new Track(this, client, streamType, sampleRate, format,
2057 channelMask, frameCount, NULL, sharedBuffer,
2058 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT);
Glenn Kasten03003332013-08-06 15:40:54 -07002059
Glenn Kasten03003332013-08-06 15:40:54 -07002060 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
2061 if (lStatus != NO_ERROR) {
Glenn Kasten0cde0762014-01-16 15:06:36 -08002062 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
Haynes Mathew George03e9e832013-12-13 15:40:13 -08002063 // track must be cleared from the caller as the caller has the AF lock
Eric Laurent81784c32012-11-19 14:55:58 -08002064 goto Exit;
2065 }
2066 mTracks.add(track);
2067
2068 sp<EffectChain> chain = getEffectChain_l(sessionId);
2069 if (chain != 0) {
2070 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
2071 track->setMainBuffer(chain->inBuffer());
2072 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
2073 chain->incTrackCnt();
2074 }
2075
Eric Laurent05067782016-06-01 18:27:28 -07002076 if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) {
Eric Laurent81784c32012-11-19 14:55:58 -08002077 pid_t callingPid = IPCThreadState::self()->getCallingPid();
2078 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
2079 // so ask activity manager to do this on our behalf
2080 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
2081 }
2082 }
2083
2084 lStatus = NO_ERROR;
2085
2086Exit:
Glenn Kasten9156ef32013-08-06 15:39:08 -07002087 *status = lStatus;
Eric Laurent81784c32012-11-19 14:55:58 -08002088 return track;
2089}
2090
2091uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
2092{
2093 return latency;
2094}
2095
2096uint32_t AudioFlinger::PlaybackThread::latency() const
2097{
2098 Mutex::Autolock _l(mLock);
2099 return latency_l();
2100}
2101uint32_t AudioFlinger::PlaybackThread::latency_l() const
2102{
2103 if (initCheck() == NO_ERROR) {
2104 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream));
2105 } else {
2106 return 0;
2107 }
2108}
2109
2110void AudioFlinger::PlaybackThread::setMasterVolume(float value)
2111{
2112 Mutex::Autolock _l(mLock);
2113 // Don't apply master volume in SW if our HAL can do it for us.
2114 if (mOutput && mOutput->audioHwDev &&
2115 mOutput->audioHwDev->canSetMasterVolume()) {
2116 mMasterVolume = 1.0;
2117 } else {
2118 mMasterVolume = value;
2119 }
2120}
2121
2122void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
2123{
2124 Mutex::Autolock _l(mLock);
2125 // Don't apply master mute in SW if our HAL can do it for us.
2126 if (mOutput && mOutput->audioHwDev &&
2127 mOutput->audioHwDev->canSetMasterMute()) {
2128 mMasterMute = false;
2129 } else {
2130 mMasterMute = muted;
2131 }
2132}
2133
2134void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
2135{
2136 Mutex::Autolock _l(mLock);
2137 mStreamTypes[stream].volume = value;
Eric Laurentede6c3b2013-09-19 14:37:46 -07002138 broadcast_l();
Eric Laurent81784c32012-11-19 14:55:58 -08002139}
2140
2141void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
2142{
2143 Mutex::Autolock _l(mLock);
2144 mStreamTypes[stream].mute = muted;
Eric Laurentede6c3b2013-09-19 14:37:46 -07002145 broadcast_l();
Eric Laurent81784c32012-11-19 14:55:58 -08002146}
2147
2148float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
2149{
2150 Mutex::Autolock _l(mLock);
2151 return mStreamTypes[stream].volume;
2152}
2153
2154// addTrack_l() must be called with ThreadBase::mLock held
2155status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
2156{
2157 status_t status = ALREADY_EXISTS;
2158
Eric Laurent81784c32012-11-19 14:55:58 -08002159 if (mActiveTracks.indexOf(track) < 0) {
2160 // the track is newly added, make sure it fills up all its
2161 // buffers before playing. This is to ensure the client will
2162 // effectively get the latency it requested.
Eric Laurent83b88082014-06-20 18:31:16 -07002163 if (track->isExternalTrack()) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08002164 TrackBase::track_state state = track->mState;
2165 mLock.unlock();
Eric Laurente83b55d2014-11-14 10:06:21 -08002166 status = AudioSystem::startOutput(mId, track->streamType(),
Glenn Kastend848eb42016-03-08 13:42:11 -08002167 track->sessionId());
Eric Laurentbfb1b832013-01-07 09:53:42 -08002168 mLock.lock();
2169 // abort track was stopped/paused while we released the lock
2170 if (state != track->mState) {
2171 if (status == NO_ERROR) {
2172 mLock.unlock();
Eric Laurente83b55d2014-11-14 10:06:21 -08002173 AudioSystem::stopOutput(mId, track->streamType(),
Glenn Kastend848eb42016-03-08 13:42:11 -08002174 track->sessionId());
Eric Laurentbfb1b832013-01-07 09:53:42 -08002175 mLock.lock();
2176 }
2177 return INVALID_OPERATION;
2178 }
2179 // abort if start is rejected by audio policy manager
2180 if (status != NO_ERROR) {
2181 return PERMISSION_DENIED;
2182 }
2183#ifdef ADD_BATTERY_DATA
2184 // to track the speaker usage
2185 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
2186#endif
2187 }
2188
Eric Laurent51716182016-02-29 18:00:56 -08002189 // set retry count for buffer fill
2190 if (track->isOffloaded()) {
Eric Laurente93cc032016-05-05 10:15:10 -07002191 if (track->isStopping_1()) {
2192 track->mRetryCount = kMaxTrackStopRetriesOffload;
2193 } else {
2194 track->mRetryCount = kMaxTrackStartupRetriesOffload;
2195 }
2196 track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED;
Eric Laurent51716182016-02-29 18:00:56 -08002197 } else {
2198 track->mRetryCount = kMaxTrackStartupRetries;
Eric Laurente93cc032016-05-05 10:15:10 -07002199 track->mFillingUpStatus =
2200 track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
Eric Laurent51716182016-02-29 18:00:56 -08002201 }
2202
Eric Laurent81784c32012-11-19 14:55:58 -08002203 track->mResetDone = false;
2204 track->mPresentationCompleteFrames = 0;
2205 mActiveTracks.add(track);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08002206 mWakeLockUids.add(track->uid());
2207 mActiveTracksGeneration++;
Eric Laurentfd477972013-10-25 18:10:40 -07002208 mLatestActiveTrack = track;
Eric Laurentd0107bc2013-06-11 14:38:48 -07002209 sp<EffectChain> chain = getEffectChain_l(track->sessionId());
2210 if (chain != 0) {
2211 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
2212 track->sessionId());
2213 chain->incActiveTrackCnt();
Eric Laurent81784c32012-11-19 14:55:58 -08002214 }
2215
2216 status = NO_ERROR;
2217 }
2218
Haynes Mathew George4c6a4332014-01-15 12:31:39 -08002219 onAddNewTrack_l();
Eric Laurent81784c32012-11-19 14:55:58 -08002220 return status;
2221}
2222
Eric Laurentbfb1b832013-01-07 09:53:42 -08002223bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
Eric Laurent81784c32012-11-19 14:55:58 -08002224{
Eric Laurentbfb1b832013-01-07 09:53:42 -08002225 track->terminate();
Eric Laurent81784c32012-11-19 14:55:58 -08002226 // active tracks are removed by threadLoop()
Eric Laurentbfb1b832013-01-07 09:53:42 -08002227 bool trackActive = (mActiveTracks.indexOf(track) >= 0);
2228 track->mState = TrackBase::STOPPED;
2229 if (!trackActive) {
Eric Laurent81784c32012-11-19 14:55:58 -08002230 removeTrack_l(track);
Eric Laurentab5cdba2014-06-09 17:22:27 -07002231 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08002232 track->mState = TrackBase::STOPPING_1;
Eric Laurent81784c32012-11-19 14:55:58 -08002233 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08002234
2235 return trackActive;
Eric Laurent81784c32012-11-19 14:55:58 -08002236}
2237
2238void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
2239{
2240 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
2241 mTracks.remove(track);
2242 deleteTrackName_l(track->name());
2243 // redundant as track is about to be destroyed, for dumpsys only
2244 track->mName = -1;
2245 if (track->isFastTrack()) {
2246 int index = track->mFastIndex;
Glenn Kastendc2c50b2016-04-21 08:13:14 -07002247 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks);
Eric Laurent81784c32012-11-19 14:55:58 -08002248 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index)));
2249 mFastTrackAvailMask |= 1 << index;
2250 // redundant as track is about to be destroyed, for dumpsys only
2251 track->mFastIndex = -1;
2252 }
2253 sp<EffectChain> chain = getEffectChain_l(track->sessionId());
2254 if (chain != 0) {
2255 chain->decTrackCnt();
2256 }
2257}
2258
Eric Laurentede6c3b2013-09-19 14:37:46 -07002259void AudioFlinger::PlaybackThread::broadcast_l()
Eric Laurentbfb1b832013-01-07 09:53:42 -08002260{
2261 // Thread could be blocked waiting for async
2262 // so signal it to handle state changes immediately
2263 // If threadLoop is currently unlocked a signal of mWaitWorkCV will
2264 // be lost so we also flag to prevent it blocking on mWaitWorkCV
2265 mSignalPending = true;
Eric Laurentede6c3b2013-09-19 14:37:46 -07002266 mWaitWorkCV.broadcast();
Eric Laurentbfb1b832013-01-07 09:53:42 -08002267}
2268
Eric Laurent81784c32012-11-19 14:55:58 -08002269String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
2270{
Eric Laurent81784c32012-11-19 14:55:58 -08002271 Mutex::Autolock _l(mLock);
2272 if (initCheck() != NO_ERROR) {
Glenn Kastend8ea6992013-07-16 14:17:15 -07002273 return String8();
Eric Laurent81784c32012-11-19 14:55:58 -08002274 }
2275
Glenn Kastend8ea6992013-07-16 14:17:15 -07002276 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
2277 const String8 out_s8(s);
Eric Laurent81784c32012-11-19 14:55:58 -08002278 free(s);
2279 return out_s8;
2280}
2281
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07002282void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) {
Eric Laurent73e26b62015-04-27 16:55:58 -07002283 sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
2284 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);
Eric Laurent81784c32012-11-19 14:55:58 -08002285
Eric Laurent73e26b62015-04-27 16:55:58 -07002286 desc->mIoHandle = mId;
Eric Laurent81784c32012-11-19 14:55:58 -08002287
2288 switch (event) {
Eric Laurent73e26b62015-04-27 16:55:58 -07002289 case AUDIO_OUTPUT_OPENED:
2290 case AUDIO_OUTPUT_CONFIG_CHANGED:
Eric Laurent296fb132015-05-01 11:38:42 -07002291 desc->mPatch = mPatch;
Eric Laurent73e26b62015-04-27 16:55:58 -07002292 desc->mChannelMask = mChannelMask;
2293 desc->mSamplingRate = mSampleRate;
2294 desc->mFormat = mFormat;
2295 desc->mFrameCount = mNormalFrameCount; // FIXME see
Eric Laurent81784c32012-11-19 14:55:58 -08002296 // AudioFlinger::frameCount(audio_io_handle_t)
Glenn Kasten4a8308b2016-04-18 14:10:01 -07002297 desc->mFrameCountHAL = mFrameCount;
Eric Laurent73e26b62015-04-27 16:55:58 -07002298 desc->mLatency = latency_l();
Eric Laurent81784c32012-11-19 14:55:58 -08002299 break;
2300
Eric Laurent73e26b62015-04-27 16:55:58 -07002301 case AUDIO_OUTPUT_CLOSED:
Eric Laurent81784c32012-11-19 14:55:58 -08002302 default:
2303 break;
2304 }
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07002305 mAudioFlinger->ioConfigChanged(event, desc, pid);
Eric Laurent81784c32012-11-19 14:55:58 -08002306}
2307
Eric Laurentbfb1b832013-01-07 09:53:42 -08002308void AudioFlinger::PlaybackThread::writeCallback()
2309{
2310 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002311 mCallbackThread->resetWriteBlocked();
Eric Laurentbfb1b832013-01-07 09:53:42 -08002312}
2313
2314void AudioFlinger::PlaybackThread::drainCallback()
2315{
2316 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002317 mCallbackThread->resetDraining();
Eric Laurentbfb1b832013-01-07 09:53:42 -08002318}
2319
Haynes Mathew George4527b9e2016-07-07 19:54:17 -07002320void AudioFlinger::PlaybackThread::errorCallback()
2321{
2322 ALOG_ASSERT(mCallbackThread != 0);
2323 mCallbackThread->setAsyncError();
2324}
2325
Eric Laurent3b4529e2013-09-05 18:09:19 -07002326void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08002327{
2328 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002329 // reject out of sequence requests
2330 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) {
2331 mWriteAckSequence &= ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002332 mWaitWorkCV.signal();
2333 }
2334}
2335
Eric Laurent3b4529e2013-09-05 18:09:19 -07002336void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08002337{
2338 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002339 // reject out of sequence requests
2340 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) {
2341 mDrainSequence &= ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002342 mWaitWorkCV.signal();
2343 }
2344}
2345
2346// static
2347int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
Glenn Kasten0f11b512014-01-31 16:18:54 -08002348 void *param __unused,
Eric Laurentbfb1b832013-01-07 09:53:42 -08002349 void *cookie)
2350{
2351 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie;
2352 ALOGV("asyncCallback() event %d", event);
2353 switch (event) {
2354 case STREAM_CBK_EVENT_WRITE_READY:
2355 me->writeCallback();
2356 break;
2357 case STREAM_CBK_EVENT_DRAIN_READY:
2358 me->drainCallback();
2359 break;
Haynes Mathew George4527b9e2016-07-07 19:54:17 -07002360 case STREAM_CBK_EVENT_ERROR:
2361 me->errorCallback();
2362 break;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002363 default:
2364 ALOGW("asyncCallback() unknown event %d", event);
2365 break;
2366 }
2367 return 0;
2368}
2369
Glenn Kastendeca2ae2014-02-07 10:25:56 -08002370void AudioFlinger::PlaybackThread::readOutputParameters_l()
Eric Laurent81784c32012-11-19 14:55:58 -08002371{
Glenn Kastenadad3d72014-02-21 14:51:43 -08002372 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
Phil Burkca5e6142015-07-14 09:42:29 -07002373 mSampleRate = mOutput->getSampleRate();
2374 mChannelMask = mOutput->getChannelMask();
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002375 if (!audio_is_output_channel(mChannelMask)) {
Glenn Kastenadad3d72014-02-21 14:51:43 -08002376 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002377 }
Andy Hung9a592762014-07-21 21:56:01 -07002378 if ((mType == MIXER || mType == DUPLICATING)
2379 && !isValidPcmSinkChannelMask(mChannelMask)) {
2380 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
2381 mChannelMask);
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002382 }
Andy Hunge5412692014-05-16 11:25:07 -07002383 mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
Phil Burkca5e6142015-07-14 09:42:29 -07002384
2385 // Get actual HAL format.
Andy Hung463be252014-07-10 16:56:07 -07002386 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
Phil Burkca5e6142015-07-14 09:42:29 -07002387 // Get format from the shim, which will be different than the HAL format
2388 // if playing compressed audio over HDMI passthrough.
2389 mFormat = mOutput->getFormat();
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002390 if (!audio_is_valid_format(mFormat)) {
Glenn Kastenadad3d72014-02-21 14:51:43 -08002391 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002392 }
Andy Hung6146c082014-03-18 11:56:15 -07002393 if ((mType == MIXER || mType == DUPLICATING)
2394 && !isValidPcmSinkFormat(mFormat)) {
2395 LOG_FATAL("HAL format %#x not supported for mixed output",
2396 mFormat);
Glenn Kasten7fc97ba2013-07-16 17:18:58 -07002397 }
Phil Burk062e67a2015-02-11 13:40:50 -08002398 mFrameSize = mOutput->getFrameSize();
Glenn Kasten70949c42013-08-06 07:40:12 -07002399 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
2400 mFrameCount = mBufferSize / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08002401 if (mFrameCount & 15) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07002402 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
Eric Laurent81784c32012-11-19 14:55:58 -08002403 mFrameCount);
2404 }
2405
Eric Laurentbfb1b832013-01-07 09:53:42 -08002406 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) &&
2407 (mOutput->stream->set_callback != NULL)) {
2408 if (mOutput->stream->set_callback(mOutput->stream,
2409 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) {
2410 mUseAsyncWrite = true;
Eric Laurent4de95592013-09-26 15:28:21 -07002411 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002412 }
2413 }
2414
Eric Laurentd1f69b02014-12-15 14:33:13 -08002415 mHwSupportsPause = false;
2416 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) {
2417 if (mOutput->stream->pause != NULL) {
2418 if (mOutput->stream->resume != NULL) {
2419 mHwSupportsPause = true;
2420 } else {
2421 ALOGW("direct output implements pause but not resume");
2422 }
2423 } else if (mOutput->stream->resume != NULL) {
2424 ALOGW("direct output implements resume but not pause");
2425 }
2426 }
Phil Burk6fc2a7c2015-04-30 16:08:10 -07002427 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) {
2428 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume");
2429 }
Eric Laurentd1f69b02014-12-15 14:33:13 -08002430
Andy Hungfbfc3952015-01-15 13:33:51 -08002431 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) {
2432 // For best precision, we use float instead of the associated output
2433 // device format (typically PCM 16 bit).
2434
2435 mFormat = AUDIO_FORMAT_PCM_FLOAT;
2436 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
2437 mBufferSize = mFrameSize * mFrameCount;
2438
2439 // TODO: We currently use the associated output device channel mask and sample rate.
2440 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads
2441 // (if a valid mask) to avoid premature downmix.
2442 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads
2443 // instead of the output device sample rate to avoid loss of high frequency information.
2444 // This may need to be updated as MixerThread/OutputTracks are added and not here.
2445 }
2446
Andy Hung09a50072014-02-27 14:30:47 -08002447 // Calculate size of normal sink buffer relative to the HAL output buffer size
Eric Laurent81784c32012-11-19 14:55:58 -08002448 double multiplier = 1.0;
2449 if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
2450 kUseFastMixer == FastMixer_Dynamic)) {
Andy Hung09a50072014-02-27 14:30:47 -08002451 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000;
2452 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000;
Haynes Mathew George227a14b2016-05-09 12:45:48 -07002453
Eric Laurent81784c32012-11-19 14:55:58 -08002454 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
2455 minNormalFrameCount = (minNormalFrameCount + 15) & ~15;
2456 maxNormalFrameCount = maxNormalFrameCount & ~15;
2457 if (maxNormalFrameCount < minNormalFrameCount) {
2458 maxNormalFrameCount = minNormalFrameCount;
2459 }
2460 multiplier = (double) minNormalFrameCount / (double) mFrameCount;
2461 if (multiplier <= 1.0) {
2462 multiplier = 1.0;
2463 } else if (multiplier <= 2.0) {
2464 if (2 * mFrameCount <= maxNormalFrameCount) {
2465 multiplier = 2.0;
2466 } else {
2467 multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
2468 }
2469 } else {
Haynes Mathew George227a14b2016-05-09 12:45:48 -07002470 multiplier = floor(multiplier);
Eric Laurent81784c32012-11-19 14:55:58 -08002471 }
2472 }
2473 mNormalFrameCount = multiplier * mFrameCount;
2474 // round up to nearest 16 frames to satisfy AudioMixer
Eric Laurentab5cdba2014-06-09 17:22:27 -07002475 if (mType == MIXER || mType == DUPLICATING) {
2476 mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
2477 }
Glenn Kastenc42e9b42016-03-21 11:35:03 -07002478 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
Eric Laurent81784c32012-11-19 14:55:58 -08002479 mNormalFrameCount);
2480
Andy Hung08fb1742015-05-31 23:22:10 -07002481 // Check if we want to throttle the processing to no more than 2x normal rate
2482 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */);
Andy Hung40eb1a12015-06-18 13:42:02 -07002483 mThreadThrottleTimeMs = 0;
2484 mThreadThrottleEndMs = 0;
Andy Hung08fb1742015-05-31 23:22:10 -07002485 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate);
2486
Andy Hung010a1a12014-03-13 13:57:33 -07002487 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames.
2488 // Originally this was int16_t[] array, need to remove legacy implications.
2489 free(mSinkBuffer);
2490 mSinkBuffer = NULL;
Andy Hung5b10a202014-03-13 13:59:29 -07002491 // For sink buffer size, we use the frame size from the downstream sink to avoid problems
2492 // with non PCM formats for compressed music, e.g. AAC, and Offload threads.
2493 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
Andy Hung010a1a12014-03-13 13:57:33 -07002494 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
Eric Laurent81784c32012-11-19 14:55:58 -08002495
Andy Hung69aed5f2014-02-25 17:24:40 -08002496 // We resize the mMixerBuffer according to the requirements of the sink buffer which
2497 // drives the output.
2498 free(mMixerBuffer);
2499 mMixerBuffer = NULL;
2500 if (mMixerBufferEnabled) {
2501 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT.
2502 mMixerBufferSize = mNormalFrameCount * mChannelCount
2503 * audio_bytes_per_sample(mMixerBufferFormat);
2504 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
2505 }
Andy Hung98ef9782014-03-04 14:46:50 -08002506 free(mEffectBuffer);
2507 mEffectBuffer = NULL;
2508 if (mEffectBufferEnabled) {
2509 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only
2510 mEffectBufferSize = mNormalFrameCount * mChannelCount
2511 * audio_bytes_per_sample(mEffectBufferFormat);
2512 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
2513 }
Andy Hung69aed5f2014-02-25 17:24:40 -08002514
Eric Laurent81784c32012-11-19 14:55:58 -08002515 // force reconfiguration of effect chains and engines to take new buffer size and audio
2516 // parameters into account
Glenn Kastendeca2ae2014-02-07 10:25:56 -08002517 // Note that mLock is not held when readOutputParameters_l() is called from the constructor
Eric Laurent81784c32012-11-19 14:55:58 -08002518 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't
2519 // matter.
2520 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
2521 Vector< sp<EffectChain> > effectChains = mEffectChains;
2522 for (size_t i = 0; i < effectChains.size(); i ++) {
2523 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
2524 }
2525}
2526
2527
Kévin PETIT377b2ec2014-02-03 12:35:36 +00002528status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
Eric Laurent81784c32012-11-19 14:55:58 -08002529{
2530 if (halFrames == NULL || dspFrames == NULL) {
2531 return BAD_VALUE;
2532 }
2533 Mutex::Autolock _l(mLock);
2534 if (initCheck() != NO_ERROR) {
2535 return INVALID_OPERATION;
2536 }
Andy Hung818e7a32016-02-16 18:08:07 -08002537 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08002538 *halFrames = framesWritten;
2539
2540 if (isSuspended()) {
2541 // return an estimation of rendered frames when the output is suspended
2542 size_t latencyFrames = (latency_l() * mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08002543 *dspFrames = (uint32_t)
2544 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0);
Eric Laurent81784c32012-11-19 14:55:58 -08002545 return NO_ERROR;
2546 } else {
Kévin PETIT377b2ec2014-02-03 12:35:36 +00002547 status_t status;
2548 uint32_t frames;
Phil Burk062e67a2015-02-11 13:40:50 -08002549 status = mOutput->getRenderPosition(&frames);
Kévin PETIT377b2ec2014-02-03 12:35:36 +00002550 *dspFrames = (size_t)frames;
2551 return status;
Eric Laurent81784c32012-11-19 14:55:58 -08002552 }
2553}
2554
Eric Laurent4c415062016-06-17 16:14:16 -07002555// hasAudioSession_l() must be called with ThreadBase::mLock held
2556uint32_t AudioFlinger::PlaybackThread::hasAudioSession_l(audio_session_t sessionId) const
Eric Laurent81784c32012-11-19 14:55:58 -08002557{
Eric Laurent81784c32012-11-19 14:55:58 -08002558 uint32_t result = 0;
2559 if (getEffectChain_l(sessionId) != 0) {
2560 result = EFFECT_SESSION;
2561 }
2562
2563 for (size_t i = 0; i < mTracks.size(); ++i) {
2564 sp<Track> track = mTracks[i];
Glenn Kasten5736c352012-12-04 12:12:34 -08002565 if (sessionId == track->sessionId() && !track->isInvalid()) {
Eric Laurent81784c32012-11-19 14:55:58 -08002566 result |= TRACK_SESSION;
Eric Laurent4c415062016-06-17 16:14:16 -07002567 if (track->isFastTrack()) {
2568 result |= FAST_SESSION;
2569 }
Eric Laurent81784c32012-11-19 14:55:58 -08002570 break;
2571 }
2572 }
2573
2574 return result;
2575}
2576
Glenn Kastend848eb42016-03-08 13:42:11 -08002577uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08002578{
2579 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
2580 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
2581 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
2582 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
2583 }
2584 for (size_t i = 0; i < mTracks.size(); i++) {
2585 sp<Track> track = mTracks[i];
Glenn Kasten5736c352012-12-04 12:12:34 -08002586 if (sessionId == track->sessionId() && !track->isInvalid()) {
Eric Laurent81784c32012-11-19 14:55:58 -08002587 return AudioSystem::getStrategyForStream(track->streamType());
2588 }
2589 }
2590 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
2591}
2592
2593
Phil Burk062e67a2015-02-11 13:40:50 -08002594AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
Eric Laurent81784c32012-11-19 14:55:58 -08002595{
2596 Mutex::Autolock _l(mLock);
2597 return mOutput;
2598}
2599
Phil Burk062e67a2015-02-11 13:40:50 -08002600AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
Eric Laurent81784c32012-11-19 14:55:58 -08002601{
2602 Mutex::Autolock _l(mLock);
2603 AudioStreamOut *output = mOutput;
2604 mOutput = NULL;
2605 // FIXME FastMixer might also have a raw ptr to mOutputSink;
2606 // must push a NULL and wait for ack
2607 mOutputSink.clear();
2608 mPipeSink.clear();
2609 mNormalSink.clear();
2610 return output;
2611}
2612
2613// this method must always be called either with ThreadBase mLock held or inside the thread loop
2614audio_stream_t* AudioFlinger::PlaybackThread::stream() const
2615{
2616 if (mOutput == NULL) {
2617 return NULL;
2618 }
2619 return &mOutput->stream->common;
2620}
2621
2622uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
2623{
2624 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
2625}
2626
2627status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
2628{
2629 if (!isValidSyncEvent(event)) {
2630 return BAD_VALUE;
2631 }
2632
2633 Mutex::Autolock _l(mLock);
2634
2635 for (size_t i = 0; i < mTracks.size(); ++i) {
2636 sp<Track> track = mTracks[i];
2637 if (event->triggerSession() == track->sessionId()) {
2638 (void) track->setSyncEvent(event);
2639 return NO_ERROR;
2640 }
2641 }
2642
2643 return NAME_NOT_FOUND;
2644}
2645
2646bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
2647{
2648 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
2649}
2650
2651void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
2652 const Vector< sp<Track> >& tracksToRemove)
2653{
2654 size_t count = tracksToRemove.size();
Glenn Kasten34fca342013-08-13 09:48:14 -07002655 if (count > 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08002656 for (size_t i = 0 ; i < count ; i++) {
2657 const sp<Track>& track = tracksToRemove.itemAt(i);
Eric Laurent83b88082014-06-20 18:31:16 -07002658 if (track->isExternalTrack()) {
Eric Laurente83b55d2014-11-14 10:06:21 -08002659 AudioSystem::stopOutput(mId, track->streamType(),
Glenn Kastend848eb42016-03-08 13:42:11 -08002660 track->sessionId());
Eric Laurentbfb1b832013-01-07 09:53:42 -08002661#ifdef ADD_BATTERY_DATA
2662 // to track the speaker usage
2663 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
2664#endif
2665 if (track->isTerminated()) {
Eric Laurente83b55d2014-11-14 10:06:21 -08002666 AudioSystem::releaseOutput(mId, track->streamType(),
Glenn Kastend848eb42016-03-08 13:42:11 -08002667 track->sessionId());
Eric Laurentbfb1b832013-01-07 09:53:42 -08002668 }
Eric Laurent81784c32012-11-19 14:55:58 -08002669 }
2670 }
2671 }
Eric Laurent81784c32012-11-19 14:55:58 -08002672}
2673
2674void AudioFlinger::PlaybackThread::checkSilentMode_l()
2675{
2676 if (!mMasterMute) {
2677 char value[PROPERTY_VALUE_MAX];
Jean-Michel Trivi32f37c22016-03-31 16:00:32 -07002678 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
2679 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX");
2680 return;
2681 }
Eric Laurent81784c32012-11-19 14:55:58 -08002682 if (property_get("ro.audio.silent", value, "0") > 0) {
2683 char *endptr;
2684 unsigned long ul = strtoul(value, &endptr, 0);
2685 if (*endptr == '\0' && ul != 0) {
2686 ALOGD("Silence is golden");
2687 // The setprop command will not allow a property to be changed after
2688 // the first time it is set, so we don't have to worry about un-muting.
2689 setMasterMute_l(true);
2690 }
2691 }
2692 }
2693}
2694
2695// shared by MIXER and DIRECT, overridden by DUPLICATING
Eric Laurentbfb1b832013-01-07 09:53:42 -08002696ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
Eric Laurent81784c32012-11-19 14:55:58 -08002697{
Eric Laurent81784c32012-11-19 14:55:58 -08002698 mInWrite = true;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002699 ssize_t bytesWritten;
Andy Hung010a1a12014-03-13 13:57:33 -07002700 const size_t offset = mCurrentWriteLength - mBytesRemaining;
Eric Laurent81784c32012-11-19 14:55:58 -08002701
2702 // If an NBAIO sink is present, use it to write the normal mixer's submix
2703 if (mNormalSink != 0) {
Glenn Kasten4c053ea2014-09-28 14:41:07 -07002704
Andy Hung010a1a12014-03-13 13:57:33 -07002705 const size_t count = mBytesRemaining / mFrameSize;
2706
Simon Wilson2d590962012-11-29 15:18:50 -08002707 ATRACE_BEGIN("write");
Eric Laurent81784c32012-11-19 14:55:58 -08002708 // update the setpoint when AudioFlinger::mScreenState changes
2709 uint32_t screenState = AudioFlinger::mScreenState;
2710 if (screenState != mScreenState) {
2711 mScreenState = screenState;
2712 MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
2713 if (pipe != NULL) {
2714 pipe->setAvgFrames((mScreenState & 1) ?
2715 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
2716 }
2717 }
Andy Hung010a1a12014-03-13 13:57:33 -07002718 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count);
Simon Wilson2d590962012-11-29 15:18:50 -08002719 ATRACE_END();
Eric Laurent81784c32012-11-19 14:55:58 -08002720 if (framesWritten > 0) {
Andy Hung010a1a12014-03-13 13:57:33 -07002721 bytesWritten = framesWritten * mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08002722 } else {
2723 bytesWritten = framesWritten;
2724 }
2725 // otherwise use the HAL / AudioStreamOut directly
2726 } else {
Eric Laurentbfb1b832013-01-07 09:53:42 -08002727 // Direct output and offload threads
Andy Hung010a1a12014-03-13 13:57:33 -07002728
Eric Laurentbfb1b832013-01-07 09:53:42 -08002729 if (mUseAsyncWrite) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07002730 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request");
2731 mWriteAckSequence += 2;
2732 mWriteAckSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002733 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002734 mCallbackThread->setWriteBlocked(mWriteAckSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002735 }
Glenn Kasten767094d2013-08-23 13:51:43 -07002736 // FIXME We should have an implementation of timestamps for direct output threads.
2737 // They are used e.g for multichannel PCM playback over HDMI.
Phil Burk062e67a2015-02-11 13:40:50 -08002738 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
Eric Laurent51716182016-02-29 18:00:56 -08002739
Eric Laurentbfb1b832013-01-07 09:53:42 -08002740 if (mUseAsyncWrite &&
2741 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
2742 // do not wait for async callback in case of error of full write
Eric Laurent3b4529e2013-09-05 18:09:19 -07002743 mWriteAckSequence &= ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002744 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002745 mCallbackThread->setWriteBlocked(mWriteAckSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002746 }
Eric Laurent81784c32012-11-19 14:55:58 -08002747 }
2748
Eric Laurent81784c32012-11-19 14:55:58 -08002749 mNumWrites++;
2750 mInWrite = false;
Eric Laurentfd477972013-10-25 18:10:40 -07002751 mStandby = false;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002752 return bytesWritten;
2753}
2754
2755void AudioFlinger::PlaybackThread::threadLoop_drain()
2756{
2757 if (mOutput->stream->drain) {
2758 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full");
2759 if (mUseAsyncWrite) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07002760 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request");
2761 mDrainSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08002762 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07002763 mCallbackThread->setDraining(mDrainSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08002764 }
2765 mOutput->stream->drain(mOutput->stream,
2766 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY
2767 : AUDIO_DRAIN_ALL);
2768 }
2769}
2770
2771void AudioFlinger::PlaybackThread::threadLoop_exit()
2772{
Eric Laurent275e8e92014-11-30 15:14:47 -08002773 {
2774 Mutex::Autolock _l(mLock);
2775 for (size_t i = 0; i < mTracks.size(); i++) {
2776 sp<Track> track = mTracks[i];
2777 track->invalidate();
2778 }
2779 }
Eric Laurent81784c32012-11-19 14:55:58 -08002780}
2781
2782/*
2783The derived values that are cached:
Andy Hung25c2dac2014-02-27 14:56:00 -08002784 - mSinkBufferSize from frame count * frame size
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002785 - mActiveSleepTimeUs from activeSleepTimeUs()
2786 - mIdleSleepTimeUs from idleSleepTimeUs()
Eric Laurent42537be2016-01-08 17:16:42 -08002787 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least
2788 kDefaultStandbyTimeInNsecs when connected to an A2DP device.
Eric Laurent81784c32012-11-19 14:55:58 -08002789 - maxPeriod from frame count and sample rate (MIXER only)
2790
2791The parameters that affect these derived values are:
2792 - frame count
2793 - frame size
2794 - sample rate
2795 - device type: A2DP or not
2796 - device latency
2797 - format: PCM or not
2798 - active sleep time
2799 - idle sleep time
2800*/
2801
2802void AudioFlinger::PlaybackThread::cacheParameters_l()
2803{
Andy Hung25c2dac2014-02-27 14:56:00 -08002804 mSinkBufferSize = mNormalFrameCount * mFrameSize;
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002805 mActiveSleepTimeUs = activeSleepTimeUs();
2806 mIdleSleepTimeUs = idleSleepTimeUs();
Eric Laurent42537be2016-01-08 17:16:42 -08002807
2808 // make sure standby delay is not too short when connected to an A2DP sink to avoid
2809 // truncating audio when going to standby.
2810 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs;
2811 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) {
2812 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) {
2813 mStandbyDelayNs = kDefaultStandbyTimeInNsecs;
2814 }
2815 }
Eric Laurent81784c32012-11-19 14:55:58 -08002816}
2817
Eric Laurent13084622016-05-17 10:51:49 -07002818bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
Eric Laurent81784c32012-11-19 14:55:58 -08002819{
Glenn Kastenc42e9b42016-03-21 11:35:03 -07002820 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu",
Eric Laurent81784c32012-11-19 14:55:58 -08002821 this, streamType, mTracks.size());
Eric Laurent13084622016-05-17 10:51:49 -07002822 bool trackMatch = false;
Eric Laurent81784c32012-11-19 14:55:58 -08002823 size_t size = mTracks.size();
2824 for (size_t i = 0; i < size; i++) {
2825 sp<Track> t = mTracks[i];
Eric Laurentd60560a2015-04-10 11:31:20 -07002826 if (t->streamType() == streamType && t->isExternalTrack()) {
Glenn Kasten5736c352012-12-04 12:12:34 -08002827 t->invalidate();
Eric Laurent13084622016-05-17 10:51:49 -07002828 trackMatch = true;
Eric Laurent81784c32012-11-19 14:55:58 -08002829 }
2830 }
Eric Laurent13084622016-05-17 10:51:49 -07002831 return trackMatch;
Eric Laurent81784c32012-11-19 14:55:58 -08002832}
2833
Haynes Mathew George05317d22016-05-03 16:34:26 -07002834void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
2835{
2836 Mutex::Autolock _l(mLock);
2837 invalidateTracks_l(streamType);
2838}
2839
Eric Laurent81784c32012-11-19 14:55:58 -08002840status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
2841{
Glenn Kastend848eb42016-03-08 13:42:11 -08002842 audio_session_t session = chain->sessionId();
Andy Hung010a1a12014-03-13 13:57:33 -07002843 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled
2844 ? mEffectBuffer : mSinkBuffer);
Eric Laurent81784c32012-11-19 14:55:58 -08002845 bool ownsBuffer = false;
2846
2847 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
Glenn Kastend848eb42016-03-08 13:42:11 -08002848 if (session > AUDIO_SESSION_OUTPUT_MIX) {
Eric Laurent81784c32012-11-19 14:55:58 -08002849 // Only one effect chain can be present in direct output thread and it uses
Andy Hung2098f272014-02-27 14:00:06 -08002850 // the sink buffer as input
Eric Laurent81784c32012-11-19 14:55:58 -08002851 if (mType != DIRECT) {
2852 size_t numSamples = mNormalFrameCount * mChannelCount;
2853 buffer = new int16_t[numSamples];
2854 memset(buffer, 0, numSamples * sizeof(int16_t));
2855 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
2856 ownsBuffer = true;
2857 }
2858
2859 // Attach all tracks with same session ID to this chain.
2860 for (size_t i = 0; i < mTracks.size(); ++i) {
2861 sp<Track> track = mTracks[i];
2862 if (session == track->sessionId()) {
2863 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(),
2864 buffer);
2865 track->setMainBuffer(buffer);
2866 chain->incTrackCnt();
2867 }
2868 }
2869
2870 // indicate all active tracks in the chain
2871 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
2872 sp<Track> track = mActiveTracks[i].promote();
2873 if (track == 0) {
2874 continue;
2875 }
2876 if (session == track->sessionId()) {
2877 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
2878 chain->incActiveTrackCnt();
2879 }
2880 }
2881 }
Eric Laurentaaa44472014-09-12 17:41:50 -07002882 chain->setThread(this);
Eric Laurent81784c32012-11-19 14:55:58 -08002883 chain->setInBuffer(buffer, ownsBuffer);
Andy Hung010a1a12014-03-13 13:57:33 -07002884 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled
2885 ? mEffectBuffer : mSinkBuffer));
Eric Laurent81784c32012-11-19 14:55:58 -08002886 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
Glenn Kastend848eb42016-03-08 13:42:11 -08002887 // chains list in order to be processed last as it contains output stage effects.
Eric Laurent81784c32012-11-19 14:55:58 -08002888 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
2889 // session AUDIO_SESSION_OUTPUT_STAGE to be processed
Glenn Kastend848eb42016-03-08 13:42:11 -08002890 // after track specific effects and before output stage.
Eric Laurent81784c32012-11-19 14:55:58 -08002891 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
Glenn Kastend848eb42016-03-08 13:42:11 -08002892 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX.
Eric Laurent81784c32012-11-19 14:55:58 -08002893 // Effect chain for other sessions are inserted at beginning of effect
2894 // chains list to be processed before output mix effects. Relative order between other
Glenn Kastend848eb42016-03-08 13:42:11 -08002895 // sessions is not important.
2896 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 &&
2897 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX,
2898 "audio_session_t constants misdefined");
Eric Laurent81784c32012-11-19 14:55:58 -08002899 size_t size = mEffectChains.size();
2900 size_t i = 0;
2901 for (i = 0; i < size; i++) {
2902 if (mEffectChains[i]->sessionId() < session) {
2903 break;
2904 }
2905 }
2906 mEffectChains.insertAt(chain, i);
2907 checkSuspendOnAddEffectChain_l(chain);
2908
2909 return NO_ERROR;
2910}
2911
2912size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
2913{
Glenn Kastend848eb42016-03-08 13:42:11 -08002914 audio_session_t session = chain->sessionId();
Eric Laurent81784c32012-11-19 14:55:58 -08002915
2916 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
2917
2918 for (size_t i = 0; i < mEffectChains.size(); i++) {
2919 if (chain == mEffectChains[i]) {
2920 mEffectChains.removeAt(i);
2921 // detach all active tracks from the chain
2922 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
2923 sp<Track> track = mActiveTracks[i].promote();
2924 if (track == 0) {
2925 continue;
2926 }
2927 if (session == track->sessionId()) {
2928 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
2929 chain.get(), session);
2930 chain->decActiveTrackCnt();
2931 }
2932 }
2933
2934 // detach all tracks with same session ID from this chain
2935 for (size_t i = 0; i < mTracks.size(); ++i) {
2936 sp<Track> track = mTracks[i];
2937 if (session == track->sessionId()) {
Andy Hung010a1a12014-03-13 13:57:33 -07002938 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer));
Eric Laurent81784c32012-11-19 14:55:58 -08002939 chain->decTrackCnt();
2940 }
2941 }
2942 break;
2943 }
2944 }
2945 return mEffectChains.size();
2946}
2947
2948status_t AudioFlinger::PlaybackThread::attachAuxEffect(
Chih-Hung Hsiehe964d4e2016-08-09 14:31:32 -07002949 const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId)
Eric Laurent81784c32012-11-19 14:55:58 -08002950{
2951 Mutex::Autolock _l(mLock);
2952 return attachAuxEffect_l(track, EffectId);
2953}
2954
2955status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
Chih-Hung Hsiehe964d4e2016-08-09 14:31:32 -07002956 const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId)
Eric Laurent81784c32012-11-19 14:55:58 -08002957{
2958 status_t status = NO_ERROR;
2959
2960 if (EffectId == 0) {
2961 track->setAuxBuffer(0, NULL);
2962 } else {
2963 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
2964 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
2965 if (effect != 0) {
2966 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
2967 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
2968 } else {
2969 status = INVALID_OPERATION;
2970 }
2971 } else {
2972 status = BAD_VALUE;
2973 }
2974 }
2975 return status;
2976}
2977
2978void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
2979{
2980 for (size_t i = 0; i < mTracks.size(); ++i) {
2981 sp<Track> track = mTracks[i];
2982 if (track->auxEffectId() == effectId) {
2983 attachAuxEffect_l(track, 0);
2984 }
2985 }
2986}
2987
2988bool AudioFlinger::PlaybackThread::threadLoop()
2989{
2990 Vector< sp<Track> > tracksToRemove;
2991
Eric Laurentad9cb8b2015-05-26 16:38:19 -07002992 mStandbyTimeNs = systemTime();
Andy Hung69488c42016-05-16 18:43:33 -07002993 nsecs_t lastWriteFinished = -1; // time last server write completed
2994 int64_t lastFramesWritten = -1; // track changes in timestamp server frames written
Eric Laurent81784c32012-11-19 14:55:58 -08002995
2996 // MIXER
2997 nsecs_t lastWarning = 0;
2998
2999 // DUPLICATING
3000 // FIXME could this be made local to while loop?
3001 writeFrames = 0;
3002
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003003 int lastGeneration = 0;
3004
Eric Laurent81784c32012-11-19 14:55:58 -08003005 cacheParameters_l();
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003006 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08003007
3008 if (mType == MIXER) {
3009 sleepTimeShift = 0;
3010 }
3011
3012 CpuStats cpuStats;
3013 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
3014
3015 acquireWakeLock();
3016
Glenn Kasten9e58b552013-01-18 15:09:48 -08003017 // mNBLogWriter->log can only be called while thread mutex mLock is held.
3018 // So if you need to log when mutex is unlocked, set logString to a non-NULL string,
3019 // and then that string will be logged at the next convenient opportunity.
3020 const char *logString = NULL;
3021
Eric Laurent664539d2013-09-23 18:24:31 -07003022 checkSilentMode_l();
3023
Eric Laurent81784c32012-11-19 14:55:58 -08003024 while (!exitPending())
3025 {
3026 cpuStats.sample(myName);
3027
3028 Vector< sp<EffectChain> > effectChains;
3029
Eric Laurent81784c32012-11-19 14:55:58 -08003030 { // scope for mLock
3031
3032 Mutex::Autolock _l(mLock);
3033
Eric Laurent021cf962014-05-13 10:18:14 -07003034 processConfigEvents_l();
Eric Laurent10351942014-05-08 18:49:52 -07003035
Glenn Kasten9e58b552013-01-18 15:09:48 -08003036 if (logString != NULL) {
3037 mNBLogWriter->logTimestamp();
3038 mNBLogWriter->log(logString);
3039 logString = NULL;
3040 }
3041
Glenn Kasten4c053ea2014-09-28 14:41:07 -07003042 // Gather the framesReleased counters for all active tracks,
Andy Hunge10393e2015-06-12 13:59:33 -07003043 // and associate with the sink frames written out. We need
3044 // this to convert the sink timestamp to the track timestamp.
Andy Hung69488c42016-05-16 18:43:33 -07003045 bool kernelLocationUpdate = false;
Andy Hunge10393e2015-06-12 13:59:33 -07003046 if (mNormalSink != 0) {
Andy Hungc54b1ff2016-02-23 14:07:07 -08003047 // Note: The DuplicatingThread may not have a mNormalSink.
Andy Hung818e7a32016-02-16 18:08:07 -08003048 // We always fetch the timestamp here because often the downstream
Andy Hung69488c42016-05-16 18:43:33 -07003049 // sink will block while writing.
Andy Hung818e7a32016-02-16 18:08:07 -08003050 ExtendedTimestamp timestamp; // use private copy to fetch
3051 (void) mNormalSink->getTimestamp(timestamp);
Andy Hung6d7b1192016-05-07 22:59:48 -07003052
3053 // We keep track of the last valid kernel position in case we are in underrun
3054 // and the normal mixer period is the same as the fast mixer period, or there
3055 // is some error from the HAL.
3056 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
3057 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
3058 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
3059 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] =
3060 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
3061
3062 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
3063 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER];
3064 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] =
3065 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER];
Andy Hung69488c42016-05-16 18:43:33 -07003066 }
3067
3068 if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) {
3069 kernelLocationUpdate = true;
Andy Hung6d7b1192016-05-07 22:59:48 -07003070 } else {
Eric Laurent122f7e72016-06-29 11:53:29 -07003071 ALOGVV("getTimestamp error - no valid kernel position");
Andy Hung6d7b1192016-05-07 22:59:48 -07003072 }
3073
Andy Hung818e7a32016-02-16 18:08:07 -08003074 // copy over kernel info
3075 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
Andy Hung238fa3d2016-07-28 10:53:22 -07003076 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]
3077 + mSuspendedFrames; // add frames discarded when suspended
Andy Hung818e7a32016-02-16 18:08:07 -08003078 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
3079 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
Andy Hungc54b1ff2016-02-23 14:07:07 -08003080 }
3081 // mFramesWritten for non-offloaded tracks are contiguous
3082 // even after standby() is called. This is useful for the track frame
3083 // to sink frame mapping.
Andy Hung69488c42016-05-16 18:43:33 -07003084 bool serverLocationUpdate = false;
3085 if (mFramesWritten != lastFramesWritten) {
3086 serverLocationUpdate = true;
3087 lastFramesWritten = mFramesWritten;
3088 }
3089 // Only update timestamps if there is a meaningful change.
3090 // Either the kernel timestamp must be valid or we have written something.
3091 if (kernelLocationUpdate || serverLocationUpdate) {
3092 if (serverLocationUpdate) {
3093 // use the time before we called the HAL write - it is a bit more accurate
3094 // to when the server last read data than the current time here.
3095 //
3096 // If we haven't written anything, mLastWriteTime will be -1
3097 // and we use systemTime().
3098 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten;
3099 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1
3100 ? systemTime() : mLastWriteTime;
3101 }
3102 const size_t size = mActiveTracks.size();
3103 for (size_t i = 0; i < size; ++i) {
3104 sp<Track> t = mActiveTracks[i].promote();
3105 if (t != 0 && !t->isFastTrack()) {
3106 t->updateTrackFrameInfo(
3107 t->mAudioTrackServerProxy->framesReleased(),
3108 mFramesWritten,
3109 mTimestamp);
3110 }
Andy Hunge10393e2015-06-12 13:59:33 -07003111 }
Glenn Kastenbd096fd2013-08-23 13:53:56 -07003112 }
3113
Eric Laurent81784c32012-11-19 14:55:58 -08003114 saveOutputTracks();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003115 if (mSignalPending) {
3116 // A signal was raised while we were unlocked
3117 mSignalPending = false;
3118 } else if (waitingAsyncCallback_l()) {
3119 if (exitPending()) {
3120 break;
3121 }
Marco Nelissen078538c2015-05-12 09:17:57 -07003122 bool released = false;
Eric Laurent64667972016-03-30 18:19:46 -07003123 if (!keepWakeLock()) {
Marco Nelissen078538c2015-05-12 09:17:57 -07003124 releaseWakeLock_l();
3125 released = true;
Mikhail Naganove94c27a2016-08-18 17:31:46 -07003126 mWakeLockUids.clear();
3127 mActiveTracksGeneration++;
Marco Nelissen078538c2015-05-12 09:17:57 -07003128 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003129 ALOGV("wait async completion");
3130 mWaitWorkCV.wait(mLock);
3131 ALOGV("async completion/wake");
Marco Nelissen078538c2015-05-12 09:17:57 -07003132 if (released) {
3133 acquireWakeLock_l();
3134 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003135 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
3136 mSleepTimeUs = 0;
Eric Laurentede6c3b2013-09-19 14:37:46 -07003137
3138 continue;
3139 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003140 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) ||
Eric Laurentbfb1b832013-01-07 09:53:42 -08003141 isSuspended()) {
3142 // put audio hardware into standby after short delay
3143 if (shouldStandby_l()) {
Eric Laurent81784c32012-11-19 14:55:58 -08003144
3145 threadLoop_standby();
3146
3147 mStandby = true;
3148 }
3149
3150 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
3151 // we're about to wait, flush the binder command buffer
3152 IPCThreadState::self()->flushCommands();
3153
3154 clearOutputTracks();
3155
3156 if (exitPending()) {
3157 break;
3158 }
3159
3160 releaseWakeLock_l();
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003161 mWakeLockUids.clear();
3162 mActiveTracksGeneration++;
Eric Laurent81784c32012-11-19 14:55:58 -08003163 // wait until we have something to do...
3164 ALOGV("%s going to sleep", myName.string());
3165 mWaitWorkCV.wait(mLock);
3166 ALOGV("%s waking up", myName.string());
3167 acquireWakeLock_l();
3168
3169 mMixerStatus = MIXER_IDLE;
3170 mMixerStatusIgnoringFastTracks = MIXER_IDLE;
3171 mBytesWritten = 0;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003172 mBytesRemaining = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08003173 checkSilentMode_l();
3174
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003175 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
3176 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08003177 if (mType == MIXER) {
3178 sleepTimeShift = 0;
3179 }
3180
3181 continue;
3182 }
3183 }
Eric Laurent81784c32012-11-19 14:55:58 -08003184 // mMixerStatusIgnoringFastTracks is also updated internally
3185 mMixerStatus = prepareTracks_l(&tracksToRemove);
3186
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003187 // compare with previously applied list
3188 if (lastGeneration != mActiveTracksGeneration) {
3189 // update wakelock
3190 updateWakeLockUids_l(mWakeLockUids);
3191 lastGeneration = mActiveTracksGeneration;
3192 }
3193
Eric Laurent81784c32012-11-19 14:55:58 -08003194 // prevent any changes in effect chain list and in each effect chain
3195 // during mixing and effect process as the audio buffers could be deleted
3196 // or modified if an effect is created or deleted
3197 lockEffectChains_l(effectChains);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003198 } // mLock scope ends
Eric Laurent81784c32012-11-19 14:55:58 -08003199
Eric Laurentbfb1b832013-01-07 09:53:42 -08003200 if (mBytesRemaining == 0) {
3201 mCurrentWriteLength = 0;
3202 if (mMixerStatus == MIXER_TRACKS_READY) {
3203 // threadLoop_mix() sets mCurrentWriteLength
3204 threadLoop_mix();
3205 } else if ((mMixerStatus != MIXER_DRAIN_TRACK)
3206 && (mMixerStatus != MIXER_DRAIN_ALL)) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003207 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data
Eric Laurentbfb1b832013-01-07 09:53:42 -08003208 // must be written to HAL
3209 threadLoop_sleepTime();
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003210 if (mSleepTimeUs == 0) {
Andy Hung25c2dac2014-02-27 14:56:00 -08003211 mCurrentWriteLength = mSinkBufferSize;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003212 }
3213 }
Andy Hung98ef9782014-03-04 14:46:50 -08003214 // Either threadLoop_mix() or threadLoop_sleepTime() should have set
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003215 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0.
Andy Hung98ef9782014-03-04 14:46:50 -08003216 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid)
3217 // or mSinkBuffer (if there are no effects).
3218 //
3219 // This is done pre-effects computation; if effects change to
3220 // support higher precision, this needs to move.
3221 //
3222 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l().
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003223 // TODO use mSleepTimeUs == 0 as an additional condition.
Andy Hung98ef9782014-03-04 14:46:50 -08003224 if (mMixerBufferValid) {
3225 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
3226 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
3227
Andy Hung2ddee192015-12-18 17:34:44 -08003228 // mono blend occurs for mixer threads only (not direct or offloaded)
3229 // and is handled here if we're going directly to the sink.
3230 if (requireMonoBlend() && !mEffectBufferValid) {
Glenn Kasten03c48d52016-01-27 17:25:17 -08003231 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount,
3232 true /*limit*/);
Andy Hung2ddee192015-12-18 17:34:44 -08003233 }
3234
Andy Hung98ef9782014-03-04 14:46:50 -08003235 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
3236 mNormalFrameCount * mChannelCount);
3237 }
3238
Eric Laurentbfb1b832013-01-07 09:53:42 -08003239 mBytesRemaining = mCurrentWriteLength;
3240 if (isSuspended()) {
Andy Hung238fa3d2016-07-28 10:53:22 -07003241 // Simulate write to HAL when suspended (e.g. BT SCO phone call).
3242 mSleepTimeUs = suspendSleepTimeUs(); // assumes full buffer.
3243 const size_t framesRemaining = mBytesRemaining / mFrameSize;
3244 mBytesWritten += mBytesRemaining;
3245 mFramesWritten += framesRemaining;
3246 mSuspendedFrames += framesRemaining; // to adjust kernel HAL position
Eric Laurentbfb1b832013-01-07 09:53:42 -08003247 mBytesRemaining = 0;
3248 }
Eric Laurent81784c32012-11-19 14:55:58 -08003249
Eric Laurentbfb1b832013-01-07 09:53:42 -08003250 // only process effects if we're going to write
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003251 if (mSleepTimeUs == 0 && mType != OFFLOAD) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08003252 for (size_t i = 0; i < effectChains.size(); i ++) {
3253 effectChains[i]->process_l();
3254 }
Eric Laurent81784c32012-11-19 14:55:58 -08003255 }
3256 }
Eric Laurent59fe0102013-09-27 18:48:26 -07003257 // Process effect chains for offloaded thread even if no audio
3258 // was read from audio track: process only updates effect state
3259 // and thus does have to be synchronized with audio writes but may have
3260 // to be called while waiting for async write callback
3261 if (mType == OFFLOAD) {
3262 for (size_t i = 0; i < effectChains.size(); i ++) {
3263 effectChains[i]->process_l();
3264 }
3265 }
Eric Laurent81784c32012-11-19 14:55:58 -08003266
Andy Hung98ef9782014-03-04 14:46:50 -08003267 // Only if the Effects buffer is enabled and there is data in the
3268 // Effects buffer (buffer valid), we need to
3269 // copy into the sink buffer.
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003270 // TODO use mSleepTimeUs == 0 as an additional condition.
Andy Hung98ef9782014-03-04 14:46:50 -08003271 if (mEffectBufferValid) {
3272 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat);
Andy Hung2ddee192015-12-18 17:34:44 -08003273
3274 if (requireMonoBlend()) {
Glenn Kasten03c48d52016-01-27 17:25:17 -08003275 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount,
3276 true /*limit*/);
Andy Hung2ddee192015-12-18 17:34:44 -08003277 }
3278
Andy Hung98ef9782014-03-04 14:46:50 -08003279 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat,
3280 mNormalFrameCount * mChannelCount);
3281 }
3282
Eric Laurent81784c32012-11-19 14:55:58 -08003283 // enable changes in effect chain
3284 unlockEffectChains(effectChains);
3285
Eric Laurentbfb1b832013-01-07 09:53:42 -08003286 if (!waitingAsyncCallback()) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003287 // mSleepTimeUs == 0 means we must write to audio hardware
3288 if (mSleepTimeUs == 0) {
Andy Hung08fb1742015-05-31 23:22:10 -07003289 ssize_t ret = 0;
Andy Hung69488c42016-05-16 18:43:33 -07003290 // We save lastWriteFinished here, as previousLastWriteFinished,
3291 // for throttling. On thread start, previousLastWriteFinished will be
3292 // set to -1, which properly results in no throttling after the first write.
3293 nsecs_t previousLastWriteFinished = lastWriteFinished;
3294 nsecs_t delta = 0;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003295 if (mBytesRemaining) {
Andy Hung69488c42016-05-16 18:43:33 -07003296 // FIXME rewrite to reduce number of system calls
3297 mLastWriteTime = systemTime(); // also used for dumpsys
Andy Hung08fb1742015-05-31 23:22:10 -07003298 ret = threadLoop_write();
Andy Hung69488c42016-05-16 18:43:33 -07003299 lastWriteFinished = systemTime();
3300 delta = lastWriteFinished - mLastWriteTime;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003301 if (ret < 0) {
3302 mBytesRemaining = 0;
3303 } else {
3304 mBytesWritten += ret;
3305 mBytesRemaining -= ret;
Andy Hungc54b1ff2016-02-23 14:07:07 -08003306 mFramesWritten += ret / mFrameSize;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003307 }
3308 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) ||
3309 (mMixerStatus == MIXER_DRAIN_ALL)) {
3310 threadLoop_drain();
Eric Laurent81784c32012-11-19 14:55:58 -08003311 }
Andy Hung08fb1742015-05-31 23:22:10 -07003312 if (mType == MIXER && !mStandby) {
Glenn Kasten4944acb2013-08-19 08:39:20 -07003313 // write blocked detection
Andy Hung08fb1742015-05-31 23:22:10 -07003314 if (delta > maxPeriod) {
Glenn Kasten4944acb2013-08-19 08:39:20 -07003315 mNumDelayedWrites++;
Andy Hung69488c42016-05-16 18:43:33 -07003316 if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) {
Glenn Kasten4944acb2013-08-19 08:39:20 -07003317 ATRACE_NAME("underrun");
3318 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
Glenn Kastenc42e9b42016-03-21 11:35:03 -07003319 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this);
Andy Hung69488c42016-05-16 18:43:33 -07003320 lastWarning = lastWriteFinished;
Glenn Kasten4944acb2013-08-19 08:39:20 -07003321 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003322 }
Andy Hung08fb1742015-05-31 23:22:10 -07003323
3324 if (mThreadThrottle
3325 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks)
3326 && ret > 0) { // we wrote something
3327 // Limit MixerThread data processing to no more than twice the
3328 // expected processing rate.
3329 //
3330 // This helps prevent underruns with NuPlayer and other applications
3331 // which may set up buffers that are close to the minimum size, or use
3332 // deep buffers, and rely on a double-buffering sleep strategy to fill.
3333 //
3334 // The throttle smooths out sudden large data drains from the device,
3335 // e.g. when it comes out of standby, which often causes problems with
3336 // (1) mixer threads without a fast mixer (which has its own warm-up)
3337 // (2) minimum buffer sized tracks (even if the track is full,
3338 // the app won't fill fast enough to handle the sudden draw).
Haynes Mathew Georgef92b2172016-05-09 11:34:15 -07003339 //
3340 // Total time spent in last processing cycle equals time spent in
3341 // 1. threadLoop_write, as well as time spent in
3342 // 2. threadLoop_mix (significant for heavy mixing, especially
3343 // on low tier processors)
Andy Hung08fb1742015-05-31 23:22:10 -07003344
Andy Hung69488c42016-05-16 18:43:33 -07003345 // it's OK if deltaMs is an overestimate.
3346 const int32_t deltaMs =
3347 (lastWriteFinished - previousLastWriteFinished) / 1000000;
Andy Hung08fb1742015-05-31 23:22:10 -07003348 const int32_t throttleMs = mHalfBufferMs - deltaMs;
3349 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) {
3350 usleep(throttleMs * 1000);
Andy Hung40eb1a12015-06-18 13:42:02 -07003351 // notify of throttle start on verbose log
3352 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs,
3353 "mixer(%p) throttle begin:"
3354 " ret(%zd) deltaMs(%d) requires sleep %d ms",
Andy Hung08fb1742015-05-31 23:22:10 -07003355 this, ret, deltaMs, throttleMs);
Andy Hung40eb1a12015-06-18 13:42:02 -07003356 mThreadThrottleTimeMs += throttleMs;
Andy Hung0a31ddd2016-07-06 19:10:29 -07003357 // Throttle must be attributed to the previous mixer loop's write time
3358 // to allow back-to-back throttling.
3359 lastWriteFinished += throttleMs * 1000000;
Andy Hung40eb1a12015-06-18 13:42:02 -07003360 } else {
3361 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs;
3362 if (diff > 0) {
3363 // notify of throttle end on debug log
Andy Hung3ea004d2016-05-05 16:48:37 -07003364 // but prevent spamming for bluetooth
3365 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()),
3366 "mixer(%p) throttle end: throttle time(%u)", this, diff);
Andy Hung40eb1a12015-06-18 13:42:02 -07003367 mThreadThrottleEndMs = mThreadThrottleTimeMs;
3368 }
Andy Hung08fb1742015-05-31 23:22:10 -07003369 }
3370 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003371 }
Eric Laurent81784c32012-11-19 14:55:58 -08003372
Eric Laurentbfb1b832013-01-07 09:53:42 -08003373 } else {
Glenn Kastene7754022014-10-31 12:11:26 -07003374 ATRACE_BEGIN("sleep");
Eric Laurente93cc032016-05-05 10:15:10 -07003375 Mutex::Autolock _l(mLock);
3376 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) {
3377 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs));
Eric Laurent51716182016-02-29 18:00:56 -08003378 }
Glenn Kastene7754022014-10-31 12:11:26 -07003379 ATRACE_END();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003380 }
Eric Laurent81784c32012-11-19 14:55:58 -08003381 }
3382
3383 // Finally let go of removed track(s), without the lock held
3384 // since we can't guarantee the destructors won't acquire that
3385 // same lock. This will also mutate and push a new fast mixer state.
3386 threadLoop_removeTracks(tracksToRemove);
3387 tracksToRemove.clear();
3388
3389 // FIXME I don't understand the need for this here;
3390 // it was in the original code but maybe the
3391 // assignment in saveOutputTracks() makes this unnecessary?
3392 clearOutputTracks();
3393
3394 // Effect chains will be actually deleted here if they were removed from
3395 // mEffectChains list during mixing or effects processing
3396 effectChains.clear();
3397
3398 // FIXME Note that the above .clear() is no longer necessary since effectChains
3399 // is now local to this block, but will keep it for now (at least until merge done).
3400 }
3401
Eric Laurentbfb1b832013-01-07 09:53:42 -08003402 threadLoop_exit();
3403
Eric Laurentcf817a22014-08-04 20:36:31 -07003404 if (!mStandby) {
3405 threadLoop_standby();
3406 mStandby = true;
Eric Laurent81784c32012-11-19 14:55:58 -08003407 }
3408
3409 releaseWakeLock();
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003410 mWakeLockUids.clear();
3411 mActiveTracksGeneration++;
Eric Laurent81784c32012-11-19 14:55:58 -08003412
3413 ALOGV("Thread %p type %d exiting", this, mType);
3414 return false;
3415}
3416
Eric Laurentbfb1b832013-01-07 09:53:42 -08003417// removeTracks_l() must be called with ThreadBase::mLock held
3418void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
3419{
3420 size_t count = tracksToRemove.size();
Glenn Kasten34fca342013-08-13 09:48:14 -07003421 if (count > 0) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08003422 for (size_t i=0 ; i<count ; i++) {
3423 const sp<Track>& track = tracksToRemove.itemAt(i);
3424 mActiveTracks.remove(track);
Marco Nelissen462fd2f2013-01-14 14:12:05 -08003425 mWakeLockUids.remove(track->uid());
3426 mActiveTracksGeneration++;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003427 ALOGV("removeTracks_l removing track on session %d", track->sessionId());
3428 sp<EffectChain> chain = getEffectChain_l(track->sessionId());
3429 if (chain != 0) {
3430 ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
3431 track->sessionId());
3432 chain->decActiveTrackCnt();
3433 }
3434 if (track->isTerminated()) {
3435 removeTrack_l(track);
3436 }
3437 }
3438 }
3439
3440}
Eric Laurent81784c32012-11-19 14:55:58 -08003441
Eric Laurentaccc1472013-09-20 09:36:34 -07003442status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
3443{
3444 if (mNormalSink != 0) {
Andy Hung818e7a32016-02-16 18:08:07 -08003445 ExtendedTimestamp ets;
3446 status_t status = mNormalSink->getTimestamp(ets);
3447 if (status == NO_ERROR) {
3448 status = ets.getBestTimestamp(&timestamp);
3449 }
3450 return status;
Eric Laurentaccc1472013-09-20 09:36:34 -07003451 }
Andy Hung9a1c8892014-12-03 11:37:42 -08003452 if ((mType == OFFLOAD || mType == DIRECT)
3453 && mOutput != NULL && mOutput->stream->get_presentation_position) {
Eric Laurentaccc1472013-09-20 09:36:34 -07003454 uint64_t position64;
Phil Burk062e67a2015-02-11 13:40:50 -08003455 int ret = mOutput->getPresentationPosition(&position64, &timestamp.mTime);
Eric Laurentaccc1472013-09-20 09:36:34 -07003456 if (ret == 0) {
3457 timestamp.mPosition = (uint32_t)position64;
3458 return NO_ERROR;
3459 }
3460 }
3461 return INVALID_OPERATION;
3462}
Eric Laurent1c333e22014-05-20 10:48:17 -07003463
Eric Laurent054d9d32015-04-24 08:48:48 -07003464status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch,
3465 audio_patch_handle_t *handle)
3466{
Andy Hungf60abce2016-08-26 11:37:54 -07003467 status_t status;
3468 if (property_get_bool("af.patch_park", false /* default_value */)) {
3469 // Park FastMixer to avoid potential DOS issues with writing to the HAL
3470 // or if HAL does not properly lock against access.
3471 AutoPark<FastMixer> park(mFastMixer);
3472 status = PlaybackThread::createAudioPatch_l(patch, handle);
3473 } else {
3474 status = PlaybackThread::createAudioPatch_l(patch, handle);
3475 }
Eric Laurent054d9d32015-04-24 08:48:48 -07003476 return status;
3477}
3478
Eric Laurent1c333e22014-05-20 10:48:17 -07003479status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
3480 audio_patch_handle_t *handle)
3481{
3482 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07003483
3484 // store new device and send to effects
3485 audio_devices_t type = AUDIO_DEVICE_NONE;
3486 for (unsigned int i = 0; i < patch->num_sinks; i++) {
3487 type |= patch->sinks[i].ext.device.type;
3488 }
3489
3490#ifdef ADD_BATTERY_DATA
3491 // when changing the audio output device, call addBatteryData to notify
3492 // the change
3493 if (mOutDevice != type) {
3494 uint32_t params = 0;
3495 // check whether speaker is on
3496 if (type & AUDIO_DEVICE_OUT_SPEAKER) {
3497 params |= IMediaPlayerService::kBatteryDataSpeakerOn;
Eric Laurent1c333e22014-05-20 10:48:17 -07003498 }
3499
Eric Laurent054d9d32015-04-24 08:48:48 -07003500 audio_devices_t deviceWithoutSpeaker
3501 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
3502 // check if any other device (except speaker) is on
3503 if (type & deviceWithoutSpeaker) {
3504 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
3505 }
3506
3507 if (params != 0) {
3508 addBatteryData(params);
3509 }
3510 }
3511#endif
3512
3513 for (size_t i = 0; i < mEffectChains.size(); i++) {
3514 mEffectChains[i]->setDevice_l(type);
3515 }
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07003516
3517 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when
3518 // the thread is created so that the first patch creation triggers an ioConfigChanged callback
3519 bool configChanged = mPrevOutDevice != type;
Eric Laurent054d9d32015-04-24 08:48:48 -07003520 mOutDevice = type;
Eric Laurent296fb132015-05-01 11:38:42 -07003521 mPatch = *patch;
Eric Laurent054d9d32015-04-24 08:48:48 -07003522
3523 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
Mikhail Naganove4f1f632016-08-31 11:35:10 -07003524 sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice();
3525 status = hwDevice->createAudioPatch(patch->num_sources,
3526 patch->sources,
3527 patch->num_sinks,
3528 patch->sinks,
3529 handle);
Eric Laurent1c333e22014-05-20 10:48:17 -07003530 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07003531 char *address;
3532 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) {
3533 //FIXME: we only support address on first sink with HAL version < 3.0
3534 address = audio_device_address_to_parameter(
3535 patch->sinks[0].ext.device.type,
3536 patch->sinks[0].ext.device.address);
3537 } else {
3538 address = (char *)calloc(1, 1);
3539 }
3540 AudioParameter param = AudioParameter(String8(address));
3541 free(address);
3542 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type);
3543 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3544 param.toString().string());
3545 *handle = AUDIO_PATCH_HANDLE_NONE;
Eric Laurent1c333e22014-05-20 10:48:17 -07003546 }
Eric Laurente8726fe2015-06-26 09:39:24 -07003547 if (configChanged) {
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07003548 mPrevOutDevice = type;
Eric Laurente8726fe2015-06-26 09:39:24 -07003549 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
3550 }
Eric Laurent1c333e22014-05-20 10:48:17 -07003551 return status;
3552}
3553
Eric Laurent054d9d32015-04-24 08:48:48 -07003554status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
3555{
Andy Hungf60abce2016-08-26 11:37:54 -07003556 status_t status;
3557 if (property_get_bool("af.patch_park", false /* default_value */)) {
3558 // Park FastMixer to avoid potential DOS issues with writing to the HAL
3559 // or if HAL does not properly lock against access.
3560 AutoPark<FastMixer> park(mFastMixer);
3561 status = PlaybackThread::releaseAudioPatch_l(handle);
3562 } else {
3563 status = PlaybackThread::releaseAudioPatch_l(handle);
3564 }
Eric Laurent054d9d32015-04-24 08:48:48 -07003565 return status;
3566}
3567
Eric Laurent1c333e22014-05-20 10:48:17 -07003568status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
3569{
3570 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07003571
3572 mOutDevice = AUDIO_DEVICE_NONE;
3573
Eric Laurent1c333e22014-05-20 10:48:17 -07003574 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
Mikhail Naganove4f1f632016-08-31 11:35:10 -07003575 sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice();
3576 status = hwDevice->releaseAudioPatch(handle);
Eric Laurent1c333e22014-05-20 10:48:17 -07003577 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07003578 AudioParameter param;
3579 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0);
3580 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
3581 param.toString().string());
Eric Laurent1c333e22014-05-20 10:48:17 -07003582 }
3583 return status;
3584}
3585
Eric Laurent83b88082014-06-20 18:31:16 -07003586void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track)
3587{
3588 Mutex::Autolock _l(mLock);
3589 mTracks.add(track);
3590}
3591
3592void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track)
3593{
3594 Mutex::Autolock _l(mLock);
3595 destroyTrack_l(track);
3596}
3597
3598void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config)
3599{
3600 ThreadBase::getAudioPortConfig(config);
3601 config->role = AUDIO_PORT_ROLE_SOURCE;
3602 config->ext.mix.hw_module = mOutput->audioHwDev->handle();
3603 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
3604}
3605
Eric Laurent81784c32012-11-19 14:55:58 -08003606// ----------------------------------------------------------------------------
3607
3608AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
Eric Laurent72e3f392015-05-20 14:43:50 -07003609 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type)
3610 : PlaybackThread(audioFlinger, output, id, device, type, systemReady),
Eric Laurent81784c32012-11-19 14:55:58 -08003611 // mAudioMixer below
3612 // mFastMixer below
Andy Hung2ddee192015-12-18 17:34:44 -08003613 mFastMixerFutex(0),
3614 mMasterMono(false)
Eric Laurent81784c32012-11-19 14:55:58 -08003615 // mOutputSink below
3616 // mPipeSink below
3617 // mNormalSink below
3618{
3619 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
Glenn Kastenc42e9b42016-03-21 11:35:03 -07003620 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, "
3621 "mFrameCount=%zu, mNormalFrameCount=%zu",
Eric Laurent81784c32012-11-19 14:55:58 -08003622 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
3623 mNormalFrameCount);
3624 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
3625
Andy Hungfbfc3952015-01-15 13:33:51 -08003626 if (type == DUPLICATING) {
3627 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks
3628 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write().
3629 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink.
3630 return;
3631 }
Eric Laurent81784c32012-11-19 14:55:58 -08003632 // create an NBAIO sink for the HAL output stream, and negotiate
3633 mOutputSink = new AudioStreamOutSink(output->stream);
3634 size_t numCounterOffers = 0;
Glenn Kastenf69f9862014-03-07 08:37:57 -08003635 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003636#if !LOG_NDEBUG
3637 ssize_t index =
3638#else
3639 (void)
3640#endif
3641 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers);
Eric Laurent81784c32012-11-19 14:55:58 -08003642 ALOG_ASSERT(index == 0);
3643
3644 // initialize fast mixer depending on configuration
3645 bool initFastMixer;
3646 switch (kUseFastMixer) {
3647 case FastMixer_Never:
3648 initFastMixer = false;
3649 break;
3650 case FastMixer_Always:
3651 initFastMixer = true;
3652 break;
3653 case FastMixer_Static:
3654 case FastMixer_Dynamic:
3655 initFastMixer = mFrameCount < mNormalFrameCount;
3656 break;
3657 }
3658 if (initFastMixer) {
Andy Hung1258c1a2014-05-23 21:22:17 -07003659 audio_format_t fastMixerFormat;
3660 if (mMixerBufferEnabled && mEffectBufferEnabled) {
3661 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT;
3662 } else {
3663 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
3664 }
3665 if (mFormat != fastMixerFormat) {
3666 // change our Sink format to accept our intermediate precision
3667 mFormat = fastMixerFormat;
3668 free(mSinkBuffer);
3669 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
3670 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize;
3671 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize);
3672 }
Eric Laurent81784c32012-11-19 14:55:58 -08003673
3674 // create a MonoPipe to connect our submix to FastMixer
3675 NBAIO_Format format = mOutputSink->format();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003676#ifdef TEE_SINK
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003677 NBAIO_Format origformat = format;
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003678#endif
Andy Hung1258c1a2014-05-23 21:22:17 -07003679 // adjust format to match that of the Fast Mixer
Glenn Kasten97b7b752014-09-28 13:04:24 -07003680 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat);
Andy Hung1258c1a2014-05-23 21:22:17 -07003681 format.mFormat = fastMixerFormat;
3682 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
3683
Eric Laurent81784c32012-11-19 14:55:58 -08003684 // This pipe depth compensates for scheduling latency of the normal mixer thread.
3685 // When it wakes up after a maximum latency, it runs a few cycles quickly before
3686 // finally blocking. Note the pipe implementation rounds up the request to a power of 2.
3687 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
3688 const NBAIO_Format offers[1] = {format};
3689 size_t numCounterOffers = 0;
Glenn Kastenfc302fd2016-04-11 14:11:26 -07003690#if !LOG_NDEBUG || defined(TEE_SINK)
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07003691 ssize_t index =
3692#else
3693 (void)
3694#endif
3695 monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
Eric Laurent81784c32012-11-19 14:55:58 -08003696 ALOG_ASSERT(index == 0);
3697 monoPipe->setAvgFrames((mScreenState & 1) ?
3698 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
3699 mPipeSink = monoPipe;
3700
Glenn Kasten46909e72013-02-26 09:20:22 -08003701#ifdef TEE_SINK
Glenn Kastenda6ef132013-01-10 12:31:01 -08003702 if (mTeeSinkOutputEnabled) {
3703 // create a Pipe to archive a copy of FastMixer's output for dumpsys
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003704 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat);
3705 const NBAIO_Format offers2[1] = {origformat};
Glenn Kastenda6ef132013-01-10 12:31:01 -08003706 numCounterOffers = 0;
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003707 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers);
Glenn Kastenda6ef132013-01-10 12:31:01 -08003708 ALOG_ASSERT(index == 0);
3709 mTeeSink = teeSink;
3710 PipeReader *teeSource = new PipeReader(*teeSink);
3711 numCounterOffers = 0;
Glenn Kastenba0b34c2014-09-28 13:06:06 -07003712 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers);
Glenn Kastenda6ef132013-01-10 12:31:01 -08003713 ALOG_ASSERT(index == 0);
3714 mTeeSource = teeSource;
3715 }
Glenn Kasten46909e72013-02-26 09:20:22 -08003716#endif
Eric Laurent81784c32012-11-19 14:55:58 -08003717
3718 // create fast mixer and configure it initially with just one fast track for our submix
3719 mFastMixer = new FastMixer();
3720 FastMixerStateQueue *sq = mFastMixer->sq();
3721#ifdef STATE_QUEUE_DUMP
3722 sq->setObserverDump(&mStateQueueObserverDump);
3723 sq->setMutatorDump(&mStateQueueMutatorDump);
3724#endif
3725 FastMixerState *state = sq->begin();
3726 FastTrack *fastTrack = &state->mFastTracks[0];
3727 // wrap the source side of the MonoPipe to make it an AudioBufferProvider
3728 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe));
3729 fastTrack->mVolumeProvider = NULL;
Andy Hunge8a1ced2014-05-09 15:02:21 -07003730 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer
3731 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
Eric Laurent81784c32012-11-19 14:55:58 -08003732 fastTrack->mGeneration++;
3733 state->mFastTracksGen++;
3734 state->mTrackMask = 1;
3735 // fast mixer will use the HAL output sink
3736 state->mOutputSink = mOutputSink.get();
3737 state->mOutputSinkGen++;
3738 state->mFrameCount = mFrameCount;
3739 state->mCommand = FastMixerState::COLD_IDLE;
3740 // already done in constructor initialization list
3741 //mFastMixerFutex = 0;
3742 state->mColdFutexAddr = &mFastMixerFutex;
3743 state->mColdGen++;
3744 state->mDumpState = &mFastMixerDumpState;
Glenn Kasten46909e72013-02-26 09:20:22 -08003745#ifdef TEE_SINK
Eric Laurent81784c32012-11-19 14:55:58 -08003746 state->mTeeSink = mTeeSink.get();
Glenn Kasten46909e72013-02-26 09:20:22 -08003747#endif
Glenn Kasten9e58b552013-01-18 15:09:48 -08003748 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
3749 state->mNBLogWriter = mFastMixerNBLogWriter.get();
Eric Laurent81784c32012-11-19 14:55:58 -08003750 sq->end();
3751 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3752
3753 // start the fast mixer
3754 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
3755 pid_t tid = mFastMixer->getTid();
Eric Laurent72e3f392015-05-20 14:43:50 -07003756 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
Eric Laurent81784c32012-11-19 14:55:58 -08003757
3758#ifdef AUDIO_WATCHDOG
3759 // create and start the watchdog
3760 mAudioWatchdog = new AudioWatchdog();
3761 mAudioWatchdog->setDump(&mAudioWatchdogDump);
3762 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
3763 tid = mAudioWatchdog->getTid();
Eric Laurent72e3f392015-05-20 14:43:50 -07003764 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
Eric Laurent81784c32012-11-19 14:55:58 -08003765#endif
3766
Eric Laurent81784c32012-11-19 14:55:58 -08003767 }
3768
3769 switch (kUseFastMixer) {
3770 case FastMixer_Never:
3771 case FastMixer_Dynamic:
3772 mNormalSink = mOutputSink;
3773 break;
3774 case FastMixer_Always:
3775 mNormalSink = mPipeSink;
3776 break;
3777 case FastMixer_Static:
3778 mNormalSink = initFastMixer ? mPipeSink : mOutputSink;
3779 break;
3780 }
3781}
3782
3783AudioFlinger::MixerThread::~MixerThread()
3784{
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003785 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003786 FastMixerStateQueue *sq = mFastMixer->sq();
3787 FastMixerState *state = sq->begin();
3788 if (state->mCommand == FastMixerState::COLD_IDLE) {
3789 int32_t old = android_atomic_inc(&mFastMixerFutex);
3790 if (old == -1) {
Elliott Hughesee499292014-05-21 17:55:51 -07003791 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
Eric Laurent81784c32012-11-19 14:55:58 -08003792 }
3793 }
3794 state->mCommand = FastMixerState::EXIT;
3795 sq->end();
3796 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3797 mFastMixer->join();
3798 // Though the fast mixer thread has exited, it's state queue is still valid.
3799 // We'll use that extract the final state which contains one remaining fast track
3800 // corresponding to our sub-mix.
3801 state = sq->begin();
3802 ALOG_ASSERT(state->mTrackMask == 1);
3803 FastTrack *fastTrack = &state->mFastTracks[0];
3804 ALOG_ASSERT(fastTrack->mBufferProvider != NULL);
3805 delete fastTrack->mBufferProvider;
3806 sq->end(false /*didModify*/);
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003807 mFastMixer.clear();
Eric Laurent81784c32012-11-19 14:55:58 -08003808#ifdef AUDIO_WATCHDOG
3809 if (mAudioWatchdog != 0) {
3810 mAudioWatchdog->requestExit();
3811 mAudioWatchdog->requestExitAndWait();
3812 mAudioWatchdog.clear();
3813 }
3814#endif
3815 }
Glenn Kasten9e58b552013-01-18 15:09:48 -08003816 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter);
Eric Laurent81784c32012-11-19 14:55:58 -08003817 delete mAudioMixer;
3818}
3819
3820
3821uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
3822{
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003823 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003824 MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
3825 latency += (pipe->getAvgFrames() * 1000) / mSampleRate;
3826 }
3827 return latency;
3828}
3829
3830
3831void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
3832{
3833 PlaybackThread::threadLoop_removeTracks(tracksToRemove);
3834}
3835
Eric Laurentbfb1b832013-01-07 09:53:42 -08003836ssize_t AudioFlinger::MixerThread::threadLoop_write()
Eric Laurent81784c32012-11-19 14:55:58 -08003837{
3838 // FIXME we should only do one push per cycle; confirm this is true
3839 // Start the fast mixer if it's not already running
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003840 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003841 FastMixerStateQueue *sq = mFastMixer->sq();
3842 FastMixerState *state = sq->begin();
3843 if (state->mCommand != FastMixerState::MIX_WRITE &&
3844 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) {
3845 if (state->mCommand == FastMixerState::COLD_IDLE) {
Eric Laurenta2ab4502015-09-09 12:25:51 -07003846
3847 // FIXME workaround for first HAL write being CPU bound on some devices
3848 ATRACE_BEGIN("write");
3849 mOutput->write((char *)mSinkBuffer, 0);
3850 ATRACE_END();
3851
Eric Laurent81784c32012-11-19 14:55:58 -08003852 int32_t old = android_atomic_inc(&mFastMixerFutex);
3853 if (old == -1) {
Elliott Hughesee499292014-05-21 17:55:51 -07003854 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
Eric Laurent81784c32012-11-19 14:55:58 -08003855 }
3856#ifdef AUDIO_WATCHDOG
3857 if (mAudioWatchdog != 0) {
3858 mAudioWatchdog->resume();
3859 }
3860#endif
3861 }
3862 state->mCommand = FastMixerState::MIX_WRITE;
Glenn Kastend797a9d2015-03-02 14:19:25 -08003863#ifdef FAST_THREAD_STATISTICS
Glenn Kasten4182c4e2013-07-15 14:45:07 -07003864 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
Glenn Kastenfbdb2ac2015-03-02 14:47:19 -08003865 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN);
Glenn Kastend797a9d2015-03-02 14:19:25 -08003866#endif
Eric Laurent81784c32012-11-19 14:55:58 -08003867 sq->end();
3868 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED);
3869 if (kUseFastMixer == FastMixer_Dynamic) {
3870 mNormalSink = mPipeSink;
3871 }
3872 } else {
3873 sq->end(false /*didModify*/);
3874 }
3875 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08003876 return PlaybackThread::threadLoop_write();
Eric Laurent81784c32012-11-19 14:55:58 -08003877}
3878
3879void AudioFlinger::MixerThread::threadLoop_standby()
3880{
3881 // Idle the fast mixer if it's currently running
Glenn Kasten4d23ca32014-05-13 10:39:51 -07003882 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003883 FastMixerStateQueue *sq = mFastMixer->sq();
3884 FastMixerState *state = sq->begin();
3885 if (!(state->mCommand & FastMixerState::IDLE)) {
3886 state->mCommand = FastMixerState::COLD_IDLE;
3887 state->mColdFutexAddr = &mFastMixerFutex;
3888 state->mColdGen++;
3889 mFastMixerFutex = 0;
3890 sq->end();
3891 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
3892 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED);
3893 if (kUseFastMixer == FastMixer_Dynamic) {
3894 mNormalSink = mOutputSink;
3895 }
3896#ifdef AUDIO_WATCHDOG
3897 if (mAudioWatchdog != 0) {
3898 mAudioWatchdog->pause();
3899 }
3900#endif
3901 } else {
3902 sq->end(false /*didModify*/);
3903 }
3904 }
3905 PlaybackThread::threadLoop_standby();
3906}
3907
Eric Laurentbfb1b832013-01-07 09:53:42 -08003908bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
3909{
3910 return false;
3911}
3912
3913bool AudioFlinger::PlaybackThread::shouldStandby_l()
3914{
3915 return !mStandby;
3916}
3917
3918bool AudioFlinger::PlaybackThread::waitingAsyncCallback()
3919{
3920 Mutex::Autolock _l(mLock);
3921 return waitingAsyncCallback_l();
3922}
3923
Eric Laurent81784c32012-11-19 14:55:58 -08003924// shared by MIXER and DIRECT, overridden by DUPLICATING
3925void AudioFlinger::PlaybackThread::threadLoop_standby()
3926{
3927 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
Phil Burk062e67a2015-02-11 13:40:50 -08003928 mOutput->standby();
Eric Laurentbfb1b832013-01-07 09:53:42 -08003929 if (mUseAsyncWrite != 0) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07003930 // discard any pending drain or write ack by incrementing sequence
3931 mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
3932 mDrainSequence = (mDrainSequence + 2) & ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08003933 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07003934 mCallbackThread->setWriteBlocked(mWriteAckSequence);
3935 mCallbackThread->setDraining(mDrainSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08003936 }
Eric Laurentd1f69b02014-12-15 14:33:13 -08003937 mHwPaused = false;
Eric Laurent81784c32012-11-19 14:55:58 -08003938}
3939
Haynes Mathew George4c6a4332014-01-15 12:31:39 -08003940void AudioFlinger::PlaybackThread::onAddNewTrack_l()
3941{
3942 ALOGV("signal playback thread");
3943 broadcast_l();
3944}
3945
Haynes Mathew George4527b9e2016-07-07 19:54:17 -07003946void AudioFlinger::PlaybackThread::onAsyncError()
3947{
3948 for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
3949 invalidateTracks((audio_stream_type_t)i);
3950 }
3951}
3952
Eric Laurent81784c32012-11-19 14:55:58 -08003953void AudioFlinger::MixerThread::threadLoop_mix()
3954{
Eric Laurent81784c32012-11-19 14:55:58 -08003955 // mix buffers...
Glenn Kastend79072e2016-01-06 08:41:20 -08003956 mAudioMixer->process();
Andy Hung25c2dac2014-02-27 14:56:00 -08003957 mCurrentWriteLength = mSinkBufferSize;
Eric Laurent81784c32012-11-19 14:55:58 -08003958 // increase sleep time progressively when application underrun condition clears.
3959 // Only increase sleep time if the mixer is ready for two consecutive times to avoid
3960 // that a steady state of alternating ready/not ready conditions keeps the sleep time
3961 // such that we would underrun the audio HAL.
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003962 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) {
Eric Laurent81784c32012-11-19 14:55:58 -08003963 sleepTimeShift--;
3964 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003965 mSleepTimeUs = 0;
3966 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
Eric Laurent81784c32012-11-19 14:55:58 -08003967 //TODO: delay standby when effects have a tail
Glenn Kasten4c053ea2014-09-28 14:41:07 -07003968
Eric Laurent81784c32012-11-19 14:55:58 -08003969}
3970
3971void AudioFlinger::MixerThread::threadLoop_sleepTime()
3972{
3973 // If no tracks are ready, sleep once for the duration of an output
3974 // buffer size, then write 0s to the output
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003975 if (mSleepTimeUs == 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08003976 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003977 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift;
3978 if (mSleepTimeUs < kMinThreadSleepTimeUs) {
3979 mSleepTimeUs = kMinThreadSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08003980 }
3981 // reduce sleep time in case of consecutive application underruns to avoid
3982 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
3983 // duration we would end up writing less data than needed by the audio HAL if
3984 // the condition persists.
3985 if (sleepTimeShift < kMaxThreadSleepTimeShift) {
3986 sleepTimeShift++;
3987 }
3988 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003989 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08003990 }
3991 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
Andy Hung98ef9782014-03-04 14:46:50 -08003992 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared
3993 // before effects processing or output.
3994 if (mMixerBufferValid) {
3995 memset(mMixerBuffer, 0, mMixerBufferSize);
3996 } else {
3997 memset(mSinkBuffer, 0, mSinkBufferSize);
3998 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07003999 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08004000 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED),
4001 "anticipated start");
4002 }
4003 // TODO add standby time extension fct of effect tail
4004}
4005
4006// prepareTracks_l() must be called with ThreadBase::mLock held
4007AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
4008 Vector< sp<Track> > *tracksToRemove)
4009{
4010
4011 mixer_state mixerStatus = MIXER_IDLE;
4012 // find out which tracks need to be processed
4013 size_t count = mActiveTracks.size();
4014 size_t mixedTracks = 0;
4015 size_t tracksWithEffect = 0;
4016 // counts only _active_ fast tracks
4017 size_t fastTracks = 0;
4018 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset
4019
4020 float masterVolume = mMasterVolume;
4021 bool masterMute = mMasterMute;
4022
4023 if (masterMute) {
4024 masterVolume = 0;
4025 }
4026 // Delegate master volume control to effect in output mix effect chain if needed
4027 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
4028 if (chain != 0) {
4029 uint32_t v = (uint32_t)(masterVolume * (1 << 24));
4030 chain->setVolume_l(&v, &v);
4031 masterVolume = (float)((v + (1 << 23)) >> 24);
4032 chain.clear();
4033 }
4034
4035 // prepare a new state to push
4036 FastMixerStateQueue *sq = NULL;
4037 FastMixerState *state = NULL;
4038 bool didModify = false;
4039 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED;
Glenn Kasten4d23ca32014-05-13 10:39:51 -07004040 if (mFastMixer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08004041 sq = mFastMixer->sq();
4042 state = sq->begin();
4043 }
4044
Andy Hung69aed5f2014-02-25 17:24:40 -08004045 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found.
Andy Hung98ef9782014-03-04 14:46:50 -08004046 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found.
Andy Hung69aed5f2014-02-25 17:24:40 -08004047
Eric Laurent81784c32012-11-19 14:55:58 -08004048 for (size_t i=0 ; i<count ; i++) {
Glenn Kasten9fdcb0a2013-06-26 16:11:36 -07004049 const sp<Track> t = mActiveTracks[i].promote();
Eric Laurent81784c32012-11-19 14:55:58 -08004050 if (t == 0) {
4051 continue;
4052 }
4053
4054 // this const just means the local variable doesn't change
4055 Track* const track = t.get();
4056
4057 // process fast tracks
4058 if (track->isFastTrack()) {
4059
4060 // It's theoretically possible (though unlikely) for a fast track to be created
4061 // and then removed within the same normal mix cycle. This is not a problem, as
4062 // the track never becomes active so it's fast mixer slot is never touched.
4063 // The converse, of removing an (active) track and then creating a new track
4064 // at the identical fast mixer slot within the same normal mix cycle,
4065 // is impossible because the slot isn't marked available until the end of each cycle.
4066 int j = track->mFastIndex;
Glenn Kastendc2c50b2016-04-21 08:13:14 -07004067 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks);
Eric Laurent81784c32012-11-19 14:55:58 -08004068 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j)));
4069 FastTrack *fastTrack = &state->mFastTracks[j];
4070
4071 // Determine whether the track is currently in underrun condition,
4072 // and whether it had a recent underrun.
4073 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j];
4074 FastTrackUnderruns underruns = ftDump->mUnderruns;
4075 uint32_t recentFull = (underruns.mBitFields.mFull -
4076 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK;
4077 uint32_t recentPartial = (underruns.mBitFields.mPartial -
4078 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK;
4079 uint32_t recentEmpty = (underruns.mBitFields.mEmpty -
4080 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK;
4081 uint32_t recentUnderruns = recentPartial + recentEmpty;
4082 track->mObservedUnderruns = underruns;
4083 // don't count underruns that occur while stopping or pausing
4084 // or stopped which can occur when flush() is called while active
Glenn Kasten82aaf942013-07-17 16:05:07 -07004085 if (!(track->isStopping() || track->isPausing() || track->isStopped()) &&
4086 recentUnderruns > 0) {
4087 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun
4088 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount);
Phil Burk2812d9e2016-01-04 10:34:30 -08004089 } else {
4090 track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
Eric Laurent81784c32012-11-19 14:55:58 -08004091 }
4092
4093 // This is similar to the state machine for normal tracks,
4094 // with a few modifications for fast tracks.
4095 bool isActive = true;
4096 switch (track->mState) {
4097 case TrackBase::STOPPING_1:
4098 // track stays active in STOPPING_1 state until first underrun
Eric Laurentbfb1b832013-01-07 09:53:42 -08004099 if (recentUnderruns > 0 || track->isTerminated()) {
Eric Laurent81784c32012-11-19 14:55:58 -08004100 track->mState = TrackBase::STOPPING_2;
4101 }
4102 break;
4103 case TrackBase::PAUSING:
4104 // ramp down is not yet implemented
4105 track->setPaused();
4106 break;
4107 case TrackBase::RESUMING:
4108 // ramp up is not yet implemented
4109 track->mState = TrackBase::ACTIVE;
4110 break;
4111 case TrackBase::ACTIVE:
4112 if (recentFull > 0 || recentPartial > 0) {
4113 // track has provided at least some frames recently: reset retry count
4114 track->mRetryCount = kMaxTrackRetries;
4115 }
4116 if (recentUnderruns == 0) {
4117 // no recent underruns: stay active
4118 break;
4119 }
4120 // there has recently been an underrun of some kind
4121 if (track->sharedBuffer() == 0) {
4122 // were any of the recent underruns "empty" (no frames available)?
4123 if (recentEmpty == 0) {
4124 // no, then ignore the partial underruns as they are allowed indefinitely
4125 break;
4126 }
4127 // there has recently been an "empty" underrun: decrement the retry counter
4128 if (--(track->mRetryCount) > 0) {
4129 break;
4130 }
4131 // indicate to client process that the track was disabled because of underrun;
4132 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08004133 track->disable();
Eric Laurent81784c32012-11-19 14:55:58 -08004134 // remove from active list, but state remains ACTIVE [confusing but true]
4135 isActive = false;
4136 break;
4137 }
4138 // fall through
4139 case TrackBase::STOPPING_2:
4140 case TrackBase::PAUSED:
Eric Laurent81784c32012-11-19 14:55:58 -08004141 case TrackBase::STOPPED:
4142 case TrackBase::FLUSHED: // flush() while active
4143 // Check for presentation complete if track is inactive
4144 // We have consumed all the buffers of this track.
4145 // This would be incomplete if we auto-paused on underrun
4146 {
4147 size_t audioHALFrames =
4148 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08004149 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08004150 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) {
4151 // track stays in active list until presentation is complete
4152 break;
4153 }
4154 }
4155 if (track->isStopping_2()) {
4156 track->mState = TrackBase::STOPPED;
4157 }
4158 if (track->isStopped()) {
4159 // Can't reset directly, as fast mixer is still polling this track
4160 // track->reset();
4161 // So instead mark this track as needing to be reset after push with ack
4162 resetMask |= 1 << i;
4163 }
4164 isActive = false;
4165 break;
4166 case TrackBase::IDLE:
4167 default:
Glenn Kastenadad3d72014-02-21 14:51:43 -08004168 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState);
Eric Laurent81784c32012-11-19 14:55:58 -08004169 }
4170
4171 if (isActive) {
4172 // was it previously inactive?
4173 if (!(state->mTrackMask & (1 << j))) {
4174 ExtendedAudioBufferProvider *eabp = track;
4175 VolumeProvider *vp = track;
4176 fastTrack->mBufferProvider = eabp;
4177 fastTrack->mVolumeProvider = vp;
Eric Laurent81784c32012-11-19 14:55:58 -08004178 fastTrack->mChannelMask = track->mChannelMask;
Andy Hunge8a1ced2014-05-09 15:02:21 -07004179 fastTrack->mFormat = track->mFormat;
Eric Laurent81784c32012-11-19 14:55:58 -08004180 fastTrack->mGeneration++;
4181 state->mTrackMask |= 1 << j;
4182 didModify = true;
4183 // no acknowledgement required for newly active tracks
4184 }
4185 // cache the combined master volume and stream type volume for fast mixer; this
4186 // lacks any synchronization or barrier so VolumeProvider may read a stale value
Glenn Kastene4756fe2012-11-29 13:38:14 -08004187 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume;
Eric Laurent81784c32012-11-19 14:55:58 -08004188 ++fastTracks;
4189 } else {
4190 // was it previously active?
4191 if (state->mTrackMask & (1 << j)) {
4192 fastTrack->mBufferProvider = NULL;
4193 fastTrack->mGeneration++;
4194 state->mTrackMask &= ~(1 << j);
4195 didModify = true;
4196 // If any fast tracks were removed, we must wait for acknowledgement
4197 // because we're about to decrement the last sp<> on those tracks.
4198 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
4199 } else {
Glenn Kastenf7d65ee2015-12-02 13:45:01 -08004200 LOG_ALWAYS_FATAL("fast track %d should have been active; "
4201 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d",
4202 j, track->mState, state->mTrackMask, recentUnderruns,
4203 track->sharedBuffer() != 0);
Eric Laurent81784c32012-11-19 14:55:58 -08004204 }
4205 tracksToRemove->add(track);
4206 // Avoids a misleading display in dumpsys
4207 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL;
4208 }
4209 continue;
4210 }
4211
4212 { // local variable scope to avoid goto warning
4213
4214 audio_track_cblk_t* cblk = track->cblk();
4215
4216 // The first time a track is added we wait
4217 // for all its buffers to be filled before processing it
4218 int name = track->name();
4219 // make sure that we have enough frames to mix one full buffer.
4220 // enforce this condition only once to enable draining the buffer in case the client
4221 // app does not call stop() and relies on underrun to stop:
4222 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
4223 // during last round
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004224 size_t desiredFrames;
Andy Hung8edb8dc2015-03-26 19:13:55 -07004225 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004226 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
Andy Hung8edb8dc2015-03-26 19:13:55 -07004227
4228 desiredFrames = sourceFramesNeededWithTimestretch(
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004229 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed);
Andy Hung8edb8dc2015-03-26 19:13:55 -07004230 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed.
4231 // add frames already consumed but not yet released by the resampler
4232 // because mAudioTrackServerProxy->framesReady() will include these frames
4233 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
4234
Eric Laurent81784c32012-11-19 14:55:58 -08004235 uint32_t minFrames = 1;
4236 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
4237 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004238 minFrames = desiredFrames;
Eric Laurent81784c32012-11-19 14:55:58 -08004239 }
Eric Laurent13e4c962013-12-20 17:36:01 -08004240
4241 size_t framesReady = track->framesReady();
Glenn Kastene7754022014-10-31 12:11:26 -07004242 if (ATRACE_ENABLED()) {
4243 // I wish we had formatted trace names
4244 char traceName[16];
4245 strcpy(traceName, "nRdy");
4246 int name = track->name();
4247 if (AudioMixer::TRACK0 <= name &&
4248 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) {
4249 name -= AudioMixer::TRACK0;
4250 traceName[4] = (name / 10) + '0';
4251 traceName[5] = (name % 10) + '0';
4252 } else {
4253 traceName[4] = '?';
4254 traceName[5] = '?';
4255 }
4256 traceName[6] = '\0';
4257 ATRACE_INT(traceName, framesReady);
4258 }
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004259 if ((framesReady >= minFrames) && track->isReady() &&
Eric Laurent81784c32012-11-19 14:55:58 -08004260 !track->isPaused() && !track->isTerminated())
4261 {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004262 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this);
Eric Laurent81784c32012-11-19 14:55:58 -08004263
4264 mixedTracks++;
4265
Andy Hung69aed5f2014-02-25 17:24:40 -08004266 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means
4267 // there is an effect chain connected to the track
Eric Laurent81784c32012-11-19 14:55:58 -08004268 chain.clear();
Andy Hung69aed5f2014-02-25 17:24:40 -08004269 if (track->mainBuffer() != mSinkBuffer &&
4270 track->mainBuffer() != mMixerBuffer) {
Andy Hung98ef9782014-03-04 14:46:50 -08004271 if (mEffectBufferEnabled) {
4272 mEffectBufferValid = true; // Later can set directly.
4273 }
Eric Laurent81784c32012-11-19 14:55:58 -08004274 chain = getEffectChain_l(track->sessionId());
4275 // Delegate volume control to effect in track effect chain if needed
4276 if (chain != 0) {
4277 tracksWithEffect++;
4278 } else {
4279 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
4280 "session %d",
4281 name, track->sessionId());
4282 }
4283 }
4284
4285
4286 int param = AudioMixer::VOLUME;
4287 if (track->mFillingUpStatus == Track::FS_FILLED) {
4288 // no ramp for the first volume setting
4289 track->mFillingUpStatus = Track::FS_ACTIVE;
4290 if (track->mState == TrackBase::RESUMING) {
4291 track->mState = TrackBase::ACTIVE;
4292 param = AudioMixer::RAMP_VOLUME;
4293 }
4294 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004295 // FIXME should not make a decision based on mServer
4296 } else if (cblk->mServer != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08004297 // If the track is stopped before the first frame was mixed,
4298 // do not apply ramp
4299 param = AudioMixer::RAMP_VOLUME;
4300 }
4301
4302 // compute volume for this track
Andy Hung6be49402014-05-30 10:42:03 -07004303 uint32_t vl, vr; // in U8.24 integer format
4304 float vlf, vrf, vaf; // in [0.0, 1.0] float format
Glenn Kastene4756fe2012-11-29 13:38:14 -08004305 if (track->isPausing() || mStreamTypes[track->streamType()].mute) {
Andy Hung6be49402014-05-30 10:42:03 -07004306 vl = vr = 0;
4307 vlf = vrf = vaf = 0.;
Eric Laurent81784c32012-11-19 14:55:58 -08004308 if (track->isPausing()) {
4309 track->setPaused();
4310 }
4311 } else {
4312
4313 // read original volumes with volume control
4314 float typeVolume = mStreamTypes[track->streamType()].volume;
4315 float v = masterVolume * typeVolume;
Eric Laurent5bba2f62016-03-18 11:14:14 -07004316 sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
Glenn Kastenc56f3422014-03-21 17:53:17 -07004317 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
Andy Hung6be49402014-05-30 10:42:03 -07004318 vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
4319 vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
Eric Laurent81784c32012-11-19 14:55:58 -08004320 // track volumes come from shared memory, so can't be trusted and must be clamped
Glenn Kastenc56f3422014-03-21 17:53:17 -07004321 if (vlf > GAIN_FLOAT_UNITY) {
4322 ALOGV("Track left volume out of range: %.3g", vlf);
4323 vlf = GAIN_FLOAT_UNITY;
Eric Laurent81784c32012-11-19 14:55:58 -08004324 }
Glenn Kastenc56f3422014-03-21 17:53:17 -07004325 if (vrf > GAIN_FLOAT_UNITY) {
4326 ALOGV("Track right volume out of range: %.3g", vrf);
4327 vrf = GAIN_FLOAT_UNITY;
Eric Laurent81784c32012-11-19 14:55:58 -08004328 }
4329 // now apply the master volume and stream type volume
Andy Hung6be49402014-05-30 10:42:03 -07004330 vlf *= v;
4331 vrf *= v;
Eric Laurent81784c32012-11-19 14:55:58 -08004332 // assuming master volume and stream type volume each go up to 1.0,
Andy Hung6be49402014-05-30 10:42:03 -07004333 // then derive vl and vr as U8.24 versions for the effect chain
4334 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT;
4335 vl = (uint32_t) (scaleto8_24 * vlf);
4336 vr = (uint32_t) (scaleto8_24 * vrf);
4337 // vl and vr are now in U8.24 format
Glenn Kastene3aa6592012-12-04 12:22:46 -08004338 uint16_t sendLevel = proxy->getSendLevel_U4_12();
Eric Laurent81784c32012-11-19 14:55:58 -08004339 // send level comes from shared memory and so may be corrupt
4340 if (sendLevel > MAX_GAIN_INT) {
4341 ALOGV("Track send level out of range: %04X", sendLevel);
4342 sendLevel = MAX_GAIN_INT;
4343 }
Andy Hung6be49402014-05-30 10:42:03 -07004344 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel
4345 vaf = v * sendLevel * (1. / MAX_GAIN_INT);
Eric Laurent81784c32012-11-19 14:55:58 -08004346 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08004347
Eric Laurent81784c32012-11-19 14:55:58 -08004348 // Delegate volume control to effect in track effect chain if needed
4349 if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
4350 // Do not ramp volume if volume is controlled by effect
4351 param = AudioMixer::VOLUME;
Bryant Liub6be7f22014-06-12 22:02:41 +08004352 // Update remaining floating point volume levels
4353 vlf = (float)vl / (1 << 24);
4354 vrf = (float)vr / (1 << 24);
Eric Laurent81784c32012-11-19 14:55:58 -08004355 track->mHasVolumeController = true;
4356 } else {
4357 // force no volume ramp when volume controller was just disabled or removed
4358 // from effect chain to avoid volume spike
4359 if (track->mHasVolumeController) {
4360 param = AudioMixer::VOLUME;
4361 }
4362 track->mHasVolumeController = false;
4363 }
4364
Eric Laurent81784c32012-11-19 14:55:58 -08004365 // XXX: these things DON'T need to be done each time
4366 mAudioMixer->setBufferProvider(name, track);
4367 mAudioMixer->enable(name);
4368
Andy Hung6be49402014-05-30 10:42:03 -07004369 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);
4370 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);
4371 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);
Eric Laurent81784c32012-11-19 14:55:58 -08004372 mAudioMixer->setParameter(
4373 name,
4374 AudioMixer::TRACK,
4375 AudioMixer::FORMAT, (void *)track->format());
4376 mAudioMixer->setParameter(
4377 name,
4378 AudioMixer::TRACK,
Kévin PETIT377b2ec2014-02-03 12:35:36 +00004379 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
Andy Hung9a592762014-07-21 21:56:01 -07004380 mAudioMixer->setParameter(
4381 name,
4382 AudioMixer::TRACK,
4383 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask);
Glenn Kastene3aa6592012-12-04 12:22:46 -08004384 // limit track sample rate to 2 x output sample rate, which changes at re-configuration
Andy Hungcd044842014-08-07 11:04:34 -07004385 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004386 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
Glenn Kastene3aa6592012-12-04 12:22:46 -08004387 if (reqSampleRate == 0) {
4388 reqSampleRate = mSampleRate;
4389 } else if (reqSampleRate > maxSampleRate) {
4390 reqSampleRate = maxSampleRate;
4391 }
Eric Laurent81784c32012-11-19 14:55:58 -08004392 mAudioMixer->setParameter(
4393 name,
4394 AudioMixer::RESAMPLE,
4395 AudioMixer::SAMPLE_RATE,
Kévin PETIT377b2ec2014-02-03 12:35:36 +00004396 (void *)(uintptr_t)reqSampleRate);
Andy Hung8edb8dc2015-03-26 19:13:55 -07004397
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004398 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
Andy Hung8edb8dc2015-03-26 19:13:55 -07004399 mAudioMixer->setParameter(
4400 name,
4401 AudioMixer::TIMESTRETCH,
4402 AudioMixer::PLAYBACK_RATE,
Ricardo Garcia5a8a95d2015-04-18 14:47:04 -07004403 &playbackRate);
Andy Hung8edb8dc2015-03-26 19:13:55 -07004404
Andy Hung69aed5f2014-02-25 17:24:40 -08004405 /*
4406 * Select the appropriate output buffer for the track.
4407 *
Andy Hung98ef9782014-03-04 14:46:50 -08004408 * Tracks with effects go into their own effects chain buffer
4409 * and from there into either mEffectBuffer or mSinkBuffer.
Andy Hung69aed5f2014-02-25 17:24:40 -08004410 *
4411 * Other tracks can use mMixerBuffer for higher precision
4412 * channel accumulation. If this buffer is enabled
4413 * (mMixerBufferEnabled true), then selected tracks will accumulate
4414 * into it.
4415 *
4416 */
4417 if (mMixerBufferEnabled
4418 && (track->mainBuffer() == mSinkBuffer
4419 || track->mainBuffer() == mMixerBuffer)) {
4420 mAudioMixer->setParameter(
4421 name,
4422 AudioMixer::TRACK,
Andy Hung78820702014-02-28 16:23:02 -08004423 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat);
Andy Hung69aed5f2014-02-25 17:24:40 -08004424 mAudioMixer->setParameter(
4425 name,
4426 AudioMixer::TRACK,
4427 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer);
4428 // TODO: override track->mainBuffer()?
4429 mMixerBufferValid = true;
4430 } else {
4431 mAudioMixer->setParameter(
4432 name,
4433 AudioMixer::TRACK,
Andy Hung78820702014-02-28 16:23:02 -08004434 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT);
Andy Hung69aed5f2014-02-25 17:24:40 -08004435 mAudioMixer->setParameter(
4436 name,
4437 AudioMixer::TRACK,
4438 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
4439 }
Eric Laurent81784c32012-11-19 14:55:58 -08004440 mAudioMixer->setParameter(
4441 name,
4442 AudioMixer::TRACK,
4443 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
4444
4445 // reset retry count
4446 track->mRetryCount = kMaxTrackRetries;
4447
4448 // If one track is ready, set the mixer ready if:
4449 // - the mixer was not ready during previous round OR
4450 // - no other track is not ready
4451 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY ||
4452 mixerStatus != MIXER_TRACKS_ENABLED) {
4453 mixerStatus = MIXER_TRACKS_READY;
4454 }
4455 } else {
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004456 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
Andy Hung08fb1742015-05-31 23:22:10 -07004457 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)",
4458 track, framesReady, desiredFrames);
Glenn Kasten82aaf942013-07-17 16:05:07 -07004459 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
Phil Burk2812d9e2016-01-04 10:34:30 -08004460 } else {
4461 track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
Glenn Kasten9f80dd22012-12-18 15:57:32 -08004462 }
Phil Burk2812d9e2016-01-04 10:34:30 -08004463
Eric Laurent81784c32012-11-19 14:55:58 -08004464 // clear effect chain input buffer if an active track underruns to avoid sending
4465 // previous audio buffer again to effects
4466 chain = getEffectChain_l(track->sessionId());
4467 if (chain != 0) {
4468 chain->clearInputBuffer();
4469 }
4470
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004471 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this);
Eric Laurent81784c32012-11-19 14:55:58 -08004472 if ((track->sharedBuffer() != 0) || track->isTerminated() ||
4473 track->isStopped() || track->isPaused()) {
4474 // We have consumed all the buffers of this track.
4475 // Remove it from the list of active tracks.
4476 // TODO: use actual buffer filling status instead of latency when available from
4477 // audio HAL
4478 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08004479 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurent81784c32012-11-19 14:55:58 -08004480 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) {
4481 if (track->isStopped()) {
4482 track->reset();
4483 }
4484 tracksToRemove->add(track);
4485 }
4486 } else {
Eric Laurent81784c32012-11-19 14:55:58 -08004487 // No buffers for this track. Give it a few chances to
4488 // fill a buffer, then remove it from active list.
4489 if (--(track->mRetryCount) <= 0) {
Glenn Kastenc9b2e202013-02-26 11:32:32 -08004490 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
Eric Laurent81784c32012-11-19 14:55:58 -08004491 tracksToRemove->add(track);
4492 // indicate to client process that the track was disabled because of underrun;
4493 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08004494 track->disable();
Eric Laurent81784c32012-11-19 14:55:58 -08004495 // If one track is not ready, mark the mixer also not ready if:
4496 // - the mixer was ready during previous round OR
4497 // - no other track is ready
4498 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY ||
4499 mixerStatus != MIXER_TRACKS_READY) {
4500 mixerStatus = MIXER_TRACKS_ENABLED;
4501 }
4502 }
4503 mAudioMixer->disable(name);
4504 }
4505
4506 } // local variable scope to avoid goto warning
Eric Laurent81784c32012-11-19 14:55:58 -08004507
4508 }
4509
4510 // Push the new FastMixer state if necessary
4511 bool pauseAudioWatchdog = false;
4512 if (didModify) {
4513 state->mFastTracksGen++;
4514 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle
4515 if (kUseFastMixer == FastMixer_Dynamic &&
4516 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) {
4517 state->mCommand = FastMixerState::COLD_IDLE;
4518 state->mColdFutexAddr = &mFastMixerFutex;
4519 state->mColdGen++;
4520 mFastMixerFutex = 0;
4521 if (kUseFastMixer == FastMixer_Dynamic) {
4522 mNormalSink = mOutputSink;
4523 }
4524 // If we go into cold idle, need to wait for acknowledgement
4525 // so that fast mixer stops doing I/O.
4526 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED;
4527 pauseAudioWatchdog = true;
4528 }
Eric Laurent81784c32012-11-19 14:55:58 -08004529 }
4530 if (sq != NULL) {
4531 sq->end(didModify);
4532 sq->push(block);
4533 }
4534#ifdef AUDIO_WATCHDOG
4535 if (pauseAudioWatchdog && mAudioWatchdog != 0) {
4536 mAudioWatchdog->pause();
4537 }
4538#endif
4539
4540 // Now perform the deferred reset on fast tracks that have stopped
4541 while (resetMask != 0) {
4542 size_t i = __builtin_ctz(resetMask);
4543 ALOG_ASSERT(i < count);
4544 resetMask &= ~(1 << i);
4545 sp<Track> t = mActiveTracks[i].promote();
4546 if (t == 0) {
4547 continue;
4548 }
4549 Track* track = t.get();
4550 ALOG_ASSERT(track->isFastTrack() && track->isStopped());
4551 track->reset();
4552 }
4553
4554 // remove all the tracks that need to be...
Eric Laurentbfb1b832013-01-07 09:53:42 -08004555 removeTracks_l(*tracksToRemove);
Eric Laurent81784c32012-11-19 14:55:58 -08004556
Eric Laurent97d547d2014-09-02 14:45:53 -07004557 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
4558 mEffectBufferValid = true;
Marco Nelissenac302142014-10-20 13:15:38 -07004559 }
4560
4561 if (mEffectBufferValid) {
Marco Nelissen57088b52014-10-17 16:39:39 -07004562 // as long as there are effects we should clear the effects buffer, to avoid
4563 // passing a non-clean buffer to the effect chain
4564 memset(mEffectBuffer, 0, mEffectBufferSize);
Eric Laurent97d547d2014-09-02 14:45:53 -07004565 }
Andy Hung69aed5f2014-02-25 17:24:40 -08004566 // sink or mix buffer must be cleared if all tracks are connected to an
4567 // effect chain as in this case the mixer will not write to the sink or mix buffer
4568 // and track effects will accumulate into it
Eric Laurentbfb1b832013-01-07 09:53:42 -08004569 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
4570 (mixedTracks == 0 && fastTracks > 0))) {
Eric Laurent81784c32012-11-19 14:55:58 -08004571 // FIXME as a performance optimization, should remember previous zero status
Andy Hung69aed5f2014-02-25 17:24:40 -08004572 if (mMixerBufferValid) {
4573 memset(mMixerBuffer, 0, mMixerBufferSize);
4574 // TODO: In testing, mSinkBuffer below need not be cleared because
4575 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer
4576 // after mixing.
4577 //
4578 // To enforce this guarantee:
4579 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
4580 // (mixedTracks == 0 && fastTracks > 0))
4581 // must imply MIXER_TRACKS_READY.
4582 // Later, we may clear buffers regardless, and skip much of this logic.
4583 }
Andy Hung98ef9782014-03-04 14:46:50 -08004584 // FIXME as a performance optimization, should remember previous zero status
Andy Hung5567aaf2014-07-17 14:00:07 -07004585 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize);
Eric Laurent81784c32012-11-19 14:55:58 -08004586 }
4587
4588 // if any fast tracks, then status is ready
4589 mMixerStatusIgnoringFastTracks = mixerStatus;
4590 if (fastTracks > 0) {
4591 mixerStatus = MIXER_TRACKS_READY;
4592 }
4593 return mixerStatus;
4594}
4595
4596// getTrackName_l() must be called with ThreadBase::mLock held
Andy Hunge8a1ced2014-05-09 15:02:21 -07004597int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask,
Glenn Kastend848eb42016-03-08 13:42:11 -08004598 audio_format_t format, audio_session_t sessionId)
Eric Laurent81784c32012-11-19 14:55:58 -08004599{
Andy Hunge8a1ced2014-05-09 15:02:21 -07004600 return mAudioMixer->getTrackName(channelMask, format, sessionId);
Eric Laurent81784c32012-11-19 14:55:58 -08004601}
4602
4603// deleteTrackName_l() must be called with ThreadBase::mLock held
4604void AudioFlinger::MixerThread::deleteTrackName_l(int name)
4605{
4606 ALOGV("remove track (%d) and delete from mixer", name);
4607 mAudioMixer->deleteTrackName(name);
4608}
4609
Eric Laurent10351942014-05-08 18:49:52 -07004610// checkForNewParameter_l() must be called with ThreadBase::mLock held
4611bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair,
4612 status_t& status)
Eric Laurent81784c32012-11-19 14:55:58 -08004613{
Eric Laurent81784c32012-11-19 14:55:58 -08004614 bool reconfig = false;
Eric Laurent42537be2016-01-08 17:16:42 -08004615 bool a2dpDeviceChanged = false;
Eric Laurent81784c32012-11-19 14:55:58 -08004616
Eric Laurent10351942014-05-08 18:49:52 -07004617 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08004618
Glenn Kastenc05b8d72016-03-24 09:48:17 -07004619 AutoPark<FastMixer> park(mFastMixer);
Eric Laurent81784c32012-11-19 14:55:58 -08004620
Eric Laurent10351942014-05-08 18:49:52 -07004621 AudioParameter param = AudioParameter(keyValuePair);
4622 int value;
4623 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
4624 reconfig = true;
4625 }
4626 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
Andy Hung9a592762014-07-21 21:56:01 -07004627 if (!isValidPcmSinkFormat((audio_format_t) value)) {
Eric Laurent10351942014-05-08 18:49:52 -07004628 status = BAD_VALUE;
4629 } else {
4630 // no need to save value, since it's constant
Eric Laurent81784c32012-11-19 14:55:58 -08004631 reconfig = true;
4632 }
Eric Laurent10351942014-05-08 18:49:52 -07004633 }
4634 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
Andy Hung9a592762014-07-21 21:56:01 -07004635 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) {
Eric Laurent10351942014-05-08 18:49:52 -07004636 status = BAD_VALUE;
4637 } else {
4638 // no need to save value, since it's constant
4639 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08004640 }
Eric Laurent10351942014-05-08 18:49:52 -07004641 }
4642 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
4643 // do not accept frame count changes if tracks are open as the track buffer
4644 // size depends on frame count and correct behavior would not be guaranteed
4645 // if frame count is changed after track creation
4646 if (!mTracks.isEmpty()) {
4647 status = INVALID_OPERATION;
4648 } else {
4649 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08004650 }
Eric Laurent10351942014-05-08 18:49:52 -07004651 }
4652 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
Eric Laurent81784c32012-11-19 14:55:58 -08004653#ifdef ADD_BATTERY_DATA
Eric Laurent10351942014-05-08 18:49:52 -07004654 // when changing the audio output device, call addBatteryData to notify
4655 // the change
4656 if (mOutDevice != value) {
4657 uint32_t params = 0;
4658 // check whether speaker is on
4659 if (value & AUDIO_DEVICE_OUT_SPEAKER) {
4660 params |= IMediaPlayerService::kBatteryDataSpeakerOn;
Eric Laurent81784c32012-11-19 14:55:58 -08004661 }
Eric Laurent10351942014-05-08 18:49:52 -07004662
4663 audio_devices_t deviceWithoutSpeaker
4664 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
4665 // check if any other device (except speaker) is on
Eric Laurent054d9d32015-04-24 08:48:48 -07004666 if (value & deviceWithoutSpeaker) {
Eric Laurent10351942014-05-08 18:49:52 -07004667 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
4668 }
4669
4670 if (params != 0) {
4671 addBatteryData(params);
4672 }
4673 }
Eric Laurent81784c32012-11-19 14:55:58 -08004674#endif
4675
Eric Laurent10351942014-05-08 18:49:52 -07004676 // forward device change to effects that have requested to be
4677 // aware of attached audio device.
4678 if (value != AUDIO_DEVICE_NONE) {
Eric Laurent42537be2016-01-08 17:16:42 -08004679 a2dpDeviceChanged =
4680 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP);
Eric Laurent10351942014-05-08 18:49:52 -07004681 mOutDevice = value;
4682 for (size_t i = 0; i < mEffectChains.size(); i++) {
4683 mEffectChains[i]->setDevice_l(mOutDevice);
Eric Laurent81784c32012-11-19 14:55:58 -08004684 }
4685 }
Eric Laurent10351942014-05-08 18:49:52 -07004686 }
Eric Laurent81784c32012-11-19 14:55:58 -08004687
Eric Laurent10351942014-05-08 18:49:52 -07004688 if (status == NO_ERROR) {
4689 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
4690 keyValuePair.string());
4691 if (!mStandby && status == INVALID_OPERATION) {
Phil Burk062e67a2015-02-11 13:40:50 -08004692 mOutput->standby();
Eric Laurent10351942014-05-08 18:49:52 -07004693 mStandby = true;
4694 mBytesWritten = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08004695 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
Eric Laurent10351942014-05-08 18:49:52 -07004696 keyValuePair.string());
Eric Laurent81784c32012-11-19 14:55:58 -08004697 }
Eric Laurent10351942014-05-08 18:49:52 -07004698 if (status == NO_ERROR && reconfig) {
4699 readOutputParameters_l();
4700 delete mAudioMixer;
4701 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
4702 for (size_t i = 0; i < mTracks.size() ; i++) {
Andy Hunge8a1ced2014-05-09 15:02:21 -07004703 int name = getTrackName_l(mTracks[i]->mChannelMask,
4704 mTracks[i]->mFormat, mTracks[i]->mSessionId);
Eric Laurent10351942014-05-08 18:49:52 -07004705 if (name < 0) {
4706 break;
4707 }
4708 mTracks[i]->mName = name;
4709 }
Eric Laurent73e26b62015-04-27 16:55:58 -07004710 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
Eric Laurent10351942014-05-08 18:49:52 -07004711 }
Eric Laurent81784c32012-11-19 14:55:58 -08004712 }
4713
Eric Laurent42537be2016-01-08 17:16:42 -08004714 return reconfig || a2dpDeviceChanged;
Eric Laurent81784c32012-11-19 14:55:58 -08004715}
4716
4717
4718void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
4719{
Eric Laurent81784c32012-11-19 14:55:58 -08004720 PlaybackThread::dumpInternals(fd, args);
Andy Hung40eb1a12015-06-18 13:42:02 -07004721 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs);
Elliott Hughes87cebad2014-05-22 10:14:43 -07004722 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
Andy Hung2ddee192015-12-18 17:34:44 -08004723 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off");
Eric Laurent81784c32012-11-19 14:55:58 -08004724
4725 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
Glenn Kasten2f90c512015-12-02 11:40:09 -08004726 // while we are dumping it. It may be inconsistent, but it won't mutate!
4727 // This is a large object so we place it on the heap.
4728 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
4729 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState);
4730 copy->dump(fd);
4731 delete copy;
Eric Laurent81784c32012-11-19 14:55:58 -08004732
4733#ifdef STATE_QUEUE_DUMP
4734 // Similar for state queue
4735 StateQueueObserverDump observerCopy = mStateQueueObserverDump;
4736 observerCopy.dump(fd);
4737 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump;
4738 mutatorCopy.dump(fd);
4739#endif
4740
Glenn Kasten46909e72013-02-26 09:20:22 -08004741#ifdef TEE_SINK
Eric Laurent81784c32012-11-19 14:55:58 -08004742 // Write the tee output to a .wav file
4743 dumpTee(fd, mTeeSource, mId);
Glenn Kasten46909e72013-02-26 09:20:22 -08004744#endif
Eric Laurent81784c32012-11-19 14:55:58 -08004745
4746#ifdef AUDIO_WATCHDOG
4747 if (mAudioWatchdog != 0) {
4748 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us
4749 AudioWatchdogDump wdCopy = mAudioWatchdogDump;
4750 wdCopy.dump(fd);
4751 }
4752#endif
4753}
4754
4755uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
4756{
4757 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
4758}
4759
4760uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
4761{
4762 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
4763}
4764
4765void AudioFlinger::MixerThread::cacheParameters_l()
4766{
4767 PlaybackThread::cacheParameters_l();
4768
4769 // FIXME: Relaxed timing because of a certain device that can't meet latency
4770 // Should be reduced to 2x after the vendor fixes the driver issue
4771 // increase threshold again due to low power audio mode. The way this warning
4772 // threshold is calculated and its usefulness should be reconsidered anyway.
4773 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
4774}
4775
4776// ----------------------------------------------------------------------------
4777
4778AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
Eric Laurente93cc032016-05-05 10:15:10 -07004779 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady)
4780 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady)
Eric Laurent81784c32012-11-19 14:55:58 -08004781 // mLeftVolFloat, mRightVolFloat
4782{
4783}
4784
Eric Laurentbfb1b832013-01-07 09:53:42 -08004785AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
4786 AudioStreamOut* output, audio_io_handle_t id, uint32_t device,
Eric Laurente93cc032016-05-05 10:15:10 -07004787 ThreadBase::type_t type, bool systemReady)
4788 : PlaybackThread(audioFlinger, output, id, device, type, systemReady)
Eric Laurentbfb1b832013-01-07 09:53:42 -08004789 // mLeftVolFloat, mRightVolFloat
4790{
4791}
4792
Eric Laurent81784c32012-11-19 14:55:58 -08004793AudioFlinger::DirectOutputThread::~DirectOutputThread()
4794{
4795}
4796
Eric Laurentbfb1b832013-01-07 09:53:42 -08004797void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack)
4798{
Eric Laurentbfb1b832013-01-07 09:53:42 -08004799 float left, right;
4800
4801 if (mMasterMute || mStreamTypes[track->streamType()].mute) {
4802 left = right = 0;
4803 } else {
4804 float typeVolume = mStreamTypes[track->streamType()].volume;
4805 float v = mMasterVolume * typeVolume;
Eric Laurent5bba2f62016-03-18 11:14:14 -07004806 sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
Glenn Kastenc56f3422014-03-21 17:53:17 -07004807 gain_minifloat_packed_t vlr = proxy->getVolumeLR();
4808 left = float_from_gain(gain_minifloat_unpack_left(vlr));
4809 if (left > GAIN_FLOAT_UNITY) {
4810 left = GAIN_FLOAT_UNITY;
4811 }
4812 left *= v;
4813 right = float_from_gain(gain_minifloat_unpack_right(vlr));
4814 if (right > GAIN_FLOAT_UNITY) {
4815 right = GAIN_FLOAT_UNITY;
4816 }
4817 right *= v;
Eric Laurentbfb1b832013-01-07 09:53:42 -08004818 }
4819
4820 if (lastTrack) {
4821 if (left != mLeftVolFloat || right != mRightVolFloat) {
4822 mLeftVolFloat = left;
4823 mRightVolFloat = right;
4824
4825 // Convert volumes from float to 8.24
4826 uint32_t vl = (uint32_t)(left * (1 << 24));
4827 uint32_t vr = (uint32_t)(right * (1 << 24));
4828
4829 // Delegate volume control to effect in track effect chain if needed
4830 // only one effect chain can be present on DirectOutputThread, so if
4831 // there is one, the track is connected to it
4832 if (!mEffectChains.isEmpty()) {
4833 mEffectChains[0]->setVolume_l(&vl, &vr);
4834 left = (float)vl / (1 << 24);
4835 right = (float)vr / (1 << 24);
4836 }
4837 if (mOutput->stream->set_volume) {
4838 mOutput->stream->set_volume(mOutput->stream, left, right);
4839 }
4840 }
4841 }
4842}
4843
Phil Burk43b4dcc2015-06-09 16:53:44 -07004844void AudioFlinger::DirectOutputThread::onAddNewTrack_l()
4845{
4846 sp<Track> previousTrack = mPreviousTrack.promote();
4847 sp<Track> latestTrack = mLatestActiveTrack.promote();
4848
Eric Laurent0f0631e2015-07-06 18:01:25 -07004849 if (previousTrack != 0 && latestTrack != 0) {
4850 if (mType == DIRECT) {
4851 if (previousTrack.get() != latestTrack.get()) {
4852 mFlushPending = true;
4853 }
4854 } else /* mType == OFFLOAD */ {
4855 if (previousTrack->sessionId() != latestTrack->sessionId()) {
4856 mFlushPending = true;
4857 }
4858 }
Phil Burk43b4dcc2015-06-09 16:53:44 -07004859 }
4860 PlaybackThread::onAddNewTrack_l();
4861}
Eric Laurentbfb1b832013-01-07 09:53:42 -08004862
Eric Laurent81784c32012-11-19 14:55:58 -08004863AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
4864 Vector< sp<Track> > *tracksToRemove
4865)
4866{
Eric Laurentd595b7c2013-04-03 17:27:56 -07004867 size_t count = mActiveTracks.size();
Eric Laurent81784c32012-11-19 14:55:58 -08004868 mixer_state mixerStatus = MIXER_IDLE;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004869 bool doHwPause = false;
4870 bool doHwResume = false;
Eric Laurent81784c32012-11-19 14:55:58 -08004871
4872 // find out which tracks need to be processed
Eric Laurentd595b7c2013-04-03 17:27:56 -07004873 for (size_t i = 0; i < count; i++) {
4874 sp<Track> t = mActiveTracks[i].promote();
Eric Laurent81784c32012-11-19 14:55:58 -08004875 // The track died recently
4876 if (t == 0) {
Eric Laurentd595b7c2013-04-03 17:27:56 -07004877 continue;
Eric Laurent81784c32012-11-19 14:55:58 -08004878 }
4879
Phil Burk43b4dcc2015-06-09 16:53:44 -07004880 if (t->isInvalid()) {
4881 ALOGW("An invalidated track shouldn't be in active list");
4882 tracksToRemove->add(t);
4883 continue;
4884 }
4885
Eric Laurent81784c32012-11-19 14:55:58 -08004886 Track* const track = t.get();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07004887#ifdef VERY_VERY_VERBOSE_LOGGING
Eric Laurent81784c32012-11-19 14:55:58 -08004888 audio_track_cblk_t* cblk = track->cblk();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07004889#endif
Eric Laurentfd477972013-10-25 18:10:40 -07004890 // Only consider last track started for volume and mixer state control.
4891 // In theory an older track could underrun and restart after the new one starts
4892 // but as we only care about the transition phase between two tracks on a
4893 // direct output, it is not a problem to ignore the underrun case.
4894 sp<Track> l = mLatestActiveTrack.promote();
4895 bool last = l.get() == track;
Eric Laurent81784c32012-11-19 14:55:58 -08004896
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004897 if (track->isPausing()) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004898 track->setPaused();
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004899 if (mHwSupportsPause && last && !mHwPaused) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004900 doHwPause = true;
4901 mHwPaused = true;
4902 }
4903 tracksToRemove->add(track);
4904 } else if (track->isFlushPending()) {
4905 track->flushAck();
4906 if (last) {
Phil Burk43b4dcc2015-06-09 16:53:44 -07004907 mFlushPending = true;
Eric Laurentd1f69b02014-12-15 14:33:13 -08004908 }
Phil Burk6fc2a7c2015-04-30 16:08:10 -07004909 } else if (track->isResumePending()) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08004910 track->resumeAck();
Eric Laurent3df841a2016-07-15 15:15:40 -07004911 if (last) {
4912 mLeftVolFloat = mRightVolFloat = -1.0;
4913 if (mHwPaused) {
4914 doHwResume = true;
4915 mHwPaused = false;
4916 }
Eric Laurentd1f69b02014-12-15 14:33:13 -08004917 }
4918 }
4919
Eric Laurent81784c32012-11-19 14:55:58 -08004920 // The first time a track is added we wait
Phil Burk99adee32014-12-10 16:46:30 -08004921 // for all its buffers to be filled before processing it.
4922 // Allow draining the buffer in case the client
4923 // app does not call stop() and relies on underrun to stop:
4924 // hence the test on (track->mRetryCount > 1).
4925 // If retryCount<=1 then track is about to underrun and be removed.
Phil Burkca5e6142015-07-14 09:42:29 -07004926 // Do not use a high threshold for compressed audio.
Eric Laurent81784c32012-11-19 14:55:58 -08004927 uint32_t minFrames;
Phil Burk99adee32014-12-10 16:46:30 -08004928 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()
Phil Burkfdb3c072016-02-09 10:47:02 -08004929 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08004930 minFrames = mNormalFrameCount;
4931 } else {
4932 minFrames = 1;
4933 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08004934
Eric Laurentab5cdba2014-06-09 17:22:27 -07004935 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
4936 !track->isStopping_2() && !track->isStopped())
Eric Laurent81784c32012-11-19 14:55:58 -08004937 {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07004938 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer);
Eric Laurent81784c32012-11-19 14:55:58 -08004939
4940 if (track->mFillingUpStatus == Track::FS_FILLED) {
4941 track->mFillingUpStatus = Track::FS_ACTIVE;
Eric Laurent3df841a2016-07-15 15:15:40 -07004942 if (last) {
4943 // make sure processVolume_l() will apply new volume even if 0
4944 mLeftVolFloat = mRightVolFloat = -1.0;
4945 }
Eric Laurentd1f69b02014-12-15 14:33:13 -08004946 if (!mHwSupportsPause) {
4947 track->resumeAck();
Eric Laurent81784c32012-11-19 14:55:58 -08004948 }
4949 }
4950
4951 // compute volume for this track
Eric Laurentbfb1b832013-01-07 09:53:42 -08004952 processVolume_l(track, last);
4953 if (last) {
Phil Burk43b4dcc2015-06-09 16:53:44 -07004954 sp<Track> previousTrack = mPreviousTrack.promote();
4955 if (previousTrack != 0) {
4956 if (track != previousTrack.get()) {
4957 // Flush any data still being written from last track
4958 mBytesRemaining = 0;
Eric Laurent0f0631e2015-07-06 18:01:25 -07004959 // Invalidate previous track to force a seek when resuming.
4960 previousTrack->invalidate();
Phil Burk43b4dcc2015-06-09 16:53:44 -07004961 }
4962 }
4963 mPreviousTrack = track;
4964
Eric Laurentd595b7c2013-04-03 17:27:56 -07004965 // reset retry count
4966 track->mRetryCount = kMaxTrackRetriesDirect;
4967 mActiveTrack = t;
4968 mixerStatus = MIXER_TRACKS_READY;
Eric Laurent5cff4032015-05-26 13:49:58 -07004969 if (mHwPaused) {
Eric Laurent0f7b5f22014-12-19 10:43:21 -08004970 doHwResume = true;
4971 mHwPaused = false;
4972 }
Eric Laurentd595b7c2013-04-03 17:27:56 -07004973 }
Eric Laurent81784c32012-11-19 14:55:58 -08004974 } else {
Eric Laurentd595b7c2013-04-03 17:27:56 -07004975 // clear effect chain input buffer if the last active track started underruns
4976 // to avoid sending previous audio buffer again to effects
Eric Laurentfd477972013-10-25 18:10:40 -07004977 if (!mEffectChains.isEmpty() && last) {
Eric Laurent81784c32012-11-19 14:55:58 -08004978 mEffectChains[0]->clearInputBuffer();
4979 }
Eric Laurentab5cdba2014-06-09 17:22:27 -07004980 if (track->isStopping_1()) {
4981 track->mState = TrackBase::STOPPING_2;
Eric Laurentb369caf2015-03-30 20:51:47 -07004982 if (last && mHwPaused) {
4983 doHwResume = true;
4984 mHwPaused = false;
4985 }
Eric Laurentab5cdba2014-06-09 17:22:27 -07004986 }
4987 if ((track->sharedBuffer() != 0) || track->isStopped() ||
4988 track->isStopping_2() || track->isPaused()) {
Eric Laurent81784c32012-11-19 14:55:58 -08004989 // We have consumed all the buffers of this track.
4990 // Remove it from the list of active tracks.
Eric Laurentab5cdba2014-06-09 17:22:27 -07004991 size_t audioHALFrames;
Phil Burkfdb3c072016-02-09 10:47:02 -08004992 if (audio_has_proportional_frames(mFormat)) {
Eric Laurentab5cdba2014-06-09 17:22:27 -07004993 audioHALFrames = (latency_l() * mSampleRate) / 1000;
4994 } else {
4995 audioHALFrames = 0;
4996 }
4997
Andy Hung818e7a32016-02-16 18:08:07 -08004998 int64_t framesWritten = mBytesWritten / mFrameSize;
Eric Laurentfd477972013-10-25 18:10:40 -07004999 if (mStandby || !last ||
5000 track->presentationComplete(framesWritten, audioHALFrames)) {
Eric Laurentab5cdba2014-06-09 17:22:27 -07005001 if (track->isStopping_2()) {
5002 track->mState = TrackBase::STOPPED;
5003 }
Eric Laurent81784c32012-11-19 14:55:58 -08005004 if (track->isStopped()) {
5005 track->reset();
5006 }
Eric Laurentd595b7c2013-04-03 17:27:56 -07005007 tracksToRemove->add(track);
Eric Laurent81784c32012-11-19 14:55:58 -08005008 }
5009 } else {
5010 // No buffers for this track. Give it a few chances to
5011 // fill a buffer, then remove it from active list.
Eric Laurentd595b7c2013-04-03 17:27:56 -07005012 // Only consider last track started for mixer state control
Eric Laurent81784c32012-11-19 14:55:58 -08005013 if (--(track->mRetryCount) <= 0) {
5014 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
Eric Laurentd595b7c2013-04-03 17:27:56 -07005015 tracksToRemove->add(track);
Eric Laurenta23f17a2013-11-05 18:22:08 -08005016 // indicate to client process that the track was disabled because of underrun;
5017 // it will then automatically call start() when data is available
Eric Laurent4d231dc2016-03-11 18:38:23 -08005018 track->disable();
Eric Laurentbfb1b832013-01-07 09:53:42 -08005019 } else if (last) {
Phil Burkca5e6142015-07-14 09:42:29 -07005020 ALOGW("pause because of UNDERRUN, framesReady = %zu,"
5021 "minFrames = %u, mFormat = %#x",
5022 track->framesReady(), minFrames, mFormat);
Eric Laurent81784c32012-11-19 14:55:58 -08005023 mixerStatus = MIXER_TRACKS_ENABLED;
Eric Laurent5cff4032015-05-26 13:49:58 -07005024 if (mHwSupportsPause && !mHwPaused && !mStandby) {
Eric Laurent0f7b5f22014-12-19 10:43:21 -08005025 doHwPause = true;
5026 mHwPaused = true;
5027 }
Eric Laurent81784c32012-11-19 14:55:58 -08005028 }
5029 }
5030 }
5031 }
5032
Eric Laurentd1f69b02014-12-15 14:33:13 -08005033 // if an active track did not command a flush, check for pending flush on stopped tracks
Phil Burk43b4dcc2015-06-09 16:53:44 -07005034 if (!mFlushPending) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08005035 for (size_t i = 0; i < mTracks.size(); i++) {
5036 if (mTracks[i]->isFlushPending()) {
5037 mTracks[i]->flushAck();
Phil Burk43b4dcc2015-06-09 16:53:44 -07005038 mFlushPending = true;
Eric Laurentd1f69b02014-12-15 14:33:13 -08005039 }
5040 }
5041 }
5042
5043 // make sure the pause/flush/resume sequence is executed in the right order.
5044 // If a flush is pending and a track is active but the HW is not paused, force a HW pause
5045 // before flush and then resume HW. This can happen in case of pause/flush/resume
5046 // if resume is received before pause is executed.
5047 if (mHwSupportsPause && !mStandby &&
Phil Burk43b4dcc2015-06-09 16:53:44 -07005048 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08005049 mOutput->stream->pause(mOutput->stream);
5050 }
Phil Burk43b4dcc2015-06-09 16:53:44 -07005051 if (mFlushPending) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08005052 flushHw_l();
5053 }
5054 if (mHwSupportsPause && !mStandby && doHwResume) {
5055 mOutput->stream->resume(mOutput->stream);
5056 }
Eric Laurent81784c32012-11-19 14:55:58 -08005057 // remove all the tracks that need to be...
Eric Laurentbfb1b832013-01-07 09:53:42 -08005058 removeTracks_l(*tracksToRemove);
Eric Laurent81784c32012-11-19 14:55:58 -08005059
5060 return mixerStatus;
5061}
5062
5063void AudioFlinger::DirectOutputThread::threadLoop_mix()
5064{
Eric Laurent81784c32012-11-19 14:55:58 -08005065 size_t frameCount = mFrameCount;
Andy Hung2098f272014-02-27 14:00:06 -08005066 int8_t *curBuf = (int8_t *)mSinkBuffer;
Eric Laurent81784c32012-11-19 14:55:58 -08005067 // output audio to hardware
5068 while (frameCount) {
Glenn Kasten34542ac2013-06-26 11:29:02 -07005069 AudioBufferProvider::Buffer buffer;
Eric Laurent81784c32012-11-19 14:55:58 -08005070 buffer.frameCount = frameCount;
Phil Burk062e67a2015-02-11 13:40:50 -08005071 status_t status = mActiveTrack->getNextBuffer(&buffer);
5072 if (status != NO_ERROR || buffer.raw == NULL) {
Eric Laurent51716182016-02-29 18:00:56 -08005073 // no need to pad with 0 for compressed audio
5074 if (audio_has_proportional_frames(mFormat)) {
5075 memset(curBuf, 0, frameCount * mFrameSize);
5076 }
Eric Laurent81784c32012-11-19 14:55:58 -08005077 break;
5078 }
5079 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
5080 frameCount -= buffer.frameCount;
5081 curBuf += buffer.frameCount * mFrameSize;
5082 mActiveTrack->releaseBuffer(&buffer);
5083 }
Andy Hung2098f272014-02-27 14:00:06 -08005084 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer;
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005085 mSleepTimeUs = 0;
5086 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
Eric Laurent81784c32012-11-19 14:55:58 -08005087 mActiveTrack.clear();
Eric Laurent81784c32012-11-19 14:55:58 -08005088}
5089
5090void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
5091{
Eric Laurentd1f69b02014-12-15 14:33:13 -08005092 // do not write to HAL when paused
Eric Laurent0f7b5f22014-12-19 10:43:21 -08005093 if (mHwPaused || (usesHwAvSync() && mStandby)) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005094 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurentd1f69b02014-12-15 14:33:13 -08005095 return;
5096 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005097 if (mSleepTimeUs == 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08005098 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
Eric Laurente93cc032016-05-05 10:15:10 -07005099 mSleepTimeUs = mActiveSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005100 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005101 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005102 }
Phil Burkfdb3c072016-02-09 10:47:02 -08005103 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) {
Andy Hung2098f272014-02-27 14:00:06 -08005104 memset(mSinkBuffer, 0, mFrameCount * mFrameSize);
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005105 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08005106 }
5107}
5108
Eric Laurentd1f69b02014-12-15 14:33:13 -08005109void AudioFlinger::DirectOutputThread::threadLoop_exit()
5110{
5111 {
5112 Mutex::Autolock _l(mLock);
Eric Laurentd1f69b02014-12-15 14:33:13 -08005113 for (size_t i = 0; i < mTracks.size(); i++) {
5114 if (mTracks[i]->isFlushPending()) {
5115 mTracks[i]->flushAck();
Phil Burk43b4dcc2015-06-09 16:53:44 -07005116 mFlushPending = true;
Eric Laurentd1f69b02014-12-15 14:33:13 -08005117 }
5118 }
Phil Burk43b4dcc2015-06-09 16:53:44 -07005119 if (mFlushPending) {
Eric Laurentd1f69b02014-12-15 14:33:13 -08005120 flushHw_l();
5121 }
5122 }
5123 PlaybackThread::threadLoop_exit();
5124}
5125
5126// must be called with thread mutex locked
5127bool AudioFlinger::DirectOutputThread::shouldStandby_l()
5128{
5129 bool trackPaused = false;
Eric Laurentb369caf2015-03-30 20:51:47 -07005130 bool trackStopped = false;
Eric Laurentd1f69b02014-12-15 14:33:13 -08005131
vivek mehta9cd7ad12016-03-17 00:18:29 -07005132 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) {
5133 return !mStandby;
5134 }
5135
Eric Laurentd1f69b02014-12-15 14:33:13 -08005136 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
5137 // after a timeout and we will enter standby then.
5138 if (mTracks.size() > 0) {
5139 trackPaused = mTracks[mTracks.size() - 1]->isPaused();
Eric Laurentb369caf2015-03-30 20:51:47 -07005140 trackStopped = mTracks[mTracks.size() - 1]->isStopped() ||
5141 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE;
Eric Laurentd1f69b02014-12-15 14:33:13 -08005142 }
5143
Eric Laurent5cff4032015-05-26 13:49:58 -07005144 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped));
Eric Laurentd1f69b02014-12-15 14:33:13 -08005145}
5146
Eric Laurent81784c32012-11-19 14:55:58 -08005147// getTrackName_l() must be called with ThreadBase::mLock held
Glenn Kasten0f11b512014-01-31 16:18:54 -08005148int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
Glenn Kastend848eb42016-03-08 13:42:11 -08005149 audio_format_t format __unused, audio_session_t sessionId __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08005150{
5151 return 0;
5152}
5153
5154// deleteTrackName_l() must be called with ThreadBase::mLock held
Glenn Kasten0f11b512014-01-31 16:18:54 -08005155void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08005156{
5157}
5158
Eric Laurent10351942014-05-08 18:49:52 -07005159// checkForNewParameter_l() must be called with ThreadBase::mLock held
5160bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair,
5161 status_t& status)
Eric Laurent81784c32012-11-19 14:55:58 -08005162{
5163 bool reconfig = false;
Eric Laurent42537be2016-01-08 17:16:42 -08005164 bool a2dpDeviceChanged = false;
Eric Laurent81784c32012-11-19 14:55:58 -08005165
Eric Laurent10351942014-05-08 18:49:52 -07005166 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08005167
Eric Laurent10351942014-05-08 18:49:52 -07005168 AudioParameter param = AudioParameter(keyValuePair);
5169 int value;
5170 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
5171 // forward device change to effects that have requested to be
5172 // aware of attached audio device.
5173 if (value != AUDIO_DEVICE_NONE) {
Eric Laurent42537be2016-01-08 17:16:42 -08005174 a2dpDeviceChanged =
5175 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP);
Eric Laurent10351942014-05-08 18:49:52 -07005176 mOutDevice = value;
5177 for (size_t i = 0; i < mEffectChains.size(); i++) {
5178 mEffectChains[i]->setDevice_l(mOutDevice);
Glenn Kastenc125f382014-04-11 18:37:33 -07005179 }
5180 }
Eric Laurent81784c32012-11-19 14:55:58 -08005181 }
Eric Laurent10351942014-05-08 18:49:52 -07005182 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
5183 // do not accept frame count changes if tracks are open as the track buffer
5184 // size depends on frame count and correct behavior would not be garantied
5185 // if frame count is changed after track creation
5186 if (!mTracks.isEmpty()) {
5187 status = INVALID_OPERATION;
5188 } else {
5189 reconfig = true;
5190 }
5191 }
5192 if (status == NO_ERROR) {
5193 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
5194 keyValuePair.string());
5195 if (!mStandby && status == INVALID_OPERATION) {
Phil Burk062e67a2015-02-11 13:40:50 -08005196 mOutput->standby();
Eric Laurent10351942014-05-08 18:49:52 -07005197 mStandby = true;
5198 mBytesWritten = 0;
5199 status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
5200 keyValuePair.string());
5201 }
5202 if (status == NO_ERROR && reconfig) {
5203 readOutputParameters_l();
Eric Laurent73e26b62015-04-27 16:55:58 -07005204 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
Eric Laurent10351942014-05-08 18:49:52 -07005205 }
5206 }
5207
Eric Laurent42537be2016-01-08 17:16:42 -08005208 return reconfig || a2dpDeviceChanged;
Eric Laurent81784c32012-11-19 14:55:58 -08005209}
5210
5211uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
5212{
5213 uint32_t time;
Phil Burkfdb3c072016-02-09 10:47:02 -08005214 if (audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08005215 time = PlaybackThread::activeSleepTimeUs();
5216 } else {
Eric Laurent51716182016-02-29 18:00:56 -08005217 time = kDirectMinSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005218 }
5219 return time;
5220}
5221
5222uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
5223{
5224 uint32_t time;
Phil Burkfdb3c072016-02-09 10:47:02 -08005225 if (audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08005226 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
5227 } else {
Eric Laurent51716182016-02-29 18:00:56 -08005228 time = kDirectMinSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005229 }
5230 return time;
5231}
5232
5233uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
5234{
5235 uint32_t time;
Phil Burkfdb3c072016-02-09 10:47:02 -08005236 if (audio_has_proportional_frames(mFormat)) {
Eric Laurent81784c32012-11-19 14:55:58 -08005237 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
5238 } else {
Eric Laurent51716182016-02-29 18:00:56 -08005239 time = kDirectMinSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005240 }
5241 return time;
5242}
5243
5244void AudioFlinger::DirectOutputThread::cacheParameters_l()
5245{
5246 PlaybackThread::cacheParameters_l();
5247
5248 // use shorter standby delay as on normal output to release
5249 // hardware resources as soon as possible
Eric Laurentb369caf2015-03-30 20:51:47 -07005250 // no delay on outputs with HW A/V sync
5251 if (usesHwAvSync()) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005252 mStandbyDelayNs = 0;
Phil Burkfdb3c072016-02-09 10:47:02 -08005253 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005254 mStandbyDelayNs = kOffloadStandbyDelayNs;
Eric Laurent5cff4032015-05-26 13:49:58 -07005255 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005256 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2);
Eric Laurent972a1732013-09-04 09:42:59 -07005257 }
Eric Laurent81784c32012-11-19 14:55:58 -08005258}
5259
Eric Laurente659ef42014-09-29 13:06:46 -07005260void AudioFlinger::DirectOutputThread::flushHw_l()
5261{
Phil Burk062e67a2015-02-11 13:40:50 -08005262 mOutput->flush();
Eric Laurentd1f69b02014-12-15 14:33:13 -08005263 mHwPaused = false;
Phil Burk43b4dcc2015-06-09 16:53:44 -07005264 mFlushPending = false;
Eric Laurente659ef42014-09-29 13:06:46 -07005265}
5266
Eric Laurent81784c32012-11-19 14:55:58 -08005267// ----------------------------------------------------------------------------
5268
Eric Laurentbfb1b832013-01-07 09:53:42 -08005269AudioFlinger::AsyncCallbackThread::AsyncCallbackThread(
Eric Laurent4de95592013-09-26 15:28:21 -07005270 const wp<AudioFlinger::PlaybackThread>& playbackThread)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005271 : Thread(false /*canCallJava*/),
Eric Laurent4de95592013-09-26 15:28:21 -07005272 mPlaybackThread(playbackThread),
Eric Laurent3b4529e2013-09-05 18:09:19 -07005273 mWriteAckSequence(0),
Haynes Mathew George4527b9e2016-07-07 19:54:17 -07005274 mDrainSequence(0),
5275 mAsyncError(false)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005276{
5277}
5278
5279AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread()
5280{
5281}
5282
5283void AudioFlinger::AsyncCallbackThread::onFirstRef()
5284{
5285 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO);
5286}
5287
5288bool AudioFlinger::AsyncCallbackThread::threadLoop()
5289{
5290 while (!exitPending()) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07005291 uint32_t writeAckSequence;
5292 uint32_t drainSequence;
Haynes Mathew George4527b9e2016-07-07 19:54:17 -07005293 bool asyncError;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005294
5295 {
5296 Mutex::Autolock _l(mLock);
Haynes Mathew George24a325d2013-12-03 21:26:02 -08005297 while (!((mWriteAckSequence & 1) ||
5298 (mDrainSequence & 1) ||
Haynes Mathew George4527b9e2016-07-07 19:54:17 -07005299 mAsyncError ||
Haynes Mathew George24a325d2013-12-03 21:26:02 -08005300 exitPending())) {
5301 mWaitWorkCV.wait(mLock);
5302 }
5303
Eric Laurentbfb1b832013-01-07 09:53:42 -08005304 if (exitPending()) {
5305 break;
5306 }
Eric Laurent3b4529e2013-09-05 18:09:19 -07005307 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d",
5308 mWriteAckSequence, mDrainSequence);
5309 writeAckSequence = mWriteAckSequence;
5310 mWriteAckSequence &= ~1;
5311 drainSequence = mDrainSequence;
5312 mDrainSequence &= ~1;
Haynes Mathew George4527b9e2016-07-07 19:54:17 -07005313 asyncError = mAsyncError;
5314 mAsyncError = false;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005315 }
5316 {
Eric Laurent4de95592013-09-26 15:28:21 -07005317 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote();
5318 if (playbackThread != 0) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07005319 if (writeAckSequence & 1) {
Eric Laurent4de95592013-09-26 15:28:21 -07005320 playbackThread->resetWriteBlocked(writeAckSequence >> 1);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005321 }
Eric Laurent3b4529e2013-09-05 18:09:19 -07005322 if (drainSequence & 1) {
Eric Laurent4de95592013-09-26 15:28:21 -07005323 playbackThread->resetDraining(drainSequence >> 1);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005324 }
Haynes Mathew George4527b9e2016-07-07 19:54:17 -07005325 if (asyncError) {
5326 playbackThread->onAsyncError();
5327 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005328 }
5329 }
5330 }
5331 return false;
5332}
5333
5334void AudioFlinger::AsyncCallbackThread::exit()
5335{
5336 ALOGV("AsyncCallbackThread::exit");
5337 Mutex::Autolock _l(mLock);
5338 requestExit();
5339 mWaitWorkCV.broadcast();
5340}
5341
Eric Laurent3b4529e2013-09-05 18:09:19 -07005342void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005343{
5344 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07005345 // bit 0 is cleared
5346 mWriteAckSequence = sequence << 1;
5347}
5348
5349void AudioFlinger::AsyncCallbackThread::resetWriteBlocked()
5350{
5351 Mutex::Autolock _l(mLock);
5352 // ignore unexpected callbacks
5353 if (mWriteAckSequence & 2) {
5354 mWriteAckSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005355 mWaitWorkCV.signal();
5356 }
5357}
5358
Eric Laurent3b4529e2013-09-05 18:09:19 -07005359void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005360{
5361 Mutex::Autolock _l(mLock);
Eric Laurent3b4529e2013-09-05 18:09:19 -07005362 // bit 0 is cleared
5363 mDrainSequence = sequence << 1;
5364}
5365
5366void AudioFlinger::AsyncCallbackThread::resetDraining()
5367{
5368 Mutex::Autolock _l(mLock);
5369 // ignore unexpected callbacks
5370 if (mDrainSequence & 2) {
5371 mDrainSequence |= 1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005372 mWaitWorkCV.signal();
5373 }
5374}
5375
Haynes Mathew George4527b9e2016-07-07 19:54:17 -07005376void AudioFlinger::AsyncCallbackThread::setAsyncError()
5377{
5378 Mutex::Autolock _l(mLock);
5379 mAsyncError = true;
5380 mWaitWorkCV.signal();
5381}
5382
Eric Laurentbfb1b832013-01-07 09:53:42 -08005383
5384// ----------------------------------------------------------------------------
5385AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
Eric Laurente93cc032016-05-05 10:15:10 -07005386 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady)
5387 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady),
Andy Hungf8044752016-07-27 14:58:11 -07005388 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true),
5389 mOffloadUnderrunPosition(~0LL)
Eric Laurentbfb1b832013-01-07 09:53:42 -08005390{
Eric Laurentfd477972013-10-25 18:10:40 -07005391 //FIXME: mStandby should be set to true by ThreadBase constructor
5392 mStandby = true;
Eric Laurent64667972016-03-30 18:19:46 -07005393 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005394}
5395
Eric Laurentbfb1b832013-01-07 09:53:42 -08005396void AudioFlinger::OffloadThread::threadLoop_exit()
5397{
5398 if (mFlushPending || mHwPaused) {
5399 // If a flush is pending or track was paused, just discard buffered data
5400 flushHw_l();
5401 } else {
5402 mMixerStatus = MIXER_DRAIN_ALL;
5403 threadLoop_drain();
5404 }
Uday Gupta56604aa2014-05-13 11:19:17 -07005405 if (mUseAsyncWrite) {
5406 ALOG_ASSERT(mCallbackThread != 0);
5407 mCallbackThread->exit();
5408 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005409 PlaybackThread::threadLoop_exit();
5410}
5411
5412AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l(
5413 Vector< sp<Track> > *tracksToRemove
5414)
5415{
Eric Laurentbfb1b832013-01-07 09:53:42 -08005416 size_t count = mActiveTracks.size();
5417
5418 mixer_state mixerStatus = MIXER_IDLE;
Eric Laurent972a1732013-09-04 09:42:59 -07005419 bool doHwPause = false;
5420 bool doHwResume = false;
5421
Glenn Kastenc42e9b42016-03-21 11:35:03 -07005422 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count);
Eric Laurentede6c3b2013-09-19 14:37:46 -07005423
Eric Laurentbfb1b832013-01-07 09:53:42 -08005424 // find out which tracks need to be processed
5425 for (size_t i = 0; i < count; i++) {
5426 sp<Track> t = mActiveTracks[i].promote();
5427 // The track died recently
5428 if (t == 0) {
5429 continue;
5430 }
5431 Track* const track = t.get();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07005432#ifdef VERY_VERY_VERBOSE_LOGGING
Eric Laurentbfb1b832013-01-07 09:53:42 -08005433 audio_track_cblk_t* cblk = track->cblk();
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07005434#endif
Eric Laurentfd477972013-10-25 18:10:40 -07005435 // Only consider last track started for volume and mixer state control.
5436 // In theory an older track could underrun and restart after the new one starts
5437 // but as we only care about the transition phase between two tracks on a
5438 // direct output, it is not a problem to ignore the underrun case.
5439 sp<Track> l = mLatestActiveTrack.promote();
5440 bool last = l.get() == track;
5441
Haynes Mathew George7844f672014-01-15 12:32:55 -08005442 if (track->isInvalid()) {
5443 ALOGW("An invalidated track shouldn't be in active list");
5444 tracksToRemove->add(track);
5445 continue;
5446 }
5447
5448 if (track->mState == TrackBase::IDLE) {
5449 ALOGW("An idle track shouldn't be in active list");
5450 continue;
5451 }
5452
Eric Laurentbfb1b832013-01-07 09:53:42 -08005453 if (track->isPausing()) {
5454 track->setPaused();
5455 if (last) {
Eric Laurent5cff4032015-05-26 13:49:58 -07005456 if (mHwSupportsPause && !mHwPaused) {
Eric Laurent972a1732013-09-04 09:42:59 -07005457 doHwPause = true;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005458 mHwPaused = true;
5459 }
5460 // If we were part way through writing the mixbuffer to
5461 // the HAL we must save this until we resume
5462 // BUG - this will be wrong if a different track is made active,
5463 // in that case we want to discard the pending data in the
5464 // mixbuffer and tell the client to present it again when the
5465 // track is resumed
5466 mPausedWriteLength = mCurrentWriteLength;
5467 mPausedBytesRemaining = mBytesRemaining;
5468 mBytesRemaining = 0; // stop writing
5469 }
5470 tracksToRemove->add(track);
Haynes Mathew George7844f672014-01-15 12:32:55 -08005471 } else if (track->isFlushPending()) {
Eric Laurente93cc032016-05-05 10:15:10 -07005472 if (track->isStopping_1()) {
5473 track->mRetryCount = kMaxTrackStopRetriesOffload;
5474 } else {
5475 track->mRetryCount = kMaxTrackRetriesOffload;
5476 }
Haynes Mathew George7844f672014-01-15 12:32:55 -08005477 track->flushAck();
5478 if (last) {
5479 mFlushPending = true;
5480 }
Haynes Mathew George2d3ca682014-03-07 13:43:49 -08005481 } else if (track->isResumePending()){
5482 track->resumeAck();
5483 if (last) {
5484 if (mPausedBytesRemaining) {
5485 // Need to continue write that was interrupted
5486 mCurrentWriteLength = mPausedWriteLength;
5487 mBytesRemaining = mPausedBytesRemaining;
5488 mPausedBytesRemaining = 0;
5489 }
5490 if (mHwPaused) {
5491 doHwResume = true;
5492 mHwPaused = false;
5493 // threadLoop_mix() will handle the case that we need to
5494 // resume an interrupted write
5495 }
5496 // enable write to audio HAL
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005497 mSleepTimeUs = 0;
Haynes Mathew George2d3ca682014-03-07 13:43:49 -08005498
Eric Laurent3df841a2016-07-15 15:15:40 -07005499 mLeftVolFloat = mRightVolFloat = -1.0;
5500
Haynes Mathew George2d3ca682014-03-07 13:43:49 -08005501 // Do not handle new data in this iteration even if track->framesReady()
5502 mixerStatus = MIXER_TRACKS_ENABLED;
5503 }
5504 } else if (track->framesReady() && track->isReady() &&
Eric Laurent3b4529e2013-09-05 18:09:19 -07005505 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07005506 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005507 if (track->mFillingUpStatus == Track::FS_FILLED) {
5508 track->mFillingUpStatus = Track::FS_ACTIVE;
Eric Laurent3df841a2016-07-15 15:15:40 -07005509 if (last) {
5510 // make sure processVolume_l() will apply new volume even if 0
5511 mLeftVolFloat = mRightVolFloat = -1.0;
5512 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005513 }
5514
5515 if (last) {
Eric Laurentd7e59222013-11-15 12:02:28 -08005516 sp<Track> previousTrack = mPreviousTrack.promote();
5517 if (previousTrack != 0) {
5518 if (track != previousTrack.get()) {
Eric Laurent9da3d952013-11-12 19:25:43 -08005519 // Flush any data still being written from last track
5520 mBytesRemaining = 0;
5521 if (mPausedBytesRemaining) {
5522 // Last track was paused so we also need to flush saved
5523 // mixbuffer state and invalidate track so that it will
5524 // re-submit that unwritten data when it is next resumed
5525 mPausedBytesRemaining = 0;
5526 // Invalidate is a bit drastic - would be more efficient
5527 // to have a flag to tell client that some of the
5528 // previously written data was lost
Eric Laurentd7e59222013-11-15 12:02:28 -08005529 previousTrack->invalidate();
Eric Laurent9da3d952013-11-12 19:25:43 -08005530 }
5531 // flush data already sent to the DSP if changing audio session as audio
5532 // comes from a different source. Also invalidate previous track to force a
5533 // seek when resuming.
Eric Laurentd7e59222013-11-15 12:02:28 -08005534 if (previousTrack->sessionId() != track->sessionId()) {
5535 previousTrack->invalidate();
Eric Laurent9da3d952013-11-12 19:25:43 -08005536 }
5537 }
5538 }
5539 mPreviousTrack = track;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005540 // reset retry count
Eric Laurente93cc032016-05-05 10:15:10 -07005541 if (track->isStopping_1()) {
5542 track->mRetryCount = kMaxTrackStopRetriesOffload;
5543 } else {
5544 track->mRetryCount = kMaxTrackRetriesOffload;
5545 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005546 mActiveTrack = t;
5547 mixerStatus = MIXER_TRACKS_READY;
5548 }
5549 } else {
Glenn Kastenf20e1d82013-07-12 09:45:18 -07005550 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005551 if (track->isStopping_1()) {
Eric Laurente93cc032016-05-05 10:15:10 -07005552 if (--(track->mRetryCount) <= 0) {
5553 // Hardware buffer can hold a large amount of audio so we must
5554 // wait for all current track's data to drain before we say
5555 // that the track is stopped.
5556 if (mBytesRemaining == 0) {
5557 // Only start draining when all data in mixbuffer
5558 // has been written
5559 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2");
5560 track->mState = TrackBase::STOPPING_2; // so presentation completes after
5561 // drain do not drain if no data was ever sent to HAL (mStandby == true)
5562 if (last && !mStandby) {
5563 // do not modify drain sequence if we are already draining. This happens
5564 // when resuming from pause after drain.
5565 if ((mDrainSequence & 1) == 0) {
5566 mSleepTimeUs = 0;
5567 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
5568 mixerStatus = MIXER_DRAIN_TRACK;
5569 mDrainSequence += 2;
5570 }
5571 if (mHwPaused) {
5572 // It is possible to move from PAUSED to STOPPING_1 without
5573 // a resume so we must ensure hardware is running
5574 doHwResume = true;
5575 mHwPaused = false;
5576 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005577 }
5578 }
Eric Laurente93cc032016-05-05 10:15:10 -07005579 } else if (last) {
5580 ALOGV("stopping1 underrun retries left %d", track->mRetryCount);
5581 mixerStatus = MIXER_TRACKS_ENABLED;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005582 }
5583 } else if (track->isStopping_2()) {
Eric Laurent6a51d7e2013-10-17 18:59:26 -07005584 // Drain has completed or we are in standby, signal presentation complete
5585 if (!(mDrainSequence & 1) || !last || mStandby) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08005586 track->mState = TrackBase::STOPPED;
5587 size_t audioHALFrames =
5588 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000;
Andy Hung818e7a32016-02-16 18:08:07 -08005589 int64_t framesWritten =
Phil Burk062e67a2015-02-11 13:40:50 -08005590 mBytesWritten / mOutput->getFrameSize();
Eric Laurentbfb1b832013-01-07 09:53:42 -08005591 track->presentationComplete(framesWritten, audioHALFrames);
5592 track->reset();
5593 tracksToRemove->add(track);
5594 }
5595 } else {
5596 // No buffers for this track. Give it a few chances to
5597 // fill a buffer, then remove it from active list.
5598 if (--(track->mRetryCount) <= 0) {
Andy Hungf8044752016-07-27 14:58:11 -07005599 bool running = false;
5600 if (mOutput->stream->get_presentation_position != nullptr) {
5601 uint64_t position = 0;
5602 struct timespec unused;
5603 // The running check restarts the retry counter at least once.
5604 int ret = mOutput->stream->get_presentation_position(
5605 mOutput->stream, &position, &unused);
5606 if (ret == NO_ERROR && position != mOffloadUnderrunPosition) {
5607 running = true;
5608 mOffloadUnderrunPosition = position;
5609 }
5610 ALOGVV("underrun counter, running(%d): %lld vs %lld", running,
5611 (long long)position, (long long)mOffloadUnderrunPosition);
5612 }
5613 if (running) { // still running, give us more time.
5614 track->mRetryCount = kMaxTrackRetriesOffload;
5615 } else {
5616 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list",
5617 track->name());
5618 tracksToRemove->add(track);
5619 // indicate to client process that the track was disabled because of underrun;
5620 // it will then automatically call start() when data is available
5621 track->disable();
5622 }
Eric Laurentbfb1b832013-01-07 09:53:42 -08005623 } else if (last){
5624 mixerStatus = MIXER_TRACKS_ENABLED;
5625 }
5626 }
5627 }
5628 // compute volume for this track
5629 processVolume_l(track, last);
5630 }
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005631
Eric Laurentea0fade2013-10-04 16:23:48 -07005632 // make sure the pause/flush/resume sequence is executed in the right order.
5633 // If a flush is pending and a track is active but the HW is not paused, force a HW pause
5634 // before flush and then resume HW. This can happen in case of pause/flush/resume
5635 // if resume is received before pause is executed.
Eric Laurentfd477972013-10-25 18:10:40 -07005636 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
Eric Laurent972a1732013-09-04 09:42:59 -07005637 mOutput->stream->pause(mOutput->stream);
5638 }
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005639 if (mFlushPending) {
5640 flushHw_l();
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005641 }
Eric Laurentfd477972013-10-25 18:10:40 -07005642 if (!mStandby && doHwResume) {
Eric Laurent972a1732013-09-04 09:42:59 -07005643 mOutput->stream->resume(mOutput->stream);
5644 }
Eric Laurent6bf9ae22013-08-30 15:12:37 -07005645
Eric Laurentbfb1b832013-01-07 09:53:42 -08005646 // remove all the tracks that need to be...
5647 removeTracks_l(*tracksToRemove);
5648
5649 return mixerStatus;
5650}
5651
Eric Laurentbfb1b832013-01-07 09:53:42 -08005652// must be called with thread mutex locked
5653bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
5654{
Eric Laurent3b4529e2013-09-05 18:09:19 -07005655 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d",
5656 mWriteAckSequence, mDrainSequence);
5657 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) {
Eric Laurentbfb1b832013-01-07 09:53:42 -08005658 return true;
5659 }
5660 return false;
5661}
5662
Eric Laurentbfb1b832013-01-07 09:53:42 -08005663bool AudioFlinger::OffloadThread::waitingAsyncCallback()
5664{
5665 Mutex::Autolock _l(mLock);
5666 return waitingAsyncCallback_l();
5667}
5668
5669void AudioFlinger::OffloadThread::flushHw_l()
5670{
Eric Laurente659ef42014-09-29 13:06:46 -07005671 DirectOutputThread::flushHw_l();
Eric Laurentbfb1b832013-01-07 09:53:42 -08005672 // Flush anything still waiting in the mixbuffer
5673 mCurrentWriteLength = 0;
5674 mBytesRemaining = 0;
5675 mPausedWriteLength = 0;
5676 mPausedBytesRemaining = 0;
Eric Laurent3eaf66b2016-04-01 14:44:17 -07005677 // reset bytes written count to reflect that DSP buffers are empty after flush.
5678 mBytesWritten = 0;
Andy Hungf8044752016-07-27 14:58:11 -07005679 mOffloadUnderrunPosition = ~0LL;
Haynes Mathew George0f02f262014-01-11 13:03:57 -08005680
Eric Laurentbfb1b832013-01-07 09:53:42 -08005681 if (mUseAsyncWrite) {
Eric Laurent3b4529e2013-09-05 18:09:19 -07005682 // discard any pending drain or write ack by incrementing sequence
5683 mWriteAckSequence = (mWriteAckSequence + 2) & ~1;
5684 mDrainSequence = (mDrainSequence + 2) & ~1;
Eric Laurentbfb1b832013-01-07 09:53:42 -08005685 ALOG_ASSERT(mCallbackThread != 0);
Eric Laurent3b4529e2013-09-05 18:09:19 -07005686 mCallbackThread->setWriteBlocked(mWriteAckSequence);
5687 mCallbackThread->setDraining(mDrainSequence);
Eric Laurentbfb1b832013-01-07 09:53:42 -08005688 }
5689}
5690
Haynes Mathew George05317d22016-05-03 16:34:26 -07005691void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType)
5692{
5693 Mutex::Autolock _l(mLock);
Eric Laurent13084622016-05-17 10:51:49 -07005694 if (PlaybackThread::invalidateTracks_l(streamType)) {
5695 mFlushPending = true;
5696 }
Haynes Mathew George05317d22016-05-03 16:34:26 -07005697}
5698
Eric Laurentbfb1b832013-01-07 09:53:42 -08005699// ----------------------------------------------------------------------------
5700
Eric Laurent81784c32012-11-19 14:55:58 -08005701AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
Eric Laurent72e3f392015-05-20 14:43:50 -07005702 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady)
Eric Laurent81784c32012-11-19 14:55:58 -08005703 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
Eric Laurent72e3f392015-05-20 14:43:50 -07005704 systemReady, DUPLICATING),
Eric Laurent81784c32012-11-19 14:55:58 -08005705 mWaitTimeMs(UINT_MAX)
5706{
5707 addOutputTrack(mainThread);
5708}
5709
5710AudioFlinger::DuplicatingThread::~DuplicatingThread()
5711{
5712 for (size_t i = 0; i < mOutputTracks.size(); i++) {
5713 mOutputTracks[i]->destroy();
5714 }
5715}
5716
5717void AudioFlinger::DuplicatingThread::threadLoop_mix()
5718{
5719 // mix buffers...
5720 if (outputsReady(outputTracks)) {
Glenn Kastend79072e2016-01-06 08:41:20 -08005721 mAudioMixer->process();
Eric Laurent81784c32012-11-19 14:55:58 -08005722 } else {
Eric Laurent02b57082014-11-07 17:28:28 -08005723 if (mMixerBufferValid) {
5724 memset(mMixerBuffer, 0, mMixerBufferSize);
5725 } else {
5726 memset(mSinkBuffer, 0, mSinkBufferSize);
5727 }
Eric Laurent81784c32012-11-19 14:55:58 -08005728 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005729 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08005730 writeFrames = mNormalFrameCount;
Andy Hung25c2dac2014-02-27 14:56:00 -08005731 mCurrentWriteLength = mSinkBufferSize;
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005732 mStandbyTimeNs = systemTime() + mStandbyDelayNs;
Eric Laurent81784c32012-11-19 14:55:58 -08005733}
5734
5735void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
5736{
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005737 if (mSleepTimeUs == 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08005738 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005739 mSleepTimeUs = mActiveSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005740 } else {
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005741 mSleepTimeUs = mIdleSleepTimeUs;
Eric Laurent81784c32012-11-19 14:55:58 -08005742 }
5743 } else if (mBytesWritten != 0) {
5744 if (mMixerStatus == MIXER_TRACKS_ENABLED) {
5745 writeFrames = mNormalFrameCount;
Andy Hung25c2dac2014-02-27 14:56:00 -08005746 memset(mSinkBuffer, 0, mSinkBufferSize);
Eric Laurent81784c32012-11-19 14:55:58 -08005747 } else {
5748 // flush remaining overflow buffers in output tracks
5749 writeFrames = 0;
5750 }
Eric Laurentad9cb8b2015-05-26 16:38:19 -07005751 mSleepTimeUs = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08005752 }
5753}
5754
Eric Laurentbfb1b832013-01-07 09:53:42 -08005755ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
Eric Laurent81784c32012-11-19 14:55:58 -08005756{
5757 for (size_t i = 0; i < outputTracks.size(); i++) {
Andy Hungc25b84a2015-01-14 19:04:10 -08005758 outputTracks[i]->write(mSinkBuffer, writeFrames);
Eric Laurent81784c32012-11-19 14:55:58 -08005759 }
Eric Laurent2c3740f2013-10-30 16:57:06 -07005760 mStandby = false;
Andy Hung25c2dac2014-02-27 14:56:00 -08005761 return (ssize_t)mSinkBufferSize;
Eric Laurent81784c32012-11-19 14:55:58 -08005762}
5763
5764void AudioFlinger::DuplicatingThread::threadLoop_standby()
5765{
5766 // DuplicatingThread implements standby by stopping all tracks
5767 for (size_t i = 0; i < outputTracks.size(); i++) {
5768 outputTracks[i]->stop();
5769 }
5770}
5771
5772void AudioFlinger::DuplicatingThread::saveOutputTracks()
5773{
5774 outputTracks = mOutputTracks;
5775}
5776
5777void AudioFlinger::DuplicatingThread::clearOutputTracks()
5778{
5779 outputTracks.clear();
5780}
5781
5782void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
5783{
5784 Mutex::Autolock _l(mLock);
Andy Hungc25b84a2015-01-14 19:04:10 -08005785 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
5786 // Adjust for thread->sampleRate() to determine minimum buffer frame count.
5787 // Then triple buffer because Threads do not run synchronously and may not be clock locked.
5788 const size_t frameCount =
5789 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate());
5790 // TODO: Consider asynchronous sample rate conversion to handle clock disparity
5791 // from different OutputTracks and their associated MixerThreads (e.g. one may
5792 // nearly empty and the other may be dropping data).
5793
5794 sp<OutputTrack> outputTrack = new OutputTrack(thread,
Eric Laurent81784c32012-11-19 14:55:58 -08005795 this,
5796 mSampleRate,
Andy Hungc25b84a2015-01-14 19:04:10 -08005797 mFormat,
Eric Laurent81784c32012-11-19 14:55:58 -08005798 mChannelMask,
Marco Nelissen462fd2f2013-01-14 14:12:05 -08005799 frameCount,
5800 IPCThreadState::self()->getCallingUid());
Eric Laurentaf3ec7c2016-08-01 11:25:19 -07005801 status_t status = outputTrack != 0 ? outputTrack->initCheck() : (status_t) NO_MEMORY;
5802 if (status != NO_ERROR) {
5803 ALOGE("addOutputTrack() initCheck failed %d", status);
5804 return;
Eric Laurent81784c32012-11-19 14:55:58 -08005805 }
Eric Laurentaf3ec7c2016-08-01 11:25:19 -07005806 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
5807 mOutputTracks.add(outputTrack);
5808 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
5809 updateWaitTime_l();
Eric Laurent81784c32012-11-19 14:55:58 -08005810}
5811
5812void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
5813{
5814 Mutex::Autolock _l(mLock);
5815 for (size_t i = 0; i < mOutputTracks.size(); i++) {
5816 if (mOutputTracks[i]->thread() == thread) {
5817 mOutputTracks[i]->destroy();
5818 mOutputTracks.removeAt(i);
5819 updateWaitTime_l();
Eric Laurentf6870ae2015-05-08 10:50:03 -07005820 if (thread->getOutput() == mOutput) {
5821 mOutput = NULL;
5822 }
Eric Laurent81784c32012-11-19 14:55:58 -08005823 return;
5824 }
5825 }
Eric Laurentf6870ae2015-05-08 10:50:03 -07005826 ALOGV("removeOutputTrack(): unknown thread: %p", thread);
Eric Laurent81784c32012-11-19 14:55:58 -08005827}
5828
5829// caller must hold mLock
5830void AudioFlinger::DuplicatingThread::updateWaitTime_l()
5831{
5832 mWaitTimeMs = UINT_MAX;
5833 for (size_t i = 0; i < mOutputTracks.size(); i++) {
5834 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
5835 if (strong != 0) {
5836 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
5837 if (waitTimeMs < mWaitTimeMs) {
5838 mWaitTimeMs = waitTimeMs;
5839 }
5840 }
5841 }
5842}
5843
5844
5845bool AudioFlinger::DuplicatingThread::outputsReady(
5846 const SortedVector< sp<OutputTrack> > &outputTracks)
5847{
5848 for (size_t i = 0; i < outputTracks.size(); i++) {
5849 sp<ThreadBase> thread = outputTracks[i]->thread().promote();
5850 if (thread == 0) {
5851 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
5852 outputTracks[i].get());
5853 return false;
5854 }
5855 PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
5856 // see note at standby() declaration
5857 if (playbackThread->standby() && !playbackThread->isSuspended()) {
5858 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
5859 thread.get());
5860 return false;
5861 }
5862 }
5863 return true;
5864}
5865
5866uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
5867{
5868 return (mWaitTimeMs * 1000) / 2;
5869}
5870
5871void AudioFlinger::DuplicatingThread::cacheParameters_l()
5872{
5873 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
5874 updateWaitTime_l();
5875
5876 MixerThread::cacheParameters_l();
5877}
5878
5879// ----------------------------------------------------------------------------
5880// Record
5881// ----------------------------------------------------------------------------
5882
5883AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
5884 AudioStreamIn *input,
Eric Laurent81784c32012-11-19 14:55:58 -08005885 audio_io_handle_t id,
Eric Laurentd3922f72013-02-01 17:57:04 -08005886 audio_devices_t outDevice,
Eric Laurent72e3f392015-05-20 14:43:50 -07005887 audio_devices_t inDevice,
5888 bool systemReady
Glenn Kasten46909e72013-02-26 09:20:22 -08005889#ifdef TEE_SINK
5890 , const sp<NBAIO_Sink>& teeSink
5891#endif
5892 ) :
Eric Laurent72e3f392015-05-20 14:43:50 -07005893 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady),
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08005894 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL),
Glenn Kasten1b291842016-07-18 14:55:21 -07005895 // mRsmpInFrames, mRsmpInFramesP2, and mRsmpInFramesOA are set by readInputParameters_l()
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08005896 mRsmpInRear(0)
Glenn Kasten46909e72013-02-26 09:20:22 -08005897#ifdef TEE_SINK
5898 , mTeeSink(teeSink)
5899#endif
Glenn Kastenb880f5e2014-05-07 08:43:45 -07005900 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize,
5901 "RecordThreadRO", MemoryHeapBase::READ_ONLY))
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005902 // mFastCapture below
5903 , mFastCaptureFutex(0)
5904 // mInputSource
5905 // mPipeSink
5906 // mPipeSource
5907 , mPipeFramesP2(0)
5908 // mPipeMemory
5909 // mFastCaptureNBLogWriter
Glenn Kasten6e6704c2014-07-03 10:20:00 -07005910 , mFastTrackAvail(false)
Eric Laurent81784c32012-11-19 14:55:58 -08005911{
Glenn Kastend7dca052015-03-05 16:05:54 -08005912 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id);
5913 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName);
Eric Laurent81784c32012-11-19 14:55:58 -08005914
Glenn Kastendeca2ae2014-02-07 10:25:56 -08005915 readInputParameters_l();
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005916
5917 // create an NBAIO source for the HAL input stream, and negotiate
5918 mInputSource = new AudioStreamInSource(input->stream);
5919 size_t numCounterOffers = 0;
5920 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)};
Glenn Kasten57c4e6f2016-03-18 14:54:07 -07005921#if !LOG_NDEBUG
5922 ssize_t index =
5923#else
5924 (void)
5925#endif
5926 mInputSource->negotiate(offers, 1, NULL, numCounterOffers);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005927 ALOG_ASSERT(index == 0);
5928
5929 // initialize fast capture depending on configuration
5930 bool initFastCapture;
5931 switch (kUseFastCapture) {
5932 case FastCapture_Never:
5933 initFastCapture = false;
5934 break;
5935 case FastCapture_Always:
5936 initFastCapture = true;
5937 break;
5938 case FastCapture_Static:
Glenn Kasteneb9487e2015-07-22 09:15:17 -07005939 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005940 break;
5941 // case FastCapture_Dynamic:
5942 }
5943
5944 if (initFastCapture) {
Glenn Kastend198b852015-03-16 14:55:53 -07005945 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005946 NBAIO_Format format = mInputSource->format();
Glenn Kasten1b291842016-07-18 14:55:21 -07005947 // quadruple-buffering of 20 ms each; this ensures we can sleep for 20ms in RecordThread
5948 size_t pipeFramesP2 = roundup(4 * FMS_20 * mSampleRate / 1000);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07005949 size_t pipeSize = pipeFramesP2 * Format_frameSize(format);
5950 void *pipeBuffer;
5951 const sp<MemoryDealer> roHeap(readOnlyHeap());
5952 sp<IMemory> pipeMemory;
5953 if ((roHeap == 0) ||
5954 (pipeMemory = roHeap->allocate(pipeSize)) == 0 ||
5955 (pipeBuffer = pipeMemory->pointer()) == NULL) {
5956 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize);
5957 goto failed;
5958 }
5959 // pipe will be shared directly with fast clients, so clear to avoid leaking old information
5960 memset(pipeBuffer, 0, pipeSize);
5961 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer);
5962 const NBAIO_Format offers[1] = {format};
5963 size_t numCounterOffers = 0;
5964 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
5965 ALOG_ASSERT(index == 0);
5966 mPipeSink = pipe;
5967 PipeReader *pipeReader = new PipeReader(*pipe);
5968 numCounterOffers = 0;
5969 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
5970 ALOG_ASSERT(index == 0);
5971 mPipeSource = pipeReader;
5972 mPipeFramesP2 = pipeFramesP2;
5973 mPipeMemory = pipeMemory;
5974
5975 // create fast capture
5976 mFastCapture = new FastCapture();
5977 FastCaptureStateQueue *sq = mFastCapture->sq();
5978#ifdef STATE_QUEUE_DUMP
5979 // FIXME
5980#endif
5981 FastCaptureState *state = sq->begin();
5982 state->mCblk = NULL;
5983 state->mInputSource = mInputSource.get();
5984 state->mInputSourceGen++;
5985 state->mPipeSink = pipe;
5986 state->mPipeSinkGen++;
5987 state->mFrameCount = mFrameCount;
5988 state->mCommand = FastCaptureState::COLD_IDLE;
5989 // already done in constructor initialization list
5990 //mFastCaptureFutex = 0;
5991 state->mColdFutexAddr = &mFastCaptureFutex;
5992 state->mColdGen++;
5993 state->mDumpState = &mFastCaptureDumpState;
5994#ifdef TEE_SINK
5995 // FIXME
5996#endif
5997 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture");
5998 state->mNBLogWriter = mFastCaptureNBLogWriter.get();
5999 sq->end();
6000 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
6001
6002 // start the fast capture
6003 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
6004 pid_t tid = mFastCapture->getTid();
Glenn Kasten8379b722016-03-18 14:54:17 -07006005 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006006#ifdef AUDIO_WATCHDOG
6007 // FIXME
6008#endif
6009
Glenn Kasten6e6704c2014-07-03 10:20:00 -07006010 mFastTrackAvail = true;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006011 }
6012failed: ;
6013
6014 // FIXME mNormalSource
Eric Laurent81784c32012-11-19 14:55:58 -08006015}
6016
Eric Laurent81784c32012-11-19 14:55:58 -08006017AudioFlinger::RecordThread::~RecordThread()
6018{
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006019 if (mFastCapture != 0) {
6020 FastCaptureStateQueue *sq = mFastCapture->sq();
6021 FastCaptureState *state = sq->begin();
6022 if (state->mCommand == FastCaptureState::COLD_IDLE) {
6023 int32_t old = android_atomic_inc(&mFastCaptureFutex);
6024 if (old == -1) {
6025 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
6026 }
6027 }
6028 state->mCommand = FastCaptureState::EXIT;
6029 sq->end();
6030 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED);
6031 mFastCapture->join();
6032 mFastCapture.clear();
6033 }
6034 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter);
Glenn Kasten481fb672013-09-30 14:39:28 -07006035 mAudioFlinger->unregisterWriter(mNBLogWriter);
Andy Hung57446612015-04-19 23:56:46 -07006036 free(mRsmpInBuffer);
Eric Laurent81784c32012-11-19 14:55:58 -08006037}
6038
6039void AudioFlinger::RecordThread::onFirstRef()
6040{
Glenn Kastend7dca052015-03-05 16:05:54 -08006041 run(mThreadName, PRIORITY_URGENT_AUDIO);
Eric Laurent81784c32012-11-19 14:55:58 -08006042}
6043
Eric Laurent81784c32012-11-19 14:55:58 -08006044bool AudioFlinger::RecordThread::threadLoop()
6045{
Eric Laurent81784c32012-11-19 14:55:58 -08006046 nsecs_t lastWarning = 0;
6047
6048 inputStandBy();
Eric Laurent81784c32012-11-19 14:55:58 -08006049
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006050reacquire_wakelock:
6051 sp<RecordTrack> activeTrack;
Glenn Kasten2b806402013-11-20 16:37:38 -08006052 int activeTracksGen;
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006053 {
6054 Mutex::Autolock _l(mLock);
Glenn Kasten2b806402013-11-20 16:37:38 -08006055 size_t size = mActiveTracks.size();
6056 activeTracksGen = mActiveTracksGen;
6057 if (size > 0) {
6058 // FIXME an arbitrary choice
6059 activeTrack = mActiveTracks[0];
6060 acquireWakeLock_l(activeTrack->uid());
6061 if (size > 1) {
6062 SortedVector<int> tmp;
6063 for (size_t i = 0; i < size; i++) {
6064 tmp.add(mActiveTracks[i]->uid());
6065 }
6066 updateWakeLockUids_l(tmp);
6067 }
6068 } else {
6069 acquireWakeLock_l(-1);
6070 }
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006071 }
6072
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006073 // used to request a deferred sleep, to be executed later while mutex is unlocked
6074 uint32_t sleepUs = 0;
6075
6076 // loop while there is work to do
Glenn Kasten4ef0b462013-08-14 13:52:27 -07006077 for (;;) {
Glenn Kastenc527a7c2013-08-13 15:43:49 -07006078 Vector< sp<EffectChain> > effectChains;
Glenn Kasten2cfbf882013-08-14 13:12:11 -07006079
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006080 // activeTracks accumulates a copy of a subset of mActiveTracks
6081 Vector< sp<RecordTrack> > activeTracks;
6082
Glenn Kasten735f45f2014-08-18 15:51:59 -07006083 // reference to the (first and only) active fast track
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006084 sp<RecordTrack> fastTrack;
Eric Laurent10351942014-05-08 18:49:52 -07006085
Glenn Kasten735f45f2014-08-18 15:51:59 -07006086 // reference to a fast track which is about to be removed
6087 sp<RecordTrack> fastTrackToRemove;
6088
Eric Laurent81784c32012-11-19 14:55:58 -08006089 { // scope for mLock
6090 Mutex::Autolock _l(mLock);
Eric Laurent000a4192014-01-29 15:17:32 -08006091
Eric Laurent021cf962014-05-13 10:18:14 -07006092 processConfigEvents_l();
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006093
Eric Laurent000a4192014-01-29 15:17:32 -08006094 // check exitPending here because checkForNewParameters_l() and
6095 // checkForNewParameters_l() can temporarily release mLock
6096 if (exitPending()) {
6097 break;
6098 }
6099
Eric Laurent5c25d562016-07-13 17:17:45 -07006100 // sleep with mutex unlocked
6101 if (sleepUs > 0) {
Glenn Kastenf9715e42016-07-13 14:02:03 -07006102 ATRACE_BEGIN("sleepC");
Eric Laurent5c25d562016-07-13 17:17:45 -07006103 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)sleepUs));
6104 ATRACE_END();
6105 sleepUs = 0;
6106 continue;
6107 }
6108
Glenn Kasten2b806402013-11-20 16:37:38 -08006109 // if no active track(s), then standby and release wakelock
6110 size_t size = mActiveTracks.size();
6111 if (size == 0) {
Glenn Kasten93e471f2013-08-19 08:40:07 -07006112 standbyIfNotAlreadyInStandby();
Glenn Kasten4ef0b462013-08-14 13:52:27 -07006113 // exitPending() can't become true here
Eric Laurent81784c32012-11-19 14:55:58 -08006114 releaseWakeLock_l();
6115 ALOGV("RecordThread: loop stopping");
6116 // go to sleep
6117 mWaitWorkCV.wait(mLock);
6118 ALOGV("RecordThread: loop starting");
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006119 goto reacquire_wakelock;
6120 }
6121
Glenn Kasten2b806402013-11-20 16:37:38 -08006122 if (mActiveTracksGen != activeTracksGen) {
6123 activeTracksGen = mActiveTracksGen;
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006124 SortedVector<int> tmp;
Glenn Kasten2b806402013-11-20 16:37:38 -08006125 for (size_t i = 0; i < size; i++) {
6126 tmp.add(mActiveTracks[i]->uid());
6127 }
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006128 updateWakeLockUids_l(tmp);
Eric Laurent81784c32012-11-19 14:55:58 -08006129 }
Glenn Kasten9e982352013-08-14 14:39:50 -07006130
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006131 bool doBroadcast = false;
Eric Laurent5c25d562016-07-13 17:17:45 -07006132 bool allStopped = true;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006133 for (size_t i = 0; i < size; ) {
Glenn Kasten9e982352013-08-14 14:39:50 -07006134
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006135 activeTrack = mActiveTracks[i];
6136 if (activeTrack->isTerminated()) {
Glenn Kasten735f45f2014-08-18 15:51:59 -07006137 if (activeTrack->isFastTrack()) {
6138 ALOG_ASSERT(fastTrackToRemove == 0);
6139 fastTrackToRemove = activeTrack;
6140 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006141 removeTrack_l(activeTrack);
Glenn Kasten2b806402013-11-20 16:37:38 -08006142 mActiveTracks.remove(activeTrack);
6143 mActiveTracksGen++;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006144 size--;
Glenn Kasten9e982352013-08-14 14:39:50 -07006145 continue;
6146 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006147
6148 TrackBase::track_state activeTrackState = activeTrack->mState;
6149 switch (activeTrackState) {
6150
6151 case TrackBase::PAUSING:
6152 mActiveTracks.remove(activeTrack);
6153 mActiveTracksGen++;
6154 doBroadcast = true;
6155 size--;
6156 continue;
6157
6158 case TrackBase::STARTING_1:
6159 sleepUs = 10000;
6160 i++;
Eric Laurent5c25d562016-07-13 17:17:45 -07006161 allStopped = false;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006162 continue;
6163
6164 case TrackBase::STARTING_2:
6165 doBroadcast = true;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006166 mStandby = false;
Glenn Kasten9e982352013-08-14 14:39:50 -07006167 activeTrack->mState = TrackBase::ACTIVE;
Eric Laurent5c25d562016-07-13 17:17:45 -07006168 allStopped = false;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006169 break;
6170
6171 case TrackBase::ACTIVE:
Eric Laurent5c25d562016-07-13 17:17:45 -07006172 allStopped = false;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006173 break;
6174
6175 case TrackBase::IDLE:
6176 i++;
6177 continue;
6178
6179 default:
Glenn Kastenadad3d72014-02-21 14:51:43 -08006180 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState);
Glenn Kasten9e982352013-08-14 14:39:50 -07006181 }
Glenn Kasten9e982352013-08-14 14:39:50 -07006182
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006183 activeTracks.add(activeTrack);
6184 i++;
Glenn Kasten9e982352013-08-14 14:39:50 -07006185
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006186 if (activeTrack->isFastTrack()) {
6187 ALOG_ASSERT(!mFastTrackAvail);
6188 ALOG_ASSERT(fastTrack == 0);
6189 fastTrack = activeTrack;
6190 }
Glenn Kasten9e982352013-08-14 14:39:50 -07006191 }
Eric Laurent5c25d562016-07-13 17:17:45 -07006192
6193 if (allStopped) {
6194 standbyIfNotAlreadyInStandby();
6195 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006196 if (doBroadcast) {
6197 mStartStopCond.broadcast();
6198 }
6199
6200 // sleep if there are no active tracks to process
6201 if (activeTracks.size() == 0) {
6202 if (sleepUs == 0) {
6203 sleepUs = kRecordThreadSleepUs;
6204 }
6205 continue;
6206 }
6207 sleepUs = 0;
Glenn Kasten9e982352013-08-14 14:39:50 -07006208
Eric Laurent81784c32012-11-19 14:55:58 -08006209 lockEffectChains_l(effectChains);
6210 }
6211
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006212 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0
Glenn Kasten71652682013-08-14 15:17:55 -07006213
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006214 size_t size = effectChains.size();
6215 for (size_t i = 0; i < size; i++) {
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006216 // thread mutex is not locked, but effect chain is locked
6217 effectChains[i]->process_l();
6218 }
6219
Glenn Kasten735f45f2014-08-18 15:51:59 -07006220 // Push a new fast capture state if fast capture is not already running, or cblk change
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006221 if (mFastCapture != 0) {
6222 FastCaptureStateQueue *sq = mFastCapture->sq();
6223 FastCaptureState *state = sq->begin();
Glenn Kasten735f45f2014-08-18 15:51:59 -07006224 bool didModify = false;
6225 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006226 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME &&
6227 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) {
6228 if (state->mCommand == FastCaptureState::COLD_IDLE) {
6229 int32_t old = android_atomic_inc(&mFastCaptureFutex);
6230 if (old == -1) {
6231 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1);
6232 }
6233 }
6234 state->mCommand = FastCaptureState::READ_WRITE;
6235#if 0 // FIXME
6236 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
Glenn Kastenfbdb2ac2015-03-02 14:47:19 -08006237 FastThreadDumpState::kSamplingNforLowRamDevice :
6238 FastThreadDumpState::kSamplingN);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006239#endif
Glenn Kasten735f45f2014-08-18 15:51:59 -07006240 didModify = true;
6241 }
6242 audio_track_cblk_t *cblkOld = state->mCblk;
6243 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL;
6244 if (cblkNew != cblkOld) {
6245 state->mCblk = cblkNew;
6246 // block until acked if removing a fast track
6247 if (cblkOld != NULL) {
6248 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED;
6249 }
6250 didModify = true;
6251 }
6252 sq->end(didModify);
6253 if (didModify) {
6254 sq->push(block);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006255#if 0
6256 if (kUseFastCapture == FastCapture_Dynamic) {
6257 mNormalSource = mPipeSource;
6258 }
6259#endif
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006260 }
6261 }
6262
Glenn Kasten735f45f2014-08-18 15:51:59 -07006263 // now run the fast track destructor with thread mutex unlocked
6264 fastTrackToRemove.clear();
6265
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006266 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one.
6267 // Only the client(s) that are too slow will overrun. But if even the fastest client is too
6268 // slow, then this RecordThread will overrun by not calling HAL read often enough.
6269 // If destination is non-contiguous, first read past the nominal end of buffer, then
6270 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated.
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006271
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006272 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006273 ssize_t framesRead;
6274
6275 // If an NBAIO source is present, use it to read the normal capture's data
6276 if (mPipeSource != 0) {
6277 size_t framesToRead = mBufferSize / mFrameSize;
Glenn Kasten1b291842016-07-18 14:55:21 -07006278 framesToRead = min(mRsmpInFramesOA - rear, mRsmpInFramesP2 / 2);
Andy Hung57446612015-04-19 23:56:46 -07006279 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
Glenn Kastend79072e2016-01-06 08:41:20 -08006280 framesToRead);
Glenn Kasten1b291842016-07-18 14:55:21 -07006281 // since pipe is non-blocking, simulate blocking input by waiting for 1/2 of
6282 // buffer size or at least for 20ms.
6283 size_t sleepFrames = max(
6284 min(mPipeFramesP2, mRsmpInFramesP2) / 2, FMS_20 * mSampleRate / 1000);
6285 if (framesRead <= (ssize_t) sleepFrames) {
6286 sleepUs = (sleepFrames * 1000000LL) / mSampleRate;
6287 }
6288 if (framesRead < 0) {
6289 status_t status = (status_t) framesRead;
6290 switch (status) {
6291 case OVERRUN:
6292 ALOGW("overrun on read from pipe");
6293 framesRead = 0;
6294 break;
6295 case NEGOTIATE:
6296 ALOGE("re-negotiation is needed");
6297 framesRead = -1; // Will cause an attempt to recover.
6298 break;
6299 default:
6300 ALOGE("unknown error %d on read from pipe", status);
6301 break;
6302 }
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006303 }
6304 // otherwise use the HAL / AudioStreamIn directly
6305 } else {
Glenn Kastenec6a7032016-03-14 07:40:23 -07006306 ATRACE_BEGIN("read");
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006307 ssize_t bytesRead = mInput->stream->read(mInput->stream,
Andy Hung57446612015-04-19 23:56:46 -07006308 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize);
Glenn Kastenec6a7032016-03-14 07:40:23 -07006309 ATRACE_END();
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006310 if (bytesRead < 0) {
6311 framesRead = bytesRead;
6312 } else {
6313 framesRead = bytesRead / mFrameSize;
6314 }
6315 }
6316
Andy Hung3f0c9022016-01-15 17:49:46 -08006317 // Update server timestamp with server stats
6318 // systemTime() is optional if the hardware supports timestamps.
6319 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead;
6320 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime();
6321
6322 // Update server timestamp with kernel stats
Andy Hung69ce44d2016-07-18 12:14:25 -07006323 if (mInput->stream->get_capture_position != nullptr
6324 && mPipeSource.get() == nullptr /* don't obtain for FastCapture, could block */) {
Andy Hung3f0c9022016-01-15 17:49:46 -08006325 int64_t position, time;
6326 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time);
6327 if (ret == NO_ERROR) {
6328 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position;
6329 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time;
6330 // Note: In general record buffers should tend to be empty in
6331 // a properly running pipeline.
6332 //
6333 // Also, it is not advantageous to call get_presentation_position during the read
6334 // as the read obtains a lock, preventing the timestamp call from executing.
6335 }
6336 }
6337 // Use this to track timestamp information
6338 // ALOGD("%s", mTimestamp.toString().c_str());
6339
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006340 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006341 ALOGE("read failed: framesRead=%zd", framesRead);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006342 // Force input into standby so that it tries to recover at next read attempt
6343 inputStandBy();
6344 sleepUs = kRecordThreadSleepUs;
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006345 }
6346 if (framesRead <= 0) {
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006347 goto unlock;
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006348 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006349 ALOG_ASSERT(framesRead > 0);
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006350
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006351 if (mTeeSink != 0) {
Andy Hung57446612015-04-19 23:56:46 -07006352 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006353 }
6354 // If destination is non-contiguous, we now correct for reading past end of buffer.
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006355 {
6356 size_t part1 = mRsmpInFramesP2 - rear;
6357 if ((size_t) framesRead > part1) {
Andy Hung57446612015-04-19 23:56:46 -07006358 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize,
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006359 (framesRead - part1) * mFrameSize);
6360 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006361 }
6362 rear = mRsmpInRear += framesRead;
6363
6364 size = activeTracks.size();
6365 // loop over each active track
6366 for (size_t i = 0; i < size; i++) {
6367 activeTrack = activeTracks[i];
6368
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006369 // skip fast tracks, as those are handled directly by FastCapture
6370 if (activeTrack->isFastTrack()) {
6371 continue;
6372 }
6373
Andy Hung73c02e42015-03-29 01:13:58 -07006374 // TODO: This code probably should be moved to RecordTrack.
Andy Hung97a893e2015-03-29 01:03:07 -07006375 // TODO: Update the activeTrack buffer converter in case of reconfigure.
6376
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006377 enum {
6378 OVERRUN_UNKNOWN,
6379 OVERRUN_TRUE,
6380 OVERRUN_FALSE
6381 } overrun = OVERRUN_UNKNOWN;
6382
6383 // loop over getNextBuffer to handle circular sink
6384 for (;;) {
6385
6386 activeTrack->mSink.frameCount = ~0;
6387 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink);
6388 size_t framesOut = activeTrack->mSink.frameCount;
6389 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0));
6390
Andy Hung73c02e42015-03-29 01:13:58 -07006391 // check available frames and handle overrun conditions
6392 // if the record track isn't draining fast enough.
6393 bool hasOverrun;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006394 size_t framesIn;
Andy Hung73c02e42015-03-29 01:13:58 -07006395 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun);
6396 if (hasOverrun) {
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006397 overrun = OVERRUN_TRUE;
6398 }
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08006399 if (framesOut == 0 || framesIn == 0) {
6400 break;
6401 }
6402
Andy Hung6770c6f2015-04-07 13:43:36 -07006403 // Don't allow framesOut to be larger than what is possible with resampling
6404 // from framesIn.
6405 // This isn't strictly necessary but helps limit buffer resizing in
6406 // RecordBufferConverter. TODO: remove when no longer needed.
6407 framesOut = min(framesOut,
6408 destinationFramesPossible(
6409 framesIn, mSampleRate, activeTrack->mSampleRate));
Andy Hung97a893e2015-03-29 01:03:07 -07006410 // process frames from the RecordThread buffer provider to the RecordTrack buffer
6411 framesOut = activeTrack->mRecordBufferConverter->convert(
6412 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006413
6414 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
6415 overrun = OVERRUN_FALSE;
6416 }
6417
6418 if (activeTrack->mFramesToDrop == 0) {
6419 if (framesOut > 0) {
6420 activeTrack->mSink.frameCount = framesOut;
6421 activeTrack->releaseBuffer(&activeTrack->mSink);
6422 }
6423 } else {
6424 // FIXME could do a partial drop of framesOut
6425 if (activeTrack->mFramesToDrop > 0) {
6426 activeTrack->mFramesToDrop -= framesOut;
6427 if (activeTrack->mFramesToDrop <= 0) {
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006428 activeTrack->clearSyncStartEvent();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006429 }
6430 } else {
6431 activeTrack->mFramesToDrop += framesOut;
6432 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 ||
6433 activeTrack->mSyncStartEvent->isCancelled()) {
6434 ALOGW("Synced record %s, session %d, trigger session %d",
6435 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled",
6436 activeTrack->sessionId(),
6437 (activeTrack->mSyncStartEvent != 0) ?
Glenn Kastend848eb42016-03-08 13:42:11 -08006438 activeTrack->mSyncStartEvent->triggerSession() :
6439 AUDIO_SESSION_NONE);
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006440 activeTrack->clearSyncStartEvent();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006441 }
6442 }
6443 }
6444
6445 if (framesOut == 0) {
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006446 break;
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006447 }
6448 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006449
6450 switch (overrun) {
6451 case OVERRUN_TRUE:
6452 // client isn't retrieving buffers fast enough
6453 if (!activeTrack->setOverflow()) {
6454 nsecs_t now = systemTime();
6455 // FIXME should lastWarning per track?
6456 if ((now - lastWarning) > kWarningThrottleNs) {
6457 ALOGW("RecordThread: buffer overflow");
6458 lastWarning = now;
6459 }
6460 }
6461 break;
6462 case OVERRUN_FALSE:
6463 activeTrack->clearOverflow();
6464 break;
6465 case OVERRUN_UNKNOWN:
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006466 break;
6467 }
6468
Andy Hung3f0c9022016-01-15 17:49:46 -08006469 // update frame information and push timestamp out
6470 activeTrack->updateTrackFrameInfo(
Andy Hung6ae58432016-02-16 18:32:24 -08006471 activeTrack->mServerProxy->framesReleased(),
Andy Hung3f0c9022016-01-15 17:49:46 -08006472 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER],
6473 mSampleRate, mTimestamp);
Glenn Kasten1ba19cd2013-08-14 14:02:21 -07006474 }
6475
Glenn Kasten3d61bc12014-06-16 10:25:20 -07006476unlock:
Eric Laurent81784c32012-11-19 14:55:58 -08006477 // enable changes in effect chain
6478 unlockEffectChains(effectChains);
Glenn Kastenc527a7c2013-08-13 15:43:49 -07006479 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end
Eric Laurent81784c32012-11-19 14:55:58 -08006480 }
6481
Glenn Kasten93e471f2013-08-19 08:40:07 -07006482 standbyIfNotAlreadyInStandby();
Eric Laurent81784c32012-11-19 14:55:58 -08006483
6484 {
6485 Mutex::Autolock _l(mLock);
Eric Laurent9a54bc22013-09-09 09:08:44 -07006486 for (size_t i = 0; i < mTracks.size(); i++) {
6487 sp<RecordTrack> track = mTracks[i];
6488 track->invalidate();
6489 }
Glenn Kasten2b806402013-11-20 16:37:38 -08006490 mActiveTracks.clear();
6491 mActiveTracksGen++;
Eric Laurent81784c32012-11-19 14:55:58 -08006492 mStartStopCond.broadcast();
6493 }
6494
6495 releaseWakeLock();
6496
6497 ALOGV("RecordThread %p exiting", this);
6498 return false;
6499}
6500
Glenn Kasten93e471f2013-08-19 08:40:07 -07006501void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
Eric Laurent81784c32012-11-19 14:55:58 -08006502{
6503 if (!mStandby) {
6504 inputStandBy();
6505 mStandby = true;
6506 }
6507}
6508
6509void AudioFlinger::RecordThread::inputStandBy()
6510{
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006511 // Idle the fast capture if it's currently running
6512 if (mFastCapture != 0) {
6513 FastCaptureStateQueue *sq = mFastCapture->sq();
6514 FastCaptureState *state = sq->begin();
6515 if (!(state->mCommand & FastCaptureState::IDLE)) {
6516 state->mCommand = FastCaptureState::COLD_IDLE;
6517 state->mColdFutexAddr = &mFastCaptureFutex;
6518 state->mColdGen++;
6519 mFastCaptureFutex = 0;
6520 sq->end();
6521 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now
6522 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED);
6523#if 0
6524 if (kUseFastCapture == FastCapture_Dynamic) {
6525 // FIXME
6526 }
6527#endif
6528#ifdef AUDIO_WATCHDOG
6529 // FIXME
6530#endif
6531 } else {
6532 sq->end(false /*didModify*/);
6533 }
6534 }
Eric Laurent81784c32012-11-19 14:55:58 -08006535 mInput->stream->common.standby(&mInput->stream->common);
Andy Hungad6d52d2016-07-18 13:42:03 -07006536
6537 // If going into standby, flush the pipe source.
6538 if (mPipeSource.get() != nullptr) {
6539 const ssize_t flushed = mPipeSource->flush();
6540 if (flushed > 0) {
6541 ALOGV("Input standby flushed PipeSource %zd frames", flushed);
6542 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += flushed;
6543 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime();
6544 }
6545 }
Eric Laurent81784c32012-11-19 14:55:58 -08006546}
6547
Glenn Kasten05997e22014-03-13 15:08:33 -07006548// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
Glenn Kastene198c362013-08-13 09:13:36 -07006549sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
Eric Laurent81784c32012-11-19 14:55:58 -08006550 const sp<AudioFlinger::Client>& client,
6551 uint32_t sampleRate,
6552 audio_format_t format,
6553 audio_channel_mask_t channelMask,
Glenn Kasten74935e42013-12-19 08:56:45 -08006554 size_t *pFrameCount,
Glenn Kastend848eb42016-03-08 13:42:11 -08006555 audio_session_t sessionId,
Glenn Kasten7df8c0b2014-07-03 12:23:29 -07006556 size_t *notificationFrames,
Marco Nelissen462fd2f2013-01-14 14:12:05 -08006557 int uid,
Eric Laurent05067782016-06-01 18:27:28 -07006558 audio_input_flags_t *flags,
Eric Laurent81784c32012-11-19 14:55:58 -08006559 pid_t tid,
6560 status_t *status)
6561{
Glenn Kasten74935e42013-12-19 08:56:45 -08006562 size_t frameCount = *pFrameCount;
Eric Laurent81784c32012-11-19 14:55:58 -08006563 sp<RecordTrack> track;
6564 status_t lStatus;
Eric Laurent05067782016-06-01 18:27:28 -07006565 audio_input_flags_t inputFlags = mInput->flags;
6566
6567 // special case for FAST flag considered OK if fast capture is present
6568 if (hasFastCapture()) {
6569 inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST);
6570 }
6571
6572 // Check if requested flags are compatible with output stream flags
6573 if ((*flags & inputFlags) != *flags) {
6574 ALOGW("createRecordTrack_l(): mismatch between requested flags (%08x) and"
6575 " input flags (%08x)",
6576 *flags, inputFlags);
6577 *flags = (audio_input_flags_t)(*flags & inputFlags);
6578 }
Eric Laurent81784c32012-11-19 14:55:58 -08006579
Glenn Kasten90e58b12013-07-31 16:16:02 -07006580 // client expresses a preference for FAST, but we get the final say
Eric Laurent05067782016-06-01 18:27:28 -07006581 if (*flags & AUDIO_INPUT_FLAG_FAST) {
Glenn Kasten90e58b12013-07-31 16:16:02 -07006582 if (
Glenn Kastenb7fbf7e2015-03-18 12:57:28 -07006583 // we formerly checked for a callback handler (non-0 tid),
6584 // but that is no longer required for TRANSFER_OBTAIN mode
6585 //
Glenn Kasten74105912014-07-03 12:28:53 -07006586 // frame count is not specified, or is exactly the pipe depth
6587 ((frameCount == 0) || (frameCount == mPipeFramesP2)) &&
Glenn Kasten3a6c90a2014-03-13 15:07:51 -07006588 // PCM data
6589 audio_is_linear_pcm(format) &&
Glenn Kasten7fd04222016-02-02 12:38:16 -08006590 // hardware format
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006591 (format == mFormat) &&
Glenn Kasten7fd04222016-02-02 12:38:16 -08006592 // hardware channel mask
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006593 (channelMask == mChannelMask) &&
Glenn Kasten7fd04222016-02-02 12:38:16 -08006594 // hardware sample rate
Glenn Kasten90e58b12013-07-31 16:16:02 -07006595 (sampleRate == mSampleRate) &&
Glenn Kasten3a6c90a2014-03-13 15:07:51 -07006596 // record thread has an associated fast capture
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006597 hasFastCapture() &&
6598 // there are sufficient fast track slots available
6599 mFastTrackAvail
Glenn Kasten90e58b12013-07-31 16:16:02 -07006600 ) {
Eric Laurent4c415062016-06-17 16:14:16 -07006601 // check compatibility with audio effects.
6602 Mutex::Autolock _l(mLock);
6603 // Do not accept FAST flag if the session has software effects
6604 sp<EffectChain> chain = getEffectChain_l(sessionId);
6605 if (chain != 0) {
Eric Laurent122f7e72016-06-29 11:53:29 -07006606 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_RAW) != 0,
Eric Laurent4c415062016-06-17 16:14:16 -07006607 "AUDIO_INPUT_FLAG_RAW denied: effect present on session");
6608 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_RAW);
6609 if (chain->hasSoftwareEffect()) {
6610 ALOGV("AUDIO_INPUT_FLAG_FAST denied: software effect present on session");
6611 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST);
6612 }
6613 }
Eric Laurent122f7e72016-06-29 11:53:29 -07006614 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0,
Eric Laurent4c415062016-06-17 16:14:16 -07006615 "AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
6616 frameCount, mFrameCount);
Glenn Kasten90e58b12013-07-31 16:16:02 -07006617 } else {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006618 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu "
Glenn Kasten74105912014-07-03 12:28:53 -07006619 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006620 "hasFastCapture=%d tid=%d mFastTrackAvail=%d",
Glenn Kasten74105912014-07-03 12:28:53 -07006621 frameCount, mFrameCount, mPipeFramesP2,
6622 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate,
6623 hasFastCapture(), tid, mFastTrackAvail);
Eric Laurent05067782016-06-01 18:27:28 -07006624 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST);
Glenn Kasten74105912014-07-03 12:28:53 -07006625 }
6626 }
6627
6628 // compute track buffer size in frames, and suggest the notification frame count
Eric Laurent05067782016-06-01 18:27:28 -07006629 if (*flags & AUDIO_INPUT_FLAG_FAST) {
Glenn Kasten74105912014-07-03 12:28:53 -07006630 // fast track: frame count is exactly the pipe depth
6631 frameCount = mPipeFramesP2;
6632 // ignore requested notificationFrames, and always notify exactly once every HAL buffer
6633 *notificationFrames = mFrameCount;
6634 } else {
Glenn Kasten49d00ad2014-07-21 11:22:03 -07006635 // not fast track: max notification period is resampled equivalent of one HAL buffer time
6636 // or 20 ms if there is a fast capture
6637 // TODO This could be a roundupRatio inline, and const
6638 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount)
6639 * sampleRate + mSampleRate - 1) / mSampleRate;
6640 // minimum number of notification periods is at least kMinNotifications,
6641 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs)
6642 static const size_t kMinNotifications = 3;
6643 static const uint32_t kMinMs = 30;
6644 // TODO This could be a roundupRatio inline
6645 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000;
6646 // TODO This could be a roundupRatio inline
6647 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) /
6648 maxNotificationFrames;
6649 const size_t minFrameCount = maxNotificationFrames *
6650 max(kMinNotifications, minNotificationsByMs);
6651 frameCount = max(frameCount, minFrameCount);
6652 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) {
6653 *notificationFrames = maxNotificationFrames;
Glenn Kasten74105912014-07-03 12:28:53 -07006654 }
Glenn Kasten90e58b12013-07-31 16:16:02 -07006655 }
Glenn Kasten74935e42013-12-19 08:56:45 -08006656 *pFrameCount = frameCount;
Glenn Kasten90e58b12013-07-31 16:16:02 -07006657
Glenn Kasten15e57982013-09-24 11:52:37 -07006658 lStatus = initCheck();
6659 if (lStatus != NO_ERROR) {
6660 ALOGE("createRecordTrack_l() audio driver not initialized");
6661 goto Exit;
6662 }
Eric Laurent81784c32012-11-19 14:55:58 -08006663
6664 { // scope for mLock
6665 Mutex::Autolock _l(mLock);
6666
6667 track = new RecordTrack(this, client, sampleRate,
Eric Laurent83b88082014-06-20 18:31:16 -07006668 format, channelMask, frameCount, NULL, sessionId, uid,
6669 *flags, TrackBase::TYPE_DEFAULT);
Eric Laurent81784c32012-11-19 14:55:58 -08006670
Glenn Kasten03003332013-08-06 15:40:54 -07006671 lStatus = track->initCheck();
6672 if (lStatus != NO_ERROR) {
Glenn Kasten35295072013-10-07 09:27:06 -07006673 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
Haynes Mathew George03e9e832013-12-13 15:40:13 -08006674 // track must be cleared from the caller as the caller has the AF lock
Eric Laurent81784c32012-11-19 14:55:58 -08006675 goto Exit;
6676 }
6677 mTracks.add(track);
6678
6679 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
6680 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
6681 mAudioFlinger->btNrecIsOff();
6682 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
6683 setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
Glenn Kasten90e58b12013-07-31 16:16:02 -07006684
Eric Laurent05067782016-06-01 18:27:28 -07006685 if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) {
Glenn Kasten90e58b12013-07-31 16:16:02 -07006686 pid_t callingPid = IPCThreadState::self()->getCallingPid();
6687 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
6688 // so ask activity manager to do this on our behalf
6689 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp);
6690 }
Eric Laurent81784c32012-11-19 14:55:58 -08006691 }
Glenn Kasten05997e22014-03-13 15:08:33 -07006692
Eric Laurent81784c32012-11-19 14:55:58 -08006693 lStatus = NO_ERROR;
6694
6695Exit:
Glenn Kasten9156ef32013-08-06 15:39:08 -07006696 *status = lStatus;
Eric Laurent81784c32012-11-19 14:55:58 -08006697 return track;
6698}
6699
6700status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
6701 AudioSystem::sync_event_t event,
Glenn Kastend848eb42016-03-08 13:42:11 -08006702 audio_session_t triggerSession)
Eric Laurent81784c32012-11-19 14:55:58 -08006703{
6704 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession);
6705 sp<ThreadBase> strongMe = this;
6706 status_t status = NO_ERROR;
6707
6708 if (event == AudioSystem::SYNC_EVENT_NONE) {
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006709 recordTrack->clearSyncStartEvent();
Eric Laurent81784c32012-11-19 14:55:58 -08006710 } else if (event != AudioSystem::SYNC_EVENT_SAME) {
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006711 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
Eric Laurent81784c32012-11-19 14:55:58 -08006712 triggerSession,
6713 recordTrack->sessionId(),
6714 syncStartEventCallback,
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006715 recordTrack);
Eric Laurent81784c32012-11-19 14:55:58 -08006716 // Sync event can be cancelled by the trigger session if the track is not in a
6717 // compatible state in which case we start record immediately
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006718 if (recordTrack->mSyncStartEvent->isCancelled()) {
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006719 recordTrack->clearSyncStartEvent();
Eric Laurent81784c32012-11-19 14:55:58 -08006720 } else {
6721 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006722 recordTrack->mFramesToDrop = -
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08006723 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000);
Eric Laurent81784c32012-11-19 14:55:58 -08006724 }
6725 }
6726
6727 {
Glenn Kasten47c20702013-08-13 15:37:35 -07006728 // This section is a rendezvous between binder thread executing start() and RecordThread
Eric Laurent81784c32012-11-19 14:55:58 -08006729 AutoMutex lock(mLock);
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006730 if (mActiveTracks.indexOf(recordTrack) >= 0) {
6731 if (recordTrack->mState == TrackBase::PAUSING) {
6732 ALOGV("active record track PAUSING -> ACTIVE");
Glenn Kastenf10ffec2013-11-20 16:40:08 -08006733 recordTrack->mState = TrackBase::ACTIVE;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006734 } else {
6735 ALOGV("active record track state %d", recordTrack->mState);
Eric Laurent81784c32012-11-19 14:55:58 -08006736 }
6737 return status;
6738 }
6739
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08006740 // TODO consider other ways of handling this, such as changing the state to :STARTING and
6741 // adding the track to mActiveTracks after returning from AudioSystem::startInput(),
6742 // or using a separate command thread
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006743 recordTrack->mState = TrackBase::STARTING_1;
Glenn Kasten2b806402013-11-20 16:37:38 -08006744 mActiveTracks.add(recordTrack);
6745 mActiveTracksGen++;
Eric Laurent83b88082014-06-20 18:31:16 -07006746 status_t status = NO_ERROR;
6747 if (recordTrack->isExternalTrack()) {
6748 mLock.unlock();
Glenn Kastend848eb42016-03-08 13:42:11 -08006749 status = AudioSystem::startInput(mId, recordTrack->sessionId());
Eric Laurent83b88082014-06-20 18:31:16 -07006750 mLock.lock();
6751 // FIXME should verify that recordTrack is still in mActiveTracks
6752 if (status != NO_ERROR) {
6753 mActiveTracks.remove(recordTrack);
6754 mActiveTracksGen++;
6755 recordTrack->clearSyncStartEvent();
6756 ALOGV("RecordThread::start error %d", status);
6757 return status;
6758 }
Eric Laurent81784c32012-11-19 14:55:58 -08006759 }
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006760 // Catch up with current buffer indices if thread is already running.
6761 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront
6762 // was initialized to some value closer to the thread's mRsmpInFront, then the track could
6763 // see previously buffered data before it called start(), but with greater risk of overrun.
6764
Andy Hung73c02e42015-03-29 01:13:58 -07006765 recordTrack->mResamplerBufferProvider->reset();
Andy Hung97a893e2015-03-29 01:03:07 -07006766 // clear any converter state as new data will be discontinuous
6767 recordTrack->mRecordBufferConverter->reset();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006768 recordTrack->mState = TrackBase::STARTING_2;
Eric Laurent81784c32012-11-19 14:55:58 -08006769 // signal thread to start
Eric Laurent81784c32012-11-19 14:55:58 -08006770 mWaitWorkCV.broadcast();
Glenn Kasten2b806402013-11-20 16:37:38 -08006771 if (mActiveTracks.indexOf(recordTrack) < 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08006772 ALOGV("Record failed to start");
6773 status = BAD_VALUE;
6774 goto startError;
6775 }
Eric Laurent81784c32012-11-19 14:55:58 -08006776 return status;
6777 }
Glenn Kasten7c027242012-12-26 14:43:16 -08006778
Eric Laurent81784c32012-11-19 14:55:58 -08006779startError:
Eric Laurent83b88082014-06-20 18:31:16 -07006780 if (recordTrack->isExternalTrack()) {
Glenn Kastend848eb42016-03-08 13:42:11 -08006781 AudioSystem::stopInput(mId, recordTrack->sessionId());
Eric Laurent83b88082014-06-20 18:31:16 -07006782 }
Glenn Kasten25f4aa82014-02-07 10:50:43 -08006783 recordTrack->clearSyncStartEvent();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006784 // FIXME I wonder why we do not reset the state here?
Eric Laurent81784c32012-11-19 14:55:58 -08006785 return status;
6786}
6787
Eric Laurent81784c32012-11-19 14:55:58 -08006788void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
6789{
6790 sp<SyncEvent> strongEvent = event.promote();
6791
6792 if (strongEvent != 0) {
Eric Laurent8ea16e42014-02-20 16:26:11 -08006793 sp<RefBase> ptr = strongEvent->cookie().promote();
6794 if (ptr != 0) {
6795 RecordTrack *recordTrack = (RecordTrack *)ptr.get();
6796 recordTrack->handleSyncStartEvent(strongEvent);
6797 }
Eric Laurent81784c32012-11-19 14:55:58 -08006798 }
6799}
6800
Glenn Kastena8356f62013-07-25 14:37:52 -07006801bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
Eric Laurent81784c32012-11-19 14:55:58 -08006802 ALOGV("RecordThread::stop");
Glenn Kastena8356f62013-07-25 14:37:52 -07006803 AutoMutex _l(mLock);
Glenn Kasten2b806402013-11-20 16:37:38 -08006804 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) {
Eric Laurent81784c32012-11-19 14:55:58 -08006805 return false;
6806 }
Glenn Kasten47c20702013-08-13 15:37:35 -07006807 // note that threadLoop may still be processing the track at this point [without lock]
Eric Laurent81784c32012-11-19 14:55:58 -08006808 recordTrack->mState = TrackBase::PAUSING;
Eric Laurent5c25d562016-07-13 17:17:45 -07006809 // signal thread to stop
6810 mWaitWorkCV.broadcast();
Eric Laurent81784c32012-11-19 14:55:58 -08006811 // do not wait for mStartStopCond if exiting
6812 if (exitPending()) {
6813 return true;
6814 }
Glenn Kasten47c20702013-08-13 15:37:35 -07006815 // FIXME incorrect usage of wait: no explicit predicate or loop
Eric Laurent81784c32012-11-19 14:55:58 -08006816 mStartStopCond.wait(mLock);
Glenn Kasten2b806402013-11-20 16:37:38 -08006817 // if we have been restarted, recordTrack is in mActiveTracks here
6818 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08006819 ALOGV("Record stopped OK");
6820 return true;
6821 }
6822 return false;
6823}
6824
Glenn Kasten0f11b512014-01-31 16:18:54 -08006825bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const
Eric Laurent81784c32012-11-19 14:55:58 -08006826{
6827 return false;
6828}
6829
Glenn Kasten0f11b512014-01-31 16:18:54 -08006830status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08006831{
6832#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
6833 if (!isValidSyncEvent(event)) {
6834 return BAD_VALUE;
6835 }
6836
Glenn Kastend848eb42016-03-08 13:42:11 -08006837 audio_session_t eventSession = event->triggerSession();
Eric Laurent81784c32012-11-19 14:55:58 -08006838 status_t ret = NAME_NOT_FOUND;
6839
6840 Mutex::Autolock _l(mLock);
6841
6842 for (size_t i = 0; i < mTracks.size(); i++) {
6843 sp<RecordTrack> track = mTracks[i];
6844 if (eventSession == track->sessionId()) {
6845 (void) track->setSyncEvent(event);
6846 ret = NO_ERROR;
6847 }
6848 }
6849 return ret;
6850#else
6851 return BAD_VALUE;
6852#endif
6853}
6854
6855// destroyTrack_l() must be called with ThreadBase::mLock held
6856void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track)
6857{
Eric Laurentbfb1b832013-01-07 09:53:42 -08006858 track->terminate();
6859 track->mState = TrackBase::STOPPED;
Eric Laurent81784c32012-11-19 14:55:58 -08006860 // active tracks are removed by threadLoop()
Glenn Kasten2b806402013-11-20 16:37:38 -08006861 if (mActiveTracks.indexOf(track) < 0) {
Eric Laurent81784c32012-11-19 14:55:58 -08006862 removeTrack_l(track);
6863 }
6864}
6865
6866void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track)
6867{
6868 mTracks.remove(track);
6869 // need anything related to effects here?
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006870 if (track->isFastTrack()) {
6871 ALOG_ASSERT(!mFastTrackAvail);
6872 mFastTrackAvail = true;
6873 }
Eric Laurent81784c32012-11-19 14:55:58 -08006874}
6875
6876void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
6877{
6878 dumpInternals(fd, args);
6879 dumpTracks(fd, args);
6880 dumpEffectChains(fd, args);
6881}
6882
6883void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
6884{
Elliott Hughes87cebad2014-05-22 10:14:43 -07006885 dprintf(fd, "\nInput thread %p:\n", this);
Eric Laurent81784c32012-11-19 14:55:58 -08006886
Glenn Kasten44182c22015-03-05 17:12:23 -08006887 dumpBase(fd, args);
6888
6889 if (mActiveTracks.size() == 0) {
Elliott Hughes87cebad2014-05-22 10:14:43 -07006890 dprintf(fd, " No active record clients\n");
Eric Laurent81784c32012-11-19 14:55:58 -08006891 }
Glenn Kasten6e6704c2014-07-03 10:20:00 -07006892 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no");
Glenn Kasten6dbb5e32014-05-13 10:38:42 -07006893 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
Glenn Kasten17c9c992015-03-02 15:53:01 -08006894
Glenn Kasten2f90c512015-12-02 11:40:09 -08006895 // Make a non-atomic copy of fast capture dump state so it won't change underneath us
6896 // while we are dumping it. It may be inconsistent, but it won't mutate!
6897 // This is a large object so we place it on the heap.
6898 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
6899 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState);
6900 copy->dump(fd);
6901 delete copy;
Eric Laurent81784c32012-11-19 14:55:58 -08006902}
6903
Glenn Kasten0f11b512014-01-31 16:18:54 -08006904void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
Eric Laurent81784c32012-11-19 14:55:58 -08006905{
6906 const size_t SIZE = 256;
6907 char buffer[SIZE];
6908 String8 result;
6909
Marco Nelissenb2208842014-02-07 14:00:50 -08006910 size_t numtracks = mTracks.size();
6911 size_t numactive = mActiveTracks.size();
6912 size_t numactiveseen = 0;
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006913 dprintf(fd, " %zu Tracks", numtracks);
Marco Nelissenb2208842014-02-07 14:00:50 -08006914 if (numtracks) {
Glenn Kastenc42e9b42016-03-21 11:35:03 -07006915 dprintf(fd, " of which %zu are active\n", numactive);
Marco Nelissenb2208842014-02-07 14:00:50 -08006916 RecordTrack::appendDumpHeader(result);
6917 for (size_t i = 0; i < numtracks ; ++i) {
6918 sp<RecordTrack> track = mTracks[i];
6919 if (track != 0) {
6920 bool active = mActiveTracks.indexOf(track) >= 0;
6921 if (active) {
6922 numactiveseen++;
6923 }
6924 track->dump(buffer, SIZE, active);
6925 result.append(buffer);
6926 }
Eric Laurent81784c32012-11-19 14:55:58 -08006927 }
Marco Nelissenb2208842014-02-07 14:00:50 -08006928 } else {
Elliott Hughes87cebad2014-05-22 10:14:43 -07006929 dprintf(fd, "\n");
Eric Laurent81784c32012-11-19 14:55:58 -08006930 }
6931
Marco Nelissenb2208842014-02-07 14:00:50 -08006932 if (numactiveseen != numactive) {
6933 snprintf(buffer, SIZE, " The following tracks are in the active list but"
6934 " not in the track list\n");
Eric Laurent81784c32012-11-19 14:55:58 -08006935 result.append(buffer);
6936 RecordTrack::appendDumpHeader(result);
Marco Nelissenb2208842014-02-07 14:00:50 -08006937 for (size_t i = 0; i < numactive; ++i) {
Glenn Kasten2b806402013-11-20 16:37:38 -08006938 sp<RecordTrack> track = mActiveTracks[i];
Marco Nelissenb2208842014-02-07 14:00:50 -08006939 if (mTracks.indexOf(track) < 0) {
6940 track->dump(buffer, SIZE, true);
6941 result.append(buffer);
6942 }
Glenn Kasten2b806402013-11-20 16:37:38 -08006943 }
Eric Laurent81784c32012-11-19 14:55:58 -08006944
6945 }
6946 write(fd, result.string(), result.size());
6947}
6948
Andy Hung73c02e42015-03-29 01:13:58 -07006949
6950void AudioFlinger::RecordThread::ResamplerBufferProvider::reset()
6951{
6952 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
6953 RecordThread *recordThread = (RecordThread *) threadBase.get();
6954 mRsmpInFront = recordThread->mRsmpInRear;
6955 mRsmpInUnrel = 0;
6956}
6957
6958void AudioFlinger::RecordThread::ResamplerBufferProvider::sync(
6959 size_t *framesAvailable, bool *hasOverrun)
6960{
6961 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
6962 RecordThread *recordThread = (RecordThread *) threadBase.get();
6963 const int32_t rear = recordThread->mRsmpInRear;
6964 const int32_t front = mRsmpInFront;
6965 const ssize_t filled = rear - front;
6966
6967 size_t framesIn;
6968 bool overrun = false;
6969 if (filled < 0) {
6970 // should not happen, but treat like a massive overrun and re-sync
6971 framesIn = 0;
6972 mRsmpInFront = rear;
6973 overrun = true;
6974 } else if ((size_t) filled <= recordThread->mRsmpInFrames) {
6975 framesIn = (size_t) filled;
6976 } else {
6977 // client is not keeping up with server, but give it latest data
6978 framesIn = recordThread->mRsmpInFrames;
6979 mRsmpInFront = /* front = */ rear - framesIn;
6980 overrun = true;
6981 }
6982 if (framesAvailable != NULL) {
6983 *framesAvailable = framesIn;
6984 }
6985 if (hasOverrun != NULL) {
6986 *hasOverrun = overrun;
6987 }
6988}
6989
Eric Laurent81784c32012-11-19 14:55:58 -08006990// AudioBufferProvider interface
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006991status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
Glenn Kastend79072e2016-01-06 08:41:20 -08006992 AudioBufferProvider::Buffer* buffer)
Eric Laurent81784c32012-11-19 14:55:58 -08006993{
Andy Hung73c02e42015-03-29 01:13:58 -07006994 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote();
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006995 if (threadBase == 0) {
6996 buffer->frameCount = 0;
Glenn Kasten607fa3e2014-02-21 14:24:58 -08006997 buffer->raw = NULL;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08006998 return NOT_ENOUGH_DATA;
6999 }
7000 RecordThread *recordThread = (RecordThread *) threadBase.get();
7001 int32_t rear = recordThread->mRsmpInRear;
Andy Hung73c02e42015-03-29 01:13:58 -07007002 int32_t front = mRsmpInFront;
Glenn Kasten85948432013-08-19 12:09:05 -07007003 ssize_t filled = rear - front;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08007004 // FIXME should not be P2 (don't want to increase latency)
7005 // FIXME if client not keeping up, discard
Glenn Kasten607fa3e2014-02-21 14:24:58 -08007006 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames));
Glenn Kasten85948432013-08-19 12:09:05 -07007007 // 'filled' may be non-contiguous, so return only the first contiguous chunk
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08007008 front &= recordThread->mRsmpInFramesP2 - 1;
7009 size_t part1 = recordThread->mRsmpInFramesP2 - front;
Glenn Kasten85948432013-08-19 12:09:05 -07007010 if (part1 > (size_t) filled) {
7011 part1 = filled;
7012 }
7013 size_t ask = buffer->frameCount;
7014 ALOG_ASSERT(ask > 0);
7015 if (part1 > ask) {
7016 part1 = ask;
7017 }
7018 if (part1 == 0) {
Andy Hung73c02e42015-03-29 01:13:58 -07007019 // out of data is fine since the resampler will return a short-count.
Glenn Kasten85948432013-08-19 12:09:05 -07007020 buffer->raw = NULL;
7021 buffer->frameCount = 0;
Andy Hung73c02e42015-03-29 01:13:58 -07007022 mRsmpInUnrel = 0;
Glenn Kasten85948432013-08-19 12:09:05 -07007023 return NOT_ENOUGH_DATA;
Eric Laurent81784c32012-11-19 14:55:58 -08007024 }
7025
Andy Hung57446612015-04-19 23:56:46 -07007026 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize;
Glenn Kasten85948432013-08-19 12:09:05 -07007027 buffer->frameCount = part1;
Andy Hung73c02e42015-03-29 01:13:58 -07007028 mRsmpInUnrel = part1;
Eric Laurent81784c32012-11-19 14:55:58 -08007029 return NO_ERROR;
7030}
7031
7032// AudioBufferProvider interface
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08007033void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
7034 AudioBufferProvider::Buffer* buffer)
Eric Laurent81784c32012-11-19 14:55:58 -08007035{
Glenn Kasten85948432013-08-19 12:09:05 -07007036 size_t stepCount = buffer->frameCount;
7037 if (stepCount == 0) {
7038 return;
7039 }
Andy Hung73c02e42015-03-29 01:13:58 -07007040 ALOG_ASSERT(stepCount <= mRsmpInUnrel);
7041 mRsmpInUnrel -= stepCount;
7042 mRsmpInFront += stepCount;
Glenn Kasten85948432013-08-19 12:09:05 -07007043 buffer->raw = NULL;
Eric Laurent81784c32012-11-19 14:55:58 -08007044 buffer->frameCount = 0;
7045}
7046
Andy Hung97a893e2015-03-29 01:03:07 -07007047AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter(
7048 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
7049 uint32_t srcSampleRate,
7050 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
7051 uint32_t dstSampleRate) :
7052 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars
7053 // mSrcFormat
7054 // mSrcSampleRate
7055 // mDstChannelMask
7056 // mDstFormat
7057 // mDstSampleRate
7058 // mSrcChannelCount
7059 // mDstChannelCount
7060 // mDstFrameSize
7061 mBuf(NULL), mBufFrames(0), mBufFrameSize(0),
Andy Hungd330ee42015-04-20 13:23:41 -07007062 mResampler(NULL),
7063 mIsLegacyDownmix(false),
7064 mIsLegacyUpmix(false),
7065 mRequiresFloat(false),
7066 mInputConverterProvider(NULL)
Andy Hung97a893e2015-03-29 01:03:07 -07007067{
7068 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate,
7069 dstChannelMask, dstFormat, dstSampleRate);
7070}
7071
7072AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() {
7073 free(mBuf);
7074 delete mResampler;
Andy Hungd330ee42015-04-20 13:23:41 -07007075 delete mInputConverterProvider;
Andy Hung97a893e2015-03-29 01:03:07 -07007076}
7077
7078size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst,
7079 AudioBufferProvider *provider, size_t frames)
7080{
Andy Hungd330ee42015-04-20 13:23:41 -07007081 if (mInputConverterProvider != NULL) {
7082 mInputConverterProvider->setBufferProvider(provider);
7083 provider = mInputConverterProvider;
7084 }
7085
7086 if (mResampler == NULL) {
Andy Hung97a893e2015-03-29 01:03:07 -07007087 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
7088 mSrcSampleRate, mSrcFormat, mDstFormat);
7089
7090 AudioBufferProvider::Buffer buffer;
7091 for (size_t i = frames; i > 0; ) {
7092 buffer.frameCount = i;
Glenn Kastend79072e2016-01-06 08:41:20 -08007093 status_t status = provider->getNextBuffer(&buffer);
Andy Hung97a893e2015-03-29 01:03:07 -07007094 if (status != OK || buffer.frameCount == 0) {
7095 frames -= i; // cannot fill request.
7096 break;
7097 }
Andy Hungd330ee42015-04-20 13:23:41 -07007098 // format convert to destination buffer
7099 convertNoResampler(dst, buffer.raw, buffer.frameCount);
Andy Hung97a893e2015-03-29 01:03:07 -07007100
7101 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize;
7102 i -= buffer.frameCount;
7103 provider->releaseBuffer(&buffer);
7104 }
7105 } else {
7106 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x",
7107 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat);
7108
Andy Hungd330ee42015-04-20 13:23:41 -07007109 // reallocate buffer if needed
7110 if (mBufFrameSize != 0 && mBufFrames < frames) {
7111 free(mBuf);
7112 mBufFrames = frames;
7113 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
7114 }
Andy Hung97a893e2015-03-29 01:03:07 -07007115 // resampler accumulates, but we only have one source track
Andy Hungd330ee42015-04-20 13:23:41 -07007116 memset(mBuf, 0, frames * mBufFrameSize);
7117 frames = mResampler->resample((int32_t*)mBuf, frames, provider);
7118 // format convert to destination buffer
7119 convertResampler(dst, mBuf, frames);
Andy Hung97a893e2015-03-29 01:03:07 -07007120 }
7121 return frames;
7122}
7123
7124status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters(
7125 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat,
7126 uint32_t srcSampleRate,
7127 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat,
7128 uint32_t dstSampleRate)
7129{
7130 // quick evaluation if there is any change.
7131 if (mSrcFormat == srcFormat
7132 && mSrcChannelMask == srcChannelMask
7133 && mSrcSampleRate == srcSampleRate
7134 && mDstFormat == dstFormat
7135 && mDstChannelMask == dstChannelMask
7136 && mDstSampleRate == dstSampleRate) {
7137 return NO_ERROR;
7138 }
7139
Andy Hungdb4c0312015-05-06 08:46:52 -07007140 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x"
7141 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u",
7142 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate);
Andy Hung97a893e2015-03-29 01:03:07 -07007143 const bool valid =
7144 audio_is_input_channel(srcChannelMask)
7145 && audio_is_input_channel(dstChannelMask)
7146 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat)
7147 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat)
7148 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX)
7149 ; // no upsampling checks for now
7150 if (!valid) {
7151 return BAD_VALUE;
7152 }
7153
7154 mSrcFormat = srcFormat;
7155 mSrcChannelMask = srcChannelMask;
7156 mSrcSampleRate = srcSampleRate;
7157 mDstFormat = dstFormat;
7158 mDstChannelMask = dstChannelMask;
7159 mDstSampleRate = dstSampleRate;
7160
7161 // compute derived parameters
7162 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask);
7163 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask);
7164 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat);
7165
Andy Hungd330ee42015-04-20 13:23:41 -07007166 // do we need to resample?
7167 delete mResampler;
7168 mResampler = NULL;
7169 if (mSrcSampleRate != mDstSampleRate) {
7170 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT,
7171 mSrcChannelCount, mDstSampleRate);
7172 mResampler->setSampleRate(mSrcSampleRate);
7173 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT);
7174 }
7175
7176 // are we running legacy channel conversion modes?
7177 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO
7178 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK)
7179 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO;
7180 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO
7181 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO
7182 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK);
7183
7184 // do we need to process in float?
7185 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix;
7186
7187 // do we need a staging buffer to convert for destination (we can still optimize this)?
7188 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity
7189 if (mResampler != NULL) {
7190 mBufFrameSize = max(mSrcChannelCount, FCC_2)
7191 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
Andy Hunga97630b2015-07-22 23:27:24 -07007192 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float
Andy Hungd330ee42015-04-20 13:23:41 -07007193 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT);
7194 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) {
Andy Hung97a893e2015-03-29 01:03:07 -07007195 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat);
7196 } else {
7197 mBufFrameSize = 0;
7198 }
7199 mBufFrames = 0; // force the buffer to be resized.
7200
Andy Hungd330ee42015-04-20 13:23:41 -07007201 // do we need an input converter buffer provider to give us float?
7202 delete mInputConverterProvider;
7203 mInputConverterProvider = NULL;
7204 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) {
7205 mInputConverterProvider = new ReformatBufferProvider(
7206 audio_channel_count_from_in_mask(mSrcChannelMask),
7207 mSrcFormat,
7208 AUDIO_FORMAT_PCM_FLOAT,
7209 256 /* provider buffer frame count */);
7210 }
7211
7212 // do we need a remixer to do channel mask conversion
7213 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) {
7214 (void) memcpy_by_index_array_initialization_from_channel_mask(
7215 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask);
Andy Hung97a893e2015-03-29 01:03:07 -07007216 }
7217 return NO_ERROR;
7218}
7219
Andy Hungd330ee42015-04-20 13:23:41 -07007220void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler(
7221 void *dst, const void *src, size_t frames)
Andy Hung97a893e2015-03-29 01:03:07 -07007222{
Andy Hungd330ee42015-04-20 13:23:41 -07007223 // src is native type unless there is legacy upmix or downmix, whereupon it is float.
Andy Hung97a893e2015-03-29 01:03:07 -07007224 if (mBufFrameSize != 0 && mBufFrames < frames) {
7225 free(mBuf);
7226 mBufFrames = frames;
7227 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize);
7228 }
Andy Hungd330ee42015-04-20 13:23:41 -07007229 // do we need to do legacy upmix and downmix?
7230 if (mIsLegacyUpmix || mIsLegacyDownmix) {
Andy Hung97a893e2015-03-29 01:03:07 -07007231 void *dstBuf = mBuf != NULL ? mBuf : dst;
Andy Hungd330ee42015-04-20 13:23:41 -07007232 if (mIsLegacyUpmix) {
7233 upmix_to_stereo_float_from_mono_float((float *)dstBuf,
7234 (const float *)src, frames);
7235 } else /*mIsLegacyDownmix */ {
7236 downmix_to_mono_float_from_stereo_float((float *)dstBuf,
7237 (const float *)src, frames);
Andy Hung97a893e2015-03-29 01:03:07 -07007238 }
Andy Hungd330ee42015-04-20 13:23:41 -07007239 if (mBuf != NULL) {
7240 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT,
7241 frames * mDstChannelCount);
7242 }
7243 return;
7244 }
7245 // do we need to do channel mask conversion?
7246 if (mSrcChannelMask != mDstChannelMask) {
Andy Hung97a893e2015-03-29 01:03:07 -07007247 void *dstBuf = mBuf != NULL ? mBuf : dst;
Andy Hungd330ee42015-04-20 13:23:41 -07007248 memcpy_by_index_array(dstBuf, mDstChannelCount,
7249 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames);
7250 if (dstBuf == dst) {
7251 return; // format is the same
7252 }
7253 }
7254 // convert to destination buffer
7255 const void *convertBuf = mBuf != NULL ? mBuf : src;
7256 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat,
7257 frames * mDstChannelCount);
7258}
7259
7260void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler(
7261 void *dst, /*not-a-const*/ void *src, size_t frames)
7262{
7263 // src buffer format is ALWAYS float when entering this routine
7264 if (mIsLegacyUpmix) {
7265 ; // mono to stereo already handled by resampler
7266 } else if (mIsLegacyDownmix
7267 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) {
7268 // the resampler outputs stereo for mono input channel (a feature?)
7269 // must convert to mono
7270 downmix_to_mono_float_from_stereo_float((float *)src,
7271 (const float *)src, frames);
7272 } else if (mSrcChannelMask != mDstChannelMask) {
7273 // convert to mono channel again for channel mask conversion (could be skipped
7274 // with further optimization).
Andy Hung97a893e2015-03-29 01:03:07 -07007275 if (mSrcChannelCount == 1) {
Andy Hungd330ee42015-04-20 13:23:41 -07007276 downmix_to_mono_float_from_stereo_float((float *)src,
7277 (const float *)src, frames);
Andy Hung97a893e2015-03-29 01:03:07 -07007278 }
Andy Hungd330ee42015-04-20 13:23:41 -07007279 // convert to destination format (in place, OK as float is larger than other types)
7280 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) {
7281 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
7282 frames * mSrcChannelCount);
7283 }
7284 // channel convert and save to dst
7285 memcpy_by_index_array(dst, mDstChannelCount,
7286 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames);
7287 return;
Andy Hung97a893e2015-03-29 01:03:07 -07007288 }
Andy Hungd330ee42015-04-20 13:23:41 -07007289 // convert to destination format and save to dst
7290 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT,
7291 frames * mDstChannelCount);
Andy Hung97a893e2015-03-29 01:03:07 -07007292}
7293
Eric Laurent10351942014-05-08 18:49:52 -07007294bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
7295 status_t& status)
Eric Laurent81784c32012-11-19 14:55:58 -08007296{
7297 bool reconfig = false;
7298
Eric Laurent10351942014-05-08 18:49:52 -07007299 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08007300
Eric Laurent10351942014-05-08 18:49:52 -07007301 audio_format_t reqFormat = mFormat;
7302 uint32_t samplingRate = mSampleRate;
Glenn Kastene1635ec2015-06-08 15:46:49 -07007303 // 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 -07007304 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount);
7305
7306 AudioParameter param = AudioParameter(keyValuePair);
7307 int value;
Haynes Mathew George9ce67b52015-09-30 11:40:47 -07007308
7309 // scope for AutoPark extends to end of method
7310 AutoPark<FastCapture> park(mFastCapture);
7311
Eric Laurent10351942014-05-08 18:49:52 -07007312 // TODO Investigate when this code runs. Check with audio policy when a sample rate and
7313 // channel count change can be requested. Do we mandate the first client defines the
7314 // HAL sampling rate and channel count or do we allow changes on the fly?
7315 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
7316 samplingRate = value;
7317 reconfig = true;
7318 }
7319 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
Andy Hung97a893e2015-03-29 01:03:07 -07007320 if (!audio_is_linear_pcm((audio_format_t) value)) {
Eric Laurent10351942014-05-08 18:49:52 -07007321 status = BAD_VALUE;
7322 } else {
7323 reqFormat = (audio_format_t) value;
Eric Laurent81784c32012-11-19 14:55:58 -08007324 reconfig = true;
7325 }
Eric Laurent10351942014-05-08 18:49:52 -07007326 }
7327 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
7328 audio_channel_mask_t mask = (audio_channel_mask_t) value;
Andy Hungd330ee42015-04-20 13:23:41 -07007329 if (!audio_is_input_channel(mask) ||
7330 audio_channel_count_from_in_mask(mask) > FCC_8) {
Eric Laurent10351942014-05-08 18:49:52 -07007331 status = BAD_VALUE;
7332 } else {
7333 channelMask = mask;
7334 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08007335 }
Eric Laurent10351942014-05-08 18:49:52 -07007336 }
7337 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
7338 // do not accept frame count changes if tracks are open as the track buffer
7339 // size depends on frame count and correct behavior would not be guaranteed
7340 // if frame count is changed after track creation
7341 if (mActiveTracks.size() > 0) {
7342 status = INVALID_OPERATION;
7343 } else {
7344 reconfig = true;
Eric Laurent81784c32012-11-19 14:55:58 -08007345 }
Eric Laurent10351942014-05-08 18:49:52 -07007346 }
7347 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
7348 // forward device change to effects that have requested to be
7349 // aware of attached audio device.
7350 for (size_t i = 0; i < mEffectChains.size(); i++) {
7351 mEffectChains[i]->setDevice_l(value);
Eric Laurent81784c32012-11-19 14:55:58 -08007352 }
Eric Laurent81784c32012-11-19 14:55:58 -08007353
Eric Laurent10351942014-05-08 18:49:52 -07007354 // store input device and output device but do not forward output device to audio HAL.
7355 // Note that status is ignored by the caller for output device
7356 // (see AudioFlinger::setParameters()
7357 if (audio_is_output_devices(value)) {
7358 mOutDevice = value;
7359 status = BAD_VALUE;
7360 } else {
7361 mInDevice = value;
Eric Laurente8726fe2015-06-26 09:39:24 -07007362 if (value != AUDIO_DEVICE_NONE) {
7363 mPrevInDevice = value;
7364 }
Eric Laurent10351942014-05-08 18:49:52 -07007365 // disable AEC and NS if the device is a BT SCO headset supporting those
7366 // pre processings
7367 if (mTracks.size() > 0) {
7368 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
7369 mAudioFlinger->btNrecIsOff();
7370 for (size_t i = 0; i < mTracks.size(); i++) {
7371 sp<RecordTrack> track = mTracks[i];
7372 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
7373 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
Eric Laurent81784c32012-11-19 14:55:58 -08007374 }
7375 }
7376 }
Eric Laurent10351942014-05-08 18:49:52 -07007377 }
7378 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
7379 mAudioSource != (audio_source_t)value) {
7380 // forward device change to effects that have requested to be
7381 // aware of attached audio device.
7382 for (size_t i = 0; i < mEffectChains.size(); i++) {
7383 mEffectChains[i]->setAudioSource_l((audio_source_t)value);
Eric Laurent81784c32012-11-19 14:55:58 -08007384 }
Eric Laurent10351942014-05-08 18:49:52 -07007385 mAudioSource = (audio_source_t)value;
7386 }
Glenn Kastene198c362013-08-13 09:13:36 -07007387
Eric Laurent10351942014-05-08 18:49:52 -07007388 if (status == NO_ERROR) {
7389 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7390 keyValuePair.string());
7391 if (status == INVALID_OPERATION) {
7392 inputStandBy();
Eric Laurent81784c32012-11-19 14:55:58 -08007393 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7394 keyValuePair.string());
Eric Laurent10351942014-05-08 18:49:52 -07007395 }
7396 if (reconfig) {
7397 if (status == BAD_VALUE &&
Andy Hung97a893e2015-03-29 01:03:07 -07007398 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) &&
7399 audio_is_linear_pcm(reqFormat) &&
Eric Laurent10351942014-05-08 18:49:52 -07007400 (mInput->stream->common.get_sample_rate(&mInput->stream->common)
Andy Hung97a893e2015-03-29 01:03:07 -07007401 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) &&
Andy Hunge5412692014-05-16 11:25:07 -07007402 audio_channel_count_from_in_mask(
Andy Hungd1abb8f2015-05-05 23:42:34 -07007403 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) {
Eric Laurent10351942014-05-08 18:49:52 -07007404 status = NO_ERROR;
Eric Laurent81784c32012-11-19 14:55:58 -08007405 }
Eric Laurent10351942014-05-08 18:49:52 -07007406 if (status == NO_ERROR) {
7407 readInputParameters_l();
Eric Laurent73e26b62015-04-27 16:55:58 -07007408 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
Eric Laurent81784c32012-11-19 14:55:58 -08007409 }
7410 }
Eric Laurent81784c32012-11-19 14:55:58 -08007411 }
Eric Laurent10351942014-05-08 18:49:52 -07007412
Eric Laurent81784c32012-11-19 14:55:58 -08007413 return reconfig;
7414}
7415
7416String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
7417{
Eric Laurent81784c32012-11-19 14:55:58 -08007418 Mutex::Autolock _l(mLock);
7419 if (initCheck() != NO_ERROR) {
Glenn Kastend8ea6992013-07-16 14:17:15 -07007420 return String8();
Eric Laurent81784c32012-11-19 14:55:58 -08007421 }
7422
Glenn Kastend8ea6992013-07-16 14:17:15 -07007423 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
7424 const String8 out_s8(s);
Eric Laurent81784c32012-11-19 14:55:58 -08007425 free(s);
7426 return out_s8;
7427}
7428
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07007429void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) {
Eric Laurent73e26b62015-04-27 16:55:58 -07007430 sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
7431
7432 desc->mIoHandle = mId;
Eric Laurent81784c32012-11-19 14:55:58 -08007433
7434 switch (event) {
Eric Laurent73e26b62015-04-27 16:55:58 -07007435 case AUDIO_INPUT_OPENED:
7436 case AUDIO_INPUT_CONFIG_CHANGED:
Eric Laurent296fb132015-05-01 11:38:42 -07007437 desc->mPatch = mPatch;
Eric Laurent73e26b62015-04-27 16:55:58 -07007438 desc->mChannelMask = mChannelMask;
7439 desc->mSamplingRate = mSampleRate;
7440 desc->mFormat = mFormat;
7441 desc->mFrameCount = mFrameCount;
Glenn Kasten4a8308b2016-04-18 14:10:01 -07007442 desc->mFrameCountHAL = mFrameCount;
Eric Laurent73e26b62015-04-27 16:55:58 -07007443 desc->mLatency = 0;
Eric Laurent81784c32012-11-19 14:55:58 -08007444 break;
7445
Eric Laurent73e26b62015-04-27 16:55:58 -07007446 case AUDIO_INPUT_CLOSED:
Eric Laurent81784c32012-11-19 14:55:58 -08007447 default:
7448 break;
7449 }
Eric Laurent7c1ec5f2015-07-09 14:52:47 -07007450 mAudioFlinger->ioConfigChanged(event, desc, pid);
Eric Laurent81784c32012-11-19 14:55:58 -08007451}
7452
Glenn Kastendeca2ae2014-02-07 10:25:56 -08007453void AudioFlinger::RecordThread::readInputParameters_l()
Eric Laurent81784c32012-11-19 14:55:58 -08007454{
Eric Laurent81784c32012-11-19 14:55:58 -08007455 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
7456 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
Andy Hunge5412692014-05-16 11:25:07 -07007457 mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
Andy Hungd330ee42015-04-20 13:23:41 -07007458 if (mChannelCount > FCC_8) {
7459 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8);
7460 }
Andy Hung463be252014-07-10 16:56:07 -07007461 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common);
7462 mFormat = mHALFormat;
Andy Hungd330ee42015-04-20 13:23:41 -07007463 if (!audio_is_linear_pcm(mFormat)) {
7464 ALOGE("HAL format %#x is not linear pcm", mFormat);
Glenn Kasten291bb6d2013-07-16 17:23:39 -07007465 }
Eric Laurent665470b2014-07-03 16:37:08 -07007466 mFrameSize = audio_stream_in_frame_size(mInput->stream);
Glenn Kasten548efc92012-11-29 08:48:51 -08007467 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
7468 mFrameCount = mBufferSize / mFrameSize;
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08007469 // This is the formula for calculating the temporary buffer size.
Glenn Kastene8426142014-02-28 16:45:03 -08007470 // 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 -07007471 // 1 full output buffer, regardless of the alignment of the available input.
Glenn Kastene8426142014-02-28 16:45:03 -08007472 // The value is somewhat arbitrary, and could probably be even larger.
Glenn Kasten6dd62fb2013-12-05 16:35:58 -08007473 // A larger value should allow more old data to be read after a track calls start(),
7474 // without increasing latency.
Andy Hung97a893e2015-03-29 01:03:07 -07007475 //
7476 // Note this is independent of the maximum downsampling ratio permitted for capture.
Glenn Kastene8426142014-02-28 16:45:03 -08007477 mRsmpInFrames = mFrameCount * 7;
Glenn Kasten85948432013-08-19 12:09:05 -07007478 mRsmpInFramesP2 = roundup(mRsmpInFrames);
Andy Hung57446612015-04-19 23:56:46 -07007479 free(mRsmpInBuffer);
Andy Hung0a01c2f2015-09-21 12:44:54 -07007480 mRsmpInBuffer = NULL;
Glenn Kasten49d00ad2014-07-21 11:22:03 -07007481
7482 // TODO optimize audio capture buffer sizes ...
7483 // Here we calculate the size of the sliding buffer used as a source
7484 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7).
7485 // For current HAL frame counts, this is usually 2048 = 40 ms. It would
7486 // be better to have it derived from the pipe depth in the long term.
7487 // The current value is higher than necessary. However it should not add to latency.
7488
Glenn Kasten85948432013-08-19 12:09:05 -07007489 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
Glenn Kasten1b291842016-07-18 14:55:21 -07007490 mRsmpInFramesOA = mRsmpInFramesP2 + mFrameCount - 1;
7491 (void)posix_memalign(&mRsmpInBuffer, 32, mRsmpInFramesOA * mFrameSize);
7492 memset(mRsmpInBuffer, 0, mRsmpInFramesOA * mFrameSize); // if posix_memalign fails, will segv here.
Eric Laurent81784c32012-11-19 14:55:58 -08007493
Glenn Kasten4cc0a6a2014-02-17 14:31:46 -08007494 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
7495 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
Eric Laurent81784c32012-11-19 14:55:58 -08007496}
7497
Glenn Kasten5f972c02014-01-13 09:59:31 -08007498uint32_t AudioFlinger::RecordThread::getInputFramesLost()
Eric Laurent81784c32012-11-19 14:55:58 -08007499{
7500 Mutex::Autolock _l(mLock);
7501 if (initCheck() != NO_ERROR) {
7502 return 0;
7503 }
7504
7505 return mInput->stream->get_input_frames_lost(mInput->stream);
7506}
7507
Eric Laurent4c415062016-06-17 16:14:16 -07007508// hasAudioSession_l() must be called with ThreadBase::mLock held
7509uint32_t AudioFlinger::RecordThread::hasAudioSession_l(audio_session_t sessionId) const
Eric Laurent81784c32012-11-19 14:55:58 -08007510{
Eric Laurent81784c32012-11-19 14:55:58 -08007511 uint32_t result = 0;
7512 if (getEffectChain_l(sessionId) != 0) {
7513 result = EFFECT_SESSION;
7514 }
7515
7516 for (size_t i = 0; i < mTracks.size(); ++i) {
7517 if (sessionId == mTracks[i]->sessionId()) {
7518 result |= TRACK_SESSION;
Eric Laurent4c415062016-06-17 16:14:16 -07007519 if (mTracks[i]->isFastTrack()) {
7520 result |= FAST_SESSION;
7521 }
Eric Laurent81784c32012-11-19 14:55:58 -08007522 break;
7523 }
7524 }
7525
7526 return result;
7527}
7528
Glenn Kastend848eb42016-03-08 13:42:11 -08007529KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const
Eric Laurent81784c32012-11-19 14:55:58 -08007530{
Glenn Kastend848eb42016-03-08 13:42:11 -08007531 KeyedVector<audio_session_t, bool> ids;
Eric Laurent81784c32012-11-19 14:55:58 -08007532 Mutex::Autolock _l(mLock);
7533 for (size_t j = 0; j < mTracks.size(); ++j) {
7534 sp<RecordThread::RecordTrack> track = mTracks[j];
Glenn Kastend848eb42016-03-08 13:42:11 -08007535 audio_session_t sessionId = track->sessionId();
Eric Laurent81784c32012-11-19 14:55:58 -08007536 if (ids.indexOfKey(sessionId) < 0) {
7537 ids.add(sessionId, true);
7538 }
7539 }
7540 return ids;
7541}
7542
7543AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
7544{
7545 Mutex::Autolock _l(mLock);
7546 AudioStreamIn *input = mInput;
7547 mInput = NULL;
7548 return input;
7549}
7550
7551// this method must always be called either with ThreadBase mLock held or inside the thread loop
7552audio_stream_t* AudioFlinger::RecordThread::stream() const
7553{
7554 if (mInput == NULL) {
7555 return NULL;
7556 }
7557 return &mInput->stream->common;
7558}
7559
7560status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
7561{
7562 // only one chain per input thread
7563 if (mEffectChains.size() != 0) {
Eric Laurentaaa44472014-09-12 17:41:50 -07007564 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this);
Eric Laurent81784c32012-11-19 14:55:58 -08007565 return INVALID_OPERATION;
7566 }
7567 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
Eric Laurentaaa44472014-09-12 17:41:50 -07007568 chain->setThread(this);
Eric Laurent81784c32012-11-19 14:55:58 -08007569 chain->setInBuffer(NULL);
7570 chain->setOutBuffer(NULL);
7571
7572 checkSuspendOnAddEffectChain_l(chain);
7573
Eric Laurent1b928682014-10-02 19:41:47 -07007574 // make sure enabled pre processing effects state is communicated to the HAL as we
7575 // just moved them to a new input stream.
7576 chain->syncHalEffectsState();
7577
Eric Laurent81784c32012-11-19 14:55:58 -08007578 mEffectChains.add(chain);
7579
7580 return NO_ERROR;
7581}
7582
7583size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
7584{
7585 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
7586 ALOGW_IF(mEffectChains.size() != 1,
Glenn Kastenc42e9b42016-03-21 11:35:03 -07007587 "removeEffectChain_l() %p invalid chain size %zu on thread %p",
Eric Laurent81784c32012-11-19 14:55:58 -08007588 chain.get(), mEffectChains.size(), this);
7589 if (mEffectChains.size() == 1) {
7590 mEffectChains.removeAt(0);
7591 }
7592 return 0;
7593}
7594
Eric Laurent1c333e22014-05-20 10:48:17 -07007595status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch,
7596 audio_patch_handle_t *handle)
7597{
7598 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07007599
7600 // store new device and send to effects
7601 mInDevice = patch->sources[0].ext.device.type;
Eric Laurent296fb132015-05-01 11:38:42 -07007602 mPatch = *patch;
Eric Laurent054d9d32015-04-24 08:48:48 -07007603 for (size_t i = 0; i < mEffectChains.size(); i++) {
7604 mEffectChains[i]->setDevice_l(mInDevice);
7605 }
7606
7607 // disable AEC and NS if the device is a BT SCO headset supporting those
7608 // pre processings
7609 if (mTracks.size() > 0) {
7610 bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
7611 mAudioFlinger->btNrecIsOff();
7612 for (size_t i = 0; i < mTracks.size(); i++) {
7613 sp<RecordTrack> track = mTracks[i];
7614 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId());
7615 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId());
7616 }
7617 }
7618
7619 // store new source and send to effects
7620 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) {
7621 mAudioSource = patch->sinks[0].ext.mix.usecase.source;
Eric Laurent1c333e22014-05-20 10:48:17 -07007622 for (size_t i = 0; i < mEffectChains.size(); i++) {
Eric Laurent054d9d32015-04-24 08:48:48 -07007623 mEffectChains[i]->setAudioSource_l(mAudioSource);
Eric Laurent1c333e22014-05-20 10:48:17 -07007624 }
Eric Laurent054d9d32015-04-24 08:48:48 -07007625 }
Eric Laurent1c333e22014-05-20 10:48:17 -07007626
Eric Laurent054d9d32015-04-24 08:48:48 -07007627 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
Mikhail Naganove4f1f632016-08-31 11:35:10 -07007628 sp<DeviceHalInterface> hwDevice = mInput->audioHwDev->hwDevice();
7629 status = hwDevice->createAudioPatch(patch->num_sources,
7630 patch->sources,
7631 patch->num_sinks,
7632 patch->sinks,
7633 handle);
Eric Laurent1c333e22014-05-20 10:48:17 -07007634 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07007635 char *address;
7636 if (strcmp(patch->sources[0].ext.device.address, "") != 0) {
7637 address = audio_device_address_to_parameter(
7638 patch->sources[0].ext.device.type,
7639 patch->sources[0].ext.device.address);
7640 } else {
7641 address = (char *)calloc(1, 1);
7642 }
7643 AudioParameter param = AudioParameter(String8(address));
7644 free(address);
7645 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING),
7646 (int)patch->sources[0].ext.device.type);
7647 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE),
7648 (int)patch->sinks[0].ext.mix.usecase.source);
7649 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7650 param.toString().string());
7651 *handle = AUDIO_PATCH_HANDLE_NONE;
Eric Laurent1c333e22014-05-20 10:48:17 -07007652 }
Eric Laurent054d9d32015-04-24 08:48:48 -07007653
Eric Laurente8726fe2015-06-26 09:39:24 -07007654 if (mInDevice != mPrevInDevice) {
7655 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
7656 mPrevInDevice = mInDevice;
7657 }
Eric Laurent296fb132015-05-01 11:38:42 -07007658
Eric Laurent1c333e22014-05-20 10:48:17 -07007659 return status;
7660}
7661
7662status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
7663{
7664 status_t status = NO_ERROR;
Eric Laurent054d9d32015-04-24 08:48:48 -07007665
7666 mInDevice = AUDIO_DEVICE_NONE;
7667
Eric Laurent1c333e22014-05-20 10:48:17 -07007668 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
Mikhail Naganove4f1f632016-08-31 11:35:10 -07007669 sp<DeviceHalInterface> hwDevice = mInput->audioHwDev->hwDevice();
7670 status = hwDevice->releaseAudioPatch(handle);
Eric Laurent1c333e22014-05-20 10:48:17 -07007671 } else {
Eric Laurent054d9d32015-04-24 08:48:48 -07007672 AudioParameter param;
7673 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0);
7674 status = mInput->stream->common.set_parameters(&mInput->stream->common,
7675 param.toString().string());
Eric Laurent1c333e22014-05-20 10:48:17 -07007676 }
7677 return status;
7678}
7679
Eric Laurent83b88082014-06-20 18:31:16 -07007680void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record)
7681{
7682 Mutex::Autolock _l(mLock);
7683 mTracks.add(record);
7684}
7685
7686void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record)
7687{
7688 Mutex::Autolock _l(mLock);
7689 destroyTrack_l(record);
7690}
7691
7692void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config)
7693{
7694 ThreadBase::getAudioPortConfig(config);
7695 config->role = AUDIO_PORT_ROLE_SINK;
7696 config->ext.mix.hw_module = mInput->audioHwDev->handle();
7697 config->ext.mix.usecase.source = mAudioSource;
7698}
Eric Laurent1c333e22014-05-20 10:48:17 -07007699
Glenn Kasten63238ef2015-03-02 15:50:29 -08007700} // namespace android