Threads.cpp revision 5a8a95de6dad1a3bcf3da5a37b35766e89086e13
1cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber/* 2cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** 3cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** Copyright 2012, The Android Open Source Project 4cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** 5cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** Licensed under the Apache License, Version 2.0 (the "License"); 6cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** you may not use this file except in compliance with the License. 7cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** You may obtain a copy of the License at 8cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** 9cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** http://www.apache.org/licenses/LICENSE-2.0 10cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** 11cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** Unless required by applicable law or agreed to in writing, software 12cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** distributed under the License is distributed on an "AS IS" BASIS, 13cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** See the License for the specific language governing permissions and 15cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber** limitations under the License. 16cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber*/ 176e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber 186e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber 196e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber#define LOG_TAG "AudioFlinger" 206e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber//#define LOG_NDEBUG 0 21cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#define ATRACE_TAG ATRACE_TAG_AUDIO 22cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 23cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include "Configuration.h" 246a63a939601645404fd98f58c19cc38ca818d99eAndreas Huber#include <math.h> 2585f12e9b9062402d6110df3f7099707912040edbAndreas Huber#include <fcntl.h> 26cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <linux/futex.h> 2785f12e9b9062402d6110df3f7099707912040edbAndreas Huber#include <sys/stat.h> 28cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <sys/syscall.h> 29cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <cutils/properties.h> 30cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <media/AudioParameter.h> 31cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <media/AudioResamplerPublic.h> 32cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <utils/Log.h> 33cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <utils/Trace.h> 34cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 358dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#include <private/media/AudioTrackShared.h> 3632f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber#include <hardware/audio.h> 37cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <audio_effects/effect_ns.h> 389bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang#include <audio_effects/effect_aec.h> 39cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <audio_utils/primitives.h> 40b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Cross#include <audio_utils/format.h> 41b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Cross#include <audio_utils/minifloat.h> 42cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 43cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber// NBAIO implementations 446e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber#include <media/nbaio/AudioStreamInSource.h> 456e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber#include <media/nbaio/AudioStreamOutSink.h> 463856b090cd04ba5dd4a59a12430ed724d5995909Steve Block#include <media/nbaio/MonoPipe.h> 476e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber#include <media/nbaio/MonoPipeReader.h> 48cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <media/nbaio/Pipe.h> 49cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <media/nbaio/PipeReader.h> 50cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <media/nbaio/SourceAudioBufferProvider.h> 51d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang 52d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang#include <powermanager/PowerManager.h> 53cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 548dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#include <common_time/cc_helper.h> 558dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#include <common_time/local_clock.h> 568dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 57540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim#include "AudioFlinger.h" 58540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim#include "AudioMixer.h" 59540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim#include "FastMixer.h" 60cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include "FastCapture.h" 6187f2a558dd12043631e12c361abef301bf603140Andreas Huber#include "ServiceUtilities.h" 6287f2a558dd12043631e12c361abef301bf603140Andreas Huber#include "SchedulingPolicyService.h" 63540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 64cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#ifdef ADD_BATTERY_DATA 6532f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber#include <media/IMediaPlayerService.h> 6632f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber#include <media/IMediaDeathNotifier.h> 6732f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber#endif 68f933441648ef6a71dee783d733aac17b9508b452Andreas Huber 692a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber#ifdef DEBUG_CPU_USAGE 70cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#include <cpustats/CentralTendencyStatistics.h> 71bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih#include <cpustats/ThreadCpuUsage.h> 72cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber#endif 73bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber 74bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber// ---------------------------------------------------------------------------- 7543c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber 7643c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber// Note: the following macro is used for extremely verbose logging message. In 7743c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 7843c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber// 0; but one side effect of this is to turn all LOGV's as well. Some messages 79386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber// are so verbose that we want to suppress them even when we have ALOG_ASSERT 80386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber// turned on. Do not uncomment the #def below unless you really know what you 81386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber// are doing and want to see all of the extremely verbose messages. 82386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber//#define VERY_VERY_VERBOSE_LOGGING 83386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber#ifdef VERY_VERY_VERBOSE_LOGGING 84386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber#define ALOGVV ALOGV 858dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#else 868dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#define ALOGVV(a...) do { } while(0) 878dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#endif 888dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 8987f2a558dd12043631e12c361abef301bf603140Andreas Huber// TODO: Move these macro/inlines to a header file. 9087f2a558dd12043631e12c361abef301bf603140Andreas Huber#define max(a, b) ((a) > (b) ? (a) : (b)) 9187f2a558dd12043631e12c361abef301bf603140Andreas Hubertemplate <typename T> 9287f2a558dd12043631e12c361abef301bf603140Andreas Huberstatic inline T min(const T& a, const T& b) 93cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber{ 949bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang return a < b ? a : b; 959bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang} 969bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang 979bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhangnamespace android { 989bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang 99f933441648ef6a71dee783d733aac17b9508b452Andreas Huber// retry counts for buffer fill timeout 100386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber// 50 * ~20msecs = 1 second 101cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huberstatic const int8_t kMaxTrackRetries = 50; 102cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huberstatic const int8_t kMaxTrackStartupRetries = 50; 103bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber// allow less retry attempts on direct output thread. 104bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber// direct outputs can be a scarce resource in audio hardware and should 105799c9682b3776a55d234396aee4a302437150c26Chong Zhang// be released as quickly as possible. 106cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huberstatic const int8_t kMaxTrackRetriesDirect = 2; 10706528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber 108799c9682b3776a55d234396aee4a302437150c26Chong Zhang// don't warn about blocked writes or record buffer overflows more often than this 1099bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhangstatic const nsecs_t kWarningThrottleNs = seconds(5); 110cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 111cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber// RecordThread loop sleep time upon application overrun or audio HAL read error 112cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huberstatic const int kRecordThreadSleepUs = 5000; 113cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 114cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber// maximum time to wait in sendConfigEvent_l() for a status to be received 11587f2a558dd12043631e12c361abef301bf603140Andreas Huberstatic const nsecs_t kConfigEventTimeoutNs = seconds(2); 11687f2a558dd12043631e12c361abef301bf603140Andreas Huber 11787f2a558dd12043631e12c361abef301bf603140Andreas Huber// minimum sleep time for the mixer thread loop when tracks are active but in underrun 11887f2a558dd12043631e12c361abef301bf603140Andreas Huberstatic const uint32_t kMinThreadSleepTimeUs = 5000; 119cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber// maximum divider applied to the active sleep time in the mixer thread loop 120bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huberstatic const uint32_t kMaxThreadSleepTimeShift = 2; 121bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber 122bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber// minimum normal sink buffer size, expressed in milliseconds rather than frames 123bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huberstatic const uint32_t kMinNormalSinkBufferSizeMs = 20; 124540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// maximum normal sink buffer size 1259558f6dd733dc450270f38b9a139d384d273ce0aWei Jiastatic const uint32_t kMaxNormalSinkBufferSizeMs = 24; 126540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 1275403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber// Offloaded output thread standby delay: allows track transition without going to standby 12887f2a558dd12043631e12c361abef301bf603140Andreas Huberstatic const nsecs_t kOffloadStandbyDelayNs = seconds(1); 129cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 130540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// Whether to use fast mixer 131540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kimstatic const enum { 132cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber FastMixer_Never, // never initialize or use: for debugging only 13332f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber FastMixer_Always, // always initialize and use, even if not needed: for debugging only 13432f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber // normal mixer multiplier is 1 13532f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber FastMixer_Static, // initialize if needed, then use all the time if initialized, 136f933441648ef6a71dee783d733aac17b9508b452Andreas Huber // multiplier is calculated based on min & max normal mixer buffer size 1372a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 138cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // multiplier is calculated based on min & max normal mixer buffer size 139cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // FIXME for FastMixer_Dynamic: 140bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih // Supporting this option will require fixing HALs that can't handle large writes. 141bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih // For example, one HAL implementation returns an error from a large write, 1420852843d304006e3ab333081fddda13b07193de8Robert Shih // and another HAL implementation corrupts memory, possibly in the sample rate converter. 143bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih // We could either fix the HAL implementations, or provide a wrapper that breaks 144cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 145cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} kUseFastMixer = FastMixer_Static; 146cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 147cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber// Whether to use fast capture 148bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huberstatic const enum { 149cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber FastCapture_Never, // never initialize or use: for debugging only 150cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber FastCapture_Always, // always initialize and use, even if not needed: for debugging only 15187f2a558dd12043631e12c361abef301bf603140Andreas Huber FastCapture_Static, // initialize if needed, then use all the time if initialized 15287f2a558dd12043631e12c361abef301bf603140Andreas Huber} kUseFastCapture = FastCapture_Static; 153cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 154cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber// Priorities for requestPriority 155cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huberstatic const int kPriorityAudioApp = 2; 156cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huberstatic const int kPriorityFastMixer = 3; 157be9634d071e79b72a42a4504f64eda9e2a0bceb8Marco Nelissenstatic const int kPriorityFastCapture = 3; 158cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 15990a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 1609558f6dd733dc450270f38b9a139d384d273ce0aWei Jia// for the track. The client then sub-divides this into smaller buffers for its use. 16190a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 162386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber// So for now we just assume that client is double-buffered for fast tracks. 1636a63a939601645404fd98f58c19cc38ca818d99eAndreas Huber// FIXME It would be better for client to tell AudioFlinger the value of N, 164540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// so AudioFlinger could allocate the right amount of memory. 165540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// See the client's minBufCount and mNotificationFramesAct calculations for details. 166540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 167540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// This is the default value, if not specified by property. 168540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kimstatic const int kFastTrackMultiplier = 2; 169540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 170540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// The minimum and maximum allowed values 171540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kimstatic const int kFastTrackMultiplierMin = 1; 172540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kimstatic const int kFastTrackMultiplierMax = 2; 173540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 174540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 175cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huberstatic int sFastTrackMultiplier = kFastTrackMultiplier; 176cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 177540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// See Thread::readOnlyHeap(). 17882f7321b03eec1e40af9d681370f754ee0279582Andreas Huber// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 179cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 180cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 181cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huberstatic const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 1828dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 1838dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber// ---------------------------------------------------------------------------- 1848dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 1858dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huberstatic pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 1869ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim 1878dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huberstatic void sFastTrackMultiplierInit() 1888dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber{ 1898dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber char value[PROPERTY_VALUE_MAX]; 1908dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 1919ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim char *endptr; 1928dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber unsigned long ul = strtoul(value, &endptr, 0); 1938dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 1948dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber sFastTrackMultiplier = (int) ul; 1958dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 1968dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 1978dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber} 1988dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 1998dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber// ---------------------------------------------------------------------------- 2008dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 2019ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim#ifdef ADD_BATTERY_DATA 2029ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim// To collect the amplifier usage 2038dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huberstatic void addBatteryData(uint32_t params) { 2048dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 2058dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (service == NULL) { 2068dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // it already logged 207540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim return; 2089558f6dd733dc450270f38b9a139d384d273ce0aWei Jia } 209540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 210540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim service->addBatteryData(params); 211540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim} 2129558f6dd733dc450270f38b9a139d384d273ce0aWei Jia#endif 213540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 214540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 215540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// ---------------------------------------------------------------------------- 216540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// CPU Stats 217540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim// ---------------------------------------------------------------------------- 2189558f6dd733dc450270f38b9a139d384d273ce0aWei Jia 2199558f6dd733dc450270f38b9a139d384d273ce0aWei Jiaclass CpuStats { 2209558f6dd733dc450270f38b9a139d384d273ce0aWei Jiapublic: 221cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber CpuStats(); 222cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber void sample(const String8 &title); 223386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber#ifdef DEBUG_CPU_USAGE 224d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhangprivate: 225d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 226f933441648ef6a71dee783d733aac17b9508b452Andreas Huber CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 227386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 228f933441648ef6a71dee783d733aac17b9508b452Andreas Huber CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 229bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber 230799c9682b3776a55d234396aee4a302437150c26Chong Zhang int mCpuNum; // thread's current CPU number 231d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang int mCpukHz; // frequency of thread's current CPU in kHz 2323856b090cd04ba5dd4a59a12430ed724d5995909Steve Block#endif 233cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber}; 234cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 2358dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas HuberCpuStats::CpuStats() 2368dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#ifdef DEBUG_CPU_USAGE 23706528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber : mCpuNum(-1), mCpukHz(-1) 23806528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber#endif 2398dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber{ 2408dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber} 2418dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 242cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Hubervoid CpuStats::sample(const String8 &title 2438dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#ifndef DEBUG_CPU_USAGE 24406528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber __unused 2458dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#endif 2468dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber ) { 2478dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber#ifdef DEBUG_CPU_USAGE 2488dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // get current thread's delta CPU time in wall clock ns 24987f2a558dd12043631e12c361abef301bf603140Andreas Huber double wcNs; 25087f2a558dd12043631e12c361abef301bf603140Andreas Huber bool valid = mCpuUsage.sampleAndEnable(wcNs); 251540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 2528dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // record sample for wall clock statistics 253cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (valid) { 254cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mWcStats.sample(wcNs); 255cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 256cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 257cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // get the current CPU number 258cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber int cpuNum = sched_getcpu(); 2595403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber 260540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim // get the current CPU frequency in kHz 261cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber int cpukHz = mCpuUsage.getCpukHz(cpuNum); 262cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 263cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // check if either CPU number or frequency changed 264cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 26532f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber mCpuNum = cpuNum; 26632f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber mCpukHz = cpukHz; 267b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas Huber // ignore sample for purposes of cycles 268b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas Huber valid = false; 269b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas Huber } 270b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas Huber 271b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas Huber // if no change in CPU number or frequency, then record sample for cycle statistics 272b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas Huber if (valid && mCpukHz > 0) { 273b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas Huber double cycles = wcNs * cpukHz * 0.000001; 274b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas Huber mHzStats.sample(cycles); 2752a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber } 27632f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber 277f933441648ef6a71dee783d733aac17b9508b452Andreas Huber unsigned n = mWcStats.n(); 278f933441648ef6a71dee783d733aac17b9508b452Andreas Huber // mCpuUsage.elapsed() is expensive, so don't call it every loop 279f933441648ef6a71dee783d733aac17b9508b452Andreas Huber if ((n & 127) == 1) { 280f933441648ef6a71dee783d733aac17b9508b452Andreas Huber long long elapsed = mCpuUsage.elapsed(); 281f933441648ef6a71dee783d733aac17b9508b452Andreas Huber if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 282f933441648ef6a71dee783d733aac17b9508b452Andreas Huber double perLoop = elapsed / (double) n; 2832a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber double perLoop100 = perLoop * 0.01; 2842a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber double perLoop1k = perLoop * 0.001; 2852a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber double mean = mWcStats.mean(); 2869bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang double stddev = mWcStats.stddev(); 2879bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang double minimum = mWcStats.minimum(); 2889bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang double maximum = mWcStats.maximum(); 2899bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang double meanCycles = mHzStats.mean(); 2909bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang double stddevCycles = mHzStats.stddev(); 2919bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang double minCycles = mHzStats.minimum(); 2929bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang double maxCycles = mHzStats.maximum(); 2939bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang mCpuUsage.resetElapsed(); 2949bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang mWcStats.reset(); 2959bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang mHzStats.reset(); 2969bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang ALOGD("CPU usage for %s over past %.1f secs\n" 2979bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang " (%u mixer loops at %.1f mean ms per loop):\n" 2989bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2999bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 3009bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 3019bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang title.string(), 3029bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang elapsed * .000000001, n, perLoop * .000001, 3039bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang mean * .001, 3049bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang stddev * .001, 3059bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang minimum * .001, 3069bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang maximum * .001, 3079bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang mean / perLoop100, 3089bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang stddev / perLoop100, 3099bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang minimum / perLoop100, 3109bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang maximum / perLoop100, 3119bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang meanCycles / perLoop1k, 3129bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang stddevCycles / perLoop1k, 3139bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang minCycles / perLoop1k, 3149bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang maxCycles / perLoop1k); 3159bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang 3169bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang } 3179bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang } 3189bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang#endif 3199bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang}; 3209bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang 3219bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang// ---------------------------------------------------------------------------- 3229bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang// ThreadBase 3239bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang// ---------------------------------------------------------------------------- 3249bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang 3259bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang// static 3269bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhangconst char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 3279bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang{ 3289bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang switch (type) { 3299bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang case MIXER: 3309bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang return "MIXER"; 3319bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang case DIRECT: 3329bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang return "DIRECT"; 333e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim case DUPLICATING: 334e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim return "DUPLICATING"; 335e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim case RECORD: 3369bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang return "RECORD"; 337e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim case OFFLOAD: 338e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim return "OFFLOAD"; 339e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim default: 3409bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang return "unknown"; 3419bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang } 3429bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang} 3439bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang 3449bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong ZhangString8 devicesToString(audio_devices_t devices) 3459bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang{ 3469bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang static const struct mapping { 3479bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang audio_devices_t mDevices; 3489bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang const char * mString; 3499bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang } mappingsOut[] = { 3509bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 3519bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 3529bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 3539bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 3549bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 355bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber AUDIO_DEVICE_NONE, "NONE", // must be last 35606528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber }, mappingsIn[] = { 357cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 3583856b090cd04ba5dd4a59a12430ed724d5995909Steve Block AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 3596456ae745e919085c5024f784aaa2703f9695f98David Yeh AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 3606456ae745e919085c5024f784aaa2703f9695f98David Yeh AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 3616456ae745e919085c5024f784aaa2703f9695f98David Yeh AUDIO_DEVICE_NONE, "NONE", // must be last 3626456ae745e919085c5024f784aaa2703f9695f98David Yeh }; 3636e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber String8 result; 3643856b090cd04ba5dd4a59a12430ed724d5995909Steve Block audio_devices_t allDevices = AUDIO_DEVICE_NONE; 3656456ae745e919085c5024f784aaa2703f9695f98David Yeh const mapping *entry; 3666456ae745e919085c5024f784aaa2703f9695f98David Yeh if (devices & AUDIO_DEVICE_BIT_IN) { 3676456ae745e919085c5024f784aaa2703f9695f98David Yeh devices &= ~AUDIO_DEVICE_BIT_IN; 3686456ae745e919085c5024f784aaa2703f9695f98David Yeh entry = mappingsIn; 369cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } else { 370e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim entry = mappingsOut; 3716e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber } 372cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 373cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber allDevices = (audio_devices_t) (allDevices | entry->mDevices); 3743856b090cd04ba5dd4a59a12430ed724d5995909Steve Block if (devices & entry->mDevices) { 375cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (!result.isEmpty()) { 3766e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber result.append("|"); 3776e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber } 3786e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber result.append(entry->mString); 3796e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber } 3806e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber } 3816e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber if (devices & ~allDevices) { 3826e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber if (!result.isEmpty()) { 38387f2a558dd12043631e12c361abef301bf603140Andreas Huber result.append("|"); 38487f2a558dd12043631e12c361abef301bf603140Andreas Huber } 38587f2a558dd12043631e12c361abef301bf603140Andreas Huber result.appendFormat("0x%X", devices & ~allDevices); 38687f2a558dd12043631e12c361abef301bf603140Andreas Huber } 3876e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber if (result.isEmpty()) { 388cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber result.append(entry->mString); 389cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 3903856b090cd04ba5dd4a59a12430ed724d5995909Steve Block return result; 391cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 392cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 393cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas HuberString8 inputFlagsToString(audio_input_flags_t flags) 394bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber{ 395bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber static const struct mapping { 396cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber audio_input_flags_t mFlag; 397cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber const char * mString; 398cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } mappings[] = { 399cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber AUDIO_INPUT_FLAG_FAST, "FAST", 400cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 401e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 402cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber }; 403cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber String8 result; 4043856b090cd04ba5dd4a59a12430ed724d5995909Steve Block audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 405cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber const mapping *entry; 4066e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 407cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 408cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (flags & entry->mFlag) { 4093856b090cd04ba5dd4a59a12430ed724d5995909Steve Block if (!result.isEmpty()) { 410cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber result.append("|"); 4116e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber } 412cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber result.append(entry->mString); 413cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 4143856b090cd04ba5dd4a59a12430ed724d5995909Steve Block } 415cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (flags & ~allFlags) { 416cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (!result.isEmpty()) { 417cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber result.append("|"); 418cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 419cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber result.appendFormat("0x%X", flags & ~allFlags); 420cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 4216e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber if (result.isEmpty()) { 422cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber result.append(entry->mString); 423cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 4243856b090cd04ba5dd4a59a12430ed724d5995909Steve Block return result; 425cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 426e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 427e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk KimString8 outputFlagsToString(audio_output_flags_t flags) 428e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim{ 429cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber static const struct mapping { 430cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber audio_output_flags_t mFlag; 431cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber const char * mString; 432cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } mappings[] = { 433cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 434cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 435bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber AUDIO_OUTPUT_FLAG_FAST, "FAST", 436bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 437bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 438bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 439cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 440cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 441cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber }; 442cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber String8 result; 443e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 444e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim const mapping *entry; 445e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 4466e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 447bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber if (flags & entry->mFlag) { 448bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber if (!result.isEmpty()) { 449bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber result.append("|"); 450bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber } 451bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber result.append(entry->mString); 452bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber } 453bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber } 454bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber if (flags & ~allFlags) { 455df64d15042bbd5e0e4933ac49bf3c177dd94752cSteve Block if (!result.isEmpty()) { 456bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber result.append("|"); 457bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber } 458bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber result.appendFormat("0x%X", flags & ~allFlags); 459bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber } 460bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber if (result.isEmpty()) { 461bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber result.append(entry->mString); 46206528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber } 463df64d15042bbd5e0e4933ac49bf3c177dd94752cSteve Block return result; 46406528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber} 46506528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber 46606528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huberconst char *sourceToString(audio_source_t source) 467df64d15042bbd5e0e4933ac49bf3c177dd94752cSteve Block{ 46806528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber switch (source) { 46906528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber case AUDIO_SOURCE_DEFAULT: return "default"; 470df64d15042bbd5e0e4933ac49bf3c177dd94752cSteve Block case AUDIO_SOURCE_MIC: return "mic"; 47106528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 47206528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 47306528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 474df64d15042bbd5e0e4933ac49bf3c177dd94752cSteve Block case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 47506528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 47606528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 47706528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 4789bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 4799bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang case AUDIO_SOURCE_HOTWORD: return "hotword"; 48006528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber default: return "unknown"; 48106528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber } 482df64d15042bbd5e0e4933ac49bf3c177dd94752cSteve Block} 48306528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber 48406528d7f18ad01377357d337eaa3e875a242bd2dAndreas HuberAudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 485bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 486bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber : Thread(false /*canCallJava*/), 487bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber mType(type), 488bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber mAudioFlinger(audioFlinger), 489bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 490bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber // are set by PlaybackThread::readOutputParameters_l() or 491bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber // RecordThread::readInputParameters_l() 492bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber //FIXME: mStandby should be true here. Is this some kind of hack? 49387f2a558dd12043631e12c361abef301bf603140Andreas Huber mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 49487f2a558dd12043631e12c361abef301bf603140Andreas Huber mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 49587f2a558dd12043631e12c361abef301bf603140Andreas Huber // mName will be set by concrete (non-virtual) subclass 496bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber mDeathRecipient(new PMDeathRecipient(this)) 497bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber{ 498bc7f5b2e56107cfeaeeab13cf8979379e3c2f139Andreas Huber} 49906528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber 50006528d7f18ad01377357d337eaa3e875a242bd2dAndreas HuberAudioFlinger::ThreadBase::~ThreadBase() 501cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber{ 502cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 503799c9682b3776a55d234396aee4a302437150c26Chong Zhang mConfigEvents.clear(); 504799c9682b3776a55d234396aee4a302437150c26Chong Zhang 505799c9682b3776a55d234396aee4a302437150c26Chong Zhang // do not lock the mutex in destructor 506799c9682b3776a55d234396aee4a302437150c26Chong Zhang releaseWakeLock_l(); 507799c9682b3776a55d234396aee4a302437150c26Chong Zhang if (mPowerManager != 0) { 5089bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang sp<IBinder> binder = IInterface::asBinder(mPowerManager); 5099bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang binder->unlinkToDeath(mDeathRecipient); 5109bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang } 5119bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang} 5129bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang 5139bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhangstatus_t AudioFlinger::ThreadBase::readyToRun() 5149bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang{ 5153728ba367f1e23e652c5539d9488aa0d0d4ec9d7Chad Brubaker status_t status = initCheck(); 5169bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang if (status == NO_ERROR) { 5179bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang ALOGI("AudioFlinger's thread %p ready to run", this); 5189bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang } else { 5199bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang ALOGE("No working audio driver found."); 5209bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang } 5215c9054bc6efc080b265e028f2ebb1abd2a2e3953Chih-Hung Hsieh return status; 5229bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang} 5239bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang 5249bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhangvoid AudioFlinger::ThreadBase::exit() 5259bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang{ 5269bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang ALOGV("ThreadBase::exit"); 527799c9682b3776a55d234396aee4a302437150c26Chong Zhang // do any cleanup required for exit to succeed 528799c9682b3776a55d234396aee4a302437150c26Chong Zhang preExit(); 529cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber { 530cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // This lock prevents the following race in thread (uniprocessor for illustration): 531cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // if (!exitPending()) { 532cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // // context switch from here to exit() 5339ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih // // exit() calls requestExit(), what exitPending() observes 534cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // // exit() calls signal(), which is dropped since no waiters 535cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // // context switch back from exit() to here 536cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // mWaitWorkCV.wait(...); 537cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // // now thread is hung 538cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // } 539cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber AutoMutex lock(mLock); 540bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih requestExit(); 541bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih mWaitWorkCV.broadcast(); 542bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih } 543bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih // When Thread::requestExitAndWait is made virtual and this method is renamed to 544bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 545bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih requestExitAndWait(); 546bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih} 5479ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih 5489ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shihstatus_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 549bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih{ 550bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih status_t status; 551bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih 552bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 553bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih Mutex::Autolock _l(mLock); 554bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih 555bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber return sendSetParameterConfigEvent_l(keyValuePairs); 556799c9682b3776a55d234396aee4a302437150c26Chong Zhang} 557799c9682b3776a55d234396aee4a302437150c26Chong Zhang 558c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber// sendConfigEvent_l() must be called with ThreadBase::mLock held 559c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 560c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huberstatus_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 561c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber{ 562c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber status_t status = NO_ERROR; 563c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber 564c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber mConfigEvents.add(event); 565c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 566c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber mWaitWorkCV.signal(); 567c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber mLock.unlock(); 568bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber { 569bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber Mutex::Autolock _l(event->mLock); 57087f2a558dd12043631e12c361abef301bf603140Andreas Huber while (event->mWaitStatus) { 57187f2a558dd12043631e12c361abef301bf603140Andreas Huber if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 57287f2a558dd12043631e12c361abef301bf603140Andreas Huber event->mStatus = TIMED_OUT; 57387f2a558dd12043631e12c361abef301bf603140Andreas Huber event->mWaitStatus = false; 57487f2a558dd12043631e12c361abef301bf603140Andreas Huber } 57587f2a558dd12043631e12c361abef301bf603140Andreas Huber } 576d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber status = event->mStatus; 577d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber } 578d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber mLock.lock(); 579d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber return status; 58087f2a558dd12043631e12c361abef301bf603140Andreas Huber} 581bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber 582bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Hubervoid AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 583cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber{ 584cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber Mutex::Autolock _l(mLock); 585bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber sendIoConfigEvent_l(event, param); 58687f2a558dd12043631e12c361abef301bf603140Andreas Huber} 58787f2a558dd12043631e12c361abef301bf603140Andreas Huber 58887f2a558dd12043631e12c361abef301bf603140Andreas Huber// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 58987f2a558dd12043631e12c361abef301bf603140Andreas Hubervoid AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 590bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber{ 591bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 592cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sendConfigEvent_l(configEvent); 59387f2a558dd12043631e12c361abef301bf603140Andreas Huber} 59487f2a558dd12043631e12c361abef301bf603140Andreas Huber 5956a63a939601645404fd98f58c19cc38ca818d99eAndreas Huber// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 596be9634d071e79b72a42a4504f64eda9e2a0bceb8Marco Nelissenvoid AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 59790a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber{ 598386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 599386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber sendConfigEvent_l(configEvent); 600386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber} 60187f2a558dd12043631e12c361abef301bf603140Andreas Huber 60287f2a558dd12043631e12c361abef301bf603140Andreas Huber// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 60387f2a558dd12043631e12c361abef301bf603140Andreas Huberstatus_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 60487f2a558dd12043631e12c361abef301bf603140Andreas Huber{ 605386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 6066e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber return sendConfigEvent_l(configEvent); 607386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber} 608386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 609386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huberstatus_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 610386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber const struct audio_patch *patch, 611386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber audio_patch_handle_t *handle) 612386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber{ 613386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber Mutex::Autolock _l(mLock); 614386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 615386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber status_t status = sendConfigEvent_l(configEvent); 616386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber if (status == NO_ERROR) { 617386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber CreateAudioPatchConfigEventData *data = 618386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 619386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber *handle = data->mHandle; 620386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber } 621386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber return status; 622386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber} 623386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 624386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huberstatus_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 625386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber const audio_patch_handle_t handle) 6269bf32f06e8971c1d3eb4fc5edd74b69557f97212Chong Zhang{ 627d3c079ae9859011d118f94616d0069c2987013edChangwan Ryu Mutex::Autolock _l(mLock); 628d3c079ae9859011d118f94616d0069c2987013edChangwan Ryu sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 629d3c079ae9859011d118f94616d0069c2987013edChangwan Ryu return sendConfigEvent_l(configEvent); 630d3c079ae9859011d118f94616d0069c2987013edChangwan Ryu} 631d3c079ae9859011d118f94616d0069c2987013edChangwan Ryu 6320852843d304006e3ab333081fddda13b07193de8Robert Shih 6330852843d304006e3ab333081fddda13b07193de8Robert Shih// post condition: mConfigEvents.isEmpty() 6340852843d304006e3ab333081fddda13b07193de8Robert Shihvoid AudioFlinger::ThreadBase::processConfigEvents_l() 6350852843d304006e3ab333081fddda13b07193de8Robert Shih{ 6360852843d304006e3ab333081fddda13b07193de8Robert Shih bool configChanged = false; 637386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 638386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber while (!mConfigEvents.isEmpty()) { 639386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 640386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber sp<ConfigEvent> event = mConfigEvents[0]; 6413856b090cd04ba5dd4a59a12430ed724d5995909Steve Block mConfigEvents.removeAt(0); 64218ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber switch (event->mType) { 64318ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber case CFG_EVENT_PRIO: { 64418ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 64518ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber // FIXME Need to understand why this has to be done asynchronously 64618ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber int err = requestPriority(data->mPid, data->mTid, data->mPrio, 647cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber true /*asynchronous*/); 648cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (err != 0) { 649cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 650386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber data->mPrio, data->mPid, data->mTid, err); 651386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber } 652cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } break; 653cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber case CFG_EVENT_IO: { 6545403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 65587f2a558dd12043631e12c361abef301bf603140Andreas Huber audioConfigChanged(data->mEvent, data->mParam); 656540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim } break; 657540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim case CFG_EVENT_SET_PARAMETER: { 65818ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 6595403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 66018ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber configChanged = true; 66118ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber } 66287f2a558dd12043631e12c361abef301bf603140Andreas Huber } break; 66387f2a558dd12043631e12c361abef301bf603140Andreas Huber case CFG_EVENT_CREATE_AUDIO_PATCH: { 66487f2a558dd12043631e12c361abef301bf603140Andreas Huber CreateAudioPatchConfigEventData *data = 66587f2a558dd12043631e12c361abef301bf603140Andreas Huber (CreateAudioPatchConfigEventData *)event->mData.get(); 66687f2a558dd12043631e12c361abef301bf603140Andreas Huber event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 6679558f6dd733dc450270f38b9a139d384d273ce0aWei Jia } break; 66887f2a558dd12043631e12c361abef301bf603140Andreas Huber case CFG_EVENT_RELEASE_AUDIO_PATCH: { 66987f2a558dd12043631e12c361abef301bf603140Andreas Huber ReleaseAudioPatchConfigEventData *data = 67087f2a558dd12043631e12c361abef301bf603140Andreas Huber (ReleaseAudioPatchConfigEventData *)event->mData.get(); 67194a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber event->mStatus = releaseAudioPatch_l(data->mHandle); 67294a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber } break; 67394a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber default: 67494a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 67594a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber break; 67694a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber } 67794a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber { 67894a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber Mutex::Autolock _l(event->mLock); 67994a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber if (event->mWaitStatus) { 68094a483bf2bd699275673d9cd57cb125d48572f30Andreas Huber event->mWaitStatus = false; 68166830855846db5c211c2da6c055ca9b4167e8974Chong Zhang event->mCond.signal(); 68266830855846db5c211c2da6c055ca9b4167e8974Chong Zhang } 68366830855846db5c211c2da6c055ca9b4167e8974Chong Zhang } 68487f2a558dd12043631e12c361abef301bf603140Andreas Huber ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 68587f2a558dd12043631e12c361abef301bf603140Andreas Huber } 68687f2a558dd12043631e12c361abef301bf603140Andreas Huber 68787f2a558dd12043631e12c361abef301bf603140Andreas Huber if (configChanged) { 688cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber cacheParameters_l(); 689540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim } 690cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 691cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 692cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas HuberString8 channelMaskToString(audio_channel_mask_t mask, bool output) { 693cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber String8 s; 694cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (output) { 695540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 6965403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 6975403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 69865959d34fdab8319dbf765be5fbf7ff8051eedf1Wonsik Kim if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 69965959d34fdab8319dbf765be5fbf7ff8051eedf1Wonsik Kim if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 7005403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 701cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 702cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 703cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 7049558f6dd733dc450270f38b9a139d384d273ce0aWei Jia if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 705cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 706cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 707cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 7085403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 709cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 710cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 711cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 712e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 713e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 714e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } else { 715e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 716cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 7173e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 7183e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 7193e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 7203e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 7213e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 722b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Cross if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 7233e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 7243e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 7253e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 7263e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 7273e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 7283e57322b332214e3cb1874e67a5704c9b2b5f6ecAndreas Huber if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 729cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 730cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 731cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber int len = s.length(); 7325403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber if (s.length() > 2) { 7335403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber char *str = s.lockBuffer(len); 734cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber s.unlockBuffer(len - 2); 735cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 7366e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber return s; 7376e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber} 7386e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber 7396e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Hubervoid AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 7406e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber{ 7416e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber const size_t SIZE = 256; 7426e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber char buffer[SIZE]; 7436e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber String8 result; 7446e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber 7456e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber bool locked = AudioFlinger::dumpTryLock(mLock); 7466e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber if (!locked) { 7476e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, "thread %p may be deadlocked\n", this); 7486e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber } 7496e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber 7506e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " Thread name: %s\n", mThreadName); 7516e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " I/O handle: %d\n", mId); 7526e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " TID: %d\n", getTid()); 7536e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 7549bf32f06e8971c1d3eb4fc5edd74b69557f97212Chong Zhang dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 755d3c079ae9859011d118f94616d0069c2987013edChangwan Ryu dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 7566e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 7576e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 7586e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " Channel count: %u\n", mChannelCount); 7596e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 7606e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber channelMaskToString(mChannelMask, mType != RECORD).string()); 7616e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 7626e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 7630852843d304006e3ab333081fddda13b07193de8Robert Shih dprintf(fd, " Pending config events:"); 7640852843d304006e3ab333081fddda13b07193de8Robert Shih size_t numConfig = mConfigEvents.size(); 7650852843d304006e3ab333081fddda13b07193de8Robert Shih if (numConfig) { 7660852843d304006e3ab333081fddda13b07193de8Robert Shih for (size_t i = 0; i < numConfig; i++) { 7670852843d304006e3ab333081fddda13b07193de8Robert Shih mConfigEvents[i]->dump(buffer, SIZE); 7680852843d304006e3ab333081fddda13b07193de8Robert Shih dprintf(fd, "\n %s", buffer); 7690852843d304006e3ab333081fddda13b07193de8Robert Shih } 77032f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber dprintf(fd, "\n"); 77132f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber } else { 7720389cc09f7b90f155a8942a0d2e1925cad1dbe2dMarco Nelissen dprintf(fd, " none\n"); 7730389cc09f7b90f155a8942a0d2e1925cad1dbe2dMarco Nelissen } 77418ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 77518ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 77618ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 77718ac5407da14dad9731f40ffc9a56bee73830019Andreas Huber 7782a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber if (locked) { 7799558f6dd733dc450270f38b9a139d384d273ce0aWei Jia mLock.unlock(); 780aabbdc7401ae24a4199f12a283985deb648673c0Robert Shih } 7812a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber} 7822a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber 7836e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Hubervoid AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 7846e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber{ 7856e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber const size_t SIZE = 256; 7866e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber char buffer[SIZE]; 7876e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber String8 result; 7886e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber 7896e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber size_t numEffectChains = mEffectChains.size(); 7906e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 7916e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber write(fd, buffer, strlen(buffer)); 7926e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber 793f933441648ef6a71dee783d733aac17b9508b452Andreas Huber for (size_t i = 0; i < numEffectChains; ++i) { 7946e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber sp<EffectChain> chain = mEffectChains[i]; 79532f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber if (chain != 0) { 7966e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber chain->dump(fd, args); 7976e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber } 7986e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber } 7996e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber} 8006e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber 8016e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Hubervoid AudioFlinger::ThreadBase::acquireWakeLock(int uid) 8026e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber{ 8036e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber Mutex::Autolock _l(mLock); 80432f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber acquireWakeLock_l(uid); 805c6cfd70f24a11b946859485ce398a189c301a4e2Wei Jia} 806e332a9181cf6a3155ed1a0fd2afc212ccb1f2753Andreas Huber 8076e3d311b6631b12aac2879d1b08c3534aece78b1Andreas HuberString16 AudioFlinger::ThreadBase::getWakeLockTag() 808f933441648ef6a71dee783d733aac17b9508b452Andreas Huber{ 8096e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber switch (mType) { 810632740c58119a132ce19f6d498e39c5c3773971aChong Zhang case MIXER: 811bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber return String16("AudioMix"); 812f933441648ef6a71dee783d733aac17b9508b452Andreas Huber case DIRECT: 813bff07d0b22a5ee2d9f044f6cb5e4be1532017ab0Andreas Huber return String16("AudioDirectOut"); 814f933441648ef6a71dee783d733aac17b9508b452Andreas Huber case DUPLICATING: 815f933441648ef6a71dee783d733aac17b9508b452Andreas Huber return String16("AudioDup"); 816f933441648ef6a71dee783d733aac17b9508b452Andreas Huber case RECORD: 8172a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber return String16("AudioIn"); 818be9634d071e79b72a42a4504f64eda9e2a0bceb8Marco Nelissen case OFFLOAD: 819540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim return String16("AudioOffload"); 8202a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber default: 8212a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber ALOG_ASSERT(false); 822540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim return String16("AudioUnknown"); 823cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 824cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 8253856b090cd04ba5dd4a59a12430ed724d5995909Steve Block 826cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Hubervoid AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 827386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber{ 8283856b090cd04ba5dd4a59a12430ed724d5995909Steve Block getPowerManager_l(); 829386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber if (mPowerManager != 0) { 8305403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber sp<IBinder> binder = new BBinder(); 8315403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber status_t status; 832386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber if (uid >= 0) { 833386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 834cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber binder, 8353856b090cd04ba5dd4a59a12430ed724d5995909Steve Block getWakeLockTag(), 836cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber String16("media"), 837cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber uid, 8383856b090cd04ba5dd4a59a12430ed724d5995909Steve Block true /* FIXME force oneway contrary to .aidl */); 839cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } else { 840cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 841cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber binder, 842cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber getWakeLockTag(), 843cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber String16("media"), 844cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber true /* FIXME force oneway contrary to .aidl */); 845cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 846cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (status == NO_ERROR) { 847cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mWakeLockToken = binder; 848e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 849e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 850e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 851cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 8526e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber 8536e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Hubervoid AudioFlinger::ThreadBase::releaseWakeLock() 8546e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber{ 8556e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber Mutex::Autolock _l(mLock); 8566e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber releaseWakeLock_l(); 857cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 858cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 8593856b090cd04ba5dd4a59a12430ed724d5995909Steve Blockvoid AudioFlinger::ThreadBase::releaseWakeLock_l() 860cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber{ 861cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mWakeLockToken != 0) { 8623856b090cd04ba5dd4a59a12430ed724d5995909Steve Block ALOGV("releaseWakeLock_l() %s", mThreadName); 863cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mPowerManager != 0) { 864cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mPowerManager->releaseWakeLock(mWakeLockToken, 0, 8653856b090cd04ba5dd4a59a12430ed724d5995909Steve Block true /* FIXME force oneway contrary to .aidl */); 866cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 867cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mWakeLockToken.clear(); 8683856b090cd04ba5dd4a59a12430ed724d5995909Steve Block } 869cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 870cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 8713856b090cd04ba5dd4a59a12430ed724d5995909Steve Blockvoid AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 872cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber Mutex::Autolock _l(mLock); 8736e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber updateWakeLockUids_l(uids); 8746e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber} 875cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 876cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Hubervoid AudioFlinger::ThreadBase::getPowerManager_l() { 8773856b090cd04ba5dd4a59a12430ed724d5995909Steve Block 878cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mPowerManager == 0) { 879cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // use checkService() to avoid blocking if power service is not up yet 880cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sp<IBinder> binder = 881cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber defaultServiceManager()->checkService(String16("power")); 882cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (binder == 0) { 883cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 884e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } else { 885e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim mPowerManager = interface_cast<IPowerManager>(binder); 886e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim binder->linkToDeath(mDeathRecipient); 887cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 8886456ae745e919085c5024f784aaa2703f9695f98David Yeh } 8896456ae745e919085c5024f784aaa2703f9695f98David Yeh} 8906456ae745e919085c5024f784aaa2703f9695f98David Yeh 891cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Hubervoid AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 892e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 893e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim getPowerManager_l(); 894e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim if (mWakeLockToken == NULL) { 895cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber ALOGE("no wake lock to update!"); 896e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim return; 897e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 898e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim if (mPowerManager != 0) { 899cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sp<IBinder> binder = new BBinder(); 900e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim status_t status; 901e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 902e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim true /* FIXME force oneway contrary to .aidl */); 903cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 904b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Cross } 905cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 906cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 907cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Hubervoid AudioFlinger::ThreadBase::clearPowerManager() 908cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber{ 909e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim Mutex::Autolock _l(mLock); 910e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim releaseWakeLock_l(); 911e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim mPowerManager.clear(); 912cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 913e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 914e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kimvoid AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 915e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim{ 916cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sp<ThreadBase> thread = mThread.promote(); 917cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (thread != 0) { 918e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim thread->clearPowerManager(); 919e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 920e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim ALOGW("power manager service died !!!"); 921cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 922e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 923e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kimvoid AudioFlinger::ThreadBase::setEffectSuspended( 924e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim const effect_uuid_t *type, bool suspend, int sessionId) 925cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber{ 926e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim Mutex::Autolock _l(mLock); 927e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim setEffectSuspended_l(type, suspend, sessionId); 928e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim} 929cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 930b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Crossvoid AudioFlinger::ThreadBase::setEffectSuspended_l( 931cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber const effect_uuid_t *type, bool suspend, int sessionId) 932cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber{ 933cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sp<EffectChain> chain = getEffectChain_l(sessionId); 934cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (chain != 0) { 935cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (type != NULL) { 936cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber chain->setEffectSuspended_l(type, suspend); 937e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } else { 938e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim chain->setEffectSuspendedAll_l(suspend); 939e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 940cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 941cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 942cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber updateSuspendedSessions_l(type, suspend, sessionId); 943cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 944e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 945e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kimvoid AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 946e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim{ 947cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 948e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim if (index < 0) { 949e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim return; 950e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 951cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 952e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 953e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim mSuspendedSessions.valueAt(index); 954e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 955cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber for (size_t i = 0; i < sessionEffects.size(); i++) { 956b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Cross sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 9576e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber for (int j = 0; j < desc->mRefCount; j++) { 958cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 959e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim chain->setEffectSuspendedAll_l(true); 960e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } else { 961e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 962cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber desc->mType.timeLow); 963cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber chain->setEffectSuspended_l(&desc->mType, true); 964cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 965cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 966cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 967e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim} 968e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 969e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kimvoid AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 970cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber bool suspend, 971e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim int sessionId) 972e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim{ 973e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 9746e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber 975e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 976e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 977e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim if (suspend) { 978cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (index >= 0) { 979cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sessionEffects = mSuspendedSessions.valueAt(index); 980cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } else { 981cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mSuspendedSessions.add(sessionId, sessionEffects); 982cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 983cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } else { 984cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (index < 0) { 985cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber return; 986cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 987e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim sessionEffects = mSuspendedSessions.valueAt(index); 988e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 989e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 990cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 991cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber int key = EffectChain::kKeyForSuspendAll; 992cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (type != NULL) { 993cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber key = type->timeLow; 9945403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber } 99529357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47Steve Block index = sessionEffects.indexOfKey(key); 996b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Cross 9975403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber sp<SuspendedSessionDesc> desc; 9985403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber if (suspend) { 9995403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber if (index >= 0) { 10005403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber desc = sessionEffects.valueAt(index); 10015403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber } else { 10020da4dab0a45a2bc1d95cbc6ef6a4850ed2569584Andreas Huber desc = new SuspendedSessionDesc(); 1003540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim if (type != NULL) { 1004cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber desc->mType = *type; 1005cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 1006cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sessionEffects.add(key, desc); 10070da4dab0a45a2bc1d95cbc6ef6a4850ed2569584Andreas Huber ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 10080da4dab0a45a2bc1d95cbc6ef6a4850ed2569584Andreas Huber } 1009540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim desc->mRefCount++; 10100da4dab0a45a2bc1d95cbc6ef6a4850ed2569584Andreas Huber } else { 1011cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (index < 0) { 1012e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim return; 1013e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 1014e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim desc = sessionEffects.valueAt(index); 1015cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (--desc->mRefCount == 0) { 1016b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Cross ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1017cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sessionEffects.removeItemsAt(index); 1018cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (sessionEffects.isEmpty()) { 1019e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim ALOGV("updateSuspendedSessions_l() restore removing session %d", 1020e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim sessionId); 1021e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim mSuspendedSessions.removeItem(sessionId); 1022cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 1023cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 1024e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 1025e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim if (!sessionEffects.isEmpty()) { 1026e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1027cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 1028cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 10295403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber 10305403129e2a2f44620f2ac8109889e5a61be08732Andreas Hubervoid AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1031cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber bool enabled, 1032cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber int sessionId) 1033540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim{ 1034f2cecd570c35d3b6422396dd51c0b4202732bceaJaesung Chung Mutex::Autolock _l(mLock); 10355403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1036cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 1037cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 1038b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Crossvoid AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1039cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber bool enabled, 1040cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber int sessionId) 10415403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber{ 1042540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim if (mType != RECORD) { 1043cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1044cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // another session. This gives the priority to well behaved effect control panels 10455403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber // and applications not using global effects. 10465403129e2a2f44620f2ac8109889e5a61be08732Andreas Huber // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1047cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // global effects 1048cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1049cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 105084333e0475bc911adc16417f4ca327c975cf6c36Andreas Huber } 1051540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim } 105290a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber 105390a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber sp<EffectChain> chain = getEffectChain_l(sessionId); 105490a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber if (chain != 0) { 105590a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber chain->checkSuspendOnEffectEnabled(effect, enabled); 105690a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber } 105790a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber} 105890a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber 105990a92053219ae50ddf4bb54e3d54db2d309e2b8dAndreas Huber// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 10603856b090cd04ba5dd4a59a12430ed724d5995909Steve Blocksp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1061cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber const sp<AudioFlinger::Client>& client, 106298a46cf165d8de3779874eba01803bbc485f45b9Andreas Huber const sp<IEffectClient>& effectClient, 106398a46cf165d8de3779874eba01803bbc485f45b9Andreas Huber int32_t priority, 106498a46cf165d8de3779874eba01803bbc485f45b9Andreas Huber int sessionId, 106598a46cf165d8de3779874eba01803bbc485f45b9Andreas Huber effect_descriptor_t *desc, 106682f7321b03eec1e40af9d681370f754ee0279582Andreas Huber int *enabled, 1067386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber status_t *status) 1068decd96988e495133e4a1728f612d4c9fdb4d218eAndreas Huber{ 1069be9634d071e79b72a42a4504f64eda9e2a0bceb8Marco Nelissen sp<EffectModule> effect; 1070be9634d071e79b72a42a4504f64eda9e2a0bceb8Marco Nelissen sp<EffectHandle> handle; 1071be9634d071e79b72a42a4504f64eda9e2a0bceb8Marco Nelissen status_t lStatus; 1072be9634d071e79b72a42a4504f64eda9e2a0bceb8Marco Nelissen sp<EffectChain> chain; 1073decd96988e495133e4a1728f612d4c9fdb4d218eAndreas Huber bool chainCreated = false; 1074decd96988e495133e4a1728f612d4c9fdb4d218eAndreas Huber bool effectCreated = false; 1075decd96988e495133e4a1728f612d4c9fdb4d218eAndreas Huber bool effectRegistered = false; 107682f7321b03eec1e40af9d681370f754ee0279582Andreas Huber 10776a63a939601645404fd98f58c19cc38ca818d99eAndreas Huber lStatus = initCheck(); 1078540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim if (lStatus != NO_ERROR) { 1079386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber ALOGW("createEffect_l() Audio driver not initialized."); 10806a63a939601645404fd98f58c19cc38ca818d99eAndreas Huber goto Exit; 1081386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber } 108282f7321b03eec1e40af9d681370f754ee0279582Andreas Huber 10836a63a939601645404fd98f58c19cc38ca818d99eAndreas Huber // Reject any effect on Direct output threads for now, since the format of 10843856b090cd04ba5dd4a59a12430ed724d5995909Steve Block // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1085386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber if (mType == DIRECT) { 1086386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1087309aa8bf5e4cd66fe988adf2654cac3fadc2a1c3Robert Shih desc->name, mThreadName); 1088309aa8bf5e4cd66fe988adf2654cac3fadc2a1c3Robert Shih lStatus = BAD_VALUE; 1089309aa8bf5e4cd66fe988adf2654cac3fadc2a1c3Robert Shih goto Exit; 1090309aa8bf5e4cd66fe988adf2654cac3fadc2a1c3Robert Shih } 1091309aa8bf5e4cd66fe988adf2654cac3fadc2a1c3Robert Shih 1092309aa8bf5e4cd66fe988adf2654cac3fadc2a1c3Robert Shih // Reject any effect on mixer or duplicating multichannel sinks. 10936a63a939601645404fd98f58c19cc38ca818d99eAndreas Huber // TODO: fix both format and multichannel issues with effects. 10946a63a939601645404fd98f58c19cc38ca818d99eAndreas Huber if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 109582f7321b03eec1e40af9d681370f754ee0279582Andreas Huber ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1096386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 10972a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber lStatus = BAD_VALUE; 10982a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber goto Exit; 10992a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber } 11005bc087c573c70c84c6a39946457590b42d392a33Andreas Huber 11015bc087c573c70c84c6a39946457590b42d392a33Andreas Huber // Allow global effects only on offloaded and mixer threads 1102386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 11035bc087c573c70c84c6a39946457590b42d392a33Andreas Huber switch (mType) { 11046a63a939601645404fd98f58c19cc38ca818d99eAndreas Huber case MIXER: 110582f7321b03eec1e40af9d681370f754ee0279582Andreas Huber case OFFLOAD: 1106540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim break; 1107540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim case DIRECT: 1108540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim case DUPLICATING: 1109540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim case RECORD: 1110540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim default: 1111540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1112540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim desc->name, mThreadName); 11139558f6dd733dc450270f38b9a139d384d273ce0aWei Jia lStatus = BAD_VALUE; 11149558f6dd733dc450270f38b9a139d384d273ce0aWei Jia goto Exit; 11159558f6dd733dc450270f38b9a139d384d273ce0aWei Jia } 1116540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim } 1117540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim 1118540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim // Only Pre processor effects are allowed on input threads and only on input threads 111982f7321b03eec1e40af9d681370f754ee0279582Andreas Huber if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 112082f7321b03eec1e40af9d681370f754ee0279582Andreas Huber ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 112182f7321b03eec1e40af9d681370f754ee0279582Andreas Huber desc->name, desc->flags, mType); 1122cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber lStatus = BAD_VALUE; 1123386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber goto Exit; 1124386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber } 1125386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 11266e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1127386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 1128386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber { // scope for mLock 1129386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber Mutex::Autolock _l(mLock); 1130386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 1131386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber // check for existing effect chain with the requested audio session 1132386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber chain = getEffectChain_l(sessionId); 1133386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber if (chain == 0) { 11346e3d311b6631b12aac2879d1b08c3534aece78b1Andreas Huber // create a new chain for this session 1135386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1136386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber chain = new EffectChain(this, sessionId); 1137386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber addEffectChain_l(chain); 1138386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber chain->setStrategy(getStrategyForSession_l(sessionId)); 1139386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber chainCreated = true; 11400852843d304006e3ab333081fddda13b07193de8Robert Shih } else { 11410852843d304006e3ab333081fddda13b07193de8Robert Shih effect = chain->getEffectFromDesc_l(desc); 11420852843d304006e3ab333081fddda13b07193de8Robert Shih } 11430852843d304006e3ab333081fddda13b07193de8Robert Shih 11440852843d304006e3ab333081fddda13b07193de8Robert Shih ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 11450852843d304006e3ab333081fddda13b07193de8Robert Shih 11460852843d304006e3ab333081fddda13b07193de8Robert Shih if (effect == 0) { 11470852843d304006e3ab333081fddda13b07193de8Robert Shih int id = mAudioFlinger->nextUniqueId(); 1148386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber // Check CPU and memory usage 1149386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1150cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (lStatus != NO_ERROR) { 1151cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber goto Exit; 1152cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 1153cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber effectRegistered = true; 1154cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // create a new effect module if none present in the chain 1155cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber effect = new EffectModule(this, chain, desc, id, sessionId); 1156cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber lStatus = effect->status(); 1157c4c17d47b674b425fb6c399822c0ab3258543c0aAndreas Huber if (lStatus != NO_ERROR) { 115887f2a558dd12043631e12c361abef301bf603140Andreas Huber goto Exit; 115987f2a558dd12043631e12c361abef301bf603140Andreas Huber } 1160d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber effect->setOffloaded(mType == OFFLOAD, mId); 1161d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber 1162d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang lStatus = chain->addEffect_l(effect); 116387f2a558dd12043631e12c361abef301bf603140Andreas Huber if (lStatus != NO_ERROR) { 116487f2a558dd12043631e12c361abef301bf603140Andreas Huber goto Exit; 11658dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 1166cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber effectCreated = true; 1167cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 1168cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber effect->setDevice(mOutDevice); 1169cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber effect->setDevice(mInDevice); 1170cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber effect->setMode(mAudioFlinger->getMode()); 1171540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim effect->setAudioSource(mAudioSource); 1172540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim } 1173e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim // create effect handle and connect it to effect module 1174e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim handle = new EffectHandle(effect, client, effectClient, priority); 1175e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim lStatus = handle->initCheck(); 1176e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim if (lStatus == OK) { 1177cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber lStatus = effect->addHandle(handle.get()); 1178cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 1179540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim if (enabled != NULL) { 1180cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber *enabled = (int)effect->isEnabled(); 1181cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 118232f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber } 118332f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber 1184b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas HuberExit: 1185d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1186d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang Mutex::Autolock _l(mLock); 1187d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang if (effectCreated) { 1188d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang chain->removeEffect_l(effect); 1189d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang } 1190d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang if (effectRegistered) { 1191d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang AudioSystem::unregisterEffect(effect->id()); 1192d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang } 1193d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang if (chainCreated) { 1194d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang removeEffectChain_l(chain); 1195d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang } 1196d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang handle.clear(); 1197d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang } 1198d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang 1199d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang *status = lStatus; 1200b7c8e91880463ff4981e3e53e98e45d68e2fe374Andreas Huber return handle; 120187f2a558dd12043631e12c361abef301bf603140Andreas Huber} 1202e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 1203e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kimsp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1204e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim{ 1205e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim Mutex::Autolock _l(mLock); 120687f2a558dd12043631e12c361abef301bf603140Andreas Huber return getEffect_l(sessionId, effectId); 1207e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim} 1208e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 1209e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kimsp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1210e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim{ 121187f2a558dd12043631e12c361abef301bf603140Andreas Huber sp<EffectChain> chain = getEffectChain_l(sessionId); 121287f2a558dd12043631e12c361abef301bf603140Andreas Huber return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1213d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber} 1214d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber 1215e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1216e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim// PlaybackThread::mLock held 1217e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kimstatus_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1218e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim{ 1219d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber // check for existing effect chain with the requested audio session 1220d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber int sessionId = effect->sessionId(); 1221d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber sp<EffectChain> chain = getEffectChain_l(sessionId); 1222d5e56231a598b180a1d898bb7dc61b75580e59a4Andreas Huber bool chainCreated = false; 122387f2a558dd12043631e12c361abef301bf603140Andreas Huber 122487f2a558dd12043631e12c361abef301bf603140Andreas Huber ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1225f933441648ef6a71dee783d733aac17b9508b452Andreas Huber "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 122632f3cefa373cd55e63deda36ca9d07c7fe22eaafAndreas Huber this, effect->desc().name, effect->desc().flags); 1227f933441648ef6a71dee783d733aac17b9508b452Andreas Huber 1228f933441648ef6a71dee783d733aac17b9508b452Andreas Huber if (chain == 0) { 1229f933441648ef6a71dee783d733aac17b9508b452Andreas Huber // create a new chain for this session 1230f933441648ef6a71dee783d733aac17b9508b452Andreas Huber ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1231e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim chain = new EffectChain(this, sessionId); 1232e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim addEffectChain_l(chain); 1233e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim chain->setStrategy(getStrategyForSession_l(sessionId)); 1234e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim chainCreated = true; 1235f933441648ef6a71dee783d733aac17b9508b452Andreas Huber } 12362a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1237f933441648ef6a71dee783d733aac17b9508b452Andreas Huber 12382a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber if (chain->getEffectFromId_l(effect->id()) != 0) { 12392a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber ALOGW("addEffect_l() %p effect %s already present in chain %p", 12402a4d22d79e927f2245537921e10fc5fda1c47a29Andreas Huber this, effect->desc().name, chain.get()); 1241cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber return BAD_VALUE; 1242cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 12433856b090cd04ba5dd4a59a12430ed724d5995909Steve Block 12446456ae745e919085c5024f784aaa2703f9695f98David Yeh effect->setOffloaded(mType == OFFLOAD, mId); 12456456ae745e919085c5024f784aaa2703f9695f98David Yeh 12466456ae745e919085c5024f784aaa2703f9695f98David Yeh status_t status = chain->addEffect_l(effect); 12476456ae745e919085c5024f784aaa2703f9695f98David Yeh if (status != NO_ERROR) { 1248cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (chainCreated) { 12493856b090cd04ba5dd4a59a12430ed724d5995909Steve Block removeEffectChain_l(chain); 1250cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 1251e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim return status; 12526e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber } 1253cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 1254cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber effect->setDevice(mOutDevice); 12553856b090cd04ba5dd4a59a12430ed724d5995909Steve Block effect->setDevice(mInDevice); 1256cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber effect->setMode(mAudioFlinger->getMode()); 12576e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber effect->setAudioSource(mAudioSource); 12586e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber return NO_ERROR; 12596e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber} 12606e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber 12616e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Hubervoid AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 12626e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber 1263cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1264cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber effect_descriptor_t desc = effect->desc(); 1265cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1266cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber detachAuxEffect_l(effect->id()); 1267cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 12683856b090cd04ba5dd4a59a12430ed724d5995909Steve Block 1269cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sp<EffectChain> chain = effect->chain().promote(); 12706e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber if (chain != 0) { 1271cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // remove effect chain if removing last effect 1272cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (chain->removeEffect_l(effect) == 0) { 12736e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber removeEffectChain_l(chain); 1274cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 1275cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } else { 1276cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 12773856b090cd04ba5dd4a59a12430ed724d5995909Steve Block } 1278cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 1279386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 1280386d609dc513e838c7e7c4c46c604493ccd560beAndreas Hubervoid AudioFlinger::ThreadBase::lockEffectChains_l( 1281386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1282386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber{ 1283386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber effectChains = mEffectChains; 1284386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber for (size_t i = 0; i < mEffectChains.size(); i++) { 1285386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber mEffectChains[i]->lock(); 1286386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber } 1287386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber} 1288386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 1289386d609dc513e838c7e7c4c46c604493ccd560beAndreas Hubervoid AudioFlinger::ThreadBase::unlockEffectChains( 1290386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1291386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber{ 1292d47dfcb5a2e5901c96fc92662cec7aa30f7f8843Chong Zhang for (size_t i = 0; i < effectChains.size(); i++) { 1293386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber effectChains[i]->unlock(); 12948dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 12958dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber} 12968dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 12978dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Hubersp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1298cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber{ 1299cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber Mutex::Autolock _l(mLock); 1300cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber return getEffectChain_l(sessionId); 13016e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber} 1302cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 1303cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Hubersp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 130406528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber{ 1305cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber size_t size = mEffectChains.size(); 130687f2a558dd12043631e12c361abef301bf603140Andreas Huber for (size_t i = 0; i < size; i++) { 1307540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim if (mEffectChains[i]->sessionId() == sessionId) { 1308540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim return mEffectChains[i]; 13098dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 13108dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 13118dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber return 0; 13124b4bb11b8747adeb2efe56c7df4ab6803dd7db41Andreas Huber} 13138dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 1314cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Hubervoid AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 13156456ae745e919085c5024f784aaa2703f9695f98David Yeh{ 13169bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang Mutex::Autolock _l(mLock); 13179bcf3ae6c9a413afc7accb5b48db3e5c3c829785Chong Zhang size_t size = mEffectChains.size(); 13186456ae745e919085c5024f784aaa2703f9695f98David Yeh for (size_t i = 0; i < size; i++) { 13198dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mEffectChains[i]->setMode_l(mode); 1320cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 13219ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim} 1322cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 1323cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Hubervoid AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 13248dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber{ 1325e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim config->type = AUDIO_PORT_TYPE_MIX; 1326e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim config->ext.mix.handle = mId; 1327e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim config->sample_rate = mSampleRate; 13288dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber config->format = mFormat; 13298dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber config->channel_mask = mChannelMask; 13308dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 13318dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber AUDIO_PORT_CONFIG_FORMAT; 13328dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber} 13338dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 13348dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 13358dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber// ---------------------------------------------------------------------------- 13368dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber// Playback 13378dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber// ---------------------------------------------------------------------------- 13389ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim 13399ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk KimAudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 13409ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim AudioStreamOut* output, 13418dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber audio_io_handle_t id, 13428dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber audio_devices_t device, 13438dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber type_t type) 13448dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 13458dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mNormalFrameCount(0), mSinkBuffer(NULL), 13468dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 13478dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mMixerBuffer(NULL), 13488dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mMixerBufferSize(0), 13498dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mMixerBufferFormat(AUDIO_FORMAT_INVALID), 13508dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mMixerBufferValid(false), 13518dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 13528dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mEffectBuffer(NULL), 13538dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mEffectBufferSize(0), 13548dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mEffectBufferFormat(AUDIO_FORMAT_INVALID), 13558dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mEffectBufferValid(false), 13568dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mSuspended(0), mBytesWritten(0), 13578dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mActiveTracksGeneration(0), 13588dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // mStreamTypes[] initialized in constructor body 13598dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mOutput(output), 13608dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 13618dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mMixerStatus(MIXER_IDLE), 13628dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mMixerStatusIgnoringFastTracks(MIXER_IDLE), 13638dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 13644b4bb11b8747adeb2efe56c7df4ab6803dd7db41Andreas Huber mBytesRemaining(0), 13658dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mCurrentWriteLength(0), 13668dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mUseAsyncWrite(false), 13678dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mWriteAckSequence(0), 13684b4bb11b8747adeb2efe56c7df4ab6803dd7db41Andreas Huber mDrainSequence(0), 13694b4bb11b8747adeb2efe56c7df4ab6803dd7db41Andreas Huber mSignalPending(false), 13704b4bb11b8747adeb2efe56c7df4ab6803dd7db41Andreas Huber mScreenState(AudioFlinger::mScreenState), 13718dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // index 0 is reserved for normal mixer's submix 137206528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1373cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1374cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // mLatchD, mLatchQ, 1375cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mLatchDValid(false), mLatchQValid(false) 1376cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber{ 137706528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1378cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 137987f2a558dd12043631e12c361abef301bf603140Andreas Huber 1380540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim // Assumes constructor is called by AudioFlinger with it's mLock held, but 138106528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber // it would be safer to explicitly pass initial masterVolume/masterMute as 138206528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber // parameter. 138306528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber // 138406528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber // If the HAL we are using has support for master volume or master mute, 1385cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // then do not attenuate or mute during mixing (just leave the volume at 1.0 1386cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // and the mute set to false). 1387cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mMasterVolume = audioFlinger->masterVolume_l(); 1388cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mMasterMute = audioFlinger->masterMute_l(); 1389cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mOutput && mOutput->audioHwDev) { 1390cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (mOutput->audioHwDev->canSetMasterVolume()) { 13913856b090cd04ba5dd4a59a12430ed724d5995909Steve Block mMasterVolume = 1.0; 1392cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 139306528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber 139406528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber if (mOutput->audioHwDev->canSetMasterMute()) { 1395cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber mMasterMute = false; 1396cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 1397e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim } 1398cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 139987f2a558dd12043631e12c361abef301bf603140Andreas Huber readOutputParameters_l(); 1400cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber 14018a1fa1ebc2375c9dcaca2b78918c6740fff2ca74Jaesung Chung // ++ operator does not compile 14028a1fa1ebc2375c9dcaca2b78918c6740fff2ca74Jaesung Chung for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 14038a1fa1ebc2375c9dcaca2b78918c6740fff2ca74Jaesung Chung stream = (audio_stream_type_t) (stream + 1)) { 14048a1fa1ebc2375c9dcaca2b78918c6740fff2ca74Jaesung Chung mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 14058a1fa1ebc2375c9dcaca2b78918c6740fff2ca74Jaesung Chung mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 140687f2a558dd12043631e12c361abef301bf603140Andreas Huber } 140787f2a558dd12043631e12c361abef301bf603140Andreas Huber} 140887f2a558dd12043631e12c361abef301bf603140Andreas Huber 140987f2a558dd12043631e12c361abef301bf603140Andreas HuberAudioFlinger::PlaybackThread::~PlaybackThread() 141087f2a558dd12043631e12c361abef301bf603140Andreas Huber{ 141187f2a558dd12043631e12c361abef301bf603140Andreas Huber mAudioFlinger->unregisterWriter(mNBLogWriter); 141287f2a558dd12043631e12c361abef301bf603140Andreas Huber free(mSinkBuffer); 141387f2a558dd12043631e12c361abef301bf603140Andreas Huber free(mMixerBuffer); 141487f2a558dd12043631e12c361abef301bf603140Andreas Huber free(mEffectBuffer); 141587f2a558dd12043631e12c361abef301bf603140Andreas Huber} 141687f2a558dd12043631e12c361abef301bf603140Andreas Huber 141787f2a558dd12043631e12c361abef301bf603140Andreas Hubervoid AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1418e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim{ 1419e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim dumpInternals(fd, args); 1420e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim dumpTracks(fd, args); 142187f2a558dd12043631e12c361abef301bf603140Andreas Huber dumpEffectChains(fd, args); 142287f2a558dd12043631e12c361abef301bf603140Andreas Huber} 142387f2a558dd12043631e12c361abef301bf603140Andreas Huber 142487f2a558dd12043631e12c361abef301bf603140Andreas Hubervoid AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 142587f2a558dd12043631e12c361abef301bf603140Andreas Huber{ 142687f2a558dd12043631e12c361abef301bf603140Andreas Huber const size_t SIZE = 256; 142787f2a558dd12043631e12c361abef301bf603140Andreas Huber char buffer[SIZE]; 142887f2a558dd12043631e12c361abef301bf603140Andreas Huber String8 result; 142987f2a558dd12043631e12c361abef301bf603140Andreas Huber 143087f2a558dd12043631e12c361abef301bf603140Andreas Huber result.appendFormat(" Stream volumes in dB: "); 143187f2a558dd12043631e12c361abef301bf603140Andreas Huber for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 143287f2a558dd12043631e12c361abef301bf603140Andreas Huber const stream_type_t *st = &mStreamTypes[i]; 143387f2a558dd12043631e12c361abef301bf603140Andreas Huber if (i > 0) { 143487f2a558dd12043631e12c361abef301bf603140Andreas Huber result.appendFormat(", "); 143587f2a558dd12043631e12c361abef301bf603140Andreas Huber } 1436b4a7a2df4c28c3f32b5d877b54831d2cc5d78f81Colin Cross result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 143787f2a558dd12043631e12c361abef301bf603140Andreas Huber if (st->mute) { 143887f2a558dd12043631e12c361abef301bf603140Andreas Huber result.append("M"); 143987f2a558dd12043631e12c361abef301bf603140Andreas Huber } 144087f2a558dd12043631e12c361abef301bf603140Andreas Huber } 144119cec89f8b05fd05f8034ee1a4cd39ee09c33f02Marco Nelissen result.append("\n"); 144219cec89f8b05fd05f8034ee1a4cd39ee09c33f02Marco Nelissen write(fd, result.string(), result.length()); 144387f2a558dd12043631e12c361abef301bf603140Andreas Huber result.clear(); 144487f2a558dd12043631e12c361abef301bf603140Andreas Huber 144587f2a558dd12043631e12c361abef301bf603140Andreas Huber // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 144687f2a558dd12043631e12c361abef301bf603140Andreas Huber FastTrackUnderruns underruns = getFastTrackUnderruns(0); 144787f2a558dd12043631e12c361abef301bf603140Andreas Huber dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 144887f2a558dd12043631e12c361abef301bf603140Andreas Huber underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 144987f2a558dd12043631e12c361abef301bf603140Andreas Huber 145087f2a558dd12043631e12c361abef301bf603140Andreas Huber size_t numtracks = mTracks.size(); 145187f2a558dd12043631e12c361abef301bf603140Andreas Huber size_t numactive = mActiveTracks.size(); 1452cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber dprintf(fd, " %d Tracks", numtracks); 1453e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim size_t numactiveseen = 0; 1454cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (numtracks) { 1455cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber dprintf(fd, " of which %d are active\n", numactive); 1456540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim Track::appendDumpHeader(result); 14573856b090cd04ba5dd4a59a12430ed724d5995909Steve Block for (size_t i = 0; i < numtracks; ++i) { 1458cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber sp<Track> track = mTracks[i]; 1459cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (track != 0) { 14606456ae745e919085c5024f784aaa2703f9695f98David Yeh bool active = mActiveTracks.indexOf(track) >= 0; 14616456ae745e919085c5024f784aaa2703f9695f98David Yeh if (active) { 14626456ae745e919085c5024f784aaa2703f9695f98David Yeh numactiveseen++; 14636456ae745e919085c5024f784aaa2703f9695f98David Yeh } 1464cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber track->dump(buffer, SIZE, active); 146552668ca838e1869676ff95e7388f93ec1858b1e9Andreas Huber result.append(buffer); 146652668ca838e1869676ff95e7388f93ec1858b1e9Andreas Huber } 146752668ca838e1869676ff95e7388f93ec1858b1e9Andreas Huber } 146852668ca838e1869676ff95e7388f93ec1858b1e9Andreas Huber } else { 1469cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber result.append("\n"); 1470cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 14713856b090cd04ba5dd4a59a12430ed724d5995909Steve Block if (numactiveseen != numactive) { 1472cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber // some tracks in the active list were not in the tracks list 14736e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber snprintf(buffer, SIZE, " The following tracks are in the active list but" 1474cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber " not in the track list\n"); 1475cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber result.append(buffer); 14763856b090cd04ba5dd4a59a12430ed724d5995909Steve Block Track::appendDumpHeader(result); 1477cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber for (size_t i = 0; i < numactive; ++i) { 14786e4c5c499999c04c2477b987f9e64f3ff2bf1a06Andreas Huber sp<Track> track = mActiveTracks[i].promote(); 1479cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber if (track != 0 && mTracks.indexOf(track) < 0) { 1480cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber track->dump(buffer, SIZE, true); 14813856b090cd04ba5dd4a59a12430ed724d5995909Steve Block result.append(buffer); 1482cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber } 14830da4dab0a45a2bc1d95cbc6ef6a4850ed2569584Andreas Huber } 148487f2a558dd12043631e12c361abef301bf603140Andreas Huber } 14850da4dab0a45a2bc1d95cbc6ef6a4850ed2569584Andreas Huber 1486df64d15042bbd5e0e4933ac49bf3c177dd94752cSteve Block write(fd, result.string(), result.size()); 1487cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 148887f2a558dd12043631e12c361abef301bf603140Andreas Huber 148987f2a558dd12043631e12c361abef301bf603140Andreas Hubervoid AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1490e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim{ 1491e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1492e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 1493e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim dumpBase(fd, args); 1494e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim 1495540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1496540006666b4191cd78391378f1c66c21bcf0c4cdWonsik Kim dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1497e314c678ea0b53dd9296ba6b5c3272c702433b47Jinsuk Kim dprintf(fd, " Total writes: %d\n", mNumWrites); 1498cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 149906528d7f18ad01377357d337eaa3e875a242bd2dAndreas Huber dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 150087f2a558dd12043631e12c361abef301bf603140Andreas Huber dprintf(fd, " Suspend count: %d\n", mSuspended); 150187f2a558dd12043631e12c361abef301bf603140Andreas Huber dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 150287f2a558dd12043631e12c361abef301bf603140Andreas Huber dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1503cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1504cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1505cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber AudioStreamOut *output = mOutput; 15069ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1507cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber String8 flagsAsString = outputFlagsToString(flags); 1508386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 15099ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih} 15109ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih 1511386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber// Thread virtuals 1512386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber 15139ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shihvoid AudioFlinger::PlaybackThread::onFirstRef() 15149ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih{ 15159ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 15169ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih} 15179ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih 15189ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih// ThreadBase virtuals 15199ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shihvoid AudioFlinger::PlaybackThread::preExit() 15209ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih{ 15219ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih ALOGV(" preExit()"); 15229ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih // FIXME this is using hard-coded strings but in the future, this functionality will be 15239ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih // converted to use audio HAL extensions required to support tunneling 1524386d609dc513e838c7e7c4c46c604493ccd560beAndreas Huber mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 15259ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih} 15269ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih 15279ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 15289ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shihsp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 15299ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih const sp<AudioFlinger::Client>& client, 15309ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih audio_stream_type_t streamType, 1531cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber uint32_t sampleRate, 15329ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih audio_format_t format, 15339ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih audio_channel_mask_t channelMask, 1534cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber size_t *pFrameCount, 1535cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber const sp<IMemory>& sharedBuffer, 1536cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber int sessionId, 15379ff1e728de605c4cbc61bc95bb36e515b93654fbRobert Shih IAudioFlinger::track_flags_t *flags, 1538cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber pid_t tid, 1539cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber int uid, 1540bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih status_t *status) 1541bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih{ 1542bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih size_t frameCount = *pFrameCount; 1543bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih sp<Track> track; 1544bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih status_t lStatus; 1545bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih 1546bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1547bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih 1548bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih // client expresses a preference for FAST, but we get the final say 1549bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih if (*flags & IAudioFlinger::TRACK_FAST) { 1550bf20727f0aaf609bc3b495b07b45822b137d21baRobert Shih if ( 155143c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber // not timed 155243c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber (!isTimed) && 155343c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber // either of these use cases: 155443c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber ( 155543c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber // use case 1: shared buffer with any frame count 155643c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber ( 155743c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber (sharedBuffer != 0) 155843c3e6ce02215ca99d506458f596cb1211639f29Andreas Huber ) || 1559c9fa35cf7c9c11a8acb77128db1a6a13f1befe3cChad Brubaker // use case 2: frame count is default or at least as large as HAL 156087f2a558dd12043631e12c361abef301bf603140Andreas Huber ( 156119cec89f8b05fd05f8034ee1a4cd39ee09c33f02Marco Nelissen // we formerly checked for a callback handler (non-0 tid), 156219cec89f8b05fd05f8034ee1a4cd39ee09c33f02Marco Nelissen // but that is no longer required for TRANSFER_OBTAIN mode 156387f2a558dd12043631e12c361abef301bf603140Andreas Huber ((frameCount == 0) || 156487f2a558dd12043631e12c361abef301bf603140Andreas Huber (frameCount >= mFrameCount)) 156587f2a558dd12043631e12c361abef301bf603140Andreas Huber ) 156687f2a558dd12043631e12c361abef301bf603140Andreas Huber ) && 156787f2a558dd12043631e12c361abef301bf603140Andreas Huber // PCM data 156887f2a558dd12043631e12c361abef301bf603140Andreas Huber audio_is_linear_pcm(format) && 156987f2a558dd12043631e12c361abef301bf603140Andreas Huber // identical channel mask to sink, or mono in and stereo sink 157087f2a558dd12043631e12c361abef301bf603140Andreas Huber (channelMask == mChannelMask || 157187f2a558dd12043631e12c361abef301bf603140Andreas Huber (channelMask == AUDIO_CHANNEL_OUT_MONO && 157287f2a558dd12043631e12c361abef301bf603140Andreas Huber mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 157387f2a558dd12043631e12c361abef301bf603140Andreas Huber // hardware sample rate 157487f2a558dd12043631e12c361abef301bf603140Andreas Huber (sampleRate == mSampleRate) && 157587f2a558dd12043631e12c361abef301bf603140Andreas Huber // normal mixer has an associated fast mixer 157687f2a558dd12043631e12c361abef301bf603140Andreas Huber hasFastMixer() && 157787f2a558dd12043631e12c361abef301bf603140Andreas Huber // there are sufficient fast track slots available 1578c9fa35cf7c9c11a8acb77128db1a6a13f1befe3cChad Brubaker (mFastTrackAvailMask != 0) 157987f2a558dd12043631e12c361abef301bf603140Andreas Huber // FIXME test that MixerThread for this fast track has a capable output HAL 158087f2a558dd12043631e12c361abef301bf603140Andreas Huber // FIXME add a permission test also? 158187f2a558dd12043631e12c361abef301bf603140Andreas Huber ) { 158287f2a558dd12043631e12c361abef301bf603140Andreas Huber // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 158387f2a558dd12043631e12c361abef301bf603140Andreas Huber if (frameCount == 0) { 158487f2a558dd12043631e12c361abef301bf603140Andreas Huber // read the fast track multiplier property the first time it is needed 158587f2a558dd12043631e12c361abef301bf603140Andreas Huber int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 15868dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (ok != 0) { 15878dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber ALOGE("%s pthread_once failed: %d", __func__, ok); 15889ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 15899ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim frameCount = mFrameCount * sFastTrackMultiplier; 15909ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 15919ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 15929ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim frameCount, mFrameCount); 15939ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } else { 15949ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 15959ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 15969ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim "sampleRate=%u mSampleRate=%u " 15979ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 15989ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 15999ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim audio_is_linear_pcm(format), 16009ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 16019ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim *flags &= ~IAudioFlinger::TRACK_FAST; 16029ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16039ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16049ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // For normal PCM streaming tracks, update minimum frame count. 16059ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // For compatibility with AudioTrack calculation, buffer depth is forced 16069ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 16079ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // This is probably too conservative, but legacy application code may depend on it. 16089ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // If you change this calculation, also review the start threshold which is related. 16099ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim if (!(*flags & IAudioFlinger::TRACK_FAST) 16109ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim && audio_is_linear_pcm(format) && sharedBuffer == 0) { 16119ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // this must match AudioTrack.cpp calculateMinFrameCount(). 16129ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // TODO: Move to a common library 16139ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 16149ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 16159ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim if (minBufCount < 2) { 16169ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim minBufCount = 2; 16179ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16189ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 16199ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // or the client should compute and pass in a larger buffer request. 16209ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim size_t minFrameCount = 16219ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim minBufCount * sourceFramesNeededWithTimestretch( 16229ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim sampleRate, mNormalFrameCount, 16239ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 16249ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim if (frameCount < minFrameCount) { // including frameCount == 0 16259ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim frameCount = minFrameCount; 16269ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16279ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16289ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim *pFrameCount = frameCount; 16299ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim 16309ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim switch (mType) { 16319ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim 16329ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim case DIRECT: 16339ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim if (audio_is_linear_pcm(format)) { 16349ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 16359ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 16369ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim "for output %p with format %#x", 16379ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim sampleRate, format, channelMask, mOutput, mFormat); 16389ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim lStatus = BAD_VALUE; 16399ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim goto Exit; 16409ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16419ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16429ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim break; 16439ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim 16449ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim case OFFLOAD: 16459ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 16469ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 16479ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim "for output %p with format %#x", 16489ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim sampleRate, format, channelMask, mOutput, mFormat); 16499ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim lStatus = BAD_VALUE; 16509ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim goto Exit; 16519ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16529ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim break; 16539ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim 16549ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim default: 16559ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim if (!audio_is_linear_pcm(format)) { 16569ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim ALOGE("createTrack_l() Bad parameter: format %#x \"" 16579ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim "for output %p with format %#x", 16589ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim format, mOutput, mFormat); 16599ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim lStatus = BAD_VALUE; 16609ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim goto Exit; 16618dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 16628dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 16638dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 16648dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber lStatus = BAD_VALUE; 16658dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber goto Exit; 16668dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 16678dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber break; 16688dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 16698dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 16708dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 16718dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber lStatus = initCheck(); 16728dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (lStatus != NO_ERROR) { 16738dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber ALOGE("createTrack_l() audio driver not initialized"); 16748dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber goto Exit; 16758dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 16768dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 16778dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber { // scope for mLock 16788dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber Mutex::Autolock _l(mLock); 16798dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 16808dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // all tracks in same audio session must share the same routing strategy otherwise 16818dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // conflicts will happen when tracks are moved from one output to another by audio policy 16828dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // manager 16838dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 16848dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber for (size_t i = 0; i < mTracks.size(); ++i) { 16858dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber sp<Track> t = mTracks[i]; 16868dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (t != 0 && t->isExternalTrack()) { 16878dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 16888dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (sessionId == t->sessionId() && strategy != actual) { 16898dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 16908dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber strategy, actual); 16919ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim lStatus = BAD_VALUE; 16929ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim goto Exit; 16939ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16949ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 16958dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 16968dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 16978dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (!isTimed) { 16988dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber track = new Track(this, client, streamType, sampleRate, format, 16999ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim channelMask, frameCount, NULL, sharedBuffer, 17008dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 17018dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } else { 17028dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber track = TimedTrack::create(this, client, streamType, sampleRate, format, 17038dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber channelMask, frameCount, sharedBuffer, sessionId, uid); 17048dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 17058dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 17068dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // new Track always returns non-NULL, 17078dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // but TimedTrack::create() is a factory that could fail by returning NULL 17088dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 17098dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (lStatus != NO_ERROR) { 17108dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 17118dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber // track must be cleared from the caller as the caller has the AF lock 17128dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber goto Exit; 17138dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 17148dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber mTracks.add(track); 17158dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber 17168dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber sp<EffectChain> chain = getEffectChain_l(sessionId); 17178dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber if (chain != 0) { 17188dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 17198dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber track->setMainBuffer(chain->inBuffer()); 17208dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 17218dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber chain->incTrackCnt(); 17228dfa228201131da0bf3ba1d74c819c27c0500f6bAndreas Huber } 17239ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim 17249ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 17259ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim pid_t callingPid = IPCThreadState::self()->getCallingPid(); 17269ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 17279ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim // so ask activity manager to do this on our behalf 17289ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 17299ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 17309ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 17319ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim 17329ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim lStatus = NO_ERROR; 17339ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim 17349ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk KimExit: 17359ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim *status = lStatus; 17369ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim return track; 1737c0c9f50d15f7b9ed539c0c6277296d083f41b293Marco Nelissen} 1738c0c9f50d15f7b9ed539c0c6277296d083f41b293Marco Nelissen 1739c0c9f50d15f7b9ed539c0c6277296d083f41b293Marco Nelissenuint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1740c0c9f50d15f7b9ed539c0c6277296d083f41b293Marco Nelissen{ 1741c0c9f50d15f7b9ed539c0c6277296d083f41b293Marco Nelissen return latency; 1742c0c9f50d15f7b9ed539c0c6277296d083f41b293Marco Nelissen} 1743c0c9f50d15f7b9ed539c0c6277296d083f41b293Marco Nelissen 17449ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kimuint32_t AudioFlinger::PlaybackThread::latency() const 17459ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim{ 17469ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim Mutex::Autolock _l(mLock); 17479ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim return latency_l(); 17489ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim} 17499ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kimuint32_t AudioFlinger::PlaybackThread::latency_l() const 17509ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim{ 17519ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim if (initCheck() == NO_ERROR) { 17529ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 17539ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } else { 17549ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim return 0; 17559ca7b9c74e882526ee5810ff62f203bf75cd3f1aJinsuk Kim } 1756cda17c606b0fe3ccda4dc68a6d43882410ea2462Andreas Huber} 1757 1758void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1759{ 1760 Mutex::Autolock _l(mLock); 1761 // Don't apply master volume in SW if our HAL can do it for us. 1762 if (mOutput && mOutput->audioHwDev && 1763 mOutput->audioHwDev->canSetMasterVolume()) { 1764 mMasterVolume = 1.0; 1765 } else { 1766 mMasterVolume = value; 1767 } 1768} 1769 1770void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1771{ 1772 Mutex::Autolock _l(mLock); 1773 // Don't apply master mute in SW if our HAL can do it for us. 1774 if (mOutput && mOutput->audioHwDev && 1775 mOutput->audioHwDev->canSetMasterMute()) { 1776 mMasterMute = false; 1777 } else { 1778 mMasterMute = muted; 1779 } 1780} 1781 1782void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1783{ 1784 Mutex::Autolock _l(mLock); 1785 mStreamTypes[stream].volume = value; 1786 broadcast_l(); 1787} 1788 1789void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1790{ 1791 Mutex::Autolock _l(mLock); 1792 mStreamTypes[stream].mute = muted; 1793 broadcast_l(); 1794} 1795 1796float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1797{ 1798 Mutex::Autolock _l(mLock); 1799 return mStreamTypes[stream].volume; 1800} 1801 1802// addTrack_l() must be called with ThreadBase::mLock held 1803status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1804{ 1805 status_t status = ALREADY_EXISTS; 1806 1807 // set retry count for buffer fill 1808 track->mRetryCount = kMaxTrackStartupRetries; 1809 if (mActiveTracks.indexOf(track) < 0) { 1810 // the track is newly added, make sure it fills up all its 1811 // buffers before playing. This is to ensure the client will 1812 // effectively get the latency it requested. 1813 if (track->isExternalTrack()) { 1814 TrackBase::track_state state = track->mState; 1815 mLock.unlock(); 1816 status = AudioSystem::startOutput(mId, track->streamType(), 1817 (audio_session_t)track->sessionId()); 1818 mLock.lock(); 1819 // abort track was stopped/paused while we released the lock 1820 if (state != track->mState) { 1821 if (status == NO_ERROR) { 1822 mLock.unlock(); 1823 AudioSystem::stopOutput(mId, track->streamType(), 1824 (audio_session_t)track->sessionId()); 1825 mLock.lock(); 1826 } 1827 return INVALID_OPERATION; 1828 } 1829 // abort if start is rejected by audio policy manager 1830 if (status != NO_ERROR) { 1831 return PERMISSION_DENIED; 1832 } 1833#ifdef ADD_BATTERY_DATA 1834 // to track the speaker usage 1835 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1836#endif 1837 } 1838 1839 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1840 track->mResetDone = false; 1841 track->mPresentationCompleteFrames = 0; 1842 mActiveTracks.add(track); 1843 mWakeLockUids.add(track->uid()); 1844 mActiveTracksGeneration++; 1845 mLatestActiveTrack = track; 1846 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1847 if (chain != 0) { 1848 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1849 track->sessionId()); 1850 chain->incActiveTrackCnt(); 1851 } 1852 1853 status = NO_ERROR; 1854 } 1855 1856 onAddNewTrack_l(); 1857 return status; 1858} 1859 1860bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1861{ 1862 track->terminate(); 1863 // active tracks are removed by threadLoop() 1864 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1865 track->mState = TrackBase::STOPPED; 1866 if (!trackActive) { 1867 removeTrack_l(track); 1868 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1869 track->mState = TrackBase::STOPPING_1; 1870 } 1871 1872 return trackActive; 1873} 1874 1875void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1876{ 1877 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1878 mTracks.remove(track); 1879 deleteTrackName_l(track->name()); 1880 // redundant as track is about to be destroyed, for dumpsys only 1881 track->mName = -1; 1882 if (track->isFastTrack()) { 1883 int index = track->mFastIndex; 1884 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1885 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1886 mFastTrackAvailMask |= 1 << index; 1887 // redundant as track is about to be destroyed, for dumpsys only 1888 track->mFastIndex = -1; 1889 } 1890 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1891 if (chain != 0) { 1892 chain->decTrackCnt(); 1893 } 1894} 1895 1896void AudioFlinger::PlaybackThread::broadcast_l() 1897{ 1898 // Thread could be blocked waiting for async 1899 // so signal it to handle state changes immediately 1900 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1901 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1902 mSignalPending = true; 1903 mWaitWorkCV.broadcast(); 1904} 1905 1906String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1907{ 1908 Mutex::Autolock _l(mLock); 1909 if (initCheck() != NO_ERROR) { 1910 return String8(); 1911 } 1912 1913 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1914 const String8 out_s8(s); 1915 free(s); 1916 return out_s8; 1917} 1918 1919void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1920 AudioSystem::OutputDescriptor desc; 1921 void *param2 = NULL; 1922 1923 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1924 param); 1925 1926 switch (event) { 1927 case AudioSystem::OUTPUT_OPENED: 1928 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1929 desc.channelMask = mChannelMask; 1930 desc.samplingRate = mSampleRate; 1931 desc.format = mFormat; 1932 desc.frameCount = mNormalFrameCount; // FIXME see 1933 // AudioFlinger::frameCount(audio_io_handle_t) 1934 desc.latency = latency_l(); 1935 param2 = &desc; 1936 break; 1937 1938 case AudioSystem::STREAM_CONFIG_CHANGED: 1939 param2 = ¶m; 1940 case AudioSystem::OUTPUT_CLOSED: 1941 default: 1942 break; 1943 } 1944 mAudioFlinger->audioConfigChanged(event, mId, param2); 1945} 1946 1947void AudioFlinger::PlaybackThread::writeCallback() 1948{ 1949 ALOG_ASSERT(mCallbackThread != 0); 1950 mCallbackThread->resetWriteBlocked(); 1951} 1952 1953void AudioFlinger::PlaybackThread::drainCallback() 1954{ 1955 ALOG_ASSERT(mCallbackThread != 0); 1956 mCallbackThread->resetDraining(); 1957} 1958 1959void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1960{ 1961 Mutex::Autolock _l(mLock); 1962 // reject out of sequence requests 1963 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1964 mWriteAckSequence &= ~1; 1965 mWaitWorkCV.signal(); 1966 } 1967} 1968 1969void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1970{ 1971 Mutex::Autolock _l(mLock); 1972 // reject out of sequence requests 1973 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1974 mDrainSequence &= ~1; 1975 mWaitWorkCV.signal(); 1976 } 1977} 1978 1979// static 1980int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1981 void *param __unused, 1982 void *cookie) 1983{ 1984 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1985 ALOGV("asyncCallback() event %d", event); 1986 switch (event) { 1987 case STREAM_CBK_EVENT_WRITE_READY: 1988 me->writeCallback(); 1989 break; 1990 case STREAM_CBK_EVENT_DRAIN_READY: 1991 me->drainCallback(); 1992 break; 1993 default: 1994 ALOGW("asyncCallback() unknown event %d", event); 1995 break; 1996 } 1997 return 0; 1998} 1999 2000void AudioFlinger::PlaybackThread::readOutputParameters_l() 2001{ 2002 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2003 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2004 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2005 if (!audio_is_output_channel(mChannelMask)) { 2006 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2007 } 2008 if ((mType == MIXER || mType == DUPLICATING) 2009 && !isValidPcmSinkChannelMask(mChannelMask)) { 2010 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2011 mChannelMask); 2012 } 2013 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2014 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2015 mFormat = mHALFormat; 2016 if (!audio_is_valid_format(mFormat)) { 2017 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2018 } 2019 if ((mType == MIXER || mType == DUPLICATING) 2020 && !isValidPcmSinkFormat(mFormat)) { 2021 LOG_FATAL("HAL format %#x not supported for mixed output", 2022 mFormat); 2023 } 2024 mFrameSize = mOutput->getFrameSize(); 2025 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2026 mFrameCount = mBufferSize / mFrameSize; 2027 if (mFrameCount & 15) { 2028 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2029 mFrameCount); 2030 } 2031 2032 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2033 (mOutput->stream->set_callback != NULL)) { 2034 if (mOutput->stream->set_callback(mOutput->stream, 2035 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2036 mUseAsyncWrite = true; 2037 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2038 } 2039 } 2040 2041 mHwSupportsPause = false; 2042 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2043 if (mOutput->stream->pause != NULL) { 2044 if (mOutput->stream->resume != NULL) { 2045 mHwSupportsPause = true; 2046 } else { 2047 ALOGW("direct output implements pause but not resume"); 2048 } 2049 } else if (mOutput->stream->resume != NULL) { 2050 ALOGW("direct output implements resume but not pause"); 2051 } 2052 } 2053 2054 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2055 // For best precision, we use float instead of the associated output 2056 // device format (typically PCM 16 bit). 2057 2058 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2059 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2060 mBufferSize = mFrameSize * mFrameCount; 2061 2062 // TODO: We currently use the associated output device channel mask and sample rate. 2063 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2064 // (if a valid mask) to avoid premature downmix. 2065 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2066 // instead of the output device sample rate to avoid loss of high frequency information. 2067 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2068 } 2069 2070 // Calculate size of normal sink buffer relative to the HAL output buffer size 2071 double multiplier = 1.0; 2072 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2073 kUseFastMixer == FastMixer_Dynamic)) { 2074 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2075 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2076 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2077 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2078 maxNormalFrameCount = maxNormalFrameCount & ~15; 2079 if (maxNormalFrameCount < minNormalFrameCount) { 2080 maxNormalFrameCount = minNormalFrameCount; 2081 } 2082 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2083 if (multiplier <= 1.0) { 2084 multiplier = 1.0; 2085 } else if (multiplier <= 2.0) { 2086 if (2 * mFrameCount <= maxNormalFrameCount) { 2087 multiplier = 2.0; 2088 } else { 2089 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2090 } 2091 } else { 2092 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2093 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2094 // track, but we sometimes have to do this to satisfy the maximum frame count 2095 // constraint) 2096 // FIXME this rounding up should not be done if no HAL SRC 2097 uint32_t truncMult = (uint32_t) multiplier; 2098 if ((truncMult & 1)) { 2099 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2100 ++truncMult; 2101 } 2102 } 2103 multiplier = (double) truncMult; 2104 } 2105 } 2106 mNormalFrameCount = multiplier * mFrameCount; 2107 // round up to nearest 16 frames to satisfy AudioMixer 2108 if (mType == MIXER || mType == DUPLICATING) { 2109 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2110 } 2111 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2112 mNormalFrameCount); 2113 2114 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2115 // Originally this was int16_t[] array, need to remove legacy implications. 2116 free(mSinkBuffer); 2117 mSinkBuffer = NULL; 2118 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2119 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2120 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2121 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2122 2123 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2124 // drives the output. 2125 free(mMixerBuffer); 2126 mMixerBuffer = NULL; 2127 if (mMixerBufferEnabled) { 2128 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2129 mMixerBufferSize = mNormalFrameCount * mChannelCount 2130 * audio_bytes_per_sample(mMixerBufferFormat); 2131 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2132 } 2133 free(mEffectBuffer); 2134 mEffectBuffer = NULL; 2135 if (mEffectBufferEnabled) { 2136 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2137 mEffectBufferSize = mNormalFrameCount * mChannelCount 2138 * audio_bytes_per_sample(mEffectBufferFormat); 2139 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2140 } 2141 2142 // force reconfiguration of effect chains and engines to take new buffer size and audio 2143 // parameters into account 2144 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2145 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2146 // matter. 2147 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2148 Vector< sp<EffectChain> > effectChains = mEffectChains; 2149 for (size_t i = 0; i < effectChains.size(); i ++) { 2150 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2151 } 2152} 2153 2154 2155status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2156{ 2157 if (halFrames == NULL || dspFrames == NULL) { 2158 return BAD_VALUE; 2159 } 2160 Mutex::Autolock _l(mLock); 2161 if (initCheck() != NO_ERROR) { 2162 return INVALID_OPERATION; 2163 } 2164 size_t framesWritten = mBytesWritten / mFrameSize; 2165 *halFrames = framesWritten; 2166 2167 if (isSuspended()) { 2168 // return an estimation of rendered frames when the output is suspended 2169 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2170 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2171 return NO_ERROR; 2172 } else { 2173 status_t status; 2174 uint32_t frames; 2175 status = mOutput->getRenderPosition(&frames); 2176 *dspFrames = (size_t)frames; 2177 return status; 2178 } 2179} 2180 2181uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2182{ 2183 Mutex::Autolock _l(mLock); 2184 uint32_t result = 0; 2185 if (getEffectChain_l(sessionId) != 0) { 2186 result = EFFECT_SESSION; 2187 } 2188 2189 for (size_t i = 0; i < mTracks.size(); ++i) { 2190 sp<Track> track = mTracks[i]; 2191 if (sessionId == track->sessionId() && !track->isInvalid()) { 2192 result |= TRACK_SESSION; 2193 break; 2194 } 2195 } 2196 2197 return result; 2198} 2199 2200uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2201{ 2202 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2203 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2204 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2205 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2206 } 2207 for (size_t i = 0; i < mTracks.size(); i++) { 2208 sp<Track> track = mTracks[i]; 2209 if (sessionId == track->sessionId() && !track->isInvalid()) { 2210 return AudioSystem::getStrategyForStream(track->streamType()); 2211 } 2212 } 2213 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2214} 2215 2216 2217AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2218{ 2219 Mutex::Autolock _l(mLock); 2220 return mOutput; 2221} 2222 2223AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2224{ 2225 Mutex::Autolock _l(mLock); 2226 AudioStreamOut *output = mOutput; 2227 mOutput = NULL; 2228 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2229 // must push a NULL and wait for ack 2230 mOutputSink.clear(); 2231 mPipeSink.clear(); 2232 mNormalSink.clear(); 2233 return output; 2234} 2235 2236// this method must always be called either with ThreadBase mLock held or inside the thread loop 2237audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2238{ 2239 if (mOutput == NULL) { 2240 return NULL; 2241 } 2242 return &mOutput->stream->common; 2243} 2244 2245uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2246{ 2247 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2248} 2249 2250status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2251{ 2252 if (!isValidSyncEvent(event)) { 2253 return BAD_VALUE; 2254 } 2255 2256 Mutex::Autolock _l(mLock); 2257 2258 for (size_t i = 0; i < mTracks.size(); ++i) { 2259 sp<Track> track = mTracks[i]; 2260 if (event->triggerSession() == track->sessionId()) { 2261 (void) track->setSyncEvent(event); 2262 return NO_ERROR; 2263 } 2264 } 2265 2266 return NAME_NOT_FOUND; 2267} 2268 2269bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2270{ 2271 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2272} 2273 2274void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2275 const Vector< sp<Track> >& tracksToRemove) 2276{ 2277 size_t count = tracksToRemove.size(); 2278 if (count > 0) { 2279 for (size_t i = 0 ; i < count ; i++) { 2280 const sp<Track>& track = tracksToRemove.itemAt(i); 2281 if (track->isExternalTrack()) { 2282 AudioSystem::stopOutput(mId, track->streamType(), 2283 (audio_session_t)track->sessionId()); 2284#ifdef ADD_BATTERY_DATA 2285 // to track the speaker usage 2286 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2287#endif 2288 if (track->isTerminated()) { 2289 AudioSystem::releaseOutput(mId, track->streamType(), 2290 (audio_session_t)track->sessionId()); 2291 } 2292 } 2293 } 2294 } 2295} 2296 2297void AudioFlinger::PlaybackThread::checkSilentMode_l() 2298{ 2299 if (!mMasterMute) { 2300 char value[PROPERTY_VALUE_MAX]; 2301 if (property_get("ro.audio.silent", value, "0") > 0) { 2302 char *endptr; 2303 unsigned long ul = strtoul(value, &endptr, 0); 2304 if (*endptr == '\0' && ul != 0) { 2305 ALOGD("Silence is golden"); 2306 // The setprop command will not allow a property to be changed after 2307 // the first time it is set, so we don't have to worry about un-muting. 2308 setMasterMute_l(true); 2309 } 2310 } 2311 } 2312} 2313 2314// shared by MIXER and DIRECT, overridden by DUPLICATING 2315ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2316{ 2317 // FIXME rewrite to reduce number of system calls 2318 mLastWriteTime = systemTime(); 2319 mInWrite = true; 2320 ssize_t bytesWritten; 2321 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2322 2323 // If an NBAIO sink is present, use it to write the normal mixer's submix 2324 if (mNormalSink != 0) { 2325 2326 const size_t count = mBytesRemaining / mFrameSize; 2327 2328 ATRACE_BEGIN("write"); 2329 // update the setpoint when AudioFlinger::mScreenState changes 2330 uint32_t screenState = AudioFlinger::mScreenState; 2331 if (screenState != mScreenState) { 2332 mScreenState = screenState; 2333 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2334 if (pipe != NULL) { 2335 pipe->setAvgFrames((mScreenState & 1) ? 2336 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2337 } 2338 } 2339 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2340 ATRACE_END(); 2341 if (framesWritten > 0) { 2342 bytesWritten = framesWritten * mFrameSize; 2343 } else { 2344 bytesWritten = framesWritten; 2345 } 2346 mLatchDValid = false; 2347 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2348 if (status == NO_ERROR) { 2349 size_t totalFramesWritten = mNormalSink->framesWritten(); 2350 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2351 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2352 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2353 mLatchDValid = true; 2354 } 2355 } 2356 // otherwise use the HAL / AudioStreamOut directly 2357 } else { 2358 // Direct output and offload threads 2359 2360 if (mUseAsyncWrite) { 2361 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2362 mWriteAckSequence += 2; 2363 mWriteAckSequence |= 1; 2364 ALOG_ASSERT(mCallbackThread != 0); 2365 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2366 } 2367 // FIXME We should have an implementation of timestamps for direct output threads. 2368 // They are used e.g for multichannel PCM playback over HDMI. 2369 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2370 if (mUseAsyncWrite && 2371 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2372 // do not wait for async callback in case of error of full write 2373 mWriteAckSequence &= ~1; 2374 ALOG_ASSERT(mCallbackThread != 0); 2375 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2376 } 2377 } 2378 2379 mNumWrites++; 2380 mInWrite = false; 2381 mStandby = false; 2382 return bytesWritten; 2383} 2384 2385void AudioFlinger::PlaybackThread::threadLoop_drain() 2386{ 2387 if (mOutput->stream->drain) { 2388 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2389 if (mUseAsyncWrite) { 2390 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2391 mDrainSequence |= 1; 2392 ALOG_ASSERT(mCallbackThread != 0); 2393 mCallbackThread->setDraining(mDrainSequence); 2394 } 2395 mOutput->stream->drain(mOutput->stream, 2396 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2397 : AUDIO_DRAIN_ALL); 2398 } 2399} 2400 2401void AudioFlinger::PlaybackThread::threadLoop_exit() 2402{ 2403 { 2404 Mutex::Autolock _l(mLock); 2405 for (size_t i = 0; i < mTracks.size(); i++) { 2406 sp<Track> track = mTracks[i]; 2407 track->invalidate(); 2408 } 2409 } 2410} 2411 2412/* 2413The derived values that are cached: 2414 - mSinkBufferSize from frame count * frame size 2415 - activeSleepTime from activeSleepTimeUs() 2416 - idleSleepTime from idleSleepTimeUs() 2417 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2418 - maxPeriod from frame count and sample rate (MIXER only) 2419 2420The parameters that affect these derived values are: 2421 - frame count 2422 - frame size 2423 - sample rate 2424 - device type: A2DP or not 2425 - device latency 2426 - format: PCM or not 2427 - active sleep time 2428 - idle sleep time 2429*/ 2430 2431void AudioFlinger::PlaybackThread::cacheParameters_l() 2432{ 2433 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2434 activeSleepTime = activeSleepTimeUs(); 2435 idleSleepTime = idleSleepTimeUs(); 2436} 2437 2438void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2439{ 2440 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2441 this, streamType, mTracks.size()); 2442 Mutex::Autolock _l(mLock); 2443 2444 size_t size = mTracks.size(); 2445 for (size_t i = 0; i < size; i++) { 2446 sp<Track> t = mTracks[i]; 2447 if (t->streamType() == streamType) { 2448 t->invalidate(); 2449 } 2450 } 2451} 2452 2453status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2454{ 2455 int session = chain->sessionId(); 2456 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2457 ? mEffectBuffer : mSinkBuffer); 2458 bool ownsBuffer = false; 2459 2460 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2461 if (session > 0) { 2462 // Only one effect chain can be present in direct output thread and it uses 2463 // the sink buffer as input 2464 if (mType != DIRECT) { 2465 size_t numSamples = mNormalFrameCount * mChannelCount; 2466 buffer = new int16_t[numSamples]; 2467 memset(buffer, 0, numSamples * sizeof(int16_t)); 2468 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2469 ownsBuffer = true; 2470 } 2471 2472 // Attach all tracks with same session ID to this chain. 2473 for (size_t i = 0; i < mTracks.size(); ++i) { 2474 sp<Track> track = mTracks[i]; 2475 if (session == track->sessionId()) { 2476 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2477 buffer); 2478 track->setMainBuffer(buffer); 2479 chain->incTrackCnt(); 2480 } 2481 } 2482 2483 // indicate all active tracks in the chain 2484 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2485 sp<Track> track = mActiveTracks[i].promote(); 2486 if (track == 0) { 2487 continue; 2488 } 2489 if (session == track->sessionId()) { 2490 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2491 chain->incActiveTrackCnt(); 2492 } 2493 } 2494 } 2495 chain->setThread(this); 2496 chain->setInBuffer(buffer, ownsBuffer); 2497 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2498 ? mEffectBuffer : mSinkBuffer)); 2499 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2500 // chains list in order to be processed last as it contains output stage effects 2501 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2502 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2503 // after track specific effects and before output stage 2504 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2505 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2506 // Effect chain for other sessions are inserted at beginning of effect 2507 // chains list to be processed before output mix effects. Relative order between other 2508 // sessions is not important 2509 size_t size = mEffectChains.size(); 2510 size_t i = 0; 2511 for (i = 0; i < size; i++) { 2512 if (mEffectChains[i]->sessionId() < session) { 2513 break; 2514 } 2515 } 2516 mEffectChains.insertAt(chain, i); 2517 checkSuspendOnAddEffectChain_l(chain); 2518 2519 return NO_ERROR; 2520} 2521 2522size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2523{ 2524 int session = chain->sessionId(); 2525 2526 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2527 2528 for (size_t i = 0; i < mEffectChains.size(); i++) { 2529 if (chain == mEffectChains[i]) { 2530 mEffectChains.removeAt(i); 2531 // detach all active tracks from the chain 2532 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2533 sp<Track> track = mActiveTracks[i].promote(); 2534 if (track == 0) { 2535 continue; 2536 } 2537 if (session == track->sessionId()) { 2538 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2539 chain.get(), session); 2540 chain->decActiveTrackCnt(); 2541 } 2542 } 2543 2544 // detach all tracks with same session ID from this chain 2545 for (size_t i = 0; i < mTracks.size(); ++i) { 2546 sp<Track> track = mTracks[i]; 2547 if (session == track->sessionId()) { 2548 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2549 chain->decTrackCnt(); 2550 } 2551 } 2552 break; 2553 } 2554 } 2555 return mEffectChains.size(); 2556} 2557 2558status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2559 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2560{ 2561 Mutex::Autolock _l(mLock); 2562 return attachAuxEffect_l(track, EffectId); 2563} 2564 2565status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2566 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2567{ 2568 status_t status = NO_ERROR; 2569 2570 if (EffectId == 0) { 2571 track->setAuxBuffer(0, NULL); 2572 } else { 2573 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2574 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2575 if (effect != 0) { 2576 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2577 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2578 } else { 2579 status = INVALID_OPERATION; 2580 } 2581 } else { 2582 status = BAD_VALUE; 2583 } 2584 } 2585 return status; 2586} 2587 2588void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2589{ 2590 for (size_t i = 0; i < mTracks.size(); ++i) { 2591 sp<Track> track = mTracks[i]; 2592 if (track->auxEffectId() == effectId) { 2593 attachAuxEffect_l(track, 0); 2594 } 2595 } 2596} 2597 2598bool AudioFlinger::PlaybackThread::threadLoop() 2599{ 2600 Vector< sp<Track> > tracksToRemove; 2601 2602 standbyTime = systemTime(); 2603 2604 // MIXER 2605 nsecs_t lastWarning = 0; 2606 2607 // DUPLICATING 2608 // FIXME could this be made local to while loop? 2609 writeFrames = 0; 2610 2611 int lastGeneration = 0; 2612 2613 cacheParameters_l(); 2614 sleepTime = idleSleepTime; 2615 2616 if (mType == MIXER) { 2617 sleepTimeShift = 0; 2618 } 2619 2620 CpuStats cpuStats; 2621 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2622 2623 acquireWakeLock(); 2624 2625 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2626 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2627 // and then that string will be logged at the next convenient opportunity. 2628 const char *logString = NULL; 2629 2630 checkSilentMode_l(); 2631 2632 while (!exitPending()) 2633 { 2634 cpuStats.sample(myName); 2635 2636 Vector< sp<EffectChain> > effectChains; 2637 2638 { // scope for mLock 2639 2640 Mutex::Autolock _l(mLock); 2641 2642 processConfigEvents_l(); 2643 2644 if (logString != NULL) { 2645 mNBLogWriter->logTimestamp(); 2646 mNBLogWriter->log(logString); 2647 logString = NULL; 2648 } 2649 2650 // Gather the framesReleased counters for all active tracks, 2651 // and latch them atomically with the timestamp. 2652 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2653 mLatchD.mFramesReleased.clear(); 2654 size_t size = mActiveTracks.size(); 2655 for (size_t i = 0; i < size; i++) { 2656 sp<Track> t = mActiveTracks[i].promote(); 2657 if (t != 0) { 2658 mLatchD.mFramesReleased.add(t.get(), 2659 t->mAudioTrackServerProxy->framesReleased()); 2660 } 2661 } 2662 if (mLatchDValid) { 2663 mLatchQ = mLatchD; 2664 mLatchDValid = false; 2665 mLatchQValid = true; 2666 } 2667 2668 saveOutputTracks(); 2669 if (mSignalPending) { 2670 // A signal was raised while we were unlocked 2671 mSignalPending = false; 2672 } else if (waitingAsyncCallback_l()) { 2673 if (exitPending()) { 2674 break; 2675 } 2676 releaseWakeLock_l(); 2677 mWakeLockUids.clear(); 2678 mActiveTracksGeneration++; 2679 ALOGV("wait async completion"); 2680 mWaitWorkCV.wait(mLock); 2681 ALOGV("async completion/wake"); 2682 acquireWakeLock_l(); 2683 standbyTime = systemTime() + standbyDelay; 2684 sleepTime = 0; 2685 2686 continue; 2687 } 2688 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2689 isSuspended()) { 2690 // put audio hardware into standby after short delay 2691 if (shouldStandby_l()) { 2692 2693 threadLoop_standby(); 2694 2695 mStandby = true; 2696 } 2697 2698 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2699 // we're about to wait, flush the binder command buffer 2700 IPCThreadState::self()->flushCommands(); 2701 2702 clearOutputTracks(); 2703 2704 if (exitPending()) { 2705 break; 2706 } 2707 2708 releaseWakeLock_l(); 2709 mWakeLockUids.clear(); 2710 mActiveTracksGeneration++; 2711 // wait until we have something to do... 2712 ALOGV("%s going to sleep", myName.string()); 2713 mWaitWorkCV.wait(mLock); 2714 ALOGV("%s waking up", myName.string()); 2715 acquireWakeLock_l(); 2716 2717 mMixerStatus = MIXER_IDLE; 2718 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2719 mBytesWritten = 0; 2720 mBytesRemaining = 0; 2721 checkSilentMode_l(); 2722 2723 standbyTime = systemTime() + standbyDelay; 2724 sleepTime = idleSleepTime; 2725 if (mType == MIXER) { 2726 sleepTimeShift = 0; 2727 } 2728 2729 continue; 2730 } 2731 } 2732 // mMixerStatusIgnoringFastTracks is also updated internally 2733 mMixerStatus = prepareTracks_l(&tracksToRemove); 2734 2735 // compare with previously applied list 2736 if (lastGeneration != mActiveTracksGeneration) { 2737 // update wakelock 2738 updateWakeLockUids_l(mWakeLockUids); 2739 lastGeneration = mActiveTracksGeneration; 2740 } 2741 2742 // prevent any changes in effect chain list and in each effect chain 2743 // during mixing and effect process as the audio buffers could be deleted 2744 // or modified if an effect is created or deleted 2745 lockEffectChains_l(effectChains); 2746 } // mLock scope ends 2747 2748 if (mBytesRemaining == 0) { 2749 mCurrentWriteLength = 0; 2750 if (mMixerStatus == MIXER_TRACKS_READY) { 2751 // threadLoop_mix() sets mCurrentWriteLength 2752 threadLoop_mix(); 2753 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2754 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2755 // threadLoop_sleepTime sets sleepTime to 0 if data 2756 // must be written to HAL 2757 threadLoop_sleepTime(); 2758 if (sleepTime == 0) { 2759 mCurrentWriteLength = mSinkBufferSize; 2760 } 2761 } 2762 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2763 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2764 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2765 // or mSinkBuffer (if there are no effects). 2766 // 2767 // This is done pre-effects computation; if effects change to 2768 // support higher precision, this needs to move. 2769 // 2770 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2771 // TODO use sleepTime == 0 as an additional condition. 2772 if (mMixerBufferValid) { 2773 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2774 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2775 2776 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2777 mNormalFrameCount * mChannelCount); 2778 } 2779 2780 mBytesRemaining = mCurrentWriteLength; 2781 if (isSuspended()) { 2782 sleepTime = suspendSleepTimeUs(); 2783 // simulate write to HAL when suspended 2784 mBytesWritten += mSinkBufferSize; 2785 mBytesRemaining = 0; 2786 } 2787 2788 // only process effects if we're going to write 2789 if (sleepTime == 0 && mType != OFFLOAD) { 2790 for (size_t i = 0; i < effectChains.size(); i ++) { 2791 effectChains[i]->process_l(); 2792 } 2793 } 2794 } 2795 // Process effect chains for offloaded thread even if no audio 2796 // was read from audio track: process only updates effect state 2797 // and thus does have to be synchronized with audio writes but may have 2798 // to be called while waiting for async write callback 2799 if (mType == OFFLOAD) { 2800 for (size_t i = 0; i < effectChains.size(); i ++) { 2801 effectChains[i]->process_l(); 2802 } 2803 } 2804 2805 // Only if the Effects buffer is enabled and there is data in the 2806 // Effects buffer (buffer valid), we need to 2807 // copy into the sink buffer. 2808 // TODO use sleepTime == 0 as an additional condition. 2809 if (mEffectBufferValid) { 2810 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2811 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2812 mNormalFrameCount * mChannelCount); 2813 } 2814 2815 // enable changes in effect chain 2816 unlockEffectChains(effectChains); 2817 2818 if (!waitingAsyncCallback()) { 2819 // sleepTime == 0 means we must write to audio hardware 2820 if (sleepTime == 0) { 2821 if (mBytesRemaining) { 2822 ssize_t ret = threadLoop_write(); 2823 if (ret < 0) { 2824 mBytesRemaining = 0; 2825 } else { 2826 mBytesWritten += ret; 2827 mBytesRemaining -= ret; 2828 } 2829 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2830 (mMixerStatus == MIXER_DRAIN_ALL)) { 2831 threadLoop_drain(); 2832 } 2833 if (mType == MIXER) { 2834 // write blocked detection 2835 nsecs_t now = systemTime(); 2836 nsecs_t delta = now - mLastWriteTime; 2837 if (!mStandby && delta > maxPeriod) { 2838 mNumDelayedWrites++; 2839 if ((now - lastWarning) > kWarningThrottleNs) { 2840 ATRACE_NAME("underrun"); 2841 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2842 ns2ms(delta), mNumDelayedWrites, this); 2843 lastWarning = now; 2844 } 2845 } 2846 } 2847 2848 } else { 2849 ATRACE_BEGIN("sleep"); 2850 usleep(sleepTime); 2851 ATRACE_END(); 2852 } 2853 } 2854 2855 // Finally let go of removed track(s), without the lock held 2856 // since we can't guarantee the destructors won't acquire that 2857 // same lock. This will also mutate and push a new fast mixer state. 2858 threadLoop_removeTracks(tracksToRemove); 2859 tracksToRemove.clear(); 2860 2861 // FIXME I don't understand the need for this here; 2862 // it was in the original code but maybe the 2863 // assignment in saveOutputTracks() makes this unnecessary? 2864 clearOutputTracks(); 2865 2866 // Effect chains will be actually deleted here if they were removed from 2867 // mEffectChains list during mixing or effects processing 2868 effectChains.clear(); 2869 2870 // FIXME Note that the above .clear() is no longer necessary since effectChains 2871 // is now local to this block, but will keep it for now (at least until merge done). 2872 } 2873 2874 threadLoop_exit(); 2875 2876 if (!mStandby) { 2877 threadLoop_standby(); 2878 mStandby = true; 2879 } 2880 2881 releaseWakeLock(); 2882 mWakeLockUids.clear(); 2883 mActiveTracksGeneration++; 2884 2885 ALOGV("Thread %p type %d exiting", this, mType); 2886 return false; 2887} 2888 2889// removeTracks_l() must be called with ThreadBase::mLock held 2890void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2891{ 2892 size_t count = tracksToRemove.size(); 2893 if (count > 0) { 2894 for (size_t i=0 ; i<count ; i++) { 2895 const sp<Track>& track = tracksToRemove.itemAt(i); 2896 mActiveTracks.remove(track); 2897 mWakeLockUids.remove(track->uid()); 2898 mActiveTracksGeneration++; 2899 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2900 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2901 if (chain != 0) { 2902 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2903 track->sessionId()); 2904 chain->decActiveTrackCnt(); 2905 } 2906 if (track->isTerminated()) { 2907 removeTrack_l(track); 2908 } 2909 } 2910 } 2911 2912} 2913 2914status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2915{ 2916 if (mNormalSink != 0) { 2917 return mNormalSink->getTimestamp(timestamp); 2918 } 2919 if ((mType == OFFLOAD || mType == DIRECT) 2920 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2921 uint64_t position64; 2922 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 2923 if (ret == 0) { 2924 timestamp.mPosition = (uint32_t)position64; 2925 return NO_ERROR; 2926 } 2927 } 2928 return INVALID_OPERATION; 2929} 2930 2931status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2932 audio_patch_handle_t *handle) 2933{ 2934 status_t status = NO_ERROR; 2935 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2936 // store new device and send to effects 2937 audio_devices_t type = AUDIO_DEVICE_NONE; 2938 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2939 type |= patch->sinks[i].ext.device.type; 2940 } 2941 mOutDevice = type; 2942 for (size_t i = 0; i < mEffectChains.size(); i++) { 2943 mEffectChains[i]->setDevice_l(mOutDevice); 2944 } 2945 2946 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2947 status = hwDevice->create_audio_patch(hwDevice, 2948 patch->num_sources, 2949 patch->sources, 2950 patch->num_sinks, 2951 patch->sinks, 2952 handle); 2953 } else { 2954 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2955 } 2956 return status; 2957} 2958 2959status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2960{ 2961 status_t status = NO_ERROR; 2962 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2963 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2964 status = hwDevice->release_audio_patch(hwDevice, handle); 2965 } else { 2966 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2967 } 2968 return status; 2969} 2970 2971void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2972{ 2973 Mutex::Autolock _l(mLock); 2974 mTracks.add(track); 2975} 2976 2977void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2978{ 2979 Mutex::Autolock _l(mLock); 2980 destroyTrack_l(track); 2981} 2982 2983void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2984{ 2985 ThreadBase::getAudioPortConfig(config); 2986 config->role = AUDIO_PORT_ROLE_SOURCE; 2987 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2988 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2989} 2990 2991// ---------------------------------------------------------------------------- 2992 2993AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2994 audio_io_handle_t id, audio_devices_t device, type_t type) 2995 : PlaybackThread(audioFlinger, output, id, device, type), 2996 // mAudioMixer below 2997 // mFastMixer below 2998 mFastMixerFutex(0) 2999 // mOutputSink below 3000 // mPipeSink below 3001 // mNormalSink below 3002{ 3003 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3004 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3005 "mFrameCount=%d, mNormalFrameCount=%d", 3006 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3007 mNormalFrameCount); 3008 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3009 3010 if (type == DUPLICATING) { 3011 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3012 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3013 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3014 return; 3015 } 3016 // create an NBAIO sink for the HAL output stream, and negotiate 3017 mOutputSink = new AudioStreamOutSink(output->stream); 3018 size_t numCounterOffers = 0; 3019 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3020 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3021 ALOG_ASSERT(index == 0); 3022 3023 // initialize fast mixer depending on configuration 3024 bool initFastMixer; 3025 switch (kUseFastMixer) { 3026 case FastMixer_Never: 3027 initFastMixer = false; 3028 break; 3029 case FastMixer_Always: 3030 initFastMixer = true; 3031 break; 3032 case FastMixer_Static: 3033 case FastMixer_Dynamic: 3034 initFastMixer = mFrameCount < mNormalFrameCount; 3035 break; 3036 } 3037 if (initFastMixer) { 3038 audio_format_t fastMixerFormat; 3039 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3040 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3041 } else { 3042 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3043 } 3044 if (mFormat != fastMixerFormat) { 3045 // change our Sink format to accept our intermediate precision 3046 mFormat = fastMixerFormat; 3047 free(mSinkBuffer); 3048 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3049 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3050 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3051 } 3052 3053 // create a MonoPipe to connect our submix to FastMixer 3054 NBAIO_Format format = mOutputSink->format(); 3055 NBAIO_Format origformat = format; 3056 // adjust format to match that of the Fast Mixer 3057 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3058 format.mFormat = fastMixerFormat; 3059 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3060 3061 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3062 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3063 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3064 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3065 const NBAIO_Format offers[1] = {format}; 3066 size_t numCounterOffers = 0; 3067 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3068 ALOG_ASSERT(index == 0); 3069 monoPipe->setAvgFrames((mScreenState & 1) ? 3070 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3071 mPipeSink = monoPipe; 3072 3073#ifdef TEE_SINK 3074 if (mTeeSinkOutputEnabled) { 3075 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3076 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3077 const NBAIO_Format offers2[1] = {origformat}; 3078 numCounterOffers = 0; 3079 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3080 ALOG_ASSERT(index == 0); 3081 mTeeSink = teeSink; 3082 PipeReader *teeSource = new PipeReader(*teeSink); 3083 numCounterOffers = 0; 3084 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3085 ALOG_ASSERT(index == 0); 3086 mTeeSource = teeSource; 3087 } 3088#endif 3089 3090 // create fast mixer and configure it initially with just one fast track for our submix 3091 mFastMixer = new FastMixer(); 3092 FastMixerStateQueue *sq = mFastMixer->sq(); 3093#ifdef STATE_QUEUE_DUMP 3094 sq->setObserverDump(&mStateQueueObserverDump); 3095 sq->setMutatorDump(&mStateQueueMutatorDump); 3096#endif 3097 FastMixerState *state = sq->begin(); 3098 FastTrack *fastTrack = &state->mFastTracks[0]; 3099 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3100 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3101 fastTrack->mVolumeProvider = NULL; 3102 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3103 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3104 fastTrack->mGeneration++; 3105 state->mFastTracksGen++; 3106 state->mTrackMask = 1; 3107 // fast mixer will use the HAL output sink 3108 state->mOutputSink = mOutputSink.get(); 3109 state->mOutputSinkGen++; 3110 state->mFrameCount = mFrameCount; 3111 state->mCommand = FastMixerState::COLD_IDLE; 3112 // already done in constructor initialization list 3113 //mFastMixerFutex = 0; 3114 state->mColdFutexAddr = &mFastMixerFutex; 3115 state->mColdGen++; 3116 state->mDumpState = &mFastMixerDumpState; 3117#ifdef TEE_SINK 3118 state->mTeeSink = mTeeSink.get(); 3119#endif 3120 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3121 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3122 sq->end(); 3123 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3124 3125 // start the fast mixer 3126 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3127 pid_t tid = mFastMixer->getTid(); 3128 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3129 if (err != 0) { 3130 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3131 kPriorityFastMixer, getpid_cached, tid, err); 3132 } 3133 3134#ifdef AUDIO_WATCHDOG 3135 // create and start the watchdog 3136 mAudioWatchdog = new AudioWatchdog(); 3137 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3138 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3139 tid = mAudioWatchdog->getTid(); 3140 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3141 if (err != 0) { 3142 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3143 kPriorityFastMixer, getpid_cached, tid, err); 3144 } 3145#endif 3146 3147 } 3148 3149 switch (kUseFastMixer) { 3150 case FastMixer_Never: 3151 case FastMixer_Dynamic: 3152 mNormalSink = mOutputSink; 3153 break; 3154 case FastMixer_Always: 3155 mNormalSink = mPipeSink; 3156 break; 3157 case FastMixer_Static: 3158 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3159 break; 3160 } 3161} 3162 3163AudioFlinger::MixerThread::~MixerThread() 3164{ 3165 if (mFastMixer != 0) { 3166 FastMixerStateQueue *sq = mFastMixer->sq(); 3167 FastMixerState *state = sq->begin(); 3168 if (state->mCommand == FastMixerState::COLD_IDLE) { 3169 int32_t old = android_atomic_inc(&mFastMixerFutex); 3170 if (old == -1) { 3171 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3172 } 3173 } 3174 state->mCommand = FastMixerState::EXIT; 3175 sq->end(); 3176 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3177 mFastMixer->join(); 3178 // Though the fast mixer thread has exited, it's state queue is still valid. 3179 // We'll use that extract the final state which contains one remaining fast track 3180 // corresponding to our sub-mix. 3181 state = sq->begin(); 3182 ALOG_ASSERT(state->mTrackMask == 1); 3183 FastTrack *fastTrack = &state->mFastTracks[0]; 3184 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3185 delete fastTrack->mBufferProvider; 3186 sq->end(false /*didModify*/); 3187 mFastMixer.clear(); 3188#ifdef AUDIO_WATCHDOG 3189 if (mAudioWatchdog != 0) { 3190 mAudioWatchdog->requestExit(); 3191 mAudioWatchdog->requestExitAndWait(); 3192 mAudioWatchdog.clear(); 3193 } 3194#endif 3195 } 3196 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3197 delete mAudioMixer; 3198} 3199 3200 3201uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3202{ 3203 if (mFastMixer != 0) { 3204 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3205 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3206 } 3207 return latency; 3208} 3209 3210 3211void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3212{ 3213 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3214} 3215 3216ssize_t AudioFlinger::MixerThread::threadLoop_write() 3217{ 3218 // FIXME we should only do one push per cycle; confirm this is true 3219 // Start the fast mixer if it's not already running 3220 if (mFastMixer != 0) { 3221 FastMixerStateQueue *sq = mFastMixer->sq(); 3222 FastMixerState *state = sq->begin(); 3223 if (state->mCommand != FastMixerState::MIX_WRITE && 3224 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3225 if (state->mCommand == FastMixerState::COLD_IDLE) { 3226 int32_t old = android_atomic_inc(&mFastMixerFutex); 3227 if (old == -1) { 3228 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3229 } 3230#ifdef AUDIO_WATCHDOG 3231 if (mAudioWatchdog != 0) { 3232 mAudioWatchdog->resume(); 3233 } 3234#endif 3235 } 3236 state->mCommand = FastMixerState::MIX_WRITE; 3237#ifdef FAST_THREAD_STATISTICS 3238 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3239 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3240#endif 3241 sq->end(); 3242 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3243 if (kUseFastMixer == FastMixer_Dynamic) { 3244 mNormalSink = mPipeSink; 3245 } 3246 } else { 3247 sq->end(false /*didModify*/); 3248 } 3249 } 3250 return PlaybackThread::threadLoop_write(); 3251} 3252 3253void AudioFlinger::MixerThread::threadLoop_standby() 3254{ 3255 // Idle the fast mixer if it's currently running 3256 if (mFastMixer != 0) { 3257 FastMixerStateQueue *sq = mFastMixer->sq(); 3258 FastMixerState *state = sq->begin(); 3259 if (!(state->mCommand & FastMixerState::IDLE)) { 3260 state->mCommand = FastMixerState::COLD_IDLE; 3261 state->mColdFutexAddr = &mFastMixerFutex; 3262 state->mColdGen++; 3263 mFastMixerFutex = 0; 3264 sq->end(); 3265 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3266 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3267 if (kUseFastMixer == FastMixer_Dynamic) { 3268 mNormalSink = mOutputSink; 3269 } 3270#ifdef AUDIO_WATCHDOG 3271 if (mAudioWatchdog != 0) { 3272 mAudioWatchdog->pause(); 3273 } 3274#endif 3275 } else { 3276 sq->end(false /*didModify*/); 3277 } 3278 } 3279 PlaybackThread::threadLoop_standby(); 3280} 3281 3282bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3283{ 3284 return false; 3285} 3286 3287bool AudioFlinger::PlaybackThread::shouldStandby_l() 3288{ 3289 return !mStandby; 3290} 3291 3292bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3293{ 3294 Mutex::Autolock _l(mLock); 3295 return waitingAsyncCallback_l(); 3296} 3297 3298// shared by MIXER and DIRECT, overridden by DUPLICATING 3299void AudioFlinger::PlaybackThread::threadLoop_standby() 3300{ 3301 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3302 mOutput->standby(); 3303 if (mUseAsyncWrite != 0) { 3304 // discard any pending drain or write ack by incrementing sequence 3305 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3306 mDrainSequence = (mDrainSequence + 2) & ~1; 3307 ALOG_ASSERT(mCallbackThread != 0); 3308 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3309 mCallbackThread->setDraining(mDrainSequence); 3310 } 3311 mHwPaused = false; 3312} 3313 3314void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3315{ 3316 ALOGV("signal playback thread"); 3317 broadcast_l(); 3318} 3319 3320void AudioFlinger::MixerThread::threadLoop_mix() 3321{ 3322 // obtain the presentation timestamp of the next output buffer 3323 int64_t pts; 3324 status_t status = INVALID_OPERATION; 3325 3326 if (mNormalSink != 0) { 3327 status = mNormalSink->getNextWriteTimestamp(&pts); 3328 } else { 3329 status = mOutputSink->getNextWriteTimestamp(&pts); 3330 } 3331 3332 if (status != NO_ERROR) { 3333 pts = AudioBufferProvider::kInvalidPTS; 3334 } 3335 3336 // mix buffers... 3337 mAudioMixer->process(pts); 3338 mCurrentWriteLength = mSinkBufferSize; 3339 // increase sleep time progressively when application underrun condition clears. 3340 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3341 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3342 // such that we would underrun the audio HAL. 3343 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3344 sleepTimeShift--; 3345 } 3346 sleepTime = 0; 3347 standbyTime = systemTime() + standbyDelay; 3348 //TODO: delay standby when effects have a tail 3349 3350} 3351 3352void AudioFlinger::MixerThread::threadLoop_sleepTime() 3353{ 3354 // If no tracks are ready, sleep once for the duration of an output 3355 // buffer size, then write 0s to the output 3356 if (sleepTime == 0) { 3357 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3358 sleepTime = activeSleepTime >> sleepTimeShift; 3359 if (sleepTime < kMinThreadSleepTimeUs) { 3360 sleepTime = kMinThreadSleepTimeUs; 3361 } 3362 // reduce sleep time in case of consecutive application underruns to avoid 3363 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3364 // duration we would end up writing less data than needed by the audio HAL if 3365 // the condition persists. 3366 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3367 sleepTimeShift++; 3368 } 3369 } else { 3370 sleepTime = idleSleepTime; 3371 } 3372 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3373 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3374 // before effects processing or output. 3375 if (mMixerBufferValid) { 3376 memset(mMixerBuffer, 0, mMixerBufferSize); 3377 } else { 3378 memset(mSinkBuffer, 0, mSinkBufferSize); 3379 } 3380 sleepTime = 0; 3381 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3382 "anticipated start"); 3383 } 3384 // TODO add standby time extension fct of effect tail 3385} 3386 3387// prepareTracks_l() must be called with ThreadBase::mLock held 3388AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3389 Vector< sp<Track> > *tracksToRemove) 3390{ 3391 3392 mixer_state mixerStatus = MIXER_IDLE; 3393 // find out which tracks need to be processed 3394 size_t count = mActiveTracks.size(); 3395 size_t mixedTracks = 0; 3396 size_t tracksWithEffect = 0; 3397 // counts only _active_ fast tracks 3398 size_t fastTracks = 0; 3399 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3400 3401 float masterVolume = mMasterVolume; 3402 bool masterMute = mMasterMute; 3403 3404 if (masterMute) { 3405 masterVolume = 0; 3406 } 3407 // Delegate master volume control to effect in output mix effect chain if needed 3408 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3409 if (chain != 0) { 3410 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3411 chain->setVolume_l(&v, &v); 3412 masterVolume = (float)((v + (1 << 23)) >> 24); 3413 chain.clear(); 3414 } 3415 3416 // prepare a new state to push 3417 FastMixerStateQueue *sq = NULL; 3418 FastMixerState *state = NULL; 3419 bool didModify = false; 3420 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3421 if (mFastMixer != 0) { 3422 sq = mFastMixer->sq(); 3423 state = sq->begin(); 3424 } 3425 3426 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3427 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3428 3429 for (size_t i=0 ; i<count ; i++) { 3430 const sp<Track> t = mActiveTracks[i].promote(); 3431 if (t == 0) { 3432 continue; 3433 } 3434 3435 // this const just means the local variable doesn't change 3436 Track* const track = t.get(); 3437 3438 // process fast tracks 3439 if (track->isFastTrack()) { 3440 3441 // It's theoretically possible (though unlikely) for a fast track to be created 3442 // and then removed within the same normal mix cycle. This is not a problem, as 3443 // the track never becomes active so it's fast mixer slot is never touched. 3444 // The converse, of removing an (active) track and then creating a new track 3445 // at the identical fast mixer slot within the same normal mix cycle, 3446 // is impossible because the slot isn't marked available until the end of each cycle. 3447 int j = track->mFastIndex; 3448 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3449 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3450 FastTrack *fastTrack = &state->mFastTracks[j]; 3451 3452 // Determine whether the track is currently in underrun condition, 3453 // and whether it had a recent underrun. 3454 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3455 FastTrackUnderruns underruns = ftDump->mUnderruns; 3456 uint32_t recentFull = (underruns.mBitFields.mFull - 3457 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3458 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3459 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3460 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3461 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3462 uint32_t recentUnderruns = recentPartial + recentEmpty; 3463 track->mObservedUnderruns = underruns; 3464 // don't count underruns that occur while stopping or pausing 3465 // or stopped which can occur when flush() is called while active 3466 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3467 recentUnderruns > 0) { 3468 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3469 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3470 } 3471 3472 // This is similar to the state machine for normal tracks, 3473 // with a few modifications for fast tracks. 3474 bool isActive = true; 3475 switch (track->mState) { 3476 case TrackBase::STOPPING_1: 3477 // track stays active in STOPPING_1 state until first underrun 3478 if (recentUnderruns > 0 || track->isTerminated()) { 3479 track->mState = TrackBase::STOPPING_2; 3480 } 3481 break; 3482 case TrackBase::PAUSING: 3483 // ramp down is not yet implemented 3484 track->setPaused(); 3485 break; 3486 case TrackBase::RESUMING: 3487 // ramp up is not yet implemented 3488 track->mState = TrackBase::ACTIVE; 3489 break; 3490 case TrackBase::ACTIVE: 3491 if (recentFull > 0 || recentPartial > 0) { 3492 // track has provided at least some frames recently: reset retry count 3493 track->mRetryCount = kMaxTrackRetries; 3494 } 3495 if (recentUnderruns == 0) { 3496 // no recent underruns: stay active 3497 break; 3498 } 3499 // there has recently been an underrun of some kind 3500 if (track->sharedBuffer() == 0) { 3501 // were any of the recent underruns "empty" (no frames available)? 3502 if (recentEmpty == 0) { 3503 // no, then ignore the partial underruns as they are allowed indefinitely 3504 break; 3505 } 3506 // there has recently been an "empty" underrun: decrement the retry counter 3507 if (--(track->mRetryCount) > 0) { 3508 break; 3509 } 3510 // indicate to client process that the track was disabled because of underrun; 3511 // it will then automatically call start() when data is available 3512 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3513 // remove from active list, but state remains ACTIVE [confusing but true] 3514 isActive = false; 3515 break; 3516 } 3517 // fall through 3518 case TrackBase::STOPPING_2: 3519 case TrackBase::PAUSED: 3520 case TrackBase::STOPPED: 3521 case TrackBase::FLUSHED: // flush() while active 3522 // Check for presentation complete if track is inactive 3523 // We have consumed all the buffers of this track. 3524 // This would be incomplete if we auto-paused on underrun 3525 { 3526 size_t audioHALFrames = 3527 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3528 size_t framesWritten = mBytesWritten / mFrameSize; 3529 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3530 // track stays in active list until presentation is complete 3531 break; 3532 } 3533 } 3534 if (track->isStopping_2()) { 3535 track->mState = TrackBase::STOPPED; 3536 } 3537 if (track->isStopped()) { 3538 // Can't reset directly, as fast mixer is still polling this track 3539 // track->reset(); 3540 // So instead mark this track as needing to be reset after push with ack 3541 resetMask |= 1 << i; 3542 } 3543 isActive = false; 3544 break; 3545 case TrackBase::IDLE: 3546 default: 3547 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3548 } 3549 3550 if (isActive) { 3551 // was it previously inactive? 3552 if (!(state->mTrackMask & (1 << j))) { 3553 ExtendedAudioBufferProvider *eabp = track; 3554 VolumeProvider *vp = track; 3555 fastTrack->mBufferProvider = eabp; 3556 fastTrack->mVolumeProvider = vp; 3557 fastTrack->mChannelMask = track->mChannelMask; 3558 fastTrack->mFormat = track->mFormat; 3559 fastTrack->mGeneration++; 3560 state->mTrackMask |= 1 << j; 3561 didModify = true; 3562 // no acknowledgement required for newly active tracks 3563 } 3564 // cache the combined master volume and stream type volume for fast mixer; this 3565 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3566 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3567 ++fastTracks; 3568 } else { 3569 // was it previously active? 3570 if (state->mTrackMask & (1 << j)) { 3571 fastTrack->mBufferProvider = NULL; 3572 fastTrack->mGeneration++; 3573 state->mTrackMask &= ~(1 << j); 3574 didModify = true; 3575 // If any fast tracks were removed, we must wait for acknowledgement 3576 // because we're about to decrement the last sp<> on those tracks. 3577 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3578 } else { 3579 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3580 } 3581 tracksToRemove->add(track); 3582 // Avoids a misleading display in dumpsys 3583 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3584 } 3585 continue; 3586 } 3587 3588 { // local variable scope to avoid goto warning 3589 3590 audio_track_cblk_t* cblk = track->cblk(); 3591 3592 // The first time a track is added we wait 3593 // for all its buffers to be filled before processing it 3594 int name = track->name(); 3595 // make sure that we have enough frames to mix one full buffer. 3596 // enforce this condition only once to enable draining the buffer in case the client 3597 // app does not call stop() and relies on underrun to stop: 3598 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3599 // during last round 3600 size_t desiredFrames; 3601 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3602 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3603 3604 desiredFrames = sourceFramesNeededWithTimestretch( 3605 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3606 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3607 // add frames already consumed but not yet released by the resampler 3608 // because mAudioTrackServerProxy->framesReady() will include these frames 3609 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3610 3611 uint32_t minFrames = 1; 3612 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3613 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3614 minFrames = desiredFrames; 3615 } 3616 3617 size_t framesReady = track->framesReady(); 3618 if (ATRACE_ENABLED()) { 3619 // I wish we had formatted trace names 3620 char traceName[16]; 3621 strcpy(traceName, "nRdy"); 3622 int name = track->name(); 3623 if (AudioMixer::TRACK0 <= name && 3624 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3625 name -= AudioMixer::TRACK0; 3626 traceName[4] = (name / 10) + '0'; 3627 traceName[5] = (name % 10) + '0'; 3628 } else { 3629 traceName[4] = '?'; 3630 traceName[5] = '?'; 3631 } 3632 traceName[6] = '\0'; 3633 ATRACE_INT(traceName, framesReady); 3634 } 3635 if ((framesReady >= minFrames) && track->isReady() && 3636 !track->isPaused() && !track->isTerminated()) 3637 { 3638 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3639 3640 mixedTracks++; 3641 3642 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3643 // there is an effect chain connected to the track 3644 chain.clear(); 3645 if (track->mainBuffer() != mSinkBuffer && 3646 track->mainBuffer() != mMixerBuffer) { 3647 if (mEffectBufferEnabled) { 3648 mEffectBufferValid = true; // Later can set directly. 3649 } 3650 chain = getEffectChain_l(track->sessionId()); 3651 // Delegate volume control to effect in track effect chain if needed 3652 if (chain != 0) { 3653 tracksWithEffect++; 3654 } else { 3655 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3656 "session %d", 3657 name, track->sessionId()); 3658 } 3659 } 3660 3661 3662 int param = AudioMixer::VOLUME; 3663 if (track->mFillingUpStatus == Track::FS_FILLED) { 3664 // no ramp for the first volume setting 3665 track->mFillingUpStatus = Track::FS_ACTIVE; 3666 if (track->mState == TrackBase::RESUMING) { 3667 track->mState = TrackBase::ACTIVE; 3668 param = AudioMixer::RAMP_VOLUME; 3669 } 3670 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3671 // FIXME should not make a decision based on mServer 3672 } else if (cblk->mServer != 0) { 3673 // If the track is stopped before the first frame was mixed, 3674 // do not apply ramp 3675 param = AudioMixer::RAMP_VOLUME; 3676 } 3677 3678 // compute volume for this track 3679 uint32_t vl, vr; // in U8.24 integer format 3680 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3681 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3682 vl = vr = 0; 3683 vlf = vrf = vaf = 0.; 3684 if (track->isPausing()) { 3685 track->setPaused(); 3686 } 3687 } else { 3688 3689 // read original volumes with volume control 3690 float typeVolume = mStreamTypes[track->streamType()].volume; 3691 float v = masterVolume * typeVolume; 3692 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3693 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3694 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3695 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3696 // track volumes come from shared memory, so can't be trusted and must be clamped 3697 if (vlf > GAIN_FLOAT_UNITY) { 3698 ALOGV("Track left volume out of range: %.3g", vlf); 3699 vlf = GAIN_FLOAT_UNITY; 3700 } 3701 if (vrf > GAIN_FLOAT_UNITY) { 3702 ALOGV("Track right volume out of range: %.3g", vrf); 3703 vrf = GAIN_FLOAT_UNITY; 3704 } 3705 // now apply the master volume and stream type volume 3706 vlf *= v; 3707 vrf *= v; 3708 // assuming master volume and stream type volume each go up to 1.0, 3709 // then derive vl and vr as U8.24 versions for the effect chain 3710 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3711 vl = (uint32_t) (scaleto8_24 * vlf); 3712 vr = (uint32_t) (scaleto8_24 * vrf); 3713 // vl and vr are now in U8.24 format 3714 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3715 // send level comes from shared memory and so may be corrupt 3716 if (sendLevel > MAX_GAIN_INT) { 3717 ALOGV("Track send level out of range: %04X", sendLevel); 3718 sendLevel = MAX_GAIN_INT; 3719 } 3720 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3721 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3722 } 3723 3724 // Delegate volume control to effect in track effect chain if needed 3725 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3726 // Do not ramp volume if volume is controlled by effect 3727 param = AudioMixer::VOLUME; 3728 // Update remaining floating point volume levels 3729 vlf = (float)vl / (1 << 24); 3730 vrf = (float)vr / (1 << 24); 3731 track->mHasVolumeController = true; 3732 } else { 3733 // force no volume ramp when volume controller was just disabled or removed 3734 // from effect chain to avoid volume spike 3735 if (track->mHasVolumeController) { 3736 param = AudioMixer::VOLUME; 3737 } 3738 track->mHasVolumeController = false; 3739 } 3740 3741 // XXX: these things DON'T need to be done each time 3742 mAudioMixer->setBufferProvider(name, track); 3743 mAudioMixer->enable(name); 3744 3745 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3746 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3747 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3748 mAudioMixer->setParameter( 3749 name, 3750 AudioMixer::TRACK, 3751 AudioMixer::FORMAT, (void *)track->format()); 3752 mAudioMixer->setParameter( 3753 name, 3754 AudioMixer::TRACK, 3755 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3756 mAudioMixer->setParameter( 3757 name, 3758 AudioMixer::TRACK, 3759 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3760 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3761 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3762 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3763 if (reqSampleRate == 0) { 3764 reqSampleRate = mSampleRate; 3765 } else if (reqSampleRate > maxSampleRate) { 3766 reqSampleRate = maxSampleRate; 3767 } 3768 mAudioMixer->setParameter( 3769 name, 3770 AudioMixer::RESAMPLE, 3771 AudioMixer::SAMPLE_RATE, 3772 (void *)(uintptr_t)reqSampleRate); 3773 3774 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3775 mAudioMixer->setParameter( 3776 name, 3777 AudioMixer::TIMESTRETCH, 3778 AudioMixer::PLAYBACK_RATE, 3779 &playbackRate); 3780 3781 /* 3782 * Select the appropriate output buffer for the track. 3783 * 3784 * Tracks with effects go into their own effects chain buffer 3785 * and from there into either mEffectBuffer or mSinkBuffer. 3786 * 3787 * Other tracks can use mMixerBuffer for higher precision 3788 * channel accumulation. If this buffer is enabled 3789 * (mMixerBufferEnabled true), then selected tracks will accumulate 3790 * into it. 3791 * 3792 */ 3793 if (mMixerBufferEnabled 3794 && (track->mainBuffer() == mSinkBuffer 3795 || track->mainBuffer() == mMixerBuffer)) { 3796 mAudioMixer->setParameter( 3797 name, 3798 AudioMixer::TRACK, 3799 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3800 mAudioMixer->setParameter( 3801 name, 3802 AudioMixer::TRACK, 3803 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3804 // TODO: override track->mainBuffer()? 3805 mMixerBufferValid = true; 3806 } else { 3807 mAudioMixer->setParameter( 3808 name, 3809 AudioMixer::TRACK, 3810 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3811 mAudioMixer->setParameter( 3812 name, 3813 AudioMixer::TRACK, 3814 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3815 } 3816 mAudioMixer->setParameter( 3817 name, 3818 AudioMixer::TRACK, 3819 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3820 3821 // reset retry count 3822 track->mRetryCount = kMaxTrackRetries; 3823 3824 // If one track is ready, set the mixer ready if: 3825 // - the mixer was not ready during previous round OR 3826 // - no other track is not ready 3827 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3828 mixerStatus != MIXER_TRACKS_ENABLED) { 3829 mixerStatus = MIXER_TRACKS_READY; 3830 } 3831 } else { 3832 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3833 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3834 } 3835 // clear effect chain input buffer if an active track underruns to avoid sending 3836 // previous audio buffer again to effects 3837 chain = getEffectChain_l(track->sessionId()); 3838 if (chain != 0) { 3839 chain->clearInputBuffer(); 3840 } 3841 3842 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3843 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3844 track->isStopped() || track->isPaused()) { 3845 // We have consumed all the buffers of this track. 3846 // Remove it from the list of active tracks. 3847 // TODO: use actual buffer filling status instead of latency when available from 3848 // audio HAL 3849 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3850 size_t framesWritten = mBytesWritten / mFrameSize; 3851 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3852 if (track->isStopped()) { 3853 track->reset(); 3854 } 3855 tracksToRemove->add(track); 3856 } 3857 } else { 3858 // No buffers for this track. Give it a few chances to 3859 // fill a buffer, then remove it from active list. 3860 if (--(track->mRetryCount) <= 0) { 3861 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3862 tracksToRemove->add(track); 3863 // indicate to client process that the track was disabled because of underrun; 3864 // it will then automatically call start() when data is available 3865 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3866 // If one track is not ready, mark the mixer also not ready if: 3867 // - the mixer was ready during previous round OR 3868 // - no other track is ready 3869 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3870 mixerStatus != MIXER_TRACKS_READY) { 3871 mixerStatus = MIXER_TRACKS_ENABLED; 3872 } 3873 } 3874 mAudioMixer->disable(name); 3875 } 3876 3877 } // local variable scope to avoid goto warning 3878track_is_ready: ; 3879 3880 } 3881 3882 // Push the new FastMixer state if necessary 3883 bool pauseAudioWatchdog = false; 3884 if (didModify) { 3885 state->mFastTracksGen++; 3886 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3887 if (kUseFastMixer == FastMixer_Dynamic && 3888 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3889 state->mCommand = FastMixerState::COLD_IDLE; 3890 state->mColdFutexAddr = &mFastMixerFutex; 3891 state->mColdGen++; 3892 mFastMixerFutex = 0; 3893 if (kUseFastMixer == FastMixer_Dynamic) { 3894 mNormalSink = mOutputSink; 3895 } 3896 // If we go into cold idle, need to wait for acknowledgement 3897 // so that fast mixer stops doing I/O. 3898 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3899 pauseAudioWatchdog = true; 3900 } 3901 } 3902 if (sq != NULL) { 3903 sq->end(didModify); 3904 sq->push(block); 3905 } 3906#ifdef AUDIO_WATCHDOG 3907 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3908 mAudioWatchdog->pause(); 3909 } 3910#endif 3911 3912 // Now perform the deferred reset on fast tracks that have stopped 3913 while (resetMask != 0) { 3914 size_t i = __builtin_ctz(resetMask); 3915 ALOG_ASSERT(i < count); 3916 resetMask &= ~(1 << i); 3917 sp<Track> t = mActiveTracks[i].promote(); 3918 if (t == 0) { 3919 continue; 3920 } 3921 Track* track = t.get(); 3922 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3923 track->reset(); 3924 } 3925 3926 // remove all the tracks that need to be... 3927 removeTracks_l(*tracksToRemove); 3928 3929 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3930 mEffectBufferValid = true; 3931 } 3932 3933 if (mEffectBufferValid) { 3934 // as long as there are effects we should clear the effects buffer, to avoid 3935 // passing a non-clean buffer to the effect chain 3936 memset(mEffectBuffer, 0, mEffectBufferSize); 3937 } 3938 // sink or mix buffer must be cleared if all tracks are connected to an 3939 // effect chain as in this case the mixer will not write to the sink or mix buffer 3940 // and track effects will accumulate into it 3941 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3942 (mixedTracks == 0 && fastTracks > 0))) { 3943 // FIXME as a performance optimization, should remember previous zero status 3944 if (mMixerBufferValid) { 3945 memset(mMixerBuffer, 0, mMixerBufferSize); 3946 // TODO: In testing, mSinkBuffer below need not be cleared because 3947 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3948 // after mixing. 3949 // 3950 // To enforce this guarantee: 3951 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3952 // (mixedTracks == 0 && fastTracks > 0)) 3953 // must imply MIXER_TRACKS_READY. 3954 // Later, we may clear buffers regardless, and skip much of this logic. 3955 } 3956 // FIXME as a performance optimization, should remember previous zero status 3957 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3958 } 3959 3960 // if any fast tracks, then status is ready 3961 mMixerStatusIgnoringFastTracks = mixerStatus; 3962 if (fastTracks > 0) { 3963 mixerStatus = MIXER_TRACKS_READY; 3964 } 3965 return mixerStatus; 3966} 3967 3968// getTrackName_l() must be called with ThreadBase::mLock held 3969int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3970 audio_format_t format, int sessionId) 3971{ 3972 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3973} 3974 3975// deleteTrackName_l() must be called with ThreadBase::mLock held 3976void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3977{ 3978 ALOGV("remove track (%d) and delete from mixer", name); 3979 mAudioMixer->deleteTrackName(name); 3980} 3981 3982// checkForNewParameter_l() must be called with ThreadBase::mLock held 3983bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3984 status_t& status) 3985{ 3986 bool reconfig = false; 3987 3988 status = NO_ERROR; 3989 3990 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3991 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3992 if (mFastMixer != 0) { 3993 FastMixerStateQueue *sq = mFastMixer->sq(); 3994 FastMixerState *state = sq->begin(); 3995 if (!(state->mCommand & FastMixerState::IDLE)) { 3996 previousCommand = state->mCommand; 3997 state->mCommand = FastMixerState::HOT_IDLE; 3998 sq->end(); 3999 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4000 } else { 4001 sq->end(false /*didModify*/); 4002 } 4003 } 4004 4005 AudioParameter param = AudioParameter(keyValuePair); 4006 int value; 4007 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4008 reconfig = true; 4009 } 4010 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4011 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4012 status = BAD_VALUE; 4013 } else { 4014 // no need to save value, since it's constant 4015 reconfig = true; 4016 } 4017 } 4018 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4019 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4020 status = BAD_VALUE; 4021 } else { 4022 // no need to save value, since it's constant 4023 reconfig = true; 4024 } 4025 } 4026 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4027 // do not accept frame count changes if tracks are open as the track buffer 4028 // size depends on frame count and correct behavior would not be guaranteed 4029 // if frame count is changed after track creation 4030 if (!mTracks.isEmpty()) { 4031 status = INVALID_OPERATION; 4032 } else { 4033 reconfig = true; 4034 } 4035 } 4036 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4037#ifdef ADD_BATTERY_DATA 4038 // when changing the audio output device, call addBatteryData to notify 4039 // the change 4040 if (mOutDevice != value) { 4041 uint32_t params = 0; 4042 // check whether speaker is on 4043 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4044 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4045 } 4046 4047 audio_devices_t deviceWithoutSpeaker 4048 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4049 // check if any other device (except speaker) is on 4050 if (value & deviceWithoutSpeaker ) { 4051 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4052 } 4053 4054 if (params != 0) { 4055 addBatteryData(params); 4056 } 4057 } 4058#endif 4059 4060 // forward device change to effects that have requested to be 4061 // aware of attached audio device. 4062 if (value != AUDIO_DEVICE_NONE) { 4063 mOutDevice = value; 4064 for (size_t i = 0; i < mEffectChains.size(); i++) { 4065 mEffectChains[i]->setDevice_l(mOutDevice); 4066 } 4067 } 4068 } 4069 4070 if (status == NO_ERROR) { 4071 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4072 keyValuePair.string()); 4073 if (!mStandby && status == INVALID_OPERATION) { 4074 mOutput->standby(); 4075 mStandby = true; 4076 mBytesWritten = 0; 4077 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4078 keyValuePair.string()); 4079 } 4080 if (status == NO_ERROR && reconfig) { 4081 readOutputParameters_l(); 4082 delete mAudioMixer; 4083 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4084 for (size_t i = 0; i < mTracks.size() ; i++) { 4085 int name = getTrackName_l(mTracks[i]->mChannelMask, 4086 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4087 if (name < 0) { 4088 break; 4089 } 4090 mTracks[i]->mName = name; 4091 } 4092 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4093 } 4094 } 4095 4096 if (!(previousCommand & FastMixerState::IDLE)) { 4097 ALOG_ASSERT(mFastMixer != 0); 4098 FastMixerStateQueue *sq = mFastMixer->sq(); 4099 FastMixerState *state = sq->begin(); 4100 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4101 state->mCommand = previousCommand; 4102 sq->end(); 4103 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4104 } 4105 4106 return reconfig; 4107} 4108 4109 4110void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4111{ 4112 const size_t SIZE = 256; 4113 char buffer[SIZE]; 4114 String8 result; 4115 4116 PlaybackThread::dumpInternals(fd, args); 4117 4118 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4119 4120 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4121 const FastMixerDumpState copy(mFastMixerDumpState); 4122 copy.dump(fd); 4123 4124#ifdef STATE_QUEUE_DUMP 4125 // Similar for state queue 4126 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4127 observerCopy.dump(fd); 4128 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4129 mutatorCopy.dump(fd); 4130#endif 4131 4132#ifdef TEE_SINK 4133 // Write the tee output to a .wav file 4134 dumpTee(fd, mTeeSource, mId); 4135#endif 4136 4137#ifdef AUDIO_WATCHDOG 4138 if (mAudioWatchdog != 0) { 4139 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4140 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4141 wdCopy.dump(fd); 4142 } 4143#endif 4144} 4145 4146uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4147{ 4148 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4149} 4150 4151uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4152{ 4153 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4154} 4155 4156void AudioFlinger::MixerThread::cacheParameters_l() 4157{ 4158 PlaybackThread::cacheParameters_l(); 4159 4160 // FIXME: Relaxed timing because of a certain device that can't meet latency 4161 // Should be reduced to 2x after the vendor fixes the driver issue 4162 // increase threshold again due to low power audio mode. The way this warning 4163 // threshold is calculated and its usefulness should be reconsidered anyway. 4164 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4165} 4166 4167// ---------------------------------------------------------------------------- 4168 4169AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4170 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4171 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4172 // mLeftVolFloat, mRightVolFloat 4173{ 4174} 4175 4176AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4177 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4178 ThreadBase::type_t type) 4179 : PlaybackThread(audioFlinger, output, id, device, type) 4180 // mLeftVolFloat, mRightVolFloat 4181{ 4182} 4183 4184AudioFlinger::DirectOutputThread::~DirectOutputThread() 4185{ 4186} 4187 4188void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4189{ 4190 audio_track_cblk_t* cblk = track->cblk(); 4191 float left, right; 4192 4193 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4194 left = right = 0; 4195 } else { 4196 float typeVolume = mStreamTypes[track->streamType()].volume; 4197 float v = mMasterVolume * typeVolume; 4198 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4199 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4200 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4201 if (left > GAIN_FLOAT_UNITY) { 4202 left = GAIN_FLOAT_UNITY; 4203 } 4204 left *= v; 4205 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4206 if (right > GAIN_FLOAT_UNITY) { 4207 right = GAIN_FLOAT_UNITY; 4208 } 4209 right *= v; 4210 } 4211 4212 if (lastTrack) { 4213 if (left != mLeftVolFloat || right != mRightVolFloat) { 4214 mLeftVolFloat = left; 4215 mRightVolFloat = right; 4216 4217 // Convert volumes from float to 8.24 4218 uint32_t vl = (uint32_t)(left * (1 << 24)); 4219 uint32_t vr = (uint32_t)(right * (1 << 24)); 4220 4221 // Delegate volume control to effect in track effect chain if needed 4222 // only one effect chain can be present on DirectOutputThread, so if 4223 // there is one, the track is connected to it 4224 if (!mEffectChains.isEmpty()) { 4225 mEffectChains[0]->setVolume_l(&vl, &vr); 4226 left = (float)vl / (1 << 24); 4227 right = (float)vr / (1 << 24); 4228 } 4229 if (mOutput->stream->set_volume) { 4230 mOutput->stream->set_volume(mOutput->stream, left, right); 4231 } 4232 } 4233 } 4234} 4235 4236 4237AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4238 Vector< sp<Track> > *tracksToRemove 4239) 4240{ 4241 size_t count = mActiveTracks.size(); 4242 mixer_state mixerStatus = MIXER_IDLE; 4243 bool doHwPause = false; 4244 bool doHwResume = false; 4245 bool flushPending = false; 4246 4247 // find out which tracks need to be processed 4248 for (size_t i = 0; i < count; i++) { 4249 sp<Track> t = mActiveTracks[i].promote(); 4250 // The track died recently 4251 if (t == 0) { 4252 continue; 4253 } 4254 4255 Track* const track = t.get(); 4256 audio_track_cblk_t* cblk = track->cblk(); 4257 // Only consider last track started for volume and mixer state control. 4258 // In theory an older track could underrun and restart after the new one starts 4259 // but as we only care about the transition phase between two tracks on a 4260 // direct output, it is not a problem to ignore the underrun case. 4261 sp<Track> l = mLatestActiveTrack.promote(); 4262 bool last = l.get() == track; 4263 4264 if (mHwSupportsPause && track->isPausing()) { 4265 track->setPaused(); 4266 if (last && !mHwPaused) { 4267 doHwPause = true; 4268 mHwPaused = true; 4269 } 4270 tracksToRemove->add(track); 4271 } else if (track->isFlushPending()) { 4272 track->flushAck(); 4273 if (last) { 4274 flushPending = true; 4275 } 4276 } else if (mHwSupportsPause && track->isResumePending()){ 4277 track->resumeAck(); 4278 if (last) { 4279 if (mHwPaused) { 4280 doHwResume = true; 4281 mHwPaused = false; 4282 } 4283 } 4284 } 4285 4286 // The first time a track is added we wait 4287 // for all its buffers to be filled before processing it. 4288 // Allow draining the buffer in case the client 4289 // app does not call stop() and relies on underrun to stop: 4290 // hence the test on (track->mRetryCount > 1). 4291 // If retryCount<=1 then track is about to underrun and be removed. 4292 uint32_t minFrames; 4293 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4294 && (track->mRetryCount > 1)) { 4295 minFrames = mNormalFrameCount; 4296 } else { 4297 minFrames = 1; 4298 } 4299 4300 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4301 !track->isStopping_2() && !track->isStopped()) 4302 { 4303 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4304 4305 if (track->mFillingUpStatus == Track::FS_FILLED) { 4306 track->mFillingUpStatus = Track::FS_ACTIVE; 4307 // make sure processVolume_l() will apply new volume even if 0 4308 mLeftVolFloat = mRightVolFloat = -1.0; 4309 if (!mHwSupportsPause) { 4310 track->resumeAck(); 4311 } 4312 } 4313 4314 // compute volume for this track 4315 processVolume_l(track, last); 4316 if (last) { 4317 // reset retry count 4318 track->mRetryCount = kMaxTrackRetriesDirect; 4319 mActiveTrack = t; 4320 mixerStatus = MIXER_TRACKS_READY; 4321 if (usesHwAvSync() && mHwPaused) { 4322 doHwResume = true; 4323 mHwPaused = false; 4324 } 4325 } 4326 } else { 4327 // clear effect chain input buffer if the last active track started underruns 4328 // to avoid sending previous audio buffer again to effects 4329 if (!mEffectChains.isEmpty() && last) { 4330 mEffectChains[0]->clearInputBuffer(); 4331 } 4332 if (track->isStopping_1()) { 4333 track->mState = TrackBase::STOPPING_2; 4334 if (last && mHwPaused) { 4335 doHwResume = true; 4336 mHwPaused = false; 4337 } 4338 } 4339 if ((track->sharedBuffer() != 0) || track->isStopped() || 4340 track->isStopping_2() || track->isPaused()) { 4341 // We have consumed all the buffers of this track. 4342 // Remove it from the list of active tracks. 4343 size_t audioHALFrames; 4344 if (audio_is_linear_pcm(mFormat)) { 4345 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4346 } else { 4347 audioHALFrames = 0; 4348 } 4349 4350 size_t framesWritten = mBytesWritten / mFrameSize; 4351 if (mStandby || !last || 4352 track->presentationComplete(framesWritten, audioHALFrames)) { 4353 if (track->isStopping_2()) { 4354 track->mState = TrackBase::STOPPED; 4355 } 4356 if (track->isStopped()) { 4357 track->reset(); 4358 } 4359 tracksToRemove->add(track); 4360 } 4361 } else { 4362 // No buffers for this track. Give it a few chances to 4363 // fill a buffer, then remove it from active list. 4364 // Only consider last track started for mixer state control 4365 if (--(track->mRetryCount) <= 0) { 4366 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4367 tracksToRemove->add(track); 4368 // indicate to client process that the track was disabled because of underrun; 4369 // it will then automatically call start() when data is available 4370 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4371 } else if (last) { 4372 mixerStatus = MIXER_TRACKS_ENABLED; 4373 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4374 doHwPause = true; 4375 mHwPaused = true; 4376 } 4377 } 4378 } 4379 } 4380 } 4381 4382 // if an active track did not command a flush, check for pending flush on stopped tracks 4383 if (!flushPending) { 4384 for (size_t i = 0; i < mTracks.size(); i++) { 4385 if (mTracks[i]->isFlushPending()) { 4386 mTracks[i]->flushAck(); 4387 flushPending = true; 4388 } 4389 } 4390 } 4391 4392 // make sure the pause/flush/resume sequence is executed in the right order. 4393 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4394 // before flush and then resume HW. This can happen in case of pause/flush/resume 4395 // if resume is received before pause is executed. 4396 if (mHwSupportsPause && !mStandby && 4397 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4398 mOutput->stream->pause(mOutput->stream); 4399 } 4400 if (flushPending) { 4401 flushHw_l(); 4402 } 4403 if (mHwSupportsPause && !mStandby && doHwResume) { 4404 mOutput->stream->resume(mOutput->stream); 4405 } 4406 // remove all the tracks that need to be... 4407 removeTracks_l(*tracksToRemove); 4408 4409 return mixerStatus; 4410} 4411 4412void AudioFlinger::DirectOutputThread::threadLoop_mix() 4413{ 4414 size_t frameCount = mFrameCount; 4415 int8_t *curBuf = (int8_t *)mSinkBuffer; 4416 // output audio to hardware 4417 while (frameCount) { 4418 AudioBufferProvider::Buffer buffer; 4419 buffer.frameCount = frameCount; 4420 status_t status = mActiveTrack->getNextBuffer(&buffer); 4421 if (status != NO_ERROR || buffer.raw == NULL) { 4422 memset(curBuf, 0, frameCount * mFrameSize); 4423 break; 4424 } 4425 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4426 frameCount -= buffer.frameCount; 4427 curBuf += buffer.frameCount * mFrameSize; 4428 mActiveTrack->releaseBuffer(&buffer); 4429 } 4430 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4431 sleepTime = 0; 4432 standbyTime = systemTime() + standbyDelay; 4433 mActiveTrack.clear(); 4434} 4435 4436void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4437{ 4438 // do not write to HAL when paused 4439 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4440 sleepTime = idleSleepTime; 4441 return; 4442 } 4443 if (sleepTime == 0) { 4444 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4445 sleepTime = activeSleepTime; 4446 } else { 4447 sleepTime = idleSleepTime; 4448 } 4449 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4450 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4451 sleepTime = 0; 4452 } 4453} 4454 4455void AudioFlinger::DirectOutputThread::threadLoop_exit() 4456{ 4457 { 4458 Mutex::Autolock _l(mLock); 4459 bool flushPending = false; 4460 for (size_t i = 0; i < mTracks.size(); i++) { 4461 if (mTracks[i]->isFlushPending()) { 4462 mTracks[i]->flushAck(); 4463 flushPending = true; 4464 } 4465 } 4466 if (flushPending) { 4467 flushHw_l(); 4468 } 4469 } 4470 PlaybackThread::threadLoop_exit(); 4471} 4472 4473// must be called with thread mutex locked 4474bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4475{ 4476 bool trackPaused = false; 4477 bool trackStopped = false; 4478 4479 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4480 // after a timeout and we will enter standby then. 4481 if (mTracks.size() > 0) { 4482 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4483 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4484 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4485 } 4486 4487 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped)); 4488} 4489 4490// getTrackName_l() must be called with ThreadBase::mLock held 4491int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4492 audio_format_t format __unused, int sessionId __unused) 4493{ 4494 return 0; 4495} 4496 4497// deleteTrackName_l() must be called with ThreadBase::mLock held 4498void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4499{ 4500} 4501 4502// checkForNewParameter_l() must be called with ThreadBase::mLock held 4503bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4504 status_t& status) 4505{ 4506 bool reconfig = false; 4507 4508 status = NO_ERROR; 4509 4510 AudioParameter param = AudioParameter(keyValuePair); 4511 int value; 4512 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4513 // forward device change to effects that have requested to be 4514 // aware of attached audio device. 4515 if (value != AUDIO_DEVICE_NONE) { 4516 mOutDevice = value; 4517 for (size_t i = 0; i < mEffectChains.size(); i++) { 4518 mEffectChains[i]->setDevice_l(mOutDevice); 4519 } 4520 } 4521 } 4522 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4523 // do not accept frame count changes if tracks are open as the track buffer 4524 // size depends on frame count and correct behavior would not be garantied 4525 // if frame count is changed after track creation 4526 if (!mTracks.isEmpty()) { 4527 status = INVALID_OPERATION; 4528 } else { 4529 reconfig = true; 4530 } 4531 } 4532 if (status == NO_ERROR) { 4533 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4534 keyValuePair.string()); 4535 if (!mStandby && status == INVALID_OPERATION) { 4536 mOutput->standby(); 4537 mStandby = true; 4538 mBytesWritten = 0; 4539 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4540 keyValuePair.string()); 4541 } 4542 if (status == NO_ERROR && reconfig) { 4543 readOutputParameters_l(); 4544 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4545 } 4546 } 4547 4548 return reconfig; 4549} 4550 4551uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4552{ 4553 uint32_t time; 4554 if (audio_is_linear_pcm(mFormat)) { 4555 time = PlaybackThread::activeSleepTimeUs(); 4556 } else { 4557 time = 10000; 4558 } 4559 return time; 4560} 4561 4562uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4563{ 4564 uint32_t time; 4565 if (audio_is_linear_pcm(mFormat)) { 4566 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4567 } else { 4568 time = 10000; 4569 } 4570 return time; 4571} 4572 4573uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4574{ 4575 uint32_t time; 4576 if (audio_is_linear_pcm(mFormat)) { 4577 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4578 } else { 4579 time = 10000; 4580 } 4581 return time; 4582} 4583 4584void AudioFlinger::DirectOutputThread::cacheParameters_l() 4585{ 4586 PlaybackThread::cacheParameters_l(); 4587 4588 // use shorter standby delay as on normal output to release 4589 // hardware resources as soon as possible 4590 // no delay on outputs with HW A/V sync 4591 if (usesHwAvSync()) { 4592 standbyDelay = 0; 4593 } else if (audio_is_linear_pcm(mFormat)) { 4594 standbyDelay = microseconds(activeSleepTime*2); 4595 } else { 4596 standbyDelay = kOffloadStandbyDelayNs; 4597 } 4598} 4599 4600void AudioFlinger::DirectOutputThread::flushHw_l() 4601{ 4602 mOutput->flush(); 4603 mHwPaused = false; 4604} 4605 4606// ---------------------------------------------------------------------------- 4607 4608AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4609 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4610 : Thread(false /*canCallJava*/), 4611 mPlaybackThread(playbackThread), 4612 mWriteAckSequence(0), 4613 mDrainSequence(0) 4614{ 4615} 4616 4617AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4618{ 4619} 4620 4621void AudioFlinger::AsyncCallbackThread::onFirstRef() 4622{ 4623 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4624} 4625 4626bool AudioFlinger::AsyncCallbackThread::threadLoop() 4627{ 4628 while (!exitPending()) { 4629 uint32_t writeAckSequence; 4630 uint32_t drainSequence; 4631 4632 { 4633 Mutex::Autolock _l(mLock); 4634 while (!((mWriteAckSequence & 1) || 4635 (mDrainSequence & 1) || 4636 exitPending())) { 4637 mWaitWorkCV.wait(mLock); 4638 } 4639 4640 if (exitPending()) { 4641 break; 4642 } 4643 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4644 mWriteAckSequence, mDrainSequence); 4645 writeAckSequence = mWriteAckSequence; 4646 mWriteAckSequence &= ~1; 4647 drainSequence = mDrainSequence; 4648 mDrainSequence &= ~1; 4649 } 4650 { 4651 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4652 if (playbackThread != 0) { 4653 if (writeAckSequence & 1) { 4654 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4655 } 4656 if (drainSequence & 1) { 4657 playbackThread->resetDraining(drainSequence >> 1); 4658 } 4659 } 4660 } 4661 } 4662 return false; 4663} 4664 4665void AudioFlinger::AsyncCallbackThread::exit() 4666{ 4667 ALOGV("AsyncCallbackThread::exit"); 4668 Mutex::Autolock _l(mLock); 4669 requestExit(); 4670 mWaitWorkCV.broadcast(); 4671} 4672 4673void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4674{ 4675 Mutex::Autolock _l(mLock); 4676 // bit 0 is cleared 4677 mWriteAckSequence = sequence << 1; 4678} 4679 4680void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4681{ 4682 Mutex::Autolock _l(mLock); 4683 // ignore unexpected callbacks 4684 if (mWriteAckSequence & 2) { 4685 mWriteAckSequence |= 1; 4686 mWaitWorkCV.signal(); 4687 } 4688} 4689 4690void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4691{ 4692 Mutex::Autolock _l(mLock); 4693 // bit 0 is cleared 4694 mDrainSequence = sequence << 1; 4695} 4696 4697void AudioFlinger::AsyncCallbackThread::resetDraining() 4698{ 4699 Mutex::Autolock _l(mLock); 4700 // ignore unexpected callbacks 4701 if (mDrainSequence & 2) { 4702 mDrainSequence |= 1; 4703 mWaitWorkCV.signal(); 4704 } 4705} 4706 4707 4708// ---------------------------------------------------------------------------- 4709AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4710 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4711 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4712 mPausedBytesRemaining(0) 4713{ 4714 //FIXME: mStandby should be set to true by ThreadBase constructor 4715 mStandby = true; 4716} 4717 4718void AudioFlinger::OffloadThread::threadLoop_exit() 4719{ 4720 if (mFlushPending || mHwPaused) { 4721 // If a flush is pending or track was paused, just discard buffered data 4722 flushHw_l(); 4723 } else { 4724 mMixerStatus = MIXER_DRAIN_ALL; 4725 threadLoop_drain(); 4726 } 4727 if (mUseAsyncWrite) { 4728 ALOG_ASSERT(mCallbackThread != 0); 4729 mCallbackThread->exit(); 4730 } 4731 PlaybackThread::threadLoop_exit(); 4732} 4733 4734AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4735 Vector< sp<Track> > *tracksToRemove 4736) 4737{ 4738 size_t count = mActiveTracks.size(); 4739 4740 mixer_state mixerStatus = MIXER_IDLE; 4741 bool doHwPause = false; 4742 bool doHwResume = false; 4743 4744 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4745 4746 // find out which tracks need to be processed 4747 for (size_t i = 0; i < count; i++) { 4748 sp<Track> t = mActiveTracks[i].promote(); 4749 // The track died recently 4750 if (t == 0) { 4751 continue; 4752 } 4753 Track* const track = t.get(); 4754 audio_track_cblk_t* cblk = track->cblk(); 4755 // Only consider last track started for volume and mixer state control. 4756 // In theory an older track could underrun and restart after the new one starts 4757 // but as we only care about the transition phase between two tracks on a 4758 // direct output, it is not a problem to ignore the underrun case. 4759 sp<Track> l = mLatestActiveTrack.promote(); 4760 bool last = l.get() == track; 4761 4762 if (track->isInvalid()) { 4763 ALOGW("An invalidated track shouldn't be in active list"); 4764 tracksToRemove->add(track); 4765 continue; 4766 } 4767 4768 if (track->mState == TrackBase::IDLE) { 4769 ALOGW("An idle track shouldn't be in active list"); 4770 continue; 4771 } 4772 4773 if (track->isPausing()) { 4774 track->setPaused(); 4775 if (last) { 4776 if (!mHwPaused) { 4777 doHwPause = true; 4778 mHwPaused = true; 4779 } 4780 // If we were part way through writing the mixbuffer to 4781 // the HAL we must save this until we resume 4782 // BUG - this will be wrong if a different track is made active, 4783 // in that case we want to discard the pending data in the 4784 // mixbuffer and tell the client to present it again when the 4785 // track is resumed 4786 mPausedWriteLength = mCurrentWriteLength; 4787 mPausedBytesRemaining = mBytesRemaining; 4788 mBytesRemaining = 0; // stop writing 4789 } 4790 tracksToRemove->add(track); 4791 } else if (track->isFlushPending()) { 4792 track->flushAck(); 4793 if (last) { 4794 mFlushPending = true; 4795 } 4796 } else if (track->isResumePending()){ 4797 track->resumeAck(); 4798 if (last) { 4799 if (mPausedBytesRemaining) { 4800 // Need to continue write that was interrupted 4801 mCurrentWriteLength = mPausedWriteLength; 4802 mBytesRemaining = mPausedBytesRemaining; 4803 mPausedBytesRemaining = 0; 4804 } 4805 if (mHwPaused) { 4806 doHwResume = true; 4807 mHwPaused = false; 4808 // threadLoop_mix() will handle the case that we need to 4809 // resume an interrupted write 4810 } 4811 // enable write to audio HAL 4812 sleepTime = 0; 4813 4814 // Do not handle new data in this iteration even if track->framesReady() 4815 mixerStatus = MIXER_TRACKS_ENABLED; 4816 } 4817 } else if (track->framesReady() && track->isReady() && 4818 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4819 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4820 if (track->mFillingUpStatus == Track::FS_FILLED) { 4821 track->mFillingUpStatus = Track::FS_ACTIVE; 4822 // make sure processVolume_l() will apply new volume even if 0 4823 mLeftVolFloat = mRightVolFloat = -1.0; 4824 } 4825 4826 if (last) { 4827 sp<Track> previousTrack = mPreviousTrack.promote(); 4828 if (previousTrack != 0) { 4829 if (track != previousTrack.get()) { 4830 // Flush any data still being written from last track 4831 mBytesRemaining = 0; 4832 if (mPausedBytesRemaining) { 4833 // Last track was paused so we also need to flush saved 4834 // mixbuffer state and invalidate track so that it will 4835 // re-submit that unwritten data when it is next resumed 4836 mPausedBytesRemaining = 0; 4837 // Invalidate is a bit drastic - would be more efficient 4838 // to have a flag to tell client that some of the 4839 // previously written data was lost 4840 previousTrack->invalidate(); 4841 } 4842 // flush data already sent to the DSP if changing audio session as audio 4843 // comes from a different source. Also invalidate previous track to force a 4844 // seek when resuming. 4845 if (previousTrack->sessionId() != track->sessionId()) { 4846 previousTrack->invalidate(); 4847 } 4848 } 4849 } 4850 mPreviousTrack = track; 4851 // reset retry count 4852 track->mRetryCount = kMaxTrackRetriesOffload; 4853 mActiveTrack = t; 4854 mixerStatus = MIXER_TRACKS_READY; 4855 } 4856 } else { 4857 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4858 if (track->isStopping_1()) { 4859 // Hardware buffer can hold a large amount of audio so we must 4860 // wait for all current track's data to drain before we say 4861 // that the track is stopped. 4862 if (mBytesRemaining == 0) { 4863 // Only start draining when all data in mixbuffer 4864 // has been written 4865 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4866 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4867 // do not drain if no data was ever sent to HAL (mStandby == true) 4868 if (last && !mStandby) { 4869 // do not modify drain sequence if we are already draining. This happens 4870 // when resuming from pause after drain. 4871 if ((mDrainSequence & 1) == 0) { 4872 sleepTime = 0; 4873 standbyTime = systemTime() + standbyDelay; 4874 mixerStatus = MIXER_DRAIN_TRACK; 4875 mDrainSequence += 2; 4876 } 4877 if (mHwPaused) { 4878 // It is possible to move from PAUSED to STOPPING_1 without 4879 // a resume so we must ensure hardware is running 4880 doHwResume = true; 4881 mHwPaused = false; 4882 } 4883 } 4884 } 4885 } else if (track->isStopping_2()) { 4886 // Drain has completed or we are in standby, signal presentation complete 4887 if (!(mDrainSequence & 1) || !last || mStandby) { 4888 track->mState = TrackBase::STOPPED; 4889 size_t audioHALFrames = 4890 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4891 size_t framesWritten = 4892 mBytesWritten / mOutput->getFrameSize(); 4893 track->presentationComplete(framesWritten, audioHALFrames); 4894 track->reset(); 4895 tracksToRemove->add(track); 4896 } 4897 } else { 4898 // No buffers for this track. Give it a few chances to 4899 // fill a buffer, then remove it from active list. 4900 if (--(track->mRetryCount) <= 0) { 4901 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4902 track->name()); 4903 tracksToRemove->add(track); 4904 // indicate to client process that the track was disabled because of underrun; 4905 // it will then automatically call start() when data is available 4906 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4907 } else if (last){ 4908 mixerStatus = MIXER_TRACKS_ENABLED; 4909 } 4910 } 4911 } 4912 // compute volume for this track 4913 processVolume_l(track, last); 4914 } 4915 4916 // make sure the pause/flush/resume sequence is executed in the right order. 4917 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4918 // before flush and then resume HW. This can happen in case of pause/flush/resume 4919 // if resume is received before pause is executed. 4920 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4921 mOutput->stream->pause(mOutput->stream); 4922 } 4923 if (mFlushPending) { 4924 flushHw_l(); 4925 mFlushPending = false; 4926 } 4927 if (!mStandby && doHwResume) { 4928 mOutput->stream->resume(mOutput->stream); 4929 } 4930 4931 // remove all the tracks that need to be... 4932 removeTracks_l(*tracksToRemove); 4933 4934 return mixerStatus; 4935} 4936 4937// must be called with thread mutex locked 4938bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4939{ 4940 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4941 mWriteAckSequence, mDrainSequence); 4942 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4943 return true; 4944 } 4945 return false; 4946} 4947 4948bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4949{ 4950 Mutex::Autolock _l(mLock); 4951 return waitingAsyncCallback_l(); 4952} 4953 4954void AudioFlinger::OffloadThread::flushHw_l() 4955{ 4956 DirectOutputThread::flushHw_l(); 4957 // Flush anything still waiting in the mixbuffer 4958 mCurrentWriteLength = 0; 4959 mBytesRemaining = 0; 4960 mPausedWriteLength = 0; 4961 mPausedBytesRemaining = 0; 4962 4963 if (mUseAsyncWrite) { 4964 // discard any pending drain or write ack by incrementing sequence 4965 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4966 mDrainSequence = (mDrainSequence + 2) & ~1; 4967 ALOG_ASSERT(mCallbackThread != 0); 4968 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4969 mCallbackThread->setDraining(mDrainSequence); 4970 } 4971} 4972 4973void AudioFlinger::OffloadThread::onAddNewTrack_l() 4974{ 4975 sp<Track> previousTrack = mPreviousTrack.promote(); 4976 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4977 4978 if (previousTrack != 0 && latestTrack != 0 && 4979 (previousTrack->sessionId() != latestTrack->sessionId())) { 4980 mFlushPending = true; 4981 } 4982 PlaybackThread::onAddNewTrack_l(); 4983} 4984 4985// ---------------------------------------------------------------------------- 4986 4987AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4988 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4989 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4990 DUPLICATING), 4991 mWaitTimeMs(UINT_MAX) 4992{ 4993 addOutputTrack(mainThread); 4994} 4995 4996AudioFlinger::DuplicatingThread::~DuplicatingThread() 4997{ 4998 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4999 mOutputTracks[i]->destroy(); 5000 } 5001} 5002 5003void AudioFlinger::DuplicatingThread::threadLoop_mix() 5004{ 5005 // mix buffers... 5006 if (outputsReady(outputTracks)) { 5007 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5008 } else { 5009 if (mMixerBufferValid) { 5010 memset(mMixerBuffer, 0, mMixerBufferSize); 5011 } else { 5012 memset(mSinkBuffer, 0, mSinkBufferSize); 5013 } 5014 } 5015 sleepTime = 0; 5016 writeFrames = mNormalFrameCount; 5017 mCurrentWriteLength = mSinkBufferSize; 5018 standbyTime = systemTime() + standbyDelay; 5019} 5020 5021void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5022{ 5023 if (sleepTime == 0) { 5024 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5025 sleepTime = activeSleepTime; 5026 } else { 5027 sleepTime = idleSleepTime; 5028 } 5029 } else if (mBytesWritten != 0) { 5030 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5031 writeFrames = mNormalFrameCount; 5032 memset(mSinkBuffer, 0, mSinkBufferSize); 5033 } else { 5034 // flush remaining overflow buffers in output tracks 5035 writeFrames = 0; 5036 } 5037 sleepTime = 0; 5038 } 5039} 5040 5041ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5042{ 5043 for (size_t i = 0; i < outputTracks.size(); i++) { 5044 outputTracks[i]->write(mSinkBuffer, writeFrames); 5045 } 5046 mStandby = false; 5047 return (ssize_t)mSinkBufferSize; 5048} 5049 5050void AudioFlinger::DuplicatingThread::threadLoop_standby() 5051{ 5052 // DuplicatingThread implements standby by stopping all tracks 5053 for (size_t i = 0; i < outputTracks.size(); i++) { 5054 outputTracks[i]->stop(); 5055 } 5056} 5057 5058void AudioFlinger::DuplicatingThread::saveOutputTracks() 5059{ 5060 outputTracks = mOutputTracks; 5061} 5062 5063void AudioFlinger::DuplicatingThread::clearOutputTracks() 5064{ 5065 outputTracks.clear(); 5066} 5067 5068void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5069{ 5070 Mutex::Autolock _l(mLock); 5071 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5072 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5073 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5074 const size_t frameCount = 5075 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5076 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5077 // from different OutputTracks and their associated MixerThreads (e.g. one may 5078 // nearly empty and the other may be dropping data). 5079 5080 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5081 this, 5082 mSampleRate, 5083 mFormat, 5084 mChannelMask, 5085 frameCount, 5086 IPCThreadState::self()->getCallingUid()); 5087 if (outputTrack->cblk() != NULL) { 5088 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5089 mOutputTracks.add(outputTrack); 5090 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5091 updateWaitTime_l(); 5092 } 5093} 5094 5095void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5096{ 5097 Mutex::Autolock _l(mLock); 5098 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5099 if (mOutputTracks[i]->thread() == thread) { 5100 mOutputTracks[i]->destroy(); 5101 mOutputTracks.removeAt(i); 5102 updateWaitTime_l(); 5103 return; 5104 } 5105 } 5106 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5107} 5108 5109// caller must hold mLock 5110void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5111{ 5112 mWaitTimeMs = UINT_MAX; 5113 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5114 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5115 if (strong != 0) { 5116 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5117 if (waitTimeMs < mWaitTimeMs) { 5118 mWaitTimeMs = waitTimeMs; 5119 } 5120 } 5121 } 5122} 5123 5124 5125bool AudioFlinger::DuplicatingThread::outputsReady( 5126 const SortedVector< sp<OutputTrack> > &outputTracks) 5127{ 5128 for (size_t i = 0; i < outputTracks.size(); i++) { 5129 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5130 if (thread == 0) { 5131 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5132 outputTracks[i].get()); 5133 return false; 5134 } 5135 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5136 // see note at standby() declaration 5137 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5138 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5139 thread.get()); 5140 return false; 5141 } 5142 } 5143 return true; 5144} 5145 5146uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5147{ 5148 return (mWaitTimeMs * 1000) / 2; 5149} 5150 5151void AudioFlinger::DuplicatingThread::cacheParameters_l() 5152{ 5153 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5154 updateWaitTime_l(); 5155 5156 MixerThread::cacheParameters_l(); 5157} 5158 5159// ---------------------------------------------------------------------------- 5160// Record 5161// ---------------------------------------------------------------------------- 5162 5163AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5164 AudioStreamIn *input, 5165 audio_io_handle_t id, 5166 audio_devices_t outDevice, 5167 audio_devices_t inDevice 5168#ifdef TEE_SINK 5169 , const sp<NBAIO_Sink>& teeSink 5170#endif 5171 ) : 5172 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5173 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5174 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5175 mRsmpInRear(0) 5176#ifdef TEE_SINK 5177 , mTeeSink(teeSink) 5178#endif 5179 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5180 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5181 // mFastCapture below 5182 , mFastCaptureFutex(0) 5183 // mInputSource 5184 // mPipeSink 5185 // mPipeSource 5186 , mPipeFramesP2(0) 5187 // mPipeMemory 5188 // mFastCaptureNBLogWriter 5189 , mFastTrackAvail(false) 5190{ 5191 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5192 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5193 5194 readInputParameters_l(); 5195 5196 // create an NBAIO source for the HAL input stream, and negotiate 5197 mInputSource = new AudioStreamInSource(input->stream); 5198 size_t numCounterOffers = 0; 5199 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5200 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5201 ALOG_ASSERT(index == 0); 5202 5203 // initialize fast capture depending on configuration 5204 bool initFastCapture; 5205 switch (kUseFastCapture) { 5206 case FastCapture_Never: 5207 initFastCapture = false; 5208 break; 5209 case FastCapture_Always: 5210 initFastCapture = true; 5211 break; 5212 case FastCapture_Static: 5213 uint32_t primaryOutputSampleRate; 5214 { 5215 AutoMutex _l(audioFlinger->mHardwareLock); 5216 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5217 } 5218 initFastCapture = 5219 // either capture sample rate is same as (a reasonable) primary output sample rate 5220 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5221 (mSampleRate == primaryOutputSampleRate)) || 5222 // or primary output sample rate is unknown, and capture sample rate is reasonable 5223 ((primaryOutputSampleRate == 0) && 5224 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5225 // and the buffer size is < 12 ms 5226 (mFrameCount * 1000) / mSampleRate < 12; 5227 break; 5228 // case FastCapture_Dynamic: 5229 } 5230 5231 if (initFastCapture) { 5232 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5233 NBAIO_Format format = mInputSource->format(); 5234 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5235 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5236 void *pipeBuffer; 5237 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5238 sp<IMemory> pipeMemory; 5239 if ((roHeap == 0) || 5240 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5241 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5242 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5243 goto failed; 5244 } 5245 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5246 memset(pipeBuffer, 0, pipeSize); 5247 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5248 const NBAIO_Format offers[1] = {format}; 5249 size_t numCounterOffers = 0; 5250 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5251 ALOG_ASSERT(index == 0); 5252 mPipeSink = pipe; 5253 PipeReader *pipeReader = new PipeReader(*pipe); 5254 numCounterOffers = 0; 5255 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5256 ALOG_ASSERT(index == 0); 5257 mPipeSource = pipeReader; 5258 mPipeFramesP2 = pipeFramesP2; 5259 mPipeMemory = pipeMemory; 5260 5261 // create fast capture 5262 mFastCapture = new FastCapture(); 5263 FastCaptureStateQueue *sq = mFastCapture->sq(); 5264#ifdef STATE_QUEUE_DUMP 5265 // FIXME 5266#endif 5267 FastCaptureState *state = sq->begin(); 5268 state->mCblk = NULL; 5269 state->mInputSource = mInputSource.get(); 5270 state->mInputSourceGen++; 5271 state->mPipeSink = pipe; 5272 state->mPipeSinkGen++; 5273 state->mFrameCount = mFrameCount; 5274 state->mCommand = FastCaptureState::COLD_IDLE; 5275 // already done in constructor initialization list 5276 //mFastCaptureFutex = 0; 5277 state->mColdFutexAddr = &mFastCaptureFutex; 5278 state->mColdGen++; 5279 state->mDumpState = &mFastCaptureDumpState; 5280#ifdef TEE_SINK 5281 // FIXME 5282#endif 5283 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5284 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5285 sq->end(); 5286 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5287 5288 // start the fast capture 5289 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5290 pid_t tid = mFastCapture->getTid(); 5291 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5292 if (err != 0) { 5293 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5294 kPriorityFastCapture, getpid_cached, tid, err); 5295 } 5296 5297#ifdef AUDIO_WATCHDOG 5298 // FIXME 5299#endif 5300 5301 mFastTrackAvail = true; 5302 } 5303failed: ; 5304 5305 // FIXME mNormalSource 5306} 5307 5308AudioFlinger::RecordThread::~RecordThread() 5309{ 5310 if (mFastCapture != 0) { 5311 FastCaptureStateQueue *sq = mFastCapture->sq(); 5312 FastCaptureState *state = sq->begin(); 5313 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5314 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5315 if (old == -1) { 5316 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5317 } 5318 } 5319 state->mCommand = FastCaptureState::EXIT; 5320 sq->end(); 5321 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5322 mFastCapture->join(); 5323 mFastCapture.clear(); 5324 } 5325 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5326 mAudioFlinger->unregisterWriter(mNBLogWriter); 5327 delete[] mRsmpInBuffer; 5328} 5329 5330void AudioFlinger::RecordThread::onFirstRef() 5331{ 5332 run(mThreadName, PRIORITY_URGENT_AUDIO); 5333} 5334 5335bool AudioFlinger::RecordThread::threadLoop() 5336{ 5337 nsecs_t lastWarning = 0; 5338 5339 inputStandBy(); 5340 5341reacquire_wakelock: 5342 sp<RecordTrack> activeTrack; 5343 int activeTracksGen; 5344 { 5345 Mutex::Autolock _l(mLock); 5346 size_t size = mActiveTracks.size(); 5347 activeTracksGen = mActiveTracksGen; 5348 if (size > 0) { 5349 // FIXME an arbitrary choice 5350 activeTrack = mActiveTracks[0]; 5351 acquireWakeLock_l(activeTrack->uid()); 5352 if (size > 1) { 5353 SortedVector<int> tmp; 5354 for (size_t i = 0; i < size; i++) { 5355 tmp.add(mActiveTracks[i]->uid()); 5356 } 5357 updateWakeLockUids_l(tmp); 5358 } 5359 } else { 5360 acquireWakeLock_l(-1); 5361 } 5362 } 5363 5364 // used to request a deferred sleep, to be executed later while mutex is unlocked 5365 uint32_t sleepUs = 0; 5366 5367 // loop while there is work to do 5368 for (;;) { 5369 Vector< sp<EffectChain> > effectChains; 5370 5371 // sleep with mutex unlocked 5372 if (sleepUs > 0) { 5373 ATRACE_BEGIN("sleep"); 5374 usleep(sleepUs); 5375 ATRACE_END(); 5376 sleepUs = 0; 5377 } 5378 5379 // activeTracks accumulates a copy of a subset of mActiveTracks 5380 Vector< sp<RecordTrack> > activeTracks; 5381 5382 // reference to the (first and only) active fast track 5383 sp<RecordTrack> fastTrack; 5384 5385 // reference to a fast track which is about to be removed 5386 sp<RecordTrack> fastTrackToRemove; 5387 5388 { // scope for mLock 5389 Mutex::Autolock _l(mLock); 5390 5391 processConfigEvents_l(); 5392 5393 // check exitPending here because checkForNewParameters_l() and 5394 // checkForNewParameters_l() can temporarily release mLock 5395 if (exitPending()) { 5396 break; 5397 } 5398 5399 // if no active track(s), then standby and release wakelock 5400 size_t size = mActiveTracks.size(); 5401 if (size == 0) { 5402 standbyIfNotAlreadyInStandby(); 5403 // exitPending() can't become true here 5404 releaseWakeLock_l(); 5405 ALOGV("RecordThread: loop stopping"); 5406 // go to sleep 5407 mWaitWorkCV.wait(mLock); 5408 ALOGV("RecordThread: loop starting"); 5409 goto reacquire_wakelock; 5410 } 5411 5412 if (mActiveTracksGen != activeTracksGen) { 5413 activeTracksGen = mActiveTracksGen; 5414 SortedVector<int> tmp; 5415 for (size_t i = 0; i < size; i++) { 5416 tmp.add(mActiveTracks[i]->uid()); 5417 } 5418 updateWakeLockUids_l(tmp); 5419 } 5420 5421 bool doBroadcast = false; 5422 for (size_t i = 0; i < size; ) { 5423 5424 activeTrack = mActiveTracks[i]; 5425 if (activeTrack->isTerminated()) { 5426 if (activeTrack->isFastTrack()) { 5427 ALOG_ASSERT(fastTrackToRemove == 0); 5428 fastTrackToRemove = activeTrack; 5429 } 5430 removeTrack_l(activeTrack); 5431 mActiveTracks.remove(activeTrack); 5432 mActiveTracksGen++; 5433 size--; 5434 continue; 5435 } 5436 5437 TrackBase::track_state activeTrackState = activeTrack->mState; 5438 switch (activeTrackState) { 5439 5440 case TrackBase::PAUSING: 5441 mActiveTracks.remove(activeTrack); 5442 mActiveTracksGen++; 5443 doBroadcast = true; 5444 size--; 5445 continue; 5446 5447 case TrackBase::STARTING_1: 5448 sleepUs = 10000; 5449 i++; 5450 continue; 5451 5452 case TrackBase::STARTING_2: 5453 doBroadcast = true; 5454 mStandby = false; 5455 activeTrack->mState = TrackBase::ACTIVE; 5456 break; 5457 5458 case TrackBase::ACTIVE: 5459 break; 5460 5461 case TrackBase::IDLE: 5462 i++; 5463 continue; 5464 5465 default: 5466 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5467 } 5468 5469 activeTracks.add(activeTrack); 5470 i++; 5471 5472 if (activeTrack->isFastTrack()) { 5473 ALOG_ASSERT(!mFastTrackAvail); 5474 ALOG_ASSERT(fastTrack == 0); 5475 fastTrack = activeTrack; 5476 } 5477 } 5478 if (doBroadcast) { 5479 mStartStopCond.broadcast(); 5480 } 5481 5482 // sleep if there are no active tracks to process 5483 if (activeTracks.size() == 0) { 5484 if (sleepUs == 0) { 5485 sleepUs = kRecordThreadSleepUs; 5486 } 5487 continue; 5488 } 5489 sleepUs = 0; 5490 5491 lockEffectChains_l(effectChains); 5492 } 5493 5494 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5495 5496 size_t size = effectChains.size(); 5497 for (size_t i = 0; i < size; i++) { 5498 // thread mutex is not locked, but effect chain is locked 5499 effectChains[i]->process_l(); 5500 } 5501 5502 // Push a new fast capture state if fast capture is not already running, or cblk change 5503 if (mFastCapture != 0) { 5504 FastCaptureStateQueue *sq = mFastCapture->sq(); 5505 FastCaptureState *state = sq->begin(); 5506 bool didModify = false; 5507 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5508 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5509 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5510 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5511 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5512 if (old == -1) { 5513 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5514 } 5515 } 5516 state->mCommand = FastCaptureState::READ_WRITE; 5517#if 0 // FIXME 5518 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5519 FastThreadDumpState::kSamplingNforLowRamDevice : 5520 FastThreadDumpState::kSamplingN); 5521#endif 5522 didModify = true; 5523 } 5524 audio_track_cblk_t *cblkOld = state->mCblk; 5525 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5526 if (cblkNew != cblkOld) { 5527 state->mCblk = cblkNew; 5528 // block until acked if removing a fast track 5529 if (cblkOld != NULL) { 5530 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5531 } 5532 didModify = true; 5533 } 5534 sq->end(didModify); 5535 if (didModify) { 5536 sq->push(block); 5537#if 0 5538 if (kUseFastCapture == FastCapture_Dynamic) { 5539 mNormalSource = mPipeSource; 5540 } 5541#endif 5542 } 5543 } 5544 5545 // now run the fast track destructor with thread mutex unlocked 5546 fastTrackToRemove.clear(); 5547 5548 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5549 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5550 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5551 // If destination is non-contiguous, first read past the nominal end of buffer, then 5552 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5553 5554 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5555 ssize_t framesRead; 5556 5557 // If an NBAIO source is present, use it to read the normal capture's data 5558 if (mPipeSource != 0) { 5559 size_t framesToRead = mBufferSize / mFrameSize; 5560 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5561 framesToRead, AudioBufferProvider::kInvalidPTS); 5562 if (framesRead == 0) { 5563 // since pipe is non-blocking, simulate blocking input 5564 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5565 } 5566 // otherwise use the HAL / AudioStreamIn directly 5567 } else { 5568 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5569 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5570 if (bytesRead < 0) { 5571 framesRead = bytesRead; 5572 } else { 5573 framesRead = bytesRead / mFrameSize; 5574 } 5575 } 5576 5577 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5578 ALOGE("read failed: framesRead=%d", framesRead); 5579 // Force input into standby so that it tries to recover at next read attempt 5580 inputStandBy(); 5581 sleepUs = kRecordThreadSleepUs; 5582 } 5583 if (framesRead <= 0) { 5584 goto unlock; 5585 } 5586 ALOG_ASSERT(framesRead > 0); 5587 5588 if (mTeeSink != 0) { 5589 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5590 } 5591 // If destination is non-contiguous, we now correct for reading past end of buffer. 5592 { 5593 size_t part1 = mRsmpInFramesP2 - rear; 5594 if ((size_t) framesRead > part1) { 5595 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5596 (framesRead - part1) * mFrameSize); 5597 } 5598 } 5599 rear = mRsmpInRear += framesRead; 5600 5601 size = activeTracks.size(); 5602 // loop over each active track 5603 for (size_t i = 0; i < size; i++) { 5604 activeTrack = activeTracks[i]; 5605 5606 // skip fast tracks, as those are handled directly by FastCapture 5607 if (activeTrack->isFastTrack()) { 5608 continue; 5609 } 5610 5611 // TODO: This code probably should be moved to RecordTrack. 5612 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5613 5614 enum { 5615 OVERRUN_UNKNOWN, 5616 OVERRUN_TRUE, 5617 OVERRUN_FALSE 5618 } overrun = OVERRUN_UNKNOWN; 5619 5620 // loop over getNextBuffer to handle circular sink 5621 for (;;) { 5622 5623 activeTrack->mSink.frameCount = ~0; 5624 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5625 size_t framesOut = activeTrack->mSink.frameCount; 5626 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5627 5628 // check available frames and handle overrun conditions 5629 // if the record track isn't draining fast enough. 5630 bool hasOverrun; 5631 size_t framesIn; 5632 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5633 if (hasOverrun) { 5634 overrun = OVERRUN_TRUE; 5635 } 5636 if (framesOut == 0 || framesIn == 0) { 5637 break; 5638 } 5639 5640 // Don't allow framesOut to be larger than what is possible with resampling 5641 // from framesIn. 5642 // This isn't strictly necessary but helps limit buffer resizing in 5643 // RecordBufferConverter. TODO: remove when no longer needed. 5644 framesOut = min(framesOut, 5645 destinationFramesPossible( 5646 framesIn, mSampleRate, activeTrack->mSampleRate)); 5647 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5648 framesOut = activeTrack->mRecordBufferConverter->convert( 5649 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5650 5651 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5652 overrun = OVERRUN_FALSE; 5653 } 5654 5655 if (activeTrack->mFramesToDrop == 0) { 5656 if (framesOut > 0) { 5657 activeTrack->mSink.frameCount = framesOut; 5658 activeTrack->releaseBuffer(&activeTrack->mSink); 5659 } 5660 } else { 5661 // FIXME could do a partial drop of framesOut 5662 if (activeTrack->mFramesToDrop > 0) { 5663 activeTrack->mFramesToDrop -= framesOut; 5664 if (activeTrack->mFramesToDrop <= 0) { 5665 activeTrack->clearSyncStartEvent(); 5666 } 5667 } else { 5668 activeTrack->mFramesToDrop += framesOut; 5669 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5670 activeTrack->mSyncStartEvent->isCancelled()) { 5671 ALOGW("Synced record %s, session %d, trigger session %d", 5672 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5673 activeTrack->sessionId(), 5674 (activeTrack->mSyncStartEvent != 0) ? 5675 activeTrack->mSyncStartEvent->triggerSession() : 0); 5676 activeTrack->clearSyncStartEvent(); 5677 } 5678 } 5679 } 5680 5681 if (framesOut == 0) { 5682 break; 5683 } 5684 } 5685 5686 switch (overrun) { 5687 case OVERRUN_TRUE: 5688 // client isn't retrieving buffers fast enough 5689 if (!activeTrack->setOverflow()) { 5690 nsecs_t now = systemTime(); 5691 // FIXME should lastWarning per track? 5692 if ((now - lastWarning) > kWarningThrottleNs) { 5693 ALOGW("RecordThread: buffer overflow"); 5694 lastWarning = now; 5695 } 5696 } 5697 break; 5698 case OVERRUN_FALSE: 5699 activeTrack->clearOverflow(); 5700 break; 5701 case OVERRUN_UNKNOWN: 5702 break; 5703 } 5704 5705 } 5706 5707unlock: 5708 // enable changes in effect chain 5709 unlockEffectChains(effectChains); 5710 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5711 } 5712 5713 standbyIfNotAlreadyInStandby(); 5714 5715 { 5716 Mutex::Autolock _l(mLock); 5717 for (size_t i = 0; i < mTracks.size(); i++) { 5718 sp<RecordTrack> track = mTracks[i]; 5719 track->invalidate(); 5720 } 5721 mActiveTracks.clear(); 5722 mActiveTracksGen++; 5723 mStartStopCond.broadcast(); 5724 } 5725 5726 releaseWakeLock(); 5727 5728 ALOGV("RecordThread %p exiting", this); 5729 return false; 5730} 5731 5732void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5733{ 5734 if (!mStandby) { 5735 inputStandBy(); 5736 mStandby = true; 5737 } 5738} 5739 5740void AudioFlinger::RecordThread::inputStandBy() 5741{ 5742 // Idle the fast capture if it's currently running 5743 if (mFastCapture != 0) { 5744 FastCaptureStateQueue *sq = mFastCapture->sq(); 5745 FastCaptureState *state = sq->begin(); 5746 if (!(state->mCommand & FastCaptureState::IDLE)) { 5747 state->mCommand = FastCaptureState::COLD_IDLE; 5748 state->mColdFutexAddr = &mFastCaptureFutex; 5749 state->mColdGen++; 5750 mFastCaptureFutex = 0; 5751 sq->end(); 5752 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5753 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5754#if 0 5755 if (kUseFastCapture == FastCapture_Dynamic) { 5756 // FIXME 5757 } 5758#endif 5759#ifdef AUDIO_WATCHDOG 5760 // FIXME 5761#endif 5762 } else { 5763 sq->end(false /*didModify*/); 5764 } 5765 } 5766 mInput->stream->common.standby(&mInput->stream->common); 5767} 5768 5769// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5770sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5771 const sp<AudioFlinger::Client>& client, 5772 uint32_t sampleRate, 5773 audio_format_t format, 5774 audio_channel_mask_t channelMask, 5775 size_t *pFrameCount, 5776 int sessionId, 5777 size_t *notificationFrames, 5778 int uid, 5779 IAudioFlinger::track_flags_t *flags, 5780 pid_t tid, 5781 status_t *status) 5782{ 5783 size_t frameCount = *pFrameCount; 5784 sp<RecordTrack> track; 5785 status_t lStatus; 5786 5787 // client expresses a preference for FAST, but we get the final say 5788 if (*flags & IAudioFlinger::TRACK_FAST) { 5789 if ( 5790 // we formerly checked for a callback handler (non-0 tid), 5791 // but that is no longer required for TRANSFER_OBTAIN mode 5792 // 5793 // frame count is not specified, or is exactly the pipe depth 5794 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5795 // PCM data 5796 audio_is_linear_pcm(format) && 5797 // native format 5798 (format == mFormat) && 5799 // native channel mask 5800 (channelMask == mChannelMask) && 5801 // native hardware sample rate 5802 (sampleRate == mSampleRate) && 5803 // record thread has an associated fast capture 5804 hasFastCapture() && 5805 // there are sufficient fast track slots available 5806 mFastTrackAvail 5807 ) { 5808 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5809 frameCount, mFrameCount); 5810 } else { 5811 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5812 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5813 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5814 frameCount, mFrameCount, mPipeFramesP2, 5815 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5816 hasFastCapture(), tid, mFastTrackAvail); 5817 *flags &= ~IAudioFlinger::TRACK_FAST; 5818 } 5819 } 5820 5821 // compute track buffer size in frames, and suggest the notification frame count 5822 if (*flags & IAudioFlinger::TRACK_FAST) { 5823 // fast track: frame count is exactly the pipe depth 5824 frameCount = mPipeFramesP2; 5825 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5826 *notificationFrames = mFrameCount; 5827 } else { 5828 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5829 // or 20 ms if there is a fast capture 5830 // TODO This could be a roundupRatio inline, and const 5831 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5832 * sampleRate + mSampleRate - 1) / mSampleRate; 5833 // minimum number of notification periods is at least kMinNotifications, 5834 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5835 static const size_t kMinNotifications = 3; 5836 static const uint32_t kMinMs = 30; 5837 // TODO This could be a roundupRatio inline 5838 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5839 // TODO This could be a roundupRatio inline 5840 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5841 maxNotificationFrames; 5842 const size_t minFrameCount = maxNotificationFrames * 5843 max(kMinNotifications, minNotificationsByMs); 5844 frameCount = max(frameCount, minFrameCount); 5845 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5846 *notificationFrames = maxNotificationFrames; 5847 } 5848 } 5849 *pFrameCount = frameCount; 5850 5851 lStatus = initCheck(); 5852 if (lStatus != NO_ERROR) { 5853 ALOGE("createRecordTrack_l() audio driver not initialized"); 5854 goto Exit; 5855 } 5856 5857 { // scope for mLock 5858 Mutex::Autolock _l(mLock); 5859 5860 track = new RecordTrack(this, client, sampleRate, 5861 format, channelMask, frameCount, NULL, sessionId, uid, 5862 *flags, TrackBase::TYPE_DEFAULT); 5863 5864 lStatus = track->initCheck(); 5865 if (lStatus != NO_ERROR) { 5866 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5867 // track must be cleared from the caller as the caller has the AF lock 5868 goto Exit; 5869 } 5870 mTracks.add(track); 5871 5872 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5873 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5874 mAudioFlinger->btNrecIsOff(); 5875 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5876 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5877 5878 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5879 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5880 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5881 // so ask activity manager to do this on our behalf 5882 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5883 } 5884 } 5885 5886 lStatus = NO_ERROR; 5887 5888Exit: 5889 *status = lStatus; 5890 return track; 5891} 5892 5893status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5894 AudioSystem::sync_event_t event, 5895 int triggerSession) 5896{ 5897 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5898 sp<ThreadBase> strongMe = this; 5899 status_t status = NO_ERROR; 5900 5901 if (event == AudioSystem::SYNC_EVENT_NONE) { 5902 recordTrack->clearSyncStartEvent(); 5903 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5904 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5905 triggerSession, 5906 recordTrack->sessionId(), 5907 syncStartEventCallback, 5908 recordTrack); 5909 // Sync event can be cancelled by the trigger session if the track is not in a 5910 // compatible state in which case we start record immediately 5911 if (recordTrack->mSyncStartEvent->isCancelled()) { 5912 recordTrack->clearSyncStartEvent(); 5913 } else { 5914 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5915 recordTrack->mFramesToDrop = - 5916 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5917 } 5918 } 5919 5920 { 5921 // This section is a rendezvous between binder thread executing start() and RecordThread 5922 AutoMutex lock(mLock); 5923 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5924 if (recordTrack->mState == TrackBase::PAUSING) { 5925 ALOGV("active record track PAUSING -> ACTIVE"); 5926 recordTrack->mState = TrackBase::ACTIVE; 5927 } else { 5928 ALOGV("active record track state %d", recordTrack->mState); 5929 } 5930 return status; 5931 } 5932 5933 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5934 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5935 // or using a separate command thread 5936 recordTrack->mState = TrackBase::STARTING_1; 5937 mActiveTracks.add(recordTrack); 5938 mActiveTracksGen++; 5939 status_t status = NO_ERROR; 5940 if (recordTrack->isExternalTrack()) { 5941 mLock.unlock(); 5942 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5943 mLock.lock(); 5944 // FIXME should verify that recordTrack is still in mActiveTracks 5945 if (status != NO_ERROR) { 5946 mActiveTracks.remove(recordTrack); 5947 mActiveTracksGen++; 5948 recordTrack->clearSyncStartEvent(); 5949 ALOGV("RecordThread::start error %d", status); 5950 return status; 5951 } 5952 } 5953 // Catch up with current buffer indices if thread is already running. 5954 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5955 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5956 // see previously buffered data before it called start(), but with greater risk of overrun. 5957 5958 recordTrack->mResamplerBufferProvider->reset(); 5959 // clear any converter state as new data will be discontinuous 5960 recordTrack->mRecordBufferConverter->reset(); 5961 recordTrack->mState = TrackBase::STARTING_2; 5962 // signal thread to start 5963 mWaitWorkCV.broadcast(); 5964 if (mActiveTracks.indexOf(recordTrack) < 0) { 5965 ALOGV("Record failed to start"); 5966 status = BAD_VALUE; 5967 goto startError; 5968 } 5969 return status; 5970 } 5971 5972startError: 5973 if (recordTrack->isExternalTrack()) { 5974 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5975 } 5976 recordTrack->clearSyncStartEvent(); 5977 // FIXME I wonder why we do not reset the state here? 5978 return status; 5979} 5980 5981void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5982{ 5983 sp<SyncEvent> strongEvent = event.promote(); 5984 5985 if (strongEvent != 0) { 5986 sp<RefBase> ptr = strongEvent->cookie().promote(); 5987 if (ptr != 0) { 5988 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5989 recordTrack->handleSyncStartEvent(strongEvent); 5990 } 5991 } 5992} 5993 5994bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5995 ALOGV("RecordThread::stop"); 5996 AutoMutex _l(mLock); 5997 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5998 return false; 5999 } 6000 // note that threadLoop may still be processing the track at this point [without lock] 6001 recordTrack->mState = TrackBase::PAUSING; 6002 // do not wait for mStartStopCond if exiting 6003 if (exitPending()) { 6004 return true; 6005 } 6006 // FIXME incorrect usage of wait: no explicit predicate or loop 6007 mStartStopCond.wait(mLock); 6008 // if we have been restarted, recordTrack is in mActiveTracks here 6009 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6010 ALOGV("Record stopped OK"); 6011 return true; 6012 } 6013 return false; 6014} 6015 6016bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6017{ 6018 return false; 6019} 6020 6021status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6022{ 6023#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6024 if (!isValidSyncEvent(event)) { 6025 return BAD_VALUE; 6026 } 6027 6028 int eventSession = event->triggerSession(); 6029 status_t ret = NAME_NOT_FOUND; 6030 6031 Mutex::Autolock _l(mLock); 6032 6033 for (size_t i = 0; i < mTracks.size(); i++) { 6034 sp<RecordTrack> track = mTracks[i]; 6035 if (eventSession == track->sessionId()) { 6036 (void) track->setSyncEvent(event); 6037 ret = NO_ERROR; 6038 } 6039 } 6040 return ret; 6041#else 6042 return BAD_VALUE; 6043#endif 6044} 6045 6046// destroyTrack_l() must be called with ThreadBase::mLock held 6047void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6048{ 6049 track->terminate(); 6050 track->mState = TrackBase::STOPPED; 6051 // active tracks are removed by threadLoop() 6052 if (mActiveTracks.indexOf(track) < 0) { 6053 removeTrack_l(track); 6054 } 6055} 6056 6057void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6058{ 6059 mTracks.remove(track); 6060 // need anything related to effects here? 6061 if (track->isFastTrack()) { 6062 ALOG_ASSERT(!mFastTrackAvail); 6063 mFastTrackAvail = true; 6064 } 6065} 6066 6067void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6068{ 6069 dumpInternals(fd, args); 6070 dumpTracks(fd, args); 6071 dumpEffectChains(fd, args); 6072} 6073 6074void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6075{ 6076 dprintf(fd, "\nInput thread %p:\n", this); 6077 6078 dumpBase(fd, args); 6079 6080 if (mActiveTracks.size() == 0) { 6081 dprintf(fd, " No active record clients\n"); 6082 } 6083 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6084 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6085 6086 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6087 const FastCaptureDumpState copy(mFastCaptureDumpState); 6088 copy.dump(fd); 6089} 6090 6091void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6092{ 6093 const size_t SIZE = 256; 6094 char buffer[SIZE]; 6095 String8 result; 6096 6097 size_t numtracks = mTracks.size(); 6098 size_t numactive = mActiveTracks.size(); 6099 size_t numactiveseen = 0; 6100 dprintf(fd, " %d Tracks", numtracks); 6101 if (numtracks) { 6102 dprintf(fd, " of which %d are active\n", numactive); 6103 RecordTrack::appendDumpHeader(result); 6104 for (size_t i = 0; i < numtracks ; ++i) { 6105 sp<RecordTrack> track = mTracks[i]; 6106 if (track != 0) { 6107 bool active = mActiveTracks.indexOf(track) >= 0; 6108 if (active) { 6109 numactiveseen++; 6110 } 6111 track->dump(buffer, SIZE, active); 6112 result.append(buffer); 6113 } 6114 } 6115 } else { 6116 dprintf(fd, "\n"); 6117 } 6118 6119 if (numactiveseen != numactive) { 6120 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6121 " not in the track list\n"); 6122 result.append(buffer); 6123 RecordTrack::appendDumpHeader(result); 6124 for (size_t i = 0; i < numactive; ++i) { 6125 sp<RecordTrack> track = mActiveTracks[i]; 6126 if (mTracks.indexOf(track) < 0) { 6127 track->dump(buffer, SIZE, true); 6128 result.append(buffer); 6129 } 6130 } 6131 6132 } 6133 write(fd, result.string(), result.size()); 6134} 6135 6136 6137void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6138{ 6139 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6140 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6141 mRsmpInFront = recordThread->mRsmpInRear; 6142 mRsmpInUnrel = 0; 6143} 6144 6145void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6146 size_t *framesAvailable, bool *hasOverrun) 6147{ 6148 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6149 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6150 const int32_t rear = recordThread->mRsmpInRear; 6151 const int32_t front = mRsmpInFront; 6152 const ssize_t filled = rear - front; 6153 6154 size_t framesIn; 6155 bool overrun = false; 6156 if (filled < 0) { 6157 // should not happen, but treat like a massive overrun and re-sync 6158 framesIn = 0; 6159 mRsmpInFront = rear; 6160 overrun = true; 6161 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6162 framesIn = (size_t) filled; 6163 } else { 6164 // client is not keeping up with server, but give it latest data 6165 framesIn = recordThread->mRsmpInFrames; 6166 mRsmpInFront = /* front = */ rear - framesIn; 6167 overrun = true; 6168 } 6169 if (framesAvailable != NULL) { 6170 *framesAvailable = framesIn; 6171 } 6172 if (hasOverrun != NULL) { 6173 *hasOverrun = overrun; 6174 } 6175} 6176 6177// AudioBufferProvider interface 6178status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6179 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6180{ 6181 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6182 if (threadBase == 0) { 6183 buffer->frameCount = 0; 6184 buffer->raw = NULL; 6185 return NOT_ENOUGH_DATA; 6186 } 6187 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6188 int32_t rear = recordThread->mRsmpInRear; 6189 int32_t front = mRsmpInFront; 6190 ssize_t filled = rear - front; 6191 // FIXME should not be P2 (don't want to increase latency) 6192 // FIXME if client not keeping up, discard 6193 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6194 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6195 front &= recordThread->mRsmpInFramesP2 - 1; 6196 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6197 if (part1 > (size_t) filled) { 6198 part1 = filled; 6199 } 6200 size_t ask = buffer->frameCount; 6201 ALOG_ASSERT(ask > 0); 6202 if (part1 > ask) { 6203 part1 = ask; 6204 } 6205 if (part1 == 0) { 6206 // out of data is fine since the resampler will return a short-count. 6207 buffer->raw = NULL; 6208 buffer->frameCount = 0; 6209 mRsmpInUnrel = 0; 6210 return NOT_ENOUGH_DATA; 6211 } 6212 6213 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6214 buffer->frameCount = part1; 6215 mRsmpInUnrel = part1; 6216 return NO_ERROR; 6217} 6218 6219// AudioBufferProvider interface 6220void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6221 AudioBufferProvider::Buffer* buffer) 6222{ 6223 size_t stepCount = buffer->frameCount; 6224 if (stepCount == 0) { 6225 return; 6226 } 6227 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6228 mRsmpInUnrel -= stepCount; 6229 mRsmpInFront += stepCount; 6230 buffer->raw = NULL; 6231 buffer->frameCount = 0; 6232} 6233 6234AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6235 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6236 uint32_t srcSampleRate, 6237 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6238 uint32_t dstSampleRate) : 6239 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6240 // mSrcFormat 6241 // mSrcSampleRate 6242 // mDstChannelMask 6243 // mDstFormat 6244 // mDstSampleRate 6245 // mSrcChannelCount 6246 // mDstChannelCount 6247 // mDstFrameSize 6248 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6249 mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0) 6250{ 6251 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6252 dstChannelMask, dstFormat, dstSampleRate); 6253} 6254 6255AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6256 free(mBuf); 6257 delete mResampler; 6258 free(mRsmpOutBuffer); 6259} 6260 6261size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6262 AudioBufferProvider *provider, size_t frames) 6263{ 6264 if (mSrcSampleRate == mDstSampleRate) { 6265 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6266 mSrcSampleRate, mSrcFormat, mDstFormat); 6267 6268 AudioBufferProvider::Buffer buffer; 6269 for (size_t i = frames; i > 0; ) { 6270 buffer.frameCount = i; 6271 status_t status = provider->getNextBuffer(&buffer, 0); 6272 if (status != OK || buffer.frameCount == 0) { 6273 frames -= i; // cannot fill request. 6274 break; 6275 } 6276 // convert to destination buffer 6277 convert(dst, buffer.raw, buffer.frameCount); 6278 6279 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6280 i -= buffer.frameCount; 6281 provider->releaseBuffer(&buffer); 6282 } 6283 } else { 6284 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6285 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6286 6287 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 6288 if (mRsmpOutFrameCount < frames) { 6289 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 6290 free(mRsmpOutBuffer); 6291 // resampler always outputs stereo (FOR NOW) 6292 (void)posix_memalign(&mRsmpOutBuffer, 32, frames * FCC_2 * sizeof(int32_t) /*Q4.27*/); 6293 mRsmpOutFrameCount = frames; 6294 } 6295 // resampler accumulates, but we only have one source track 6296 memset(mRsmpOutBuffer, 0, frames * FCC_2 * sizeof(int32_t)); 6297 frames = mResampler->resample((int32_t*)mRsmpOutBuffer, frames, provider); 6298 6299 // convert to destination buffer 6300 convert(dst, mRsmpOutBuffer, frames); 6301 } 6302 return frames; 6303} 6304 6305status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6306 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6307 uint32_t srcSampleRate, 6308 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6309 uint32_t dstSampleRate) 6310{ 6311 // quick evaluation if there is any change. 6312 if (mSrcFormat == srcFormat 6313 && mSrcChannelMask == srcChannelMask 6314 && mSrcSampleRate == srcSampleRate 6315 && mDstFormat == dstFormat 6316 && mDstChannelMask == dstChannelMask 6317 && mDstSampleRate == dstSampleRate) { 6318 return NO_ERROR; 6319 } 6320 6321 const bool valid = 6322 audio_is_input_channel(srcChannelMask) 6323 && audio_is_input_channel(dstChannelMask) 6324 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6325 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6326 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6327 ; // no upsampling checks for now 6328 if (!valid) { 6329 return BAD_VALUE; 6330 } 6331 6332 mSrcFormat = srcFormat; 6333 mSrcChannelMask = srcChannelMask; 6334 mSrcSampleRate = srcSampleRate; 6335 mDstFormat = dstFormat; 6336 mDstChannelMask = dstChannelMask; 6337 mDstSampleRate = dstSampleRate; 6338 6339 // compute derived parameters 6340 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6341 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6342 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6343 6344 // do we need a format buffer? 6345 if (mSrcFormat != mDstFormat && mDstChannelCount != mSrcChannelCount) { 6346 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6347 } else { 6348 mBufFrameSize = 0; 6349 } 6350 mBufFrames = 0; // force the buffer to be resized. 6351 6352 // do we need to resample? 6353 if (mSrcSampleRate != mDstSampleRate) { 6354 if (mResampler != NULL) { 6355 delete mResampler; 6356 } 6357 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, 6358 mSrcChannelCount, mDstSampleRate); // may seem confusing... 6359 mResampler->setSampleRate(mSrcSampleRate); 6360 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6361 } 6362 return NO_ERROR; 6363} 6364 6365void AudioFlinger::RecordThread::RecordBufferConverter::convert( 6366 void *dst, /*const*/ void *src, size_t frames) 6367{ 6368 // check if a memcpy will do 6369 if (mResampler == NULL 6370 && mSrcChannelCount == mDstChannelCount 6371 && mSrcFormat == mDstFormat) { 6372 memcpy(dst, src, 6373 frames * mDstChannelCount * audio_bytes_per_sample(mDstFormat)); 6374 return; 6375 } 6376 // reallocate buffer if needed 6377 if (mBufFrameSize != 0 && mBufFrames < frames) { 6378 free(mBuf); 6379 mBufFrames = frames; 6380 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6381 } 6382 // do processing 6383 if (mResampler != NULL) { 6384 // src channel count is always >= 2. 6385 void *dstBuf = mBuf != NULL ? mBuf : dst; 6386 // ditherAndClamp() works as long as all buffers returned by 6387 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 6388 if (mDstChannelCount == 1) { 6389 // the resampler always outputs stereo samples. 6390 // FIXME: this rewrites back into src 6391 ditherAndClamp((int32_t *)src, (const int32_t *)src, frames); 6392 downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf, 6393 (const int16_t *)src, frames); 6394 } else { 6395 ditherAndClamp((int32_t *)dstBuf, (const int32_t *)src, frames); 6396 } 6397 } else if (mSrcChannelCount != mDstChannelCount) { 6398 void *dstBuf = mBuf != NULL ? mBuf : dst; 6399 if (mSrcChannelCount == 1) { 6400 upmix_to_stereo_i16_from_mono_i16((int16_t *)dstBuf, (const int16_t *)src, 6401 frames); 6402 } else { 6403 downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf, 6404 (const int16_t *)src, frames); 6405 } 6406 } 6407 if (mSrcFormat != mDstFormat) { 6408 void *srcBuf = mBuf != NULL ? mBuf : src; 6409 memcpy_by_audio_format(dst, mDstFormat, srcBuf, mSrcFormat, 6410 frames * mDstChannelCount); 6411 } 6412} 6413 6414bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6415 status_t& status) 6416{ 6417 bool reconfig = false; 6418 6419 status = NO_ERROR; 6420 6421 audio_format_t reqFormat = mFormat; 6422 uint32_t samplingRate = mSampleRate; 6423 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6424 6425 AudioParameter param = AudioParameter(keyValuePair); 6426 int value; 6427 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6428 // channel count change can be requested. Do we mandate the first client defines the 6429 // HAL sampling rate and channel count or do we allow changes on the fly? 6430 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6431 samplingRate = value; 6432 reconfig = true; 6433 } 6434 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6435 if (!audio_is_linear_pcm((audio_format_t) value)) { 6436 status = BAD_VALUE; 6437 } else { 6438 reqFormat = (audio_format_t) value; 6439 reconfig = true; 6440 } 6441 } 6442 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6443 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6444 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6445 status = BAD_VALUE; 6446 } else { 6447 channelMask = mask; 6448 reconfig = true; 6449 } 6450 } 6451 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6452 // do not accept frame count changes if tracks are open as the track buffer 6453 // size depends on frame count and correct behavior would not be guaranteed 6454 // if frame count is changed after track creation 6455 if (mActiveTracks.size() > 0) { 6456 status = INVALID_OPERATION; 6457 } else { 6458 reconfig = true; 6459 } 6460 } 6461 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6462 // forward device change to effects that have requested to be 6463 // aware of attached audio device. 6464 for (size_t i = 0; i < mEffectChains.size(); i++) { 6465 mEffectChains[i]->setDevice_l(value); 6466 } 6467 6468 // store input device and output device but do not forward output device to audio HAL. 6469 // Note that status is ignored by the caller for output device 6470 // (see AudioFlinger::setParameters() 6471 if (audio_is_output_devices(value)) { 6472 mOutDevice = value; 6473 status = BAD_VALUE; 6474 } else { 6475 mInDevice = value; 6476 // disable AEC and NS if the device is a BT SCO headset supporting those 6477 // pre processings 6478 if (mTracks.size() > 0) { 6479 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6480 mAudioFlinger->btNrecIsOff(); 6481 for (size_t i = 0; i < mTracks.size(); i++) { 6482 sp<RecordTrack> track = mTracks[i]; 6483 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6484 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6485 } 6486 } 6487 } 6488 } 6489 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6490 mAudioSource != (audio_source_t)value) { 6491 // forward device change to effects that have requested to be 6492 // aware of attached audio device. 6493 for (size_t i = 0; i < mEffectChains.size(); i++) { 6494 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6495 } 6496 mAudioSource = (audio_source_t)value; 6497 } 6498 6499 if (status == NO_ERROR) { 6500 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6501 keyValuePair.string()); 6502 if (status == INVALID_OPERATION) { 6503 inputStandBy(); 6504 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6505 keyValuePair.string()); 6506 } 6507 if (reconfig) { 6508 if (status == BAD_VALUE && 6509 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6510 audio_is_linear_pcm(reqFormat) && 6511 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6512 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6513 audio_channel_count_from_in_mask( 6514 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6515 (channelMask == AUDIO_CHANNEL_IN_MONO || 6516 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6517 status = NO_ERROR; 6518 } 6519 if (status == NO_ERROR) { 6520 readInputParameters_l(); 6521 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6522 } 6523 } 6524 } 6525 6526 return reconfig; 6527} 6528 6529String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6530{ 6531 Mutex::Autolock _l(mLock); 6532 if (initCheck() != NO_ERROR) { 6533 return String8(); 6534 } 6535 6536 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6537 const String8 out_s8(s); 6538 free(s); 6539 return out_s8; 6540} 6541 6542void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6543 AudioSystem::OutputDescriptor desc; 6544 const void *param2 = NULL; 6545 6546 switch (event) { 6547 case AudioSystem::INPUT_OPENED: 6548 case AudioSystem::INPUT_CONFIG_CHANGED: 6549 desc.channelMask = mChannelMask; 6550 desc.samplingRate = mSampleRate; 6551 desc.format = mFormat; 6552 desc.frameCount = mFrameCount; 6553 desc.latency = 0; 6554 param2 = &desc; 6555 break; 6556 6557 case AudioSystem::INPUT_CLOSED: 6558 default: 6559 break; 6560 } 6561 mAudioFlinger->audioConfigChanged(event, mId, param2); 6562} 6563 6564void AudioFlinger::RecordThread::readInputParameters_l() 6565{ 6566 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6567 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6568 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6569 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6570 mFormat = mHALFormat; 6571 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6572 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6573 } 6574 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6575 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6576 mFrameCount = mBufferSize / mFrameSize; 6577 // This is the formula for calculating the temporary buffer size. 6578 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6579 // 1 full output buffer, regardless of the alignment of the available input. 6580 // The value is somewhat arbitrary, and could probably be even larger. 6581 // A larger value should allow more old data to be read after a track calls start(), 6582 // without increasing latency. 6583 // 6584 // Note this is independent of the maximum downsampling ratio permitted for capture. 6585 mRsmpInFrames = mFrameCount * 7; 6586 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6587 delete[] mRsmpInBuffer; 6588 6589 // TODO optimize audio capture buffer sizes ... 6590 // Here we calculate the size of the sliding buffer used as a source 6591 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6592 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6593 // be better to have it derived from the pipe depth in the long term. 6594 // The current value is higher than necessary. However it should not add to latency. 6595 6596 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6597 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6598 6599 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6600 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6601} 6602 6603uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6604{ 6605 Mutex::Autolock _l(mLock); 6606 if (initCheck() != NO_ERROR) { 6607 return 0; 6608 } 6609 6610 return mInput->stream->get_input_frames_lost(mInput->stream); 6611} 6612 6613uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6614{ 6615 Mutex::Autolock _l(mLock); 6616 uint32_t result = 0; 6617 if (getEffectChain_l(sessionId) != 0) { 6618 result = EFFECT_SESSION; 6619 } 6620 6621 for (size_t i = 0; i < mTracks.size(); ++i) { 6622 if (sessionId == mTracks[i]->sessionId()) { 6623 result |= TRACK_SESSION; 6624 break; 6625 } 6626 } 6627 6628 return result; 6629} 6630 6631KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6632{ 6633 KeyedVector<int, bool> ids; 6634 Mutex::Autolock _l(mLock); 6635 for (size_t j = 0; j < mTracks.size(); ++j) { 6636 sp<RecordThread::RecordTrack> track = mTracks[j]; 6637 int sessionId = track->sessionId(); 6638 if (ids.indexOfKey(sessionId) < 0) { 6639 ids.add(sessionId, true); 6640 } 6641 } 6642 return ids; 6643} 6644 6645AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6646{ 6647 Mutex::Autolock _l(mLock); 6648 AudioStreamIn *input = mInput; 6649 mInput = NULL; 6650 return input; 6651} 6652 6653// this method must always be called either with ThreadBase mLock held or inside the thread loop 6654audio_stream_t* AudioFlinger::RecordThread::stream() const 6655{ 6656 if (mInput == NULL) { 6657 return NULL; 6658 } 6659 return &mInput->stream->common; 6660} 6661 6662status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6663{ 6664 // only one chain per input thread 6665 if (mEffectChains.size() != 0) { 6666 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6667 return INVALID_OPERATION; 6668 } 6669 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6670 chain->setThread(this); 6671 chain->setInBuffer(NULL); 6672 chain->setOutBuffer(NULL); 6673 6674 checkSuspendOnAddEffectChain_l(chain); 6675 6676 // make sure enabled pre processing effects state is communicated to the HAL as we 6677 // just moved them to a new input stream. 6678 chain->syncHalEffectsState(); 6679 6680 mEffectChains.add(chain); 6681 6682 return NO_ERROR; 6683} 6684 6685size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6686{ 6687 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6688 ALOGW_IF(mEffectChains.size() != 1, 6689 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6690 chain.get(), mEffectChains.size(), this); 6691 if (mEffectChains.size() == 1) { 6692 mEffectChains.removeAt(0); 6693 } 6694 return 0; 6695} 6696 6697status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6698 audio_patch_handle_t *handle) 6699{ 6700 status_t status = NO_ERROR; 6701 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6702 // store new device and send to effects 6703 mInDevice = patch->sources[0].ext.device.type; 6704 for (size_t i = 0; i < mEffectChains.size(); i++) { 6705 mEffectChains[i]->setDevice_l(mInDevice); 6706 } 6707 6708 // disable AEC and NS if the device is a BT SCO headset supporting those 6709 // pre processings 6710 if (mTracks.size() > 0) { 6711 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6712 mAudioFlinger->btNrecIsOff(); 6713 for (size_t i = 0; i < mTracks.size(); i++) { 6714 sp<RecordTrack> track = mTracks[i]; 6715 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6716 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6717 } 6718 } 6719 6720 // store new source and send to effects 6721 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6722 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6723 for (size_t i = 0; i < mEffectChains.size(); i++) { 6724 mEffectChains[i]->setAudioSource_l(mAudioSource); 6725 } 6726 } 6727 6728 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6729 status = hwDevice->create_audio_patch(hwDevice, 6730 patch->num_sources, 6731 patch->sources, 6732 patch->num_sinks, 6733 patch->sinks, 6734 handle); 6735 } else { 6736 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6737 } 6738 return status; 6739} 6740 6741status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6742{ 6743 status_t status = NO_ERROR; 6744 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6745 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6746 status = hwDevice->release_audio_patch(hwDevice, handle); 6747 } else { 6748 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6749 } 6750 return status; 6751} 6752 6753void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6754{ 6755 Mutex::Autolock _l(mLock); 6756 mTracks.add(record); 6757} 6758 6759void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6760{ 6761 Mutex::Autolock _l(mLock); 6762 destroyTrack_l(record); 6763} 6764 6765void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6766{ 6767 ThreadBase::getAudioPortConfig(config); 6768 config->role = AUDIO_PORT_ROLE_SINK; 6769 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6770 config->ext.mix.usecase.source = mAudioSource; 6771} 6772 6773} // namespace android 6774